| 00:10.45 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 00:10.45 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.11, Asterisk-addons 1.2.4, and Zaptel 1.2.8 released! (August 22, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 00:10.47 | *** join/#asterisk jpeeler (n=jpeeler@130-127-132-164.wireless.clemson.edu) |
| 00:10.54 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
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| 00:11.57 | mmurdock | howdy all. |
| 00:11.57 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 00:12.03 | mmurdock | Is anyone here using Snom phones? |
| 00:12.12 | lordbaron | yes |
| 00:12.35 | *** join/#asterisk EyeCue (n=eyecue@220-253-90-201.QLD.netspace.net.au) |
| 00:12.36 | mmurdock | lordbaron: Are you using the park orbit button? |
| 00:12.51 | s0lid | can anyone point me on how to restrict dialing on registered user only |
| 00:12.52 | lordbaron | no |
| 00:13.04 | mmurdock | lordbaron: How are you parking your calls? |
| 00:13.21 | lordbaron | xfer to an extension |
| 00:13.21 | *** join/#asterisk awe6 (n=lba@69.91.84.176) |
| 00:13.41 | lordbaron | we xfer into a queue |
| 00:13.43 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 00:13.48 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-59.dslextreme.com) |
| 00:14.00 | mmurdock | lordbaron: I see, your not using park feature in Asterisk then. |
| 00:14.19 | lordbaron | no, but I believe you can setup a park ext, and xfer there |
| 00:14.59 | mmurdock | lordbaron: Well the problem I'm having is that my snom phone is hanging up before Asterisk announces which extension it parked the call on. |
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| 00:15.16 | *** join/#asterisk _Ellipse_ (n=Not@cpe-69-135-193-16.woh.res.rr.com) |
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| 00:15.24 | lordbaron | mmurdock: which snom phone? |
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| 00:15.41 | _Ellipse_ | Hello everyone |
| 00:16.01 | mmurdock | lordbaron: I have both a 300 and 320 |
| 00:16.07 | _Ellipse_ | Has anyone here used an SPA-3102 with a FAX machine for sending AND recieving faxes with Asterisk? |
| 00:16.29 | lordbaron | mmurdock: are you using a DP Ext for parking? |
| 00:17.32 | lordbaron | mmurdock: have you turned off autodial, and tested? |
| 00:17.48 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 00:17.52 | mmurdock | lordbaron: I have one of the function puttons set to parkorbit 70 |
| 00:18.28 | mmurdock | lordbaron: Autodial is off. |
| 00:18.36 | *** join/#asterisk tessier (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 00:18.42 | awe6 | Which codec for my Grandstream phones that only allow GSM, PCMU, PCMA, G.723.1 or G.729A/B? Which qualifies as G.711 codec (either alaw or ulaw) as that is a codec that is known to work with Asterisk? |
| 00:19.10 | Snake-Eyes | I was wondering if I have a clear understanding of what 'hints' are in Asterisk. Is hints used to find out what extensions are online/offline/busy, as in user X is on a call or is free to accept a call ? |
| 00:19.38 | mmurdock | lordbaron: It parks it fine, but when watching the Asterisk cli it show it trying to play the first digit of the extension it parks to and that it. |
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| 00:19.55 | mmurdock | Snake-Eyes: Yes |
| 00:19.58 | [TK]D-Fender | awe6 All of those codec CAN work with *, but * has native support for all except G.729 and G.723 so pick any of the others |
| 00:20.08 | Snake-Eyes | mmurdock, cool thanks |
| 00:20.19 | mmurdock | Snake-Eyes: np |
| 00:20.25 | [TK]D-Fender | awe6 : PCMA = ALAW, PCMU = ULAW. Both are G.711 varients |
| 00:20.41 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.20) |
| 00:20.56 | lordbaron | mmurdock: I assume you checked the call completion settings, etc under "phone behavior", and tried various settings? I don't have a specific answer, as I don't use that feature. Sorry. |
| 00:20.58 | *** join/#asterisk Synoptic (i=Synoptic@modemcable236.180-203-24.mc.videotron.ca) |
| 00:21.03 | [TK]D-Fender | mmurdock : I'd suggest you take a look at app_valetparking. Vastly superior app for that purpose. Easy to "stack" you calls with. |
| 00:21.04 | Snake-Eyes | awe6, I use ulaw, alaw and g729(lic) for our grandstreams, all work welll |
| 00:21.09 | _Ellipse_ | hello? |
| 00:21.21 | Synoptic | hello there |
| 00:21.44 | Synoptic | I have a question concerning inbound routes with two asterisk connected in IAX user/peer mode |
| 00:21.50 | awe6 | [TK]D-Fender: Thanks. I'll set the phones up to use PCMA, PCMU and GSM in that order unless you suggest differently. |
| 00:22.56 | JT | awe6: what country are you located in? |
| 00:22.59 | *** part/#asterisk ViTa (n=vita@host.200.47.122.173.static.itcsa.net) |
| 00:23.08 | Synoptic | BoxA has extensions in the 800's, BoxB has extensions in the 600's. I want to dial from boxA 601 and make 601 and only 601 rings on BoxB. |
| 00:23.14 | awe6 | JT: USA. Well, Texas <g> |
| 00:23.27 | JT | awe6: PCMU should be higher priority then |
| 00:23.46 | JT | usa uses Mu ("u") law |
| 00:23.58 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 00:24.07 | lordbaron | Anyone have an idea why my Digium IAXY will not connect to the zap line? It dials, the zap rings, and I can answer the other side, but dead air. On the Iaxy side..constant ringing. Sip->iaxy works fine, as does iaxy->sip |
| 00:24.08 | awe6 | JT: OK I'll change to PCMU, then PCMA, then GSM. Is correct? |
| 00:24.09 | JT | most of the rest of the world uses a law companding |
| 00:24.27 | JT | awe6: looks fine to me |
| 00:24.50 | *** join/#asterisk Nivex (n=kjotte@user-0c8hq6g.cable.mindspring.com) |
| 00:24.50 | awe6 | JT: Thanks much. |
| 00:25.36 | *** join/#asterisk fafnir (n=notfaf@unaffiliated/fafnir) |
| 00:26.38 | [TK]D-Fender | awe6 : are you phones on a local LAN w/ 8? |
| 00:26.39 | Synoptic | anyone have time to explain me how the inboud routing works ? |
| 00:26.40 | [TK]D-Fender | *? |
| 00:27.35 | *** join/#asterisk mishehu (i=mishehu@207.229.186.131) |
| 00:27.45 | awe6 | [TK]D-Fender: Yes on local lan. Don't under "w/ 8?" |
| 00:27.57 | [TK]D-Fender | Synoptic : http://www.voip-info.org/wiki-Asterisk+-+dual+servers |
| 00:27.58 | awe6 | [TK]D-Fender: under=understand |
| 00:28.08 | JT | "with asterisk" i think he meant |
| 00:28.09 | [TK]D-Fender | awe6 : Typo. Mean "with Asterisk" |
| 00:28.32 | JT | g.711/pcm should be fine on most lans |
| 00:28.34 | Synoptic | [TK]D-Fender : read that and it is working |
| 00:28.36 | [TK]D-Fender | awe6 : So if you are local, forget everything else, and jsut use ULAW if you're in North America, and ALAW otherwise |
| 00:28.45 | awe6 | [TK]D-Fender: Asterisk and the phones are all inside a lan. |
| 00:28.55 | mishehu | hey guys, I'm trying to debug a speex issue on x86_64. I don't know if this is sufficient info, but I've run asterisk thru valgrin, and have some output here: http://pastebin.com/777562 |
| 00:29.17 | Synoptic | [TK]D-Fender: my question is not how to connect two boxes, is how do I change the behaviour of dialing between the two boxes |
| 00:29.17 | [TK]D-Fender | Synoptic : What do you mean by "inbound routing"? |
| 00:30.03 | Synoptic | [TK]D-Fender, for example, when from boxA I dial an extension of boxB, in box B, how do I configure the route so the dialed extension rings and not having to specify a specific destination in the inboud route. |
| 00:30.12 | Synoptic | i'm using the FreePBX interface though |
| 00:30.16 | [TK]D-Fender | Synoptic : Well just include a pattern to send calls to the other box like "exten => _6XX,1,Dial(SIP/BoxB/${EXTEN}) |
| 00:30.20 | Synoptic | and novbody in freepbs can help me right now |
| 00:30.35 | Synoptic | [TK]D-Fender : i'm using IAX |
| 00:30.42 | [TK]D-Fender | Synoptic : And doubtful anyone HERE will either.... rad the channel topic... |
| 00:30.52 | [TK]D-Fender | Synoptic : read* |
| 00:31.01 | [TK]D-Fender | Synoptic : And IAX2 isn't much different from SIP. |
| 00:31.53 | Synoptic | [TK]D-Fender, ok, well, i'm going to modify the conf files manually then to make some tests. is the syntax different for IAX ? or the "exten => _6XX,1,Dial(SIP/BoxB/${EXTEN}) will work ? |
| 00:32.22 | Synoptic | instead of SIP/BoxB having IAX2/BoxB ? |
| 00:34.24 | [TK]D-Fender | Synoptic : Pretty mucht he same. |
| 00:34.26 | Synoptic | ok |
| 00:34.53 | Synoptic | this line you pasted, in which conf file do I put it ? |
| 00:35.08 | Synoptic | i'm trying to matcht he conf file with the freepbx gui |
| 00:35.14 | Synoptic | the |
| 00:35.17 | Synoptic | and match |
| 00:35.19 | Synoptic | ;P |
| 00:35.45 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 00:35.45 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:36.02 | Strom_C | Synoptic: either go to #freepbx or wipe out freepbx and learn /real/ asterisk |
| 00:36.18 | *** part/#asterisk Synoptic (i=Synoptic@modemcable236.180-203-24.mc.videotron.ca) |
| 00:36.57 | hmmhesays | you might be able to persuade me to fix your config |
| 00:38.01 | hmmhesays | oops |
| 00:38.02 | hmmhesays | hah |
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| 00:42.44 | *** part/#asterisk lordbaron (n=redbaron@host55-226.rancor.birch.net) |
| 00:46.20 | rg1_ | anyone know if you are allowed to use nested While() loops? |
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| 00:47.26 | Strom_C | rg1_: I don't believe you are |
| 00:49.38 | rg1_ | thank you |
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| 00:51.15 | *** join/#asterisk ANTILOCAS (n=uoiuyiu@200.87.51.226) |
| 00:53.22 | ANTILOCAS | Im trying to install the Grandstream HandyTone HT-496. My network arquitecture has star topology where even the modem-router is conected to my switch. The HandyTone manual says |
| 00:53.32 | ANTILOCAS | a) insert the eternet cavble into the WAN port of HandyTone and conect the other end of the eternet cable to an uplink port" |
| 00:53.39 | ANTILOCAS | b) connect a PC to the LAN port of HandyTone |
| 00:53.47 | ANTILOCAS | Knowing my architecture how would this be? I have to configure throgh LAN port first cause WAN port configuration is disabled by default |
| 00:54.19 | ANTILOCAS | what kind of cables do i have to use for each conection? |
| 00:54.40 | Qwell | You don't need to put a PC on the LAN port |
| 00:54.48 | Qwell | don't have to put anything there |
| 00:54.54 | Qwell | Just plug the WAN port into the switch |
| 00:55.01 | file | power cables... |
| 00:55.12 | ANTILOCAS | i did but i cant enter the page to configure it |
| 00:55.39 | ANTILOCAS | the manual says that Wan configuration is disabled |
| 00:55.54 | ANTILOCAS | by default and u have to activate it through lan port first |
| 00:56.06 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:56.41 | ANTILOCAS | i tried with the 2 ips the one that says the manual and the one that is in the menu phone option |
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| 01:13.24 | hmmhesays | so file: sorry for the retarded questions last night |
| 01:14.09 | file | awww, were you drunk? maybe? |
| 01:14.17 | hmmhesays | no just really tired |
| 01:14.21 | file | bah |
| 01:14.22 | file | that's no fun |
| 01:15.44 | hmmhesays | yeah no doubt |
| 01:16.07 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 01:16.54 | Dr-Linux | question, what framing and coding i can use in Gautemala for E1 pri signalling? |
| 01:17.02 | Dr-Linux | span=1,1,0,ccs,hdb3,crc4 << is that correct? |
| 01:17.03 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 01:19.05 | Skyelar | ANTILOCAS: you have to plug the WAN port into the LAN (with the DHCP server), then a PC into the LAN port, get a DHCP address from the HT486, then log into the interface and change it to "bridge mode" and enable server-side HTTP access |
| 01:19.56 | Skyelar | ANTILOCAS: once that's done, you can unplug the PC from the LAN port, and access the web interface as you'd expect |
| 01:21.20 | Dr-Linux | anybody please answer my question? |
| 01:24.06 | [TK]D-Fender | Dr-Linux : What do you think the odds are that someone knows that answer off the top of their head? How about calling up a telco down there and ask...... Better idea |
| 01:25.19 | Skyelar | ... especially as it can differ from telco to telco |
| 01:25.21 | Dr-Linux | [TK]D-Fender: aww i first time heard that country name .. and you are talking about telco .. |
| 01:25.35 | Dr-Linux | Skyelar? |
| 01:25.46 | Dr-Linux | Skyelar: sunday bot? |
| 01:25.53 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-152.sd.sd.cox.net) |
| 01:26.15 | Skyelar | Dr-Linux: although generally the settings are the same throughout a country, it's entirely possible that different telcos within the same country will use different settings |
| 01:26.42 | tessier | Will extensions.ael eventually replace the old extensions.conf? |
| 01:27.28 | Dr-Linux | hhm.. |
| 01:28.00 | Dr-Linux | anybody knows most comon central america settings for e1 |
| 01:28.08 | [TK]D-Fender | tessier : AGI sorta did taht already, though the complications suck for many |
| 01:28.14 | hmmhesays | heh common? I dont' think there are |
| 01:28.25 | hmmhesays | r2 is popular in central america unfortunately though |
| 01:28.27 | tessier | [TK]D-Fender: Yeah, AGI was rather hard to deal with. |
| 01:28.32 | [TK]D-Fender | tessier : And its not a whole new language. AEL jsut gets down-mixed to extensions.conf logic. |
| 01:29.20 | tessier | It's been a couple years since I messed with AGI |
| 01:29.25 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 01:29.34 | [TK]D-Fender | Dr-Linux : Time to get off your butt and CALL........You want to know something ask the people who really know. |
| 01:29.37 | file | it's all interpreted and changed into the same internal thing in memory |
| 01:29.49 | *** part/#asterisk mmurdock (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net) |
| 01:30.22 | ANTILOCAS | skyelar |
| 01:30.42 | ANTILOCAS | do i have to enter using 192.168.2.1 or the one that is in my phone menu |
| 01:30.55 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 01:31.43 | Skyelar | ANTILOCAS: once you've got a DHCP lease on your PC plugged into the LAN port, look at what your default route is - which is likely to be the 192.168.2.1 address, but not necessarily. Then enter that |
| 01:33.03 | ANTILOCAS | u mean the default gateway address? |
| 01:33.14 | ANTILOCAS | thats the address of my modemrouter |
| 01:33.22 | ANTILOCAS | do i have to configure my modem first? |
| 01:34.17 | Dr-Linux | [TK]D-Fender: sir :P |
| 01:34.45 | Skyelar | ANTILOCAS: plug your PC into the LAN port, do whatever your PC's OS requires you to do to get a new IP address, then check your route settings - "netstat -rn" under most unices, "route print" under Windows |
| 01:36.28 | [TK]D-Fender | Route should be irrelevent when you're on the same subnet as the device you're looking for |
| 01:36.53 | Skyelar | [TK]D-Fender: the device becomes the default gateway - point being it's the easiest way to find out what the internal address of it is |
| 01:37.36 | [TK]D-Fender | Skyelar : If it enables DHCP sure. |
| 01:37.45 | ANTILOCAS | skyelar first Do i have to use ethernet 6wires cables for both LAN and WAN port right? |
| 01:38.03 | [TK]D-Fender | Skyelar : then again if you're already behind another router then just nmap it from a *nix box |
| 01:38.31 | Skyelar | [TK]D-Fender: which it does by default. |
| 01:39.08 | Skyelar | ANTILOCAS: err, standard RJ45 ethernet cables should be fine - and you can probably get away with only plugging in the HT486 LAN port <-> PC link |
| 01:39.32 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 01:40.01 | Skyelar | ANTILOCAS: the unit is still set to factory defaults, correct? |
| 01:40.59 | ANTILOCAS | yes |
| 01:41.13 | ANTILOCAS | ok everything is plug and my pc has ip that ends with 4 |
| 01:41.17 | ANTILOCAS | and my ATA ends with 5 |
| 01:41.32 | Skyelar | ANTILOCAS: can you reach it via the browser now? |
| 01:41.50 | ANTILOCAS | wich address? the one that ends with 5? |
| 01:41.58 | Skyelar | in theory, yes |
| 01:42.27 | hmmhesays | so they just killed an innocent guy on cold case |
| 01:43.06 | ANTILOCAS | skyelar i lost internet in the other PC |
| 01:43.29 | Skyelar | ANTILOCAS: on the other PC, can you get up the HT486's web interface? |
| 01:43.45 | awe6 | For sip phones and Asterisk behind the same NAT, do calls to extensions go out to the voip provider or are they direct inside to inside? |
| 01:44.03 | hmmhesays | they go where you send them |
| 01:44.31 | awe6 | hmmhesays: Example: I am at extension 120 and dial 122. |
| 01:45.05 | hmmhesays | it would go where you have the dialplan set up for it to go |
| 01:46.03 | ANTILOCAS | skyelar i cant |
| 01:46.30 | Skyelar | ANTILOCAS: let me reset one of my test units here, since I can't remember exactly what they come up on by default |
| 01:47.08 | awe6 | hmmhesays: The dialplan being extensions.conf? |
| 01:47.32 | ANTILOCAS | u sure i have to use same ethernet cables with Wan Port , it says uplink, doesnt that mean i have to use the 4 wire cable one? |
| 01:47.46 | ANTILOCAS | the one that is similar to the modem dsl cable signal |
| 01:48.04 | ANTILOCAS | i mean the one that uses the modem to conect to switch |
| 01:50.09 | Skyelar | ANTILOCAS: this unit has 2 x RJ45 and 2 x RJ11/RJ12 sockets, correct? |
| 01:51.05 | ANTILOCAS | yes |
| 01:51.30 | *** join/#asterisk awannabe (n=brad@ip24-251-135-202.ph.ph.cox.net) |
| 01:53.33 | awannabe | is there a way to ring multiple extensions, but if a extension is already on a call that it does not ring that extension, and then rings the rest? |
| 01:54.12 | ANTILOCAS | skyelar the manual says that i have to type ipconfig /release, but i got an error |
| 01:55.45 | hmmhesays | ipconfig? windows networking? |
| 01:56.51 | *** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net) |
| 01:57.25 | Skyelar | ANTILOCAS: what's the OS on the PC? |
| 01:57.36 | ANTILOCAS | winxp pro |
| 01:58.02 | ANTILOCAS | im not sure if the ATA came with that 4 wire cable that i told u |
| 01:58.22 | ANTILOCAS | isnt that the one i have to plug from WAN port to swicth? |
| 01:58.28 | awe6 | hmmhesays: This is what is executed in CLI debug when ext 124 calls ext 120: Executing Macro ("sip124/GDCA" "stdexten/120/sip/120") in new stack |
| 01:58.50 | awe6 | hmmhesays: Is this going to the SIP provider? IOW, does it go outside? |
| 01:59.07 | hmmhesays | you must be using trixbox or freepbx |
| 01:59.12 | Skyelar | ANTILOCAS: if you're not sure, avoid it, and use a known-good straight-through cat5 cable |
| 01:59.21 | awe6 | hmmhesays: No just plain old Asterisk |
| 01:59.29 | hmmhesays | who set up extensions.conf? |
| 01:59.34 | ANTILOCAS | im using that |
| 01:59.50 | ANTILOCAS | the usual ethernet cables |
| 01:59.58 | Skyelar | ANTILOCAS: connect one end of cable to LAN port on HT496, other end to PC. Turn on HT496. Turn on PC (if you can't figure out how to renew DHCP lease) |
| 02:00.01 | awe6 | hmmhesays: Combo of some local advice, net reading, Future of Tel book and me |
| 02:00.03 | hmmhesays | that is going to macro stdexten with args 120 sip and 120 |
| 02:00.07 | Skyelar | ANTILOCAS: then go to 192.168.2.1 in browser |
| 02:00.08 | hmmhesays | i see |
| 02:00.13 | hmmhesays | well you are calling a macro |
| 02:00.30 | awe6 | hmmhesays: yes |
| 02:01.24 | awe6 | hmmhesays: These are what I call "intercom" calls. |
| 02:01.36 | ANTILOCAS | skyelar just that without the WAN-Switch conection? |
| 02:02.20 | Skyelar | ANTILOCAS: it should work without it, yeah (just tested it with a HT486) |
| 02:04.32 | awe6 | hmmhesays: Macro stdexten starts with: exten => s,1,Dial(${ARG2},20) where ${ARG2} - Device(s) to ring |
| 02:05.25 | awe6 | hmmhesays: So I would think I'm dialing the device named 120 which is a Grandstream phone. |
| 02:06.39 | awe6 | hmmhesays: I don't understand WHY or IF these calls are coming into sip.conf |
| 02:07.38 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 02:10.17 | ANTILOCAS | skyelar is not working |
| 02:10.20 | ANTILOCAS | cant conect |
| 02:10.26 | ANTILOCAS | theres no conection |
| 02:10.36 | ANTILOCAS | in internet explorer |
| 02:11.07 | ANTILOCAS | just the LAN with the ATA cant work |
| 02:11.08 | Skyelar | ANTILOCAS: please explain how things are set up, and if possible use Firefox or something that gives better error messages |
| 02:11.37 | ANTILOCAS | does the PC need to be conected to the internet? |
| 02:12.10 | Skyelar | ANTILOCAS: PC's LAN port <-> HT496's LAN port |
| 02:12.35 | Skyelar | using a straight through cable, not a crossover cable |
| 02:16.04 | ANTILOCAS | how can you have internet conection with only that |
| 02:16.17 | ANTILOCAS | my PC is conected to the HT496 |
| 02:16.22 | ANTILOCAS | but firefox |
| 02:16.25 | ANTILOCAS | cant either |
| 02:16.43 | Skyelar | ANTILOCAS: you don't. You use that to change the setup of the HT496 so that you can connect it directly to the LAN without all this rigmarole |
| 02:16.52 | Skyelar | ANTILOCAS: going to http://192.168.2.1/ ? |
| 02:17.02 | ANTILOCAS | yeah that address is not working |
| 02:17.11 | Skyelar | ANTILOCAS: what's the IP address of the PC now? |
| 02:18.02 | hmmhesays | sounds like someone needs to read up on networking |
| 02:18.12 | ANTILOCAS | 192.168.1.4 |
| 02:18.23 | ANTILOCAS | and the ATA is 0.0.0.0 |
| 02:18.46 | Skyelar | hmmhesays: yeah, Grandstream. The HT486/HT496 beasties come out-of-the-box with a completely unworkable configuration |
| 02:19.16 | Skyelar | ANTILOCAS: check that the PC is set to DHCP (ie. "Obtain IP address automatically") |
| 02:20.24 | tessier | Anyone here done Snom mass-deployment? I want to protect the snom configs on my http server with a password but I don't see how to make the phone do http auth when getting its configs |
| 02:21.14 | ANTILOCAS | f*cksake it was because the ip was static one |
| 02:21.21 | ANTILOCAS | now i can enter |
| 02:21.40 | tessier | Not sure I want to rely on noindex and not being able to guess the mac address for security. If someone just browed the server with the configs they could get the registration password. No good. |
| 02:22.02 | Skyelar | ANTILOCAS: ok - under basic settings, change it from "NAT router" to "Bridge mode", then under advanced, enable "WAN side HTTP access" and "Reply to ICMP on WAN interface" |
| 02:22.30 | Skyelar | ANTILOCAS: save, and reboot it - you should then be able to plug the WAN interface into your switch and have the darn thing work as expected |
| 02:23.04 | awe6 | hmmhesays: If you are still here: It seems that I can dial an extension without being connected to my provider although Asterisk _tries_ the provider. If it times out, then it simply dials the extension. |
| 02:23.59 | awe6 | hmmhesays: Tested by simply disconnecting the network cable to the outside. |
| 02:26.13 | hmmhesays | i ee |
| 02:26.16 | hmmhesays | *i see |
| 02:26.25 | hmmhesays | well i suggest you start looking at your dialplan to figure out why |
| 02:30.47 | *** join/#asterisk Crashsys (n=kumba@office.crashsys.com) |
| 02:31.36 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 02:32.37 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 02:33.10 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 02:34.19 | *** join/#asterisk Doctor (n=mage@24.83.37.254) |
| 02:36.13 | tengulre | hi,all |
| 02:36.17 | ANTILOCAS | skyelar i cant enter Advance settings |
| 02:36.17 | tengulre | anybody here? |
| 02:36.20 | ANTILOCAS | it says access denied |
| 02:36.36 | ANTILOCAS | is it because is not pluged to the switch? |
| 02:37.14 | ANTILOCAS | wan port <-> switch |
| 02:37.59 | tengulre | how to encrypt SIP data?? |
| 02:38.39 | JT | check voip-info.org |
| 02:38.47 | JT | easiest is with an encrypted vpn/tunnel |
| 02:40.18 | tessier | I notice the Snom320 supports srtp |
| 02:40.19 | Druken | tengulre: define SIP data.... |
| 02:40.51 | tengulre | Druken, pls give me some tips, is it need SIP client support? |
| 02:41.02 | JT | of course |
| 02:41.11 | JT | unless you use a vpn |
| 02:42.36 | Druken | mmmmm, vpn's are fun.... |
| 02:42.48 | tengulre | ?... I can not still understand.. |
| 02:43.16 | JT | i suggest doing some research in that case |
| 02:43.25 | JT | of calling in a consultant who does understand |
| 02:43.28 | JT | s/of/or/ |
| 02:45.31 | tessier | hmm...my snom 320 is successfully downloading configs but I don't see the configs actually appearing in the phone settings. |
| 02:47.13 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 02:48.54 | Dovid | tessier:from what version to what version r u upgrading ? |
| 02:49.03 | *** join/#asterisk Here_And_There (n=oldfart2@pool-71-246-6-164.phlapa.fios.verizon.net) |
| 02:49.34 | tessier | Dovid: Not upgrading firmware. Just trying to be able to download config files via http from my server. |
| 02:49.47 | tessier | I am going to roll out around 30 phones and don't want to have to key all of the data in manually |
| 02:50.04 | Dovid | hmm. never did that. i only have a handfull |
| 02:50.19 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 02:50.48 | Dovid | r u running the most current firmware ? |
| 02:51.41 | tessier | Dovid: I don't know. I just bought the phone so it's probably something recent. |
| 02:52.14 | Dovid | tessier: not so. they ship with old firmware |
| 02:52.22 | Dovid | what version is on the phone ? |
| 02:52.32 | awannabe | are the snoms decent phones? |
| 02:52.39 | Dovid | depends for what |
| 02:52.55 | tessier | awannabe: I really like them |
| 02:53.01 | Dovid | in ways they are easier than polycom in wasy not. it deff. has features that others dont |
| 02:53.10 | tessier | And it runs Linux. :) |
| 02:53.15 | awannabe | ahh, intresting |
| 02:53.21 | awannabe | i like the blue screen on the 360, heh |
| 02:53.30 | tessier | They are the easiest to deal with so far. I've used Snom, Cisco, Grandstream. Haven't used Polycom yet. We're gonna stick with Snom. |
| 02:53.35 | awannabe | i want to get some and test |
| 02:54.04 | Dovid | tessier: polycom has a better feel to it and if u get the sidecar its the same as snom |
| 02:54.19 | tessier | How can I tell what firmware version is on the phone? |
| 02:54.22 | Dovid | the only issue with polycom that i saw is that u need the sidecar for BLF |
| 02:54.24 | awannabe | ive heard the polycom sidecards can only monitor like 7 extensions, not all of em |
| 02:54.27 | tessier | snom320-SIP 4.1 |
| 02:54.29 | tessier | Is that it? |
| 02:54.34 | Druken | anyone here know about 802.11g ? |
| 02:54.38 | awannabe | and the snoms support more lines then the polycoms do |
| 02:54.40 | Dovid | tessier: u need to upgrade |
| 02:54.50 | Dovid | Drunken: i know a lil |
| 02:54.57 | tessier | Bugger. I was hoping to avoid that. |
| 02:55.12 | Dovid | awannabe: not so, u can have lots of lines with the polycoms too |
| 02:55.20 | awannabe | intresting |
| 02:55.37 | Dovid | if u get the sidecar |
| 02:55.41 | Druken | well, i need to pick someone's brain who knows alot... hehe not a little |
| 02:55.56 | Dovid | i u can get like i think 30+ |
| 02:56.06 | awannabe | polycom is 24 max i thought |
| 02:56.11 | Druken | i want to pick someone's brain about blanketing a small city in it... |
| 02:56.16 | Dovid | Drunken: u can ask if i dont know i'll tell u |
| 02:56.57 | Dovid | awannabe: i never tried it, but in theory..... |
| 02:56.57 | awannabe | 48 lines max, 3 sidecards |
| 02:57.15 | awannabe | i read an article on voip-info.org and said 7 lines is all that person got to work |
| 02:57.18 | awannabe | who knows though |
| 02:57.22 | Dovid | awannabe: max 6 sip lines *for 601) |
| 02:57.29 | awannabe | yeah |
| 02:57.29 | Dovid | (for 601)* |
| 02:57.45 | Dovid | abnd then each sip line can have i think 5 calls |
| 02:58.05 | Dovid | my bad |
| 02:58.05 | awannabe | i have 2 501s ive been testing with |
| 02:58.06 | Dovid | 6 |
| 02:58.10 | Dovid | i did it with 6 |
| 02:58.21 | Dovid | i like thr 601 a lot more |
| 02:58.25 | awannabe | have u done the presensce keys? |
| 02:59.01 | Dovid | u mean BLF ? |
| 02:59.15 | awannabe | BLF? |
| 02:59.29 | Dovid | u do but only on the side car. for the phone itself u have to navigate with the screen which is a pain |
| 03:00.12 | awannabe | gotcha |
| 03:02.05 | awannabe | i cant stand the cisco's, but ill have to order the polycom 601s and look at em |
| 03:04.51 | *** join/#asterisk Here_And_There (n=Here_And@pool-71-246-6-164.phlapa.fios.verizon.net) |
| 03:04.52 | Crashsys | Polycom's provisioning rules... and once you figure the first phone out, they're a snap... |
| 03:05.39 | awannabe | yeah |
| 03:05.40 | Crashsys | Dont suppose anyone has any experience with Grandstreams do they? (I'm getting no dial tone from it) |
| 03:06.05 | awannabe | im buidling a web based config system right now, next item is putting the data into the polycom file |
| 03:06.46 | Crashsys | I'd love to help... but I know jack about programming... |
| 03:07.19 | Crashsys | I've got some polycom 301's (for starters)... prolly gonna get a few 601's for receptionist/sales... |
| 03:08.58 | awannabe | im still pretty new to *, so im learning as I go on everything |
| 03:09.24 | Crashsys | I'm just trying to figure out why this stupid grandstream looses dialtone as soon as it connects to the SIP Server... |
| 03:09.35 | Crashsys | like the sip server should be providing dialtone instead of it generating it's own... |
| 03:09.42 | tessier | Dovid: I followed the firmware upgrade instructions at http://www.snom.com/wiki/index.php/Automatic_Update_Service but it still says 4.1. It does not seem to have retrieved any new firmware. |
| 03:09.42 | awannabe | ive heard mixed thing about the grandstreams |
| 03:10.27 | Crashsys | I hate it so far... |
| 03:10.34 | Crashsys | they're cheap... |
| 03:10.39 | Crashsys | only real benefit... |
| 03:10.45 | awannabe | lol |
| 03:10.48 | tessier | Dovid: The instructions just say to set the update url. But shouldn't the update policy be changed from the default "Never update, load settings only"? I tried changing it to "ask for update" and it seems to not do anything differently. |
| 03:10.55 | tessier | Crashsys: Indeed. They are cheap. That's it. |
| 03:11.22 | Crashsys | I tried talking them into 301's, showed then atacomm's website so they would know how much they cost, then click on something else on a website, and these POS grandstream's pop up... |
| 03:11.32 | awannabe | lol |
| 03:11.41 | Crashsys | "Hey, that's only $60!"... I could just tell them that the phone doesn't generate dial tone :) |
| 03:12.03 | awannabe | there ya go! |
| 03:12.10 | file | you go for cheap... you get chap... |
| 03:12.21 | awannabe | ding ding! |
| 03:12.23 | Crashsys | for $60... it is what it is... i'm just glad they aren't making their business from phone's ringing... |
| 03:12.30 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 03:12.36 | Crashsys | they just want a smart answering machine... |
| 03:12.47 | Crashsys | SOHO... which is fine... |
| 03:12.49 | awannabe | do u guys know if * can be set to ring a group ph phones, but if one phone is busy, it will not ring that phone but still the rest? |
| 03:13.01 | Crashsys | awannabe: Queues |
| 03:13.11 | Newbie___ | hi all, when i call from POTs -> Voip -> POTs i get background static/noise. any ideas ? |
| 03:13.14 | Crashsys | add the extensions to a queue, and set the ringing method... |
| 03:13.18 | file | or call them all at the same time... |
| 03:13.35 | file | Dial(SIP/1&SIP/2&SIP/3) it'll call all 3, but if one is busy... it doesn't matter, the other 2 will keep ringing |
| 03:13.39 | Crashsys | file: if one phone picks up will the rest quit ringing tho? |
| 03:13.44 | file | yes |
| 03:13.57 | awannabe | Crashsys, hrmm, ok. now when ringing multiple phones, i can create something like a hunt group/phantom that will ring a bunch of phones when you dial a certain extension, right? |
| 03:14.45 | awannabe | file, yeah, thats what i was gonna do, just trying to not have the one phone ring (so that person doesnt have to hear it ring) |
| 03:15.18 | file | then it's not exactly busy :) |
| 03:15.30 | awannabe | true |
| 03:16.38 | awannabe | so i guess just ring em all, lol |
| 03:16.42 | Crashsys | awannabe: Yes, you can do it in dial plan... |
| 03:16.55 | Crashsys | if it's high volume a queue would prolly work better |
| 03:17.01 | awannabe | i guess u just make a new extension that dials multiple phones...right? |
| 03:17.11 | file | indeed, depends on the purpose of what you're trying to do |
| 03:17.19 | awannabe | a ques the same as setting up a ACD then? |
| 03:17.32 | Crashsys | If you've only got 2 pots lines... like for a home... then who cares, dial(all phones) would work... |
| 03:17.44 | Crashsys | You dont need ACD... |
| 03:17.57 | Crashsys | You can just add queue extensions like SIP/1001 |
| 03:18.14 | Crashsys | or you can do agents |
| 03:18.24 | awannabe | wiki time! |
| 03:19.18 | Crashsys | queue's and voicemail and agents was pretty straight-forward from looking in the .conf's... |
| 03:19.28 | Crashsys | the dialplan was a mystery for me when I started... |
| 03:19.38 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 03:19.46 | awannabe | i think a que is really overkill for this |
| 03:19.47 | file | it's a [TK]D-Fender |
| 03:19.51 | Crashsys | then I figured the best way to learn the dial plan was to delete the file that came with asterisk and start from a blank one :) |
| 03:19.54 | [TK]D-Fender | OMGZ!!! |
| 03:20.03 | file | it's bad luck to say good luck on opening night! |
| 03:21.02 | [TK]D-Fender | Crashsys : Good call.... |
| 03:21.12 | [TK]D-Fender | awannabe : For what? |
| 03:21.26 | awannabe | dialing multiple extensions |
| 03:21.49 | Crashsys | ok... put new firmware in this shitstream... here's to hoping I can get dialtone... |
| 03:21.50 | trelane | file, break a leg, or your head, or something entertaining |
| 03:22.02 | [TK]D-Fender | awannabe :Dial(SIP/1&SIP/2&Zap/1,30) |
| 03:22.30 | tessier | hmmm |
| 03:22.33 | awannabe | yeh, i was wanting somehting that would not ring a phone if someone was on it, so you have to use a que for that, but i think ill just have it ring everyone, lol |
| 03:23.05 | tessier | It seems I cannot connect to provisioning.snom.com. That could be why I can't update firmware. Can anyone else connect to that site? |
| 03:23.05 | Crashsys | awannabe: No, you can use the busy context in that extension's dial-plan... |
| 03:23.21 | Crashsys | when it comes back with busy... you can have it route to another extension... |
| 03:23.40 | awannabe | and still have it ring multiple extensions? |
| 03:24.10 | Crashsys | Yeah... if it tries to dial sip/1001, and it doesn't answer, or is busy, it'll come back to the dialplan to look for what to do next... |
| 03:24.19 | Crashsys | lemme find the wiki page on it (if I can) |
| 03:24.31 | awannabe | yeah, but that wont ring em all to start with, right? |
| 03:24.52 | Crashsys | You can use file's suggestion and do dial(ext1,ext2,ext3,etc) |
| 03:24.55 | file | that's provided you loop through the dialplan using chan_local |
| 03:25.08 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 03:25.15 | awannabe | i guess lots of stuff to test, heh |
| 03:25.33 | file | experiment... have fun |
| 03:25.50 | file | just don't set anything on fire please |
| 03:25.53 | awannabe | lol |
| 03:25.59 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:26.03 | file | blitzrage: !!! |
| 03:26.10 | blitzrage | file: !!! |
| 03:26.14 | blitzrage | wuz up homey? |
| 03:26.25 | file | I grabbed some pepsi, am a wee bit hyper |
| 03:26.33 | file | watching The Producers again and again and again! |
| 03:26.35 | file | you? |
| 03:26.36 | blitzrage | lol |
| 03:26.44 | blitzrage | trying to get a new website up... |
| 03:26.49 | tessier | ah-hah |
| 03:26.57 | tessier | The instructions on the snom website seem to be incorrect. How bizarre. |
| 03:27.00 | tessier | There si a |
| 03:27.09 | tessier | There is a "software update" link on the phones web interface |
| 03:27.26 | Crashsys | hmm... upgrading to a new shitstream firmware fixed the problem... joy... |
| 03:27.34 | awannabe | shitstreams lol |
| 03:28.01 | Crashsys | wonder what new problems i'm gonna have now... |
| 03:28.10 | file | Crashsys: spontaneous combustion |
| 03:28.15 | Crashsys | phone loosing registrationg every 5 seconds or when someone farts... |
| 03:28.29 | Crashsys | dood, seriously, the back of this thing is hot... |
| 03:28.34 | Crashsys | glad i'm not putting them in my office :) |
| 03:29.01 | tessier | ugh...I am going to have to manually update the firmware on all of these phones...this is the sort of thing mass-deployment is supposed to AVOID. :( |
| 03:29.07 | file | I'm surprised I didn't fry one of the video phones we had in Italy for the developer conference... I accidentally plugged it in with an adapter that only did American voltage |
| 03:29.32 | file | but the Grandstream just made a large spark, got a good adapter and it worked fine |
| 03:30.05 | *** join/#asterisk KaiHanari (n=Kai@stjhnf01-22-142163031131.nf.sympatico.ca) |
| 03:30.18 | awannabe | lol |
| 03:30.24 | KaiHanari | how do i connect to an IVR from an extension? |
| 03:30.32 | Strom_C | you dial it |
| 03:30.32 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 03:30.44 | file | KaiHanari: Newfoundland? neat |
| 03:30.50 | *** join/#asterisk sumasuma (n=sumkji@61.14.86.23) |
| 03:31.13 | sumasuma | I want to have symmetric nat using iptables |
| 03:31.28 | sumasuma | can anyone please help me how to have the same ? |
| 03:31.31 | KaiHanari | Strom_C, yea, but how do i dial it? im using freepbx/trixbox... and theres nothing there that indicates it gets an extension # |
| 03:31.40 | Crashsys | Hmmm... if i'm giving the phones static IP Address'... should I bother with SIP Registration? |
| 03:31.44 | Strom_C | KaiHanari: oh hell, please read the topic |
| 03:31.51 | file | Crashsys: same network? |
| 03:31.55 | Crashsys | yeah... |
| 03:31.56 | Crashsys | local lan |
| 03:32.01 | KaiHanari | bleh, |
| 03:32.04 | Crashsys | prolly set it all static... |
| 03:32.07 | KaiHanari | file, yea. |
| 03:32.10 | file | Crashsys: then you should be fine |
| 03:32.17 | Crashsys | not have to worry about it loosing registration :) |
| 03:33.57 | awannabe | extensions are 6 digits max, right? |
| 03:34.04 | file | awannabe: internally? ha no |
| 03:34.16 | awannabe | lesss? |
| 03:34.22 | file | 80 actually |
| 03:34.24 | Crashsys | I didn't know they had limitations other then programmatical ones... |
| 03:34.33 | awannabe | 80 digits long? |
| 03:34.40 | file | 80 characters long |
| 03:34.44 | awannabe | well i guess a 11 digit integer in mysql wont cut it, LOL |
| 03:34.53 | Crashsys | For sales, please dial extension 19471947391371823481892124 |
| 03:34.56 | Crashsys | lol |
| 03:34.57 | file | extensions don't have to be numbers |
| 03:35.15 | awannabe | i was gonna setup ring groups, and use like 6 digit extensions |
| 03:35.22 | trelane | if you are calling from a telemarketer please dial file's extension |
| 03:35.26 | awannabe | lol |
| 03:35.30 | file | pleasekthxno |
| 03:35.43 | *** join/#asterisk BlepsoaF (n=pbaker@ool-457805b1.dyn.optonline.net) |
| 03:35.46 | file | nobody calls me though, so it's all good |
| 03:36.02 | Newbie___ | hi all, when i call from POTs -> Voip -> POTs i get background static/noise. any ideas ? |
| 03:36.04 | Crashsys | I know... record that samuel L jackson snakes on a plane call... and have that played to telemarketers |
| 03:36.18 | BlepsoaF | Hello all, would anyone have an idea why I would be receiving " chan_zap.c:8963 pri_dchannel: PRI Error: We think we're the network, but they think they're the network, too. " all of a sudden? I'm using PRI_NET for connecting a nortel system |
| 03:36.37 | file | BlepsoaF: you might be in loopback, or you should be pri_cpe |
| 03:36.45 | Crashsys | blepsoaf: your set as master? |
| 03:36.52 | Strom_C | BlepsoaF: well, your nortel is either misconfigured, or you're plugging into the wrong port on the nortel |
| 03:37.05 | BlepsoaF | file: I ran a loop test in zttool, then it started |
| 03:37.09 | BlepsoaF | otherwise it was working fine before |
| 03:37.17 | Strom_C | BlepsoaF: unloop the isdn card |
| 03:37.22 | file | you need to get it out of loopback |
| 03:37.39 | file | Strom_C: will you join me at Der Waffle Haus? |
| 03:37.47 | Strom_C | definitely |
| 03:38.03 | awannabe | is there a wiki page on ques i cant find?/ |
| 03:38.17 | BlepsoaF | how do I get it out of loop back? |
| 03:38.41 | file | unfortunately I do not know |
| 03:38.54 | file | let me accidentally access technical support's database |
| 03:38.55 | Strom_C | select "loop" again in zttool perhaps? |
| 03:39.28 | BlepsoaF | tried that |
| 03:39.50 | Crashsys | You guys think a P3-450 with 128-megs of ram can handle 2-pots lines and 6 sip phones? (all u-law) |
| 03:40.05 | Strom_C | yes |
| 03:40.08 | file | Crashsys: should be ok |
| 03:40.20 | Crashsys | whoo hoo |
| 03:41.20 | file | what'cha going to use for PSTN side? |
| 03:41.35 | Crashsys | el cheapo X100p's |
| 03:41.41 | Strom_C | Crashsys: EW |
| 03:41.42 | file | Strom_C: Der Waffle Haus in Dallas! |
| 03:41.46 | trelane | Crashsys, eww |
| 03:41.47 | file | Crashsys: ...have fun |
| 03:41.50 | trelane | Crashsys, good luck |
| 03:41.51 | Crashsys | thanks |
| 03:41.54 | Strom_C | Crashsys: you'll be in echo hell |
| 03:41.56 | trelane | (he'll need it) |
| 03:42.15 | Strom_C | might I recommend a TDM02B? |
| 03:42.16 | awannabe | the digium analog cards are good right? for FXO ports |
| 03:42.22 | Strom_C | awannabe: yes |
| 03:42.22 | Crashsys | I'm taking bets on wether the calls I get will be in regards to the shitstreams catching on fire or the echo from the 100's :) |
| 03:42.23 | trelane | Strom_C, what do you mean (what do you mean), they really work fine (work fine), I can't tell a difference, can you (can you)? |
| 03:42.37 | BlepsoaF | so theres not setting I can put in place to remove the loop? |
| 03:42.50 | Strom_C | Crashsys: give up now if you're not going to put decent equipment into your installation |
| 03:42.51 | Strom_C | please |
| 03:42.54 | file | BlepsoaF: dunno, I'm not in support so I don't know this stuff... haven't done zaptel much |
| 03:43.05 | Strom_C | just hire a consultant who knows what he's doing instead |
| 03:43.26 | Strom_C | BlepsoaF: you could always just try unloading and reloading the drivers |
| 03:43.34 | trelane | BlepsoaF, pri_cpe not pri_net |
| 03:43.43 | trelane | Strom_C, I got this one :) |
| 03:43.50 | BlepsoaF | trelane: u have to use pri_net to connect to the nortel system |
| 03:43.58 | BlepsoaF | its using a patch cable |
| 03:44.13 | trelane | BlepsoaF, umm you're the DCE and they're the DCE = no data goes anywhere, switch it please. |
| 03:44.18 | Strom_C | trelane: he put the span in loopback mode |
| 03:44.30 | Strom_C | trelane: with asterisk running |
| 03:44.52 | trelane | Strom_C, it's a T1 signaling issue, both sides think they're the timing source. |
| 03:45.15 | Strom_C | trelane: I know that, and I think you missed the bit where he said it started when he put the span in loopback mode |
| 03:45.22 | trelane | eek |
| 03:45.26 | trelane | why'd he do that? |
| 03:45.32 | Strom_C | i don't know...ask him! |
| 03:45.32 | Strom_C | :) |
| 03:45.37 | trelane | Strom_C, do I have to? |
| 03:45.39 | BlepsoaF | I was playing around with zttool |
| 03:45.52 | trelane | BlepsoaF, take it out of loopback now please. |
| 03:46.05 | BlepsoaF | trelane: thats the problem |
| 03:46.06 | Strom_C | BlepsoaF is obviously the kind of person who presses things labeled "DO NOT PRESS" just to see what happens |
| 03:46.17 | BlepsoaF | trelane: i dont know who :S |
| 03:46.19 | BlepsoaF | how |
| 03:46.46 | trelane | zttool and click loopback |
| 03:46.58 | trelane | Strom_C, no, no! not the wed won! |
| 03:47.55 | BlepsoaF | trelane: I tried that, it didnt work |
| 03:48.16 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 03:48.22 | Strom_C | BlepsoaF: stop asterisk |
| 03:48.29 | Strom_C | then unload the zaptel modules |
| 03:48.33 | Strom_C | then reload the zaptel modules |
| 03:48.36 | Strom_C | then start asterisk |
| 03:48.54 | file | also run around in circles |
| 03:50.35 | BlepsoaF | Strom_C: I just tried that |
| 03:50.37 | BlepsoaF | and it didnt work |
| 03:50.40 | blitzrage | new website up finally |
| 03:50.47 | file | hold me hold me! echo cancel me! |
| 03:50.50 | file | blitzrage: yay |
| 03:51.01 | blitzrage | only took 3 hours of time... but like a year before I got around to it :) |
| 03:52.38 | Strom_C | BlepsoaF: ok...um...try restarting the system |
| 03:52.43 | BlepsoaF | tried that |
| 03:52.47 | BlepsoaF | trying to turn it off |
| 03:53.08 | Strom_C | what kind of nub just randomly loops back a T1? |
| 03:53.12 | Strom_C | sigh |
| 03:53.52 | BlepsoaF | im an idiot |
| 03:54.10 | file | it's okay |
| 03:55.27 | BlepsoaF | powering off the system didnt work either |
| 03:55.39 | Strom_C | BlepsoaF: this isn't a production system yet, I hope |
| 03:55.43 | BlepsoaF | so is this effecting the card more or the actual provider |
| 03:55.43 | Crashsys | I'd be happy if my CAS T1 would quit getting random red alarms... |
| 03:57.50 | *** join/#asterisk _Vile (n=vile@90.b160.bendtel.net) |
| 03:58.38 | BlepsoaF | it seems to be only effecting the nortel system |
| 03:58.47 | BlepsoaF | because I can dial into the asterisk system directly |
| 03:58.59 | BlepsoaF | but anything going to the nortel system gets a fast busy |
| 03:59.24 | Strom_C | BlepsoaF: because you only looped back a single span |
| 03:59.25 | Strom_C | duh |
| 03:59.40 | BlepsoaF | so there isnt a way to fix it? |
| 03:59.44 | Strom_C | there should be |
| 03:59.51 | Strom_C | i just dont have a T1 card here I can test with |
| 03:59.57 | BlepsoaF | I wonder if it requires rebooting the nortel system |
| 04:00.08 | Strom_C | what happens if you unplug the cable |
| 04:00.11 | Strom_C | do the errors stop? |
| 04:00.21 | BlepsoaF | dunno, I'm going to have to drive down to my office now |
| 04:00.30 | Strom_C | you're not even at the premises??? |
| 04:00.37 | BlepsoaF | its only 10 min away |
| 04:01.42 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 04:02.19 | BlepsoaF | going to drive down there, thanks guys |
| 04:03.20 | Strom_C | i didnt realize this was #emo-asterisk |
| 04:04.53 | JT | Strom_C: it is? |
| 04:05.05 | Strom_C | according to blepsoaf's quit message, it is |
| 04:05.23 | Strom_C | :) |
| 04:06.34 | JT | ah |
| 04:06.38 | JT | didn't notice that |
| 04:15.11 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 04:19.03 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 04:19.09 | awannabe | is there a base polycom XML file somewhere that someone knows of? |
| 04:23.49 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 04:26.28 | Strom_C | hey, here's a silly question |
| 04:26.56 | Strom_C | how do you configure a sip station to show MWI for a voicemail box on a completely different PBX? |
| 04:27.08 | file | yes. |
| 04:27.22 | [TK]D-Fender | awannabe : there is a complete sample with the firmware pack |
| 04:28.41 | _Vile | FILE |
| 04:28.46 | file | VILE |
| 04:28.50 | _Vile | hi |
| 04:29.14 | awannabe | [TK]D-Fender, nice, ill check that out |
| 04:29.32 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 04:30.20 | *** join/#asterisk Rahail (n=dd@ip66-104-51-162.z51-104-66.customer.algx.net) |
| 04:30.25 | *** join/#asterisk num000 (n=numerobi@e177180225.adsl.alicedsl.de) |
| 04:31.12 | num000 | i'm running asterisk on openwrt and unfortunately asterisk does not load the modules on start, has anyone got an idea where this could be caused from? |
| 04:31.26 | awannabe | isnt the polycom firmware uber hard to get? |
| 04:31.39 | awannabe | i thought i had a copy, i gotta see, heh |
| 04:32.50 | *** join/#asterisk ma_dzen (n=ma_dzen@217.66.17.141) |
| 04:33.57 | awannabe | ugg, i need a login |
| 04:34.13 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 04:34.28 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 04:38.12 | num000 | has anyone got asterisk on openwrt running? |
| 04:39.53 | awannabe | not a clue |
| 04:40.27 | docelmo | why in the hell would you wanna? |
| 04:40.35 | Strom_C | i did it once just for kicks |
| 04:42.39 | *** join/#asterisk cbrake (n=cbrake@oh-69-34-21-229.sta.embarqhsd.net) |
| 04:43.12 | cbrake | how do I get UNREACHABLE notices to show up in the logs? |
| 04:43.25 | *** join/#asterisk Frogdude (i=Frogdude@c-24-16-72-159.hsd1.mn.comcast.net) |
| 04:43.43 | awannabe | wow the polycom config is a cluster, we ever heard of friggin carriage returns? |
| 04:44.13 | [TK]D-Fender | awannabe : CAll your reseller, or check out http://www.freedomphones.net/polycom/files/ |
| 04:44.27 | [TK]D-Fender | awannabe : Try using a non-windows standard editor... |
| 04:44.38 | awannabe | i would think vim would be good |
| 04:44.45 | [TK]D-Fender | awannabe : One that supports LF terminated lines. |
| 04:44.50 | [TK]D-Fender | awannabe : Should be. |
| 04:45.08 | awannabe | phone.cfg is all on one line, heh |
| 04:45.52 | *** join/#asterisk Tebi_ (n=rantis@gw.aller.fi) |
| 04:47.52 | [TK]D-Fender | <PROTECTED> |
| 04:48.25 | awannabe | wow, sip.cfg is friggin long |
| 04:49.19 | tessier | Yeay! |
| 04:49.31 | tessier | I finally got the firmware upgraded on this Snom 320 and the auto-configuration works |
| 04:49.38 | awannabe | haha nice |
| 04:49.44 | tessier | It sucks that the first thing you have to do when you buy a new piece of hardware these days is to upgrade the firmware. |
| 04:49.55 | ANTILOCAS | i cant access Advance Settings in my HandyTone 486 ATA web page configuration |
| 04:50.04 | ANTILOCAS | it says access denied |
| 04:50.05 | KaiHanari | how do i forward a call from a softphone? |
| 04:50.17 | ANTILOCAS | not even show the login page |
| 04:51.39 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 04:51.46 | *** part/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 04:51.55 | *** join/#asterisk ComputerWarm (n=dan@209.29.157.205) |
| 04:52.40 | ComputerWarm | hello anyone know of a way to get ztdummy to work with out the us hsi or whatever it is called on kernel 2.4 |
| 04:52.48 | tessier | Does MS Netmeeting do SIP? Or just H323? |
| 04:53.53 | ComputerWarm | tessier i believe just h323 |
| 04:54.01 | Strom_C | ComputerWarm: nope. on 2.4, you need a UHCI USB controller |
| 04:54.07 | tessier | ComputerWarm: Yeah, I think you are right. |
| 04:54.13 | JT | netmeeting is deprecated |
| 04:54.26 | ComputerWarm | Strom_C my system shows it but not as used when i do lmosed |
| 04:54.31 | tessier | JT: In favor of? |
| 04:54.38 | ComputerWarm | or whatever the command is |
| 04:54.38 | JT | ms messenger |
| 04:54.42 | JT | where they moved to sip |
| 04:54.50 | tessier | ah |
| 04:56.26 | ComputerWarm | Strom_C do you know of anyway to get 3 way calling working through asterisk. my phone doesn`t support it. with out the conference rooms |
| 04:56.40 | Strom_C | what kind of phone? |
| 04:56.52 | ComputerWarm | Budget done 101 |
| 04:57.06 | ComputerWarm | it has conference button but i can never get it to do anything |
| 04:57.16 | Strom_C | maybe you have it misconfigured |
| 04:57.21 | tessier | Can someone with a SIP phone do me a favor and call: sip://102@75.5.99.100 ? |
| 04:57.24 | tessier | I want to see if that works. |
| 04:57.28 | ComputerWarm | nothing in the config for it |
| 04:57.29 | Strom_C | ive never used a budgetone. I bought real phones instead ;) |
| 04:57.53 | ComputerWarm | Strom_C ya i am going to be but i need to figure this out right away |
| 04:58.30 | ComputerWarm | Strom_C do u know anyway to get asterisk to support three way calling |
| 04:59.07 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 04:59.47 | Strom_C | ComputerWarm: the SIP handset has to implement it |
| 05:00.01 | Strom_C | alternatively, zaptel supports it |
| 05:00.02 | ComputerWarm | Strom_C k so there is no way to get asterisk to |
| 05:00.06 | Strom_C | but asterisk itself does not |
| 05:00.14 | Strom_C | its a channel-dependent feature |
| 05:01.45 | ComputerWarm | ok so sip to sip it won`t work unless the phone supports it.. ok thanks |
| 05:01.57 | Strom_C | your phone should support it |
| 05:02.00 | Strom_C | read the manual |
| 05:02.28 | ComputerWarm | ya i have according to everything i have read one of the upgrades will but as of now it doesn`t |
| 05:08.03 | ComputerWarm | Strom_C do you know any pbx software that does support 3 way |
| 05:08.26 | Strom_C | ComputerWarm: seriously, dude. Get it working in your phone, or get a real phone. |
| 05:08.36 | Strom_C | or get a zaptel card |
| 05:09.07 | JT | won't a meetme conference do it? |
| 05:09.11 | ComputerWarm | Strom_C zaptel card is a issue its a hosted server i have a zaptel card here so really it has nothing to do with the costs.. |
| 05:09.30 | Strom_C | JT: it's a kludge to do threeway with meetme |
| 05:09.43 | Strom_C | a HUGE GIANT ZIT-COVERED KLUDGE |
| 05:09.52 | Strom_C | ...covered in pralines |
| 05:09.58 | JT | howso? |
| 05:10.05 | JT | it's a 3 person conference |
| 05:10.45 | [TK]D-Fender | ComputerWarm : You just don't know how.... It does support 3-way call on the phone itself. Re0read the instructions. |
| 05:11.03 | Strom_C | it is not, strictly speaking, the same thing as "three-way calling", and it doesn't behave at all the same way |
| 05:11.10 | ComputerWarm | [TK]D-Fender GS does? |
| 05:11.28 | JT | explain |
| 05:11.29 | JT | ? |
| 05:11.33 | ANTILOCAS | I have my ATA LAN port conected to the PC, when i login to the webpage cofiguration I cant enter Advance SEttings and also i dont have the 2 ports Tags that appears on the manual |
| 05:11.40 | Strom_C | JT: have you ever used three-way calling on a regular phone? |
| 05:11.45 | JT | yeah |
| 05:11.49 | ANTILOCAS | do i need to reset or download something_ |
| 05:11.49 | ANTILOCAS | ? |
| 05:11.51 | Strom_C | so you know how simple it is to operate |
| 05:11.56 | JT | sure |
| 05:12.01 | ANTILOCAS | the ATA is handytone 496 |
| 05:12.02 | Strom_C | dial one party, flash, dial a second party, flash, talk to all three |
| 05:12.10 | Strom_C | flash to drop second party off the connection |
| 05:12.13 | [TK]D-Fender | ComputerWarm : Yes. its in there somewhere... |
| 05:12.15 | JT | yep |
| 05:12.37 | Strom_C | JT: I would like to see you figure out how to do that with meetme and no hookflashing :) |
| 05:12.39 | ComputerWarm | [TK]D-Fender ok thanks i thought i read everything |
| 05:13.10 | JT | hookflashing is pretty much an analogue only feature though, isn't it? |
| 05:13.31 | [TK]D-Fender | JT : Some stupid phones use that to enable SIP features still... |
| 05:13.51 | Strom_C | yes, but without doing threeway in the phone on a digital set, figure out how you'd signal the PBX that you want a new call |
| 05:13.53 | [TK]D-Fender | So warning to all of you out there : STOP BUYING THE ^&%#$ GRANDSUCK CRAP! |
| 05:14.06 | JT | tap the hpook and it will execute a sip feature? |
| 05:14.30 | Strom_C | [TK]D-Fender: let's buy grandstream and liquidate them ;) |
| 05:14.31 | JT | Strom_C: dtmf or a voip phone button i'm guessing |
| 05:14.36 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 05:14.38 | [TK]D-Fender | JT : this is where I get to putt a Qwell/RoyK |
| 05:14.39 | ComputerWarm | [TK]D-Fender no kidding... i already have a cisco ordered |
| 05:14.42 | [TK]D-Fender | ~rtfm |
| 05:14.48 | jbot | somebody said rtfm was Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 05:15.03 | JT | ? |
| 05:15.21 | [TK]D-Fender | ComputerWarm : Cisco = overpriced. Polycom is best $/value choice in Northa America. Snom & Linksys are mostly decent |
| 05:15.49 | ComputerWarm | ya i know its over priced it was recommended from this channel |
| 05:16.01 | [TK]D-Fender | ComputerWarm : But a distant runner up in my eyes |
| 05:16.10 | [TK]D-Fender | ComputerWarm : Oh well.... |
| 05:16.12 | JT | without actions analoguous to a hookflash on digital sets, there is no clear equivalent for executing a three way call |
| 05:16.28 | Qwell | [TK]D-Fender: putt me? |
| 05:16.47 | [TK]D-Fender | JT : And what phones have you used before? Vast majority have hard/soft buttons with e big shiny word called "CONFERENCE" on it... |
| 05:16.49 | KaiHanari | how do i forward a call from a softphone to another extension? |
| 05:16.55 | [TK]D-Fender | Qwell : zomg! ;) |
| 05:17.03 | [TK]D-Fender | pull* |
| 05:17.12 | [TK]D-Fender | lol |
| 05:17.13 | JT | [TK]D-Fender: but the question is how to do it serverside :) |
| 05:17.18 | Qwell | please don't pull me - that's rude |
| 05:17.41 | [TK]D-Fender | JT : you DON'T. Time for you to get an impression . its the PHONE'S JOB. |
| 05:17.43 | Qwell | Strom_C: You can ;) |
| 05:17.50 | Strom_C | hehe |
| 05:17.51 | Strom_C | <3 |
| 05:18.18 | [TK]D-Fender | KaiHanari : check the softphone's manual. |
| 05:18.26 | JT | [TK]D-Fender: hrm |
| 05:18.33 | *** join/#asterisk kumamoto (n=eryco@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 05:18.41 | Jason99 | I just downloaded the latest trunk for asterisk-addon.. It says it depends on mysqlclient... What exactly is that? |
| 05:18.53 | Strom_C | the mysql client |
| 05:18.54 | Strom_C | duh |
| 05:18.56 | tzafrir | Jason99, what distro? |
| 05:18.56 | JT | analogue extensions do it exchange side... but yeah, we agree there's no obvious way to do it serverside with voip |
| 05:19.11 | KaiHanari | [TK]D-Fender, the button isnt enabled |
| 05:19.11 | JT | s/voip/asterisk/ |
| 05:19.13 | Strom_C | JT: no, they don't do it exchange side...they do it line module side |
| 05:19.17 | Jason99 | well.. I have mysql-5.0.22 installed |
| 05:19.24 | Jason99 | Distro is Slackware |
| 05:19.39 | [TK]D-Fender | JT : Ask yourself where the dialtone is coming from... that tells you what is responisble for controlling calls. |
| 05:19.42 | *** join/#asterisk mcnobody (n=laaksola@laaksola.net) |
| 05:19.48 | JT | Strom_C: in the line card? is that not part of the exchange? |
| 05:19.50 | JT | [TK]D-Fender: heh ok |
| 05:20.00 | Strom_C | it's peripheral to the central switching module |
| 05:20.19 | JT | is that always the case? |
| 05:20.21 | Jason99 | tzafrir: Do you know what I need to get ? |
| 05:20.24 | Strom_C | I believe so |
| 05:20.32 | [TK]D-Fender | JT : So what exactly is providing tone for this call of yours? |
| 05:21.08 | JT | you guys seem to be getting pretty defensive over this issue |
| 05:21.50 | [TK]D-Fender | JT : Just leting a little of thr frustration show on something that didn't seem to be sinking in too well. |
| 05:22.05 | [TK]D-Fender | JT : So again, what kind of equipment are you using? |
| 05:22.43 | JT | i understand that normally this happens in the voip terminal |
| 05:22.48 | JT | i'm just exploring possibilities |
| 05:23.00 | JT | and it was ComputerWarm that originally was interested in doing 3 way calling it |
| 05:23.07 | JT | i'm just curious about the matter |
| 05:23.29 | JT | i'm setting up a T100P card + zhone channel bank @ home |
| 05:23.52 | ComputerWarm | sorry didn`t mean to start a big debat, just frustrated that someone told me asterisk could support 3 way calling on the server side and it can`t.. oh well |
| 05:24.08 | [TK]D-Fender | JT : Ah well your case is somewhat different. Zaptel IS in charge for which I do believe the standard hook-flash method IS required. |
| 05:24.36 | JT | ah ok |
| 05:24.47 | JT | zaptel supports it? |
| 05:24.50 | Strom_C | ComputerWarm: whoever told you that was wrong :) |
| 05:24.55 | [TK]D-Fender | JT : But on any gateway involving SIP its the ATA/phone/whatever itself that is resposible as * is talking to the SIP devince, not the phone attached to it. |
| 05:25.06 | ComputerWarm | Strom_C been getting alot of that from this channel. |
| 05:25.11 | JT | [TK]D-Fender: ok fair enough |
| 05:25.28 | ComputerWarm | :-/ oh well time to find something that can handle it |
| 05:25.32 | [TK]D-Fender | JT : at which point the device must have its own way to enact such options. This is typically first signalled through hook-flash on the normal popular ATA's |
| 05:25.54 | JT | was wondering why my voip provider did client side conferences, guess that's why it's so |
| 05:26.02 | JT | would prefer a server side option |
| 05:26.13 | JT | especially with the pstn and my limited bandwidth home connection |
| 05:26.25 | JT | i guess something closer to a conference server would be required |
| 05:26.25 | [TK]D-Fender | Strom_C : Actually on Zap channels * DOES technically do it "server-side" (sot of?) with "threewaycalling=yes" in zapata.conf no? |
| 05:26.58 | ComputerWarm | [TK]D-Fender ya but with that you would be able to place a 3 way sip call server side |
| 05:27.27 | Strom_C | it's handled in zaptel; so technically it's still outside of the asterisk core |
| 05:28.01 | JT | do hook flashes carry over pcm? |
| 05:28.07 | JT | or tdm i should say |
| 05:28.18 | [TK]D-Fender | ComputerWarm : No. What I mentioned is only on a Zap Channel direct on * and has nothing to do with SIP. SIP has no way of telling the server side to conference so the only way they could do it is to listen to DTMF which is INCREDIBLY kludgy at BEST assuming anyone is crazy enoughto try and implement it. |
| 05:28.27 | Strom_C | JT: what do you mean by that? |
| 05:28.34 | JT | with a channel bank |
| 05:28.39 | [TK]D-Fender | ComputerWarm : SIP does not support passing hook-flash as a parameter. Period. |
| 05:28.49 | JT | you don't have an analogue link firectly to asterisk |
| 05:28.50 | [TK]D-Fender | JT see above |
| 05:28.51 | JT | directly |
| 05:29.03 | Strom_C | JT: yes, you signal supervision with bit-robbed CAS |
| 05:29.20 | JT | what about ccs? |
| 05:29.33 | Strom_C | over a T1-channel bank link? |
| 05:29.35 | Strom_C | uh |
| 05:29.41 | JT | yeah |
| 05:29.41 | Strom_C | do you even know how CCS works? :) |
| 05:29.42 | ComputerWarm | [TK]D-Fender ya i tried the flash hook thing yesterday never even came close to doing it. i am wondering if gnugk will do it, because i actually need this on h323. |
| 05:29.57 | JT | yes, one channel is reserved for a D channel |
| 05:29.58 | [TK]D-Fender | ok, well it's check-out time here, I'm off.... |
| 05:30.17 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 05:30.21 | [TK]D-Fender | ComputerWarm : Virtually no chance in hell. Good luck. |
| 05:30.41 | ComputerWarm | [TK]D-Fender shit didn`t want to hear that :-/ |
| 05:30.44 | Strom_C | JT: if its a channel bank, its purpose is to break a T1 out into 24 channels |
| 05:30.49 | [TK]D-Fender | ComputerWarm : (sorry to rain on your parade, but be prepared to spend your time on something else more productive) |
| 05:30.52 | Strom_C | ISDN D-channel signaling is completely different |
| 05:31.04 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 05:31.09 | JT | hrm |
| 05:31.28 | [TK]D-Fender | ComputerWarm : You may ahve a chance with MGCP as its a "dumb" protocol that DOES have the server do everything for it. |
| 05:31.35 | [TK]D-Fender | *MAY* |
| 05:31.39 | Strom_C | "hookflash" doesn't exist in ISDN |
| 05:31.41 | JT | i haven't got it hooked up yet, but won't bit robbing turn it into 7bits quantification, reducing audio quality? |
| 05:31.48 | [TK]D-Fender | anyways I'm off, good luck everyone. |
| 05:31.59 | ComputerWarm | [TK]D-Fender thanks |
| 05:32.05 | Strom_C | JT: no, it only robs one bit every six frames or something, and even then only the least significant bit |
| 05:32.06 | JT | quantitisation, even |
| 05:32.17 | JT | hrm |
| 05:32.28 | JT | so it's not recommended to run channel banks in CCS? |
| 05:33.11 | Strom_C | why would you need to? |
| 05:33.41 | Strom_C | for analog endpoints, there's no need at all |
| 05:33.58 | JT | if you wanted to sacrifice one channel so there's no bit robbing... i'm sure it'd also be useful if you were doing something insane like running faxes over it |
| 05:34.17 | Strom_C | fax is designed to work with bit robbing |
| 05:34.24 | JT | ah ok |
| 05:34.32 | Strom_C | you find enough CAS in the PSTN itself that fax has to work with it |
| 05:35.00 | JT | hrm ok |
| 05:35.01 | ComputerWarm | hrm |
| 05:35.11 | ComputerWarm | asterisk supports assisted call transfer right? |
| 05:35.23 | JT | so 99% of the time, people get their T1 channel banks to talk CAS to asterisk, right? |
| 05:35.24 | ComputerWarm | where you can transfer a call and stay on the line? |
| 05:35.30 | Strom_C | JT: YES |
| 05:35.36 | JT | yeah ok |
| 05:35.41 | Strom_C | ComputerWarm: that's called attended transfer |
| 05:35.44 | Strom_C | and yes, you can do it in asterisk |
| 05:35.55 | JT | was wondering if there was a good reason why people don't use CCS :) |
| 05:36.14 | Strom_C | JT: for analog handsets, there's no point |
| 05:36.25 | JT | fair enough |
| 05:36.25 | Strom_C | everything has to go inband to the analog set anyway |
| 05:36.27 | ComputerWarm | ok so call 1 calls in and i can transfer them and stay on the like to station 2 and talk until i han up right? |
| 05:36.34 | JT | true |
| 05:36.38 | Strom_C | ComputerWarm: YES!!!! |
| 05:36.58 | Strom_C | it must be earwax night in #asterisk |
| 05:37.06 | JT | ok, just to annoy you one little bit more, hookflashes do go over CAS, Strom_C? :) |
| 05:37.11 | JT | hehe |
| 05:37.14 | ComputerWarm | Strom_C for that would i need a zaptel device.. |
| 05:37.16 | JT | take a poison of your choice :) |
| 05:37.31 | ComputerWarm | Strom_C sorry i just needed to clearify that didn`t mean to make you repeat yourself... sorry |
| 05:37.47 | Strom_C | ComputerWarm: no...assisted transfer is handled just like three way calling: IN THE GOD DAMNED DEVICE :) |
| 05:38.38 | JT | hey, i can retalliate with a Scitec multiplexer, so i wouldn't do that if i was you... |
| 05:39.36 | Strom_C | and I can retaliate with an entire 1A2 frame |
| 05:39.38 | Strom_C | so there |
| 05:39.39 | Strom_C | :) |
| 05:40.30 | JT | i don't know what that is |
| 05:40.42 | JT | but i hope my E1 satellite earth station modem is heavier |
| 05:41.11 | Strom_C | JT: it takes up an entire wall |
| 05:41.38 | JT | hard to throw |
| 05:41.59 | Jason99 | The lastest trunk of add-on requires mysqlclient. I have mysql installed but it still wants mysql-client.. Does anyone know where to get that? |
| 05:43.55 | Juggie | sigh |
| 05:43.59 | Juggie | mysqlclient-devel |
| 05:44.01 | Juggie | next. |
| 05:44.37 | Jason99 | Tanks Juggie |
| 05:44.40 | Jason99 | Thanks* |
| 05:44.52 | *** join/#asterisk num000 (n=numerobi@e177180225.adsl.alicedsl.de) |
| 05:46.58 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 05:47.27 | Dovid | i am having echo issues |
| 05:48.22 | Dovid | i have the sangoma a200 with echo cancel if i have echocancel=no in zapata.conf then both zap1 and zap2 work except there is terrible echo. if i turn on echo cancelation then zapt2 starts giving one way audio |
| 05:53.01 | Strom_C | balls |
| 05:53.16 | JT | heh |
| 05:53.22 | fafnir | Strom_C! |
| 05:53.34 | Strom_C | que |
| 05:53.53 | fafnir | youre needed |
| 05:54.12 | fafnir | Nevermind |
| 05:54.20 | Strom_C | uh |
| 05:54.22 | Strom_C | ok? |
| 05:55.00 | ANTILOCAS | strom thanks for yer help |
| 05:55.41 | EyeCue | strim is the man |
| 05:55.44 | EyeCue | -i+o |
| 05:56.06 | tzafrir | Dovid, hardware echo can? |
| 05:57.18 | tzafrir | Dovid, one voodoo trick: try setting echo training to a really low value. I believe something lower than 10 should do the trick. Please let me know if this has helped you. |
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| 05:59.26 | JT | so yeah anyway, about 3 way calling and hook flashes and CCS... |
| 05:59.27 | JT | :D |
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| 06:11.05 | Newbie___ | hi all, when i call from POTs -> Voip -> POTs i get background static/noise. any ideas ? |
| 06:11.22 | Strom_C | Newbie___: duh |
| 06:11.36 | Strom_C | you're getting line noise |
| 06:11.42 | Strom_C | that's what happens with POTS |
| 06:12.52 | Strom_C | :) |
| 06:13.41 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
| 06:14.56 | Newbie___ | Strom_C: is there a way to reduce the noise? |
| 06:15.04 | *** join/#asterisk ProJektD (n=projekt_@216-99-199-104.cust.aracnet.com) |
| 06:15.11 | Strom_C | dexcribe the exact call path |
| 06:15.30 | Dovid | tzafrir: where do i set it below 10 ? |
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| 06:15.55 | Strom_C | gah |
| 06:15.58 | Strom_C | describe |
| 06:16.02 | Strom_C | not dexcribe |
| 06:16.03 | Dovid | tzafrir: can i PM ? |
| 06:16.09 | Newbie___ | Strom_C: caller call in to T1, calls are transfer to voipbuster using sip protocol |
| 06:16.25 | Strom_C | well no wonder it sounds like hell - you have voipbuster in there |
| 06:16.40 | Dovid | lol |
| 06:16.42 | Newbie___ | Strom_C: meaning? |
| 06:16.46 | Strom_C | IIRC, voipbuster restrics to really bad low-bitrate codecs |
| 06:16.55 | Dovid | tzafrir: ? |
| 06:16.56 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 06:17.04 | Newbie___ | ?? |
| 06:17.33 | Dovid | Newbie___: Meaning that they arent the best provider ;) |
| 06:18.12 | littleball | hello, who can recommend an hosting center which has E1/T1 voip provider there? |
| 06:18.28 | Newbie___ | Dovid: ok, i use it beacause of the free calls. :) well, i guess there is no such thing as free and good |
| 06:18.57 | littleball | Strom_C, any recommendation? |
| 06:19.14 | Strom_C | littleball: I have no clue what you mean by "T1 voip provider" |
| 06:19.35 | littleball | voip providers which provide E1 connectivities in hosting center |
| 06:19.53 | Strom_C | uh |
| 06:19.53 | Strom_C | yeah |
| 06:19.54 | littleball | my system /asterisk can connect to voip provide over E1/T1 |
| 06:20.08 | Strom_C | sure |
| 06:20.11 | JT | why not connect over voip? |
| 06:20.31 | littleball | I need DDI numbers and super good voice quality. price is negotiable |
| 06:20.34 | Strom_C | because, oh i dont know, you'd make better use of availble bandwidth by using a PRI instead? |
| 06:20.43 | littleball | PRI yes |
| 06:20.44 | Strom_C | it's called DID, littleball |
| 06:20.48 | Strom_C | PRI is not voip |
| 06:20.51 | littleball | DID. |
| 06:20.56 | JT | no ip |
| 06:21.23 | littleball | because i need DID numbers, so i think i must use PRI, right? SIP ok? |
| 06:21.32 | JT | argh |
| 06:21.36 | JT | you're making little sense |
| 06:21.42 | JT | so you want an E1/T1 |
| 06:21.45 | JT | not a voip provider |
| 06:21.50 | Strom_C | littleball: hire a consultant. |
| 06:21.59 | Strom_C | you obviously have no clue what you're doing. |
| 06:22.08 | littleball | Strom_C, i am familiar with such things. just looking for providers |
| 06:22.20 | Strom_C | littleball: you dont even know the difference between PRI and SIP |
| 06:22.25 | JT | well you don't seem to be differentiating between sip and pri |
| 06:22.36 | littleball | actually, i have 8 E1 line in Singapore. but i want better quality |
| 06:22.52 | JT | better quality |
| 06:23.03 | JT | won't get much better than E1 |
| 06:23.15 | Strom_C | are they channelized voice E1s or are they data E1s? |
| 06:23.18 | JT | i assume you aren't delivering the E1s over voip or something |
| 06:23.22 | littleball | voice E1 |
| 06:24.00 | Strom_C | what kind of voice quality do you want, exactly? compact disc audio? |
| 06:24.06 | littleball | over cable |
| 06:24.21 | *** part/#asterisk ProJektD (n=projekt_@216-99-199-104.cust.aracnet.com) |
| 06:24.28 | littleball | because i have a fantastic application. but need super good quality of route |
| 06:24.51 | littleball | currently, my system is hosted in Equinix. (any idea of this company)? |
| 06:25.16 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 06:25.42 | littleball | Equinix is good. but it is too expensive. so, i am looking for voip provider, which has its own hosting rack for rent. |
| 06:26.04 | littleball | so, i can connect to this voip provider switch over E1 directly. |
| 06:26.11 | Strom_C | sigh |
| 06:26.15 | Strom_C | you just dont get it |
| 06:26.20 | littleball | i only focus on application and no need to worry about connectivities |
| 06:26.59 | JT | then you should focus on hiring someone who worries about connectivity |
| 06:27.25 | littleball | JT, if i can connect to a provider who can provide good quality, why need hire a person ? |
| 06:27.49 | littleball | and i don't mind buy the voice at a bit high price |
| 06:28.12 | JT | because we have been trying to explain the issues to you, and you don't seem to be grasping them, and then tell us you don't worry about the connectivity |
| 06:28.36 | ANTILOCAS | My Handytone ATA has 2 fxo port, does that mean i need to different phone lines? |
| 06:28.39 | littleball | JT, explain to me why need to hire a person? |
| 06:28.54 | JT | littleball: i just did |
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| 06:30.07 | littleball | the reason i post the quastion here is because i dont want to hire a person to handle the connectivies, but looking for a provider who can provide A-Z voice with high quality. My system only need to connect their switch over E1/PRI cable |
| 06:30.34 | ANTILOCAS | little and what is BRI? |
| 06:30.44 | Strom_C | littleball: go read a telecom book |
| 06:30.46 | Dovid | do i need to install ztmonitor seperatley ? |
| 06:30.47 | JT | there are two types of telephony circuits generally, littleball. ones that are delivered over an IP nework - which is voip. others which connect via TDM such as E1/PRI or old analogue POTS, traditional telcos |
| 06:30.59 | littleball | TDM |
| 06:31.06 | Dovid | i did ztmonitor 3 -v and i get an error that there is no such program |
| 06:31.07 | ANTILOCAS | TDM is ISDN right? |
| 06:31.12 | JT | littleball: so you are after a telco |
| 06:31.24 | JT | ANTILOCAS: ISDN/T1 CAS, etc |
| 06:31.31 | JT | time division multiplexing |
| 06:31.32 | littleball | JT, i am using digium card TE411P card |
| 06:31.52 | *** join/#asterisk Bambr (n=Bambr@213-35-236-25-dsl.end.estpak.ee) |
| 06:32.09 | ANTILOCAS | arent those cards for analog lines? |
| 06:32.25 | Strom_C | ANTILOCAS: no, those are cards for T1 lines |
| 06:32.29 | JT | no, it's a quad T1 |
| 06:32.29 | Strom_C | all of you: |
| 06:32.32 | Strom_C | ~docs |
| 06:32.33 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 06:32.37 | Strom_C | ~thebook |
| 06:32.39 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:32.40 | littleball | i say voip providers because most of providers should provide both ip and tdm , right? |
| 06:32.44 | Strom_C | READ ALL OF THAT |
| 06:32.50 | Strom_C | GO NOW |
| 06:32.55 | JT | littleball: no, most providers provider one or the other |
| 06:33.40 | JT | Strom_C: even me? :( |
| 06:33.48 | ANTILOCAS | haha |
| 06:33.54 | Strom_C | hey, I read it :) |
| 06:33.59 | ANTILOCAS | i need that Book in Spanish |
| 06:34.04 | JT | heh, yeah i refer to it |
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| 06:34.15 | Dovid | Strom_C: have u worked with the zangoma a200 ? |
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| 06:34.18 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 06:35.00 | ANTILOCAS | T1 only in dreams here |
| 06:35.17 | Strom_C | Dovid: nope. I work with digium hardware. |
| 06:35.43 | *** join/#asterisk mredz (n=mail@ppp167-250-67.lns2.syd6.internode.on.net) |
| 06:35.48 | Dovid | kk |
| 06:35.50 | Dovid | kk |
| 06:36.52 | mredz | Can anybody help out with this. I am not getting a calling tone when dialing out through a voip provider. Is there an obvious setting that I have missed? |
| 06:37.08 | mredz | I am using Trixbox |
| 06:37.19 | Strom_C | mredz: please read the topic |
| 06:37.23 | Strom_C | go to #freepbx |
| 06:37.24 | Strom_C | thanks |
| 06:37.25 | littleball | Qwell, can recommend a TDM provider who can provide A-Z route with good quality? In addition, i need to get DID numbers |
| 06:37.32 | Strom_C | Qwell: dont answer it |
| 06:37.42 | Strom_C | Qwell: he's totally clueless! :) |
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| 06:38.14 | ANTILOCAS | haha |
| 06:38.54 | littleball | Strom_C, why? i don't understand how you get such a conclusion. tell you the trueth, i am runing three small voip companies... i think i am quite familiar with you are talking about. just used the "maybe" inproper sentence |
| 06:39.09 | mredz | Ok no problems thanks |
| 06:39.22 | Strom_C | littleball: you don't seem to understand the difference between SIP, PRI, TDM, and VoIP |
| 06:39.37 | Strom_C | and you seem uninterested in learning |
| 06:39.40 | ANTILOCAS | stromc and in that book is explained that? |
| 06:39.45 | Strom_C | ANTILOCAS: yes |
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| 06:40.18 | littleball | Strom_C, you can say "seem". right? i have my own SIP service which cluster express route and asterisk... and do sip to phone, firefly to pstn ..... etc |
| 06:40.53 | littleball | do u think i doon;t understand ? rediculous. just ask a question. I think you know what i am asking but just question the question itself. |
| 06:40.54 | Strom_C | In that case, prove you know what you're talking about. |
| 06:40.55 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:41.15 | Strom_C | this is ridiculous |
| 06:41.18 | Strom_C | I'm going to get food |
| 06:41.19 | Strom_C | later |
| 06:41.36 | littleball | if you cannot recommend, then just keep silent. it is not my questions to how to understand the "terms" |
| 06:42.22 | littleball | i am looking for TDM provider/voip provider, which can provide DID numbers and be able to terminate calls to A-Z routes. |
| 06:42.26 | littleball | is this clean? |
| 06:42.28 | littleball | clear |
| 06:42.49 | JT | NO |
| 06:42.54 | mitcheloc | littleball: www.voip-info.org has everything you need to know |
| 06:42.56 | littleball | how to "connect" is not important because i can adjust the system myself |
| 06:43.06 | JT | you are still asking a fundamentally CONTRADICTORY question, littleball |
| 06:43.08 | mitcheloc | as well as www.google.com |
| 06:43.13 | JT | VoIP != TDM |
| 06:43.26 | JT | VoIP DOES NOT EQUAL TDM |
| 06:43.40 | littleball | JT, i know. VOIP /TDM. it is not my question. |
| 06:43.50 | JT | you either want a voip provider (eg. SIP) |
| 06:43.53 | JT | or TDM |
| 06:44.12 | littleball | (1)which can provide DID numbers, (2)can terminate calls A-Z with good quality. (3)TDM or IP is not important |
| 06:44.34 | littleball | because i can adjust my system either use VOIP or TDM. but the first two are important |
| 06:45.40 | littleball | JT, VOIP and TDM has some relationship. Although they are not the same. Example, you connect to provider A over TDM, provider A could terminate calls to some countries over VOIP. |
| 06:45.52 | JT | i understand that |
| 06:46.02 | littleball | so, no need to bother the terms. |
| 06:46.06 | JT | but what concerns you is how you connect to the provider |
| 06:46.18 | littleball | only care the connectivies betweeen ssytem to your provider. |
| 06:46.21 | JT | so what you mean to ask is "what is a good VoIP OR TDM provider?" |
| 06:46.43 | littleball | good provider means famous or branded |
| 06:47.04 | Strom_C | littleball: let me answer your question quickly and simply for you: |
| 06:47.13 | Strom_C | AT&T, Level3, or Global Crossing |
| 06:47.16 | Strom_C | end of story :) |
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| 06:47.51 | littleball | do they provide PRI. I talk to German telecom, they only support SS7 |
| 06:47.56 | littleball | germany |
| 06:48.00 | Strom_C | Yes |
| 06:48.07 | Strom_C | All three of them do |
| 06:48.11 | Strom_C | they also do voip |
| 06:48.13 | Strom_C | so have fun |
| 06:48.17 | littleball | DID also? |
| 06:48.21 | Strom_C | obviously |
| 06:48.24 | cornet | is anyone running a Voicetronix OpenPCI |
| 06:48.56 | littleball | this is what i expected. thanks. |
| 06:49.34 | Strom_C | cornet: what is a voicetronix openpci? |
| 06:50.21 | cornet | Strom_C: the openpci is a POTS card by voicetronix |
| 06:50.29 | MGSsancho | http://bash.org/?654797 |
| 06:51.03 | Strom_C | hahahaha |
| 06:51.08 | Strom_C | cornet: it looks like junk |
| 06:51.36 | JT | rofl |
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| 06:59.51 | tessier | I got so burned out on VOIP a year or so ago that I totally gave up on it. But building this system makes me want to get back into it again. |
| 07:00.09 | tessier | Learned a lot. Not nearly as likely to get burned again (I hope). |
| 07:00.30 | tessier | Alas, I currenly have a 9 to 5 |
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| 07:09.13 | kumamoto | what is the deal with asterisk and HT (hyperthreading) |
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| 07:12.05 | JT | some people say HT's bad, some people don't |
| 07:12.13 | JT | that's about the long and the short of it |
| 07:12.23 | JT | so to "be safe" you could disable it |
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| 07:16.20 | num000 | my asterisk on openwrt is not loading the modules during start, does anyone had this problem? |
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| 07:47.39 | *** join/#asterisk num000 (n=numerobi@e177180225.adsl.alicedsl.de) |
| 07:52.57 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:53.00 | Chris-NB | hi |
| 07:53.21 | Chris-NB | what do I have to do to surpress cdr from asterisk ? |
| 07:53.27 | Strom_C | NoCdr() |
| 07:53.58 | Chris-NB | Strom_C, that is for every call, right? |
| 07:54.14 | Strom_C | what do you mean |
| 07:54.29 | Chris-NB | where do I have to say that NoCdr() ? |
| 07:54.35 | Chris-NB | in dialplan ? |
| 07:54.51 | Chris-NB | is it possible to generaly surpress cdr? |
| 07:55.04 | Chris-NB | i thought this is possible with noload => app_cdr.so in modules.conf ? |
| 07:55.10 | Chris-NB | but ... its not working : / |
| 07:58.41 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 08:00.14 | ANTILOCAS | is asterisk a voip provider as well as a free pbx? |
| 08:00.23 | Strom_C | ANTILOCAS: read the book. |
| 08:00.30 | mitcheloc | ~book |
| 08:00.34 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 08:00.45 | ANTILOCAS | it has too many pages |
| 08:00.53 | JT | so does life |
| 08:00.55 | ANTILOCAS | i read the BRI part though |
| 08:01.43 | Chris-NB | Strom_C, is it possible to generaly disable cdr's ? |
| 08:02.20 | Strom_C | sure |
| 08:02.26 | Strom_C | you don't load the cdr modules |
| 08:03.32 | Chris-NB | with noload => app_cdr.so in modules.conf ? |
| 08:03.38 | *** join/#asterisk L-info (n=Adam@62.69.102.99) |
| 08:04.16 | Strom_C | there is no such thing as app_cdr.so |
| 08:05.10 | Strom_C | <PROTECTED> |
| 08:05.10 | Strom_C | cdr_custom.so cdr_manager.so |
| 08:07.53 | Chris-NB | oh, thx |
| 08:08.50 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 08:09.01 | Chris-NB | Strom_C, that worx : D |
| 08:09.37 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:22.53 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 08:23.23 | *** join/#asterisk vistacdd (n=vistac@59-125-115-136.HINET-IP.hinet.net) |
| 08:24.24 | *** join/#asterisk alib80 (n=chatzill@196.35.242.16) |
| 08:24.40 | vistacdd | Does any know the different between Asterisk an Asterisk Business Edtion? |
| 08:25.20 | *** part/#asterisk boobee2 (n=real@host28-166-static.40-88-b.business.telecomitalia.it) |
| 08:26.28 | tessier | vistacdd: The second costs money? |
| 08:27.00 | vistacdd | and Asterisk have two module, Single Edtion and 5 License, what's the difference? |
| 08:28.30 | alib80 | Asterisk Business Edition is also an older version |
| 08:28.36 | vistacdd | yesterday, I install the Asterisk that I download from internet, and now I install the asterisk from the CD(Asterisk Bussiness Edition) that I bought from reseller? |
| 08:28.52 | alib80 | It has been tested for a particular type of hardware as well. |
| 08:28.54 | vistacdd | but I don't know the difference |
| 08:28.58 | benjk_ | apparently, ABE comes without source code, which means you cant replace and fix things |
| 08:29.11 | mitcheloc | for good reason |
| 08:29.21 | vistacdd | ohh ... |
| 08:29.25 | benjk_ | good reason or not, it can be a serious problem |
| 08:29.48 | benjk_ | especially if you are using it outside of the United States |
| 08:30.08 | vistacdd | = = |
| 08:30.13 | benjk_ | because you are much more likely to tweak something to make it support your local telco environment |
| 08:30.20 | vistacdd | I'm at Taiwan |
| 08:30.28 | benjk_ | yeah, don't use ABE then |
| 08:30.50 | benjk_ | Taiwan used to have the same standards we have here in Japan |
| 08:30.55 | benjk_ | and there is still some of that left |
| 08:31.08 | mitcheloc | benjk_: how is it a serious problem? if you are *talented* enough to modify the source code, you don't need ABE |
| 08:31.08 | benjk_ | Digium have no interest in supporting that |
| 08:31.14 | vistacdd | I have already buy it >"< |
| 08:31.22 | benjk_ | I say, it is a problem if you use ABE |
| 08:31.52 | vistacdd | Now, I found a problem. |
| 08:32.05 | vistacdd | ABE seems can't access mysql directly ... |
| 08:32.12 | benjk_ | of course not |
| 08:32.22 | vistacdd | ABE access mysql use unixODBC = = |
| 08:32.28 | benjk_ | Digium don't want to have to pay 500 USD to MySQL AB |
| 08:33.19 | benjk_ | mitcheloc, it's also not a matter of "talented enough to modify" |
| 08:33.44 | *** join/#asterisk ltd (n=z@202-161-16-50.dyn.iinet.net.au) |
| 08:33.53 | benjk_ | its a matter of "can you hire somebody local to make customisations needed in the local environment" |
| 08:34.08 | benjk_ | with ABE the answer is "no" |
| 08:34.17 | benjk_ | your only supplier is Digium |
| 08:34.33 | benjk_ | in other words, you just got yourself a nice vendor lock-in again |
| 08:35.05 | benjk_ | only this time a vendor who may not care about the place you are |
| 08:35.28 | benjk_ | at least if you have vendor-lock-in by a local vendor, you can be sure they support local standards |
| 08:36.52 | benjk_ | so if you are outside of the United States and perhaps Canada, don't even think about ABE |
| 08:39.40 | vistacdd | in the feature, is there any difference with free edition and ABE ? |
| 08:40.09 | benjk_ | I don't think so |
| 08:40.36 | benjk_ | you are better off with the GPL version |
| 08:40.54 | benjk_ | then find a local company in Taiwan that can support you |
| 08:41.16 | benjk_ | or in Hong Kong even |
| 08:41.25 | *** join/#asterisk Anotsu (n=kvirc@a81-14-180-246.net-htp.de) |
| 08:42.50 | benjk_ | vistacdd what kind of telco hardware do you need? PRI ? |
| 08:43.11 | vistacdd | FXO or PRI T1/E1 |
| 08:43.31 | benjk_ | you may want to go with the GPL version and buy your cards from OpenVox |
| 08:43.34 | inspired | how can I use asterisk's manager interface to get the status of a channel I know the uniqueID of? |
| 08:43.47 | benjk_ | they are in Shenzen and now a lot more about Chinese telco environments than Digium does |
| 08:44.03 | benjk_ | and they can read the Chinese standards documentation, too |
| 08:45.15 | benjk_ | Digium neither have the expertise, nor do they have an interest in acquiring it |
| 08:45.23 | *** join/#asterisk kshyvy (i=kshyvy@ip-82-177-96-22.nm.e-zet.pl) |
| 08:45.39 | benjk_ | they are doing well enough in their home market not to have to care about Asia |
| 08:45.39 | kumamoto | how is can this variable be used " ${CALLERIDNAME} " ? |
| 08:46.26 | vistacdd | OK, I see |
| 08:46.38 | vistacdd | Thank you very much .... |
| 08:47.11 | benjk_ | I am in the same boat as you, have been for three years, just i am in Japan, but for Digium, Japan, Taiwan, China, its all the same |
| 08:47.46 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 08:48.06 | kumamoto | oh that that variable is dead in as of 1.2.0 now it is ${CALLERID(name) |
| 08:51.33 | alib80 | Hi ben_jk do you know whether ABE has the queue bugs sorted out |
| 08:55.18 | alib80 | Hi all |
| 08:55.39 | alib80 | i'm hoping to start a discussion around asterisk queues |
| 08:55.43 | *** join/#asterisk lionelp (n=lionel@ip-128.net-82-216-65.rev.numericable.fr) |
| 08:56.18 | alib80 | namely the setting of the voice recording names and their position in the filesystem |
| 08:56.43 | alib80 | i've noticed that this is easy to do by setting the MONITOR_FILENAME in the dial plan |
| 08:57.50 | *** join/#asterisk Fraeggl (n=Fraeggl@rkom.r-kom.de) |
| 08:57.56 | alib80 | for static members of a queue (e.g. member => SIP/1000) |
| 08:58.07 | alib80 | and that works fine |
| 08:58.35 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 08:59.04 | alib80 | you can set the filename and the folder where it is written to |
| 08:59.56 | alib80 | However as soon as you use agents whether call back or normal it immediatley uses the default filename and positon no matter what you set |
| 09:00.18 | alib80 | I am using the latest version of asterisk, zaptel drivers etc. |
| 09:00.28 | alib80 | sorry latest stable version |
| 09:00.46 | alib80 | Has anyone else run into this problem? |
| 09:00.58 | alib80 | and has anyone figured out how to sort it out. |
| 09:01.00 | alib80 | ?? |
| 09:03.47 | alib80 | i am using the wav49 file format |
| 09:04.52 | *** join/#asterisk lionelp (n=lionel@ip-128.net-82-216-65.rev.numericable.fr) |
| 09:05.33 | *** join/#asterisk sskyles (n=sskyles@adsl-154-244-219.jax.bellsouth.net) |
| 09:07.40 | sskyles | Can anyone tell me where Asterisk stores the current number of messages in a mailbox and how I can get it to re-enumerate that value? |
| 09:10.44 | alib80 | Hi sskyles |
| 09:10.55 | sskyles | Hello. |
| 09:10.57 | alib80 | i'm not sure where in the code this takes place |
| 09:11.08 | alib80 | or if it stores it int he * db |
| 09:11.11 | alib80 | but |
| 09:11.23 | alib80 | i do know that depending on how voicemail.conf |
| 09:11.26 | alib80 | is setup |
| 09:11.30 | sskyles | I looked in astDB... doesn't seem to be there. |
| 09:11.37 | alib80 | the voicemails are stored in |
| 09:12.13 | alib80 | var/spool/asterisk/voicemail/ |
| 09:12.27 | alib80 | in the INBOX of the voicemail box |
| 09:12.33 | alib80 | it enumertaes based on |
| 09:12.37 | alib80 | what is in the folder |
| 09:13.08 | alib80 | if you set delete=yes |
| 09:13.12 | alib80 | in voicemail.conf |
| 09:13.13 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:13.24 | alib80 | for a voicemail2email solution |
| 09:13.28 | sskyles | I rely mostly on the messages that are delivered via email, but sometimes I only have my phone. I have a script that I wrote which prunes the voicemail files in /var/spool/asterisk/voicemail, but asterisk still believes there are many more messages than there really are when I do this. |
| 09:13.29 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 09:14.00 | EmleyMoor | Is FWD being particularly crap at the moment or is it my setup that's wrong? |
| 09:14.32 | EmleyMoor | I managed to get a call from a friend over iax last night but couldn't call him as it claimed his number was busy/congested |
| 09:14.43 | sskyles | The script deletes files that are older than 7 days and moves everything up so that the numbers are still sequential and everything. I just want to tell asterisk to do a recount on the message after I do that. |
| 09:14.48 | sskyles | I think this would fix the problem. |
| 09:14.56 | EmleyMoor | I tried a SIP connection to FWD and got error 500 when I tried to call |
| 09:15.04 | EmleyMoor | Now their iax has gone down again |
| 09:15.17 | alib80 | hmmm |
| 09:16.45 | alib80 | sskyles does the script delete both the wav and taxt file |
| 09:16.56 | alib80 | sorry i know this is a lame question |
| 09:17.03 | sskyles | Yes, it does everything in that respect quite nicely. |
| 09:17.15 | sskyles | I think the final task is to have asterisk update it's count. |
| 09:17.21 | sskyles | I just don't know where that is. |
| 09:19.06 | sskyles | The script even renumbers the files so that it starts at msg0000.* |
| 09:19.17 | alib80 | ja I have seen that |
| 09:19.31 | alib80 | you delete old files and it starts from scratch |
| 09:19.40 | alib80 | ast msg000. |
| 09:20.09 | alib80 | i know this is a hack but could you create empty files that |
| 09:20.23 | alib80 | take the place of the old?? |
| 09:20.40 | sskyles | Well, that's what I think it's supposed to do, but if I had 20 messages, asterisk still thinks I have 20 even when there is now less. |
| 09:21.01 | alib80 | true |
| 09:21.03 | alib80 | that won't help |
| 09:21.28 | sskyles | What it does is tries to play the old files for me anyway, which is now delivered as very short silent messages. |
| 09:21.53 | sskyles | The idea was to avoid the old messages alltogether. |
| 09:22.05 | alib80 | true, tell me something are you using a modified version of the the perl script supplied by asterisk to do this? |
| 09:22.15 | sskyles | No... |
| 09:22.26 | alib80 | and does it matter that the messages are sequential in their numbering |
| 09:22.40 | sskyles | I wrote my own and it's a shell script. |
| 09:23.14 | sskyles | It doesn't seem to matter about the sequencing. It was just something I tried and still didn't work. |
| 09:23.50 | sskyles | The only difference is makes is where asterisk plays empty messages for me. |
| 09:24.00 | sskyles | The count is always wrong after I delete files. |
| 09:24.47 | sskyles | I probably should have a look at the source and see how this is being done... I just thought someone might know off the top of their head. |
| 09:25.05 | alib80 | i'm just looking at a contrib script |
| 09:25.10 | alib80 | that handles vmail |
| 09:25.17 | alib80 | there may be some clues in there |
| 09:25.36 | *** part/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 09:26.11 | sskyles | Send it to me. I'd like to have a look at it... When I wrote mine a couple of years ago, there didn't seem to be any other available solutions. |
| 09:26.28 | sskyles | I've just been to lazy to fix it. |
| 09:26.39 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 09:26.48 | EmleyMoor | FWD over SIP is working now :-) |
| 09:27.02 | Aurs | ERROR[2116]: chan_sip.c:11348 sipsock_read: BAD! BAD! BAD! |
| 09:27.06 | Aurs | that cannot be good? |
| 09:27.24 | Aurs | ERROR[2116]: chan_sip.c:11347 sipsock_read: SIP MESSAGE JUST IGNORED: ACK |
| 09:27.49 | sskyles | My script is only 13 lines in length and that's with the renumbering of the files. |
| 09:30.00 | *** join/#asterisk sambal (n=ivo@sd5116ceb.adsl.wanadoo.nl) |
| 09:30.03 | EmleyMoor | Anyone prepared to help me do a "live person call" test using my asterisk box and fwd? |
| 09:30.21 | sambal | is there a application to stress test asterisk? |
| 09:31.51 | Aurs | sambal: sipsak.. i think |
| 09:32.11 | sambal | ok tnx |
| 09:33.00 | alib80 | Hi all as posted earlier i'm hoping to start a discussion around asterisk queues |
| 09:33.17 | alib80 | namely voice recording filename settings |
| 09:33.27 | vlt | Hello. Anyone here using chan_capi? I try to dial out but always get CONNECT_CONF. Kernel module capi is loaded, asterisk modle chan_capi is loaded, too. There's an entry for controller1 in capi.conf. I use Dial(CAPI/g1/${EXTEN}). Any idea why no channel available? |
| 09:36.30 | vlt | CLI> capi info shows "Contr1: 2 B channels total, 2 B channels free \n Contr2: 1 B channel total, 1 B channel free" (It's an AVM fcdsl, maybe that's why I see two controllers). |
| 09:36.36 | alib80 | hi sskyles |
| 09:36.47 | sskyles | Hi. |
| 09:36.56 | alib80 | i think the renumbering is because the voicemail box may not use a db to track |
| 09:37.00 | alib80 | the filenames |
| 09:37.24 | sskyles | It's possible. I will look though just as soon as I get the other script. |
| 09:37.32 | alib80 | cool |
| 09:38.35 | *** join/#asterisk num000 (n=numer@e177187222.adsl.alicedsl.de) |
| 09:38.42 | EmleyMoor | If someone would like to do a live person call test, can they please call FWD 794933? |
| 09:42.12 | *** join/#asterisk num000 (n=numer@e177187222.adsl.alicedsl.de) |
| 09:42.15 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 09:42.41 | *** join/#asterisk vistacdd (n=vistac@59-125-115-136.HINET-IP.hinet.net) |
| 09:43.49 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 09:44.26 | num000 | is anyone running asterisk on an openwrt box? |
| 09:45.57 | RoyK | i beleive it's been done |
| 09:45.58 | RoyK | yes |
| 09:46.15 | JT | it's definately been done |
| 09:53.19 | X-Rob | openwrt has asterisk packages |
| 09:53.22 | X-Rob | ipkg install asterisk |
| 09:55.48 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 10:01.17 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 10:01.20 | jalsot | hi |
| 10:02.10 | jalsot | does anybody know if the 1.2 branch supports SIP re-INVITEs? We are getting crazy ring-back problem in asterisk-SER-ciscogw configuration |
| 10:04.06 | jalsot | as I can see, we are getting back an INVITE from the SER which is handled by asterisk as a new call, however it belongs [according to CallId] to an existing call |
| 10:09.45 | vlt | [chan_capi] PS: I *can* answer incoming Calls on the CAPI device (if I create an extension for the MSN). What could be a reason that I cannot dial outß |
| 10:09.58 | vlt | *out? |
| 10:10.58 | *** join/#asterisk E-bola (i=bola@rbii-valhalla.mrseb.co.uk) |
| 10:13.07 | E-bola | if i append a /1234 behind a register line |
| 10:13.20 | E-bola | shouldnt all calls voer that line ring 1234? |
| 10:13.45 | E-bola | im having problems controlling what phonenumbers rings on what phones, everything seam to come in on the s extension |
| 10:22.06 | *** part/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 10:25.51 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 10:31.01 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 10:31.37 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 10:37.37 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:38.15 | puzzled | morning |
| 10:42.04 | RoyK | morning |
| 10:46.12 | RoyK | does anyone have the Q.931 spec? |
| 10:49.57 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 10:50.48 | vlt | E-bola: Works here. Did you reload your config? |
| 10:51.36 | florz | RoyK: http://www.vsi.ru/library/ITU-T/original/recs/q/q0931e.zip |
| 10:52.22 | RoyK | florz: thanks |
| 10:54.49 | vlt | Any idea why I can answer calls on a CAPI channel but not dial out? |
| 10:55.46 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 11:11.03 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 11:12.40 | benjk_ | vlt, which country? |
| 11:13.53 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:14.05 | backblue | hi*morning |
| 11:15.20 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 11:16.00 | *** part/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 11:16.45 | vlt | benjk_: Germany. |
| 11:17.08 | *** join/#asterisk motu (n=motu@192.165.166.143) |
| 11:17.56 | vlt | benjk_: CAPI is connected directly to "external" S0 from Telco. |
| 11:18.21 | benjk_ | do you have a means to see the Q.931 messages |
| 11:18.34 | benjk_ | like with PRI you can do pri intense debug on the asterisk console |
| 11:18.52 | benjk_ | if you have some equivalent of that for CAPI, then that would help a great deal |
| 11:18.59 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 11:19.38 | benjk_ | here in Japan, I couldn't make certain outgoing calls on BRI when I had the TON and network type settings wrong |
| 11:19.46 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 11:19.52 | benjk_ | it depends what the telco expects you to send |
| 11:20.19 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 11:20.23 | vlt | benjk_: Mmh, I'll have to look at what Q.931 is ... |
| 11:20.23 | benjk_ | if you have the wrong settings there, they may reject your calls, while this won't impact your ability to receive calls |
| 11:20.42 | benjk_ | its the standard that describes the messages that ISDN nodes exchange |
| 11:21.14 | benjk_ | if you use PRI or BRIstuff both of which use libpri, then you can do "intense debug" on the asterisk console |
| 11:21.23 | vlt | benjk_: So if any changes neccessary I'll have to edit capi.conf? |
| 11:21.32 | benjk_ | this will show you the exact data and cleartext meanings of the messages between your box and the telco |
| 11:22.16 | benjk_ | it will likely have some message that says your type of network or type of number is wrong and your call is being rejected because of it |
| 11:22.29 | vlt | benjk_: At the moment I'm trying with an AVM ISDN controller (Fritz!Card) that can only be driven vai CAPI I think ... |
| 11:22.39 | vlt | *via |
| 11:23.09 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 11:23.10 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:24.35 | JT | arrgh |
| 11:24.49 | JT | all the wires look the same in 25 pair cable |
| 11:24.51 | benjk_ | that's fine, all I am saying is that you should try to find the equivalent command which gives you all the debug output from the ISDN signaling between your box and the telco |
| 11:24.57 | JT | colours are repeated |
| 11:25.08 | JT | and the twiste rate is solawit's hard to see which pair is witch |
| 11:25.11 | JT | which |
| 11:25.54 | benjk_ | maybe there is a command capi intense debug, I dont know |
| 11:26.15 | benjk_ | but whatever it is, you have to get that output to know whats happening |
| 11:26.59 | benjk_ | it will tell you why your outbound calls don't proceed |
| 11:30.06 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 11:35.54 | *** join/#asterisk ctaloi (n=ctaloi@cpe-24-58-22-17.twcny.res.rr.com) |
| 11:36.45 | vlt | benjk_: There's a command `capi debug` but there's no output. Two things that confuse me now: When I try to Dial(CAPI/g1/55512345) I get a CLI line: "data=g1/55512345" and "capi request group = 2". g1=2? Then I get a mesage "ISDN1: received CONNECT_CONF PLCI = 0x101". What does CONNECT_CONF mean? |
| 11:38.12 | backblue | can you connect b2b pri cards? |
| 11:38.46 | benjk_ | I dont know |
| 11:38.52 | vlt | backblue: Mmh, can I??? |
| 11:38.54 | benjk_ | this is a capi specific message |
| 11:39.00 | backblue | vlt: that card have HFC chipset, please use misdn. |
| 11:39.08 | benjk_ | it doesn't, its AVM |
| 11:39.14 | backblue | vlt: was not for you. |
| 11:39.32 | vlt | backblue: Ah, ;) |
| 11:41.05 | backblue | anyone with realtime? |
| 11:41.44 | benjk_ | vlt, is there also a capi intense debug? |
| 11:42.01 | benjk_ | if so, maybe that will display the ISDN signaling |
| 11:42.06 | EmleyMoor | If I install FXS ports in the UK, some phones need a capacitor in their connection to ring - would using master lineboxes be a suitable way of providing this? |
| 11:42.08 | vlt | benjk_: I tried this on CLI but get "unknown command ..." |
| 11:42.18 | benjk_ | too bad |
| 11:42.45 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 11:42.52 | benjk_ | you may want to experiment with the dialplan setting in your conf |
| 11:42.54 | vlt | benjk_: Is my dialstring "Dial(CAPI/g1/55512345)" correct? In wikis I read something like CAPI/:b... |
| 11:42.57 | benjk_ | if there is such a thing |
| 11:43.23 | benjk_ | with zaptel there is dialplan = unknown etc |
| 11:43.39 | benjk_ | if you have such a setting, try all the possibilities |
| 11:44.18 | benjk_ | note that this "dialplan" setting has nothing to do with the asterisk dialplan |
| 11:44.41 | benjk_ | its the type of network information exchanged between your box and the telco over ISDN |
| 11:51.17 | benjk_ | vlt, the dialstring looks alright to me, but I am not a capi expert, if they do something in chan_capi that doesn't follow the format of chan_zap, then it may not be correct, I wouldn't know |
| 11:51.44 | puzzled | vlt: looks ok to me |
| 11:51.49 | benjk_ | my hunch is that the type of network and type of number transmitted to the telco is different from what the telco expects |
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| 11:57.54 | *** join/#asterisk Rahail (n=rahail1@209.19.88.243) |
| 11:58.12 | Sonderblade | using the voicemail ivr, im recording a file that should be played when the user is busy, but the sound is never played why? |
| 11:58.26 | Aurs | NOTICE[16202]: channel.c:1923 ast_read: Dropping incompatible voice frame on Local/45003500@nummerplan-ba88,2 of format alaw since our native format has changed to slin |
| 11:58.32 | Aurs | tips? |
| 12:00.11 | E-bola | I have some stupid phones which doesnt have any forward functionality |
| 12:00.19 | E-bola | is there any numeric codes to transfer a call? |
| 12:00.51 | *** join/#asterisk s1gny|wrk (n=s1gny@p54916B48.dip.t-dialin.net) |
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| 12:03.55 | Rahail | i think you can hit ## |
| 12:03.59 | Rahail | then you can trasfer |
| 12:04.07 | Rahail | or maby flash |
| 12:04.27 | E-bola | so ## and then the number? |
| 12:04.37 | Rahail | yes |
| 12:05.10 | E-bola | mmm nope doesnt work |
| 12:05.17 | Rahail | try flash |
| 12:05.29 | Rahail | do you flash function |
| 12:05.47 | Rahail | have* |
| 12:05.49 | E-bola | phones dont got a flash button |
| 12:05.59 | benjk_ | if anything else fails, smash the phone very violently against a concrete wall |
| 12:06.19 | benjk_ | that won't make it work either BUT |
| 12:06.31 | benjk_ | it will give you a reason to buy better gear |
| 12:06.36 | Rahail | :) |
| 12:06.41 | E-bola | well is it phone dependant or something i config on asterisk? |
| 12:06.53 | benjk_ | it should work out of the box |
| 12:07.04 | benjk_ | btw, every phone has a flash "button" |
| 12:07.07 | E-bola | it works fine with normal phones |
| 12:07.12 | bXi | can somebody recommend a livecd which includes asterisk+isdn support? |
| 12:07.26 | benjk_ | the flash button is the thing you press to hang up |
| 12:07.39 | benjk_ | the hook |
| 12:08.22 | benjk_ | ifyou go off-hook and immediately back on-hook again, it'll do a flash |
| 12:08.28 | E-bola | hmmm |
| 12:08.32 | E-bola | its a wireless pstn phone |
| 12:08.35 | E-bola | and digital |
| 12:08.39 | E-bola | im pretty sure it just hangs u |
| 12:08.39 | Rahail | :) |
| 12:08.41 | Rahail | eh |
| 12:08.42 | E-bola | +p |
| 12:09.09 | benjk_ | well, like I said, there is always the concrete wall method |
| 12:09.13 | Rahail | maby you can go to menu |
| 12:09.19 | Rahail | and find New call |
| 12:09.36 | Rahail | then dail the new number and hit join |
| 12:12.15 | benjk_ | you can always do a three way conference if every participant has got two phones |
| 12:13.18 | benjk_ | a calls b on phone 1 and c on phone 2, b receives a on phone 1 calls c on phone 2, c receives b on phone 1 calls a on phone 2 |
| 12:13.36 | benjk_ | then everybody puts phone 1 to one ear and phone 2 to the other ear |
| 12:13.55 | benjk_ | then you can drop out of the conference |
| 12:14.03 | benjk_ | et voila your transfer is complete |
| 12:16.41 | Rahail | lol |
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| 12:20.08 | vlt | Anyone here with an AVM Fritz on German ISDN where dialling out works? Do I need special settings in capi.conf? |
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| 12:20.30 | Druken | anyone know what time digium opens? |
| 12:21.51 | benjk_ | vlt, capi is very much out of fashion, you may have more luck on some german asterisk forum |
| 12:22.34 | benjk_ | besides, did you find any such dialplan setting in your capi config? |
| 12:22.48 | benjk_ | if it is there, just try all the possibilities |
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| 12:25.24 | vlt | benjk_: No, in the meantime I STFW but couldn't find a solution. The only thing I recognized was that most of expample capi.conf files look different: There's an [interfaces] section. |
| 12:26.53 | benjk_ | interfaces doesn't sound right |
| 12:27.11 | benjk_ | try to find out if there is a dialplan=foobar setting somewhere |
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| 12:43.48 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 12:43.50 | l-fy | hello |
| 12:44.02 | l-fy | what's the status of t.38 in asterisk? |
| 12:44.06 | myiagy | hi, i'm doing monitor on some calls, this is the line i use: |
| 12:44.13 | myiagy | Monitor(wav49|${CALLERIDNUM}_${data}_${EXTEN}|mb) |
| 12:44.29 | myiagy | sometimes, i get a "." in the middle of the callerid or the exten tho.. |
| 12:44.47 | myiagy | and then it won't mix, i get both in and out files, with this dot in the middle |
| 12:45.02 | myiagy | like this: 7702_1154409081_9986.2730-in.WAV |
| 12:45.09 | myiagy | anyone knows why that happens? |
| 12:47.19 | benjk_ | l-fy, asterisk does t38 passthrough |
| 12:47.50 | benjk_ | and that's likely to remain so, according to the guy who wrote the t38 stuff |
| 12:47.59 | l-fy | benjk_ > but real t38 gateway? |
| 12:48.03 | benjk_ | no |
| 12:48.07 | l-fy | o well |
| 12:48.13 | l-fy | anyway benjk_, coppice is a moron |
| 12:48.15 | benjk_ | openpbx |
| 12:48.16 | l-fy | always was |
| 12:48.36 | benjk_ | seems like you only heard one side of the story then |
| 12:48.46 | l-fy | benjk_ > well, i suppose so |
| 12:48.53 | l-fy | but i've saw his code |
| 12:48.57 | l-fy | ugly mess |
| 12:49.14 | benjk_ | he's done all the work, but the folks who make the decisions didn't like it |
| 12:49.31 | l-fy | benjk_ > are you a developer? |
| 12:49.31 | benjk_ | so he's contributed it elsewhere |
| 12:49.42 | benjk_ | yes |
| 12:49.46 | l-fy | good |
| 12:49.53 | l-fy | did you ever tried to read spandsp? |
| 12:50.04 | l-fy | because i did |
| 12:50.19 | benjk_ | my level of expertise isn't in dsp and signaling |
| 12:50.31 | l-fy | dosen't matter |
| 12:50.36 | l-fy | i don't comment on the dsp side |
| 12:50.43 | l-fy | i comment on the programming side |
| 12:50.49 | l-fy | and coppice stinks |
| 12:51.21 | benjk_ | well, I dont comment on code written for a field I don't know, I only comment on code that is written for areas which I do know |
| 12:51.24 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:51.50 | l-fy | thanks god the guys from asterisk didn't based on spandsp and libunicall |
| 12:52.41 | benjk_ | thank god we have choices, because I don't like the 1960s BASIC peek and poke style of asterisk |
| 12:53.10 | l-fy | benjk_ > nither do i |
| 12:53.14 | benjk_ | abstraction is more important than perceived optimisation which in the end does nothing to get you better performance after all |
| 12:53.50 | l-fy | benjk_ > i know that |
| 12:53.50 | benjk_ | or in the words of Hoare and Knuth... "premature optimisation is the root of all evil" |
| 12:53.54 | l-fy | but than |
| 12:54.04 | l-fy | this is why i've chosen something else |
| 12:54.29 | benjk_ | llike I said, thank god there are choices |
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| 12:55.05 | benjk_ | the world would be a bad place if there was only one of a kind for everything and we couldn't make choices |
| 12:55.27 | l-fy | i know :) |
| 12:55.40 | l-fy | this is why gnu/linux, bsd, unix, and windows are so cool |
| 12:55.45 | l-fy | are so many ways to do things |
| 12:56.56 | caio1982 | what are the choices for people who needs MFC/R2 support in asterisk (like, millions of people) without unicall? |
| 12:58.02 | Druken | benjk_: not a bad place, just a very boring one.... |
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| 13:15.13 | sxpert | heya |
| 13:15.35 | sxpert | Strom_C: so here it is: http://bugs.digium.com/view.php?id=7814 |
| 13:16.06 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 13:16.13 | vlt | benjk_: [outgoing capi don't work] For testing I installed hylafax on this machine. It can receive faxes but gets an error when trying to send: "Connection dropped with Reason 0x34A2 (No circuit / channel available)." So I think it's not an asterisk problem :( |
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| 13:19.15 | TheCops | There's a way to make work Meridian telephone (from nortstar meridian 6x16) with Asterisk ? |
| 13:19.51 | sxpert | TheCops: not that I know of |
| 13:20.11 | sxpert | unless you can put your hands on a <whatever> to sip gateway |
| 13:20.12 | l-fy | hey sx-wks |
| 13:20.16 | l-fy | hey sxpert |
| 13:20.17 | sxpert | l-fy: hey |
| 13:20.24 | l-fy | still lonely? |
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| 13:20.29 | sxpert | l-fy: yeah |
| 13:20.33 | l-fy | good |
| 13:20.36 | l-fy | that made my day |
| 13:20.42 | sxpert | hehe |
| 13:20.42 | sxpert | lol |
| 13:20.54 | *** join/#asterisk rata (n=rcampos@princed/developer/rata) |
| 13:20.56 | sxpert | or mebbe not ;D |
| 13:21.01 | sxpert | who knows |
| 13:21.13 | sxpert | mebbe I'm just talking shit |
| 13:21.28 | l-fy | sxpert > that will not be something new for you right? |
| 13:21.36 | sxpert | hehe |
| 13:21.41 | sxpert | guess not |
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| 14:06.55 | a1fa | heh |
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| 14:10.40 | vlt | Hello. I try to initiate a call by putting a file to /var/spool/asterisk/outgoing. But I get the following error: "pbx_spool.c:346 scan_service: Unable to open /var/spool/asterisk/outgoing/fax: Permission denied, deleting". File is asterisk:asterisk -rw-r--r--. What permission is denied here? |
| 14:10.50 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 14:12.14 | florz | vlt: For unlinking you need write access to the directory |
| 14:12.53 | vlt | florz: asterisk immediately unlinks this file, so it has w access ... |
| 14:13.34 | florz | vlt: Erm, IC, so it is unlinking it _because_ it can't read it? |
| 14:14.22 | vlt | florz: Yes, it rather like "permission denied --> deleting" |
| 14:14.39 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 14:14.42 | florz | vlt: Yeah, indeed. Well, no, doesn't make sense to me, then :-) |
| 14:15.07 | florz | vlt: how are you creating the file? |
| 14:15.51 | *** join/#asterisk mercestes (n=merceste@216.54.143.2) |
| 14:16.39 | vlt | florz: touch /var/spool/asterisk/putgoing/../fax; chown asterisk:asterisk !$; nano !$; cp !$ /var/spool/asterisk/outgoing; |
| 14:17.28 | florz | vlt: you should never copy into the outgoing dir, because that's not atomic |
| 14:17.56 | florz | vlt: When asterisk is trying to open the file, it doesn't belong to asterisk:asterisk yet, probably |
| 14:17.58 | vlt | florz: so it's better moving it there? |
| 14:18.13 | florz | vlt: yep, from the same filesystem, of course |
| 14:18.26 | vlt | florz: I'll try that ... |
| 14:19.07 | florz | linking might work, too, hard linking at least ... |
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| 14:21.05 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 14:22.35 | vlt | florz: Ok, that worked, indeed ... nearly. Next problem: "No such host: myvoipprovider". Can't I use a peer defined in sip.conf? How must the "dial string" be formatted? |
| 14:23.15 | florz | vlt: You really should have a look at http://www.voip-info.org/ |
| 14:23.43 | vlt | florz: Now I have "Channel:SIP/5554277401@myvoipprovider" |
| 14:24.06 | vlt | florz: Ok, there's a wiki entry for txfax ... I'll read ... |
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| 14:32.17 | dhill | Does Voicemail() support people to check their voicemail remotely? |
| 14:32.38 | dhill | like, during the intro.. press * to enter password and check? |
| 14:32.52 | Strom_M | yes |
| 14:33.06 | *** join/#asterisk fabsoft (n=fx@host226-130.pool80104.interbusiness.it) |
| 14:33.07 | Strom_M | when they press *, the "a" extension is called in the current context |
| 14:33.17 | Strom_M | from there you can execute VoicemailMain() |
| 14:33.33 | dhill | nice |
| 14:33.36 | dhill | thanks! |
| 14:34.07 | dhill | wait, hmm |
| 14:34.31 | dhill | 6165551212,2,Voicemail(${EXTEN}) |
| 14:34.35 | dhill | how would i do 'a' ? |
| 14:34.54 | dhill | i need a context for the voicemail? |
| 14:35.04 | Strom_M | exten => a,1,VoicemailMain($CALLERID(num)} |
| 14:35.07 | Strom_M | or something |
| 14:35.35 | Strom_M | alternatively, you can set a variable before calling voicemail containing whatever value you want to pass to voicemailmain |
| 14:35.45 | *** join/#asterisk mover (n=dlu@83.125.8.7) |
| 14:35.52 | mover | hiall |
| 14:36.00 | Strom_M | hi |
| 14:36.15 | dhill | i see |
| 14:36.25 | mover | exist any backport from format_h264.c (trunk) to 1.2.x ? |
| 14:36.27 | dhill | sweet, thanks |
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| 14:41.35 | vlt | florz: [spool/.../outgoing] It works now. I tested on the wrong * server, omfg! |
| 14:45.12 | RoyK | ~lart vlt |
| 14:45.13 | RoyK | :) |
| 14:46.05 | florz | *gg* |
| 14:48.19 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 14:50.38 | dhill | i am hitting "*" during Voicemail(), and it is not doing anything |
| 14:50.40 | dhill | debug shows nothing |
| 14:50.46 | dhill | # works |
| 14:50.57 | Strom_M | what version of asterisk are you using? |
| 14:51.01 | dhill | 1.2.9.1 |
| 14:51.05 | Strom_M | odd |
| 14:51.13 | *** join/#asterisk _Mega_ (n=tiago@200-210-129-133.ultrawave.com.br) |
| 14:51.18 | Strom_M | did you put an 'a" extension into the context yet? |
| 14:51.39 | dhill | oh wait, i am a dumbshit |
| 14:51.39 | dhill | hold |
| 14:52.41 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 14:52.53 | dhill | yup |
| 14:52.55 | dhill | i am a dumbshit |
| 14:52.58 | dhill | put it in the wrong context :) |
| 14:53.02 | dhill | thanks |
| 14:53.28 | docelmo | At least you can admit it. Most of us are just duh...... |
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| 14:56.19 | *** join/#asterisk Nebukadneza (n=daddel9@i3ED6F307.versanet.de) |
| 14:56.21 | Nebukadneza | hi |
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| 14:57.02 | Nebukadneza | i got some probs installing asterisk on a archlinux box. thing is i checked out asterisk 1.2 from svn and i dont want to install it "into the system" but rather to /usr/local/asterisk/<foo> or something |
| 14:57.21 | macTijn | what's wrong with that ? |
| 14:57.26 | Nebukadneza | so - how do i do that? i tried PREFIX="/usr/local/asterisk/" make install in zaptel, but it didnt work |
| 14:57.27 | macTijn | (and try modifying the Makefile) |
| 14:57.38 | Nebukadneza | it installed to /etc/, /lib, and so on |
| 14:57.49 | macTijn | Nebukadneza: you also need to do that buildtime |
| 14:58.02 | Nebukadneza | i did that after doing make clean |
| 14:58.28 | macTijn | see the Makefile |
| 14:58.35 | QbY | what would cause Asterisk to not respond to the CLI. I type in a command hit return, and it gives me another CLI> prompt..never executing the command..? |
| 14:59.12 | Nebukadneza | why is there no configure btw? |
| 15:00.25 | backblue | QbY: do you have terminal? |
| 15:00.38 | backblue | are you in a correct session? |
| 15:01.49 | QbY | yes |
| 15:02.00 | QbY | i merely just did asterisk -r to reconnect |
| 15:02.55 | QbY | but it won't do anything.. i did show channels, it showed me a ton of channels in use, but ceased to show me anything else. |
| 15:03.13 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 15:03.46 | RoyK | does anyone know how i can tell meetme to use a certain MoH program? |
| 15:03.50 | RoyK | it just uses the default one |
| 15:04.05 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:04.45 | *** join/#asterisk tarmot (n=tarmo@turing.uiah.fi) |
| 15:06.59 | tarmot | Hi all, I'm trying to get Asterisk to initiate an outgoing call on a Siemens Hicom 300H private PBX. No luck so far, although it does answer calls, and otherwise works nicely (with a Digium clone FXO card)... |
| 15:07.44 | Strom_M | why is it always the clone fxo card people that seem to have the most trouble |
| 15:07.44 | tarmot | What would be easiest way in CLI to test whether dialing out works? The syntax for the "dial" command is a bit baffling. |
| 15:07.57 | tarmot | Strom_M: It's cheaper, so people get that. :) |
| 15:08.33 | coppice | Strom_M perhaps they think all clonmes are made equal |
| 15:08.36 | Strom_M | tarmot, what do you have for your current dial statement? |
| 15:08.59 | QbY | I'm looking at http://bugs2.digium.com/view.php?id=6196 -- Can anyone tell me an alternative way of weighting queues? |
| 15:09.38 | tarmot | Strom_M: I don't have a dial statement yet. I'm using Fast_AGI to initiate the calls from a python program. But the command syntax for API apparently isn't the same as in CLI. |
| 15:09.57 | Strom_M | oy |
| 15:10.15 | Strom_M | dont you have a phone attached to the pbx you can place test calls from? |
| 15:10.30 | tarmot | Strom_M: My Zap is configured properly, since the audio plays nicely via the phone line (if I use a second phone on the same line to listen in) |
| 15:10.43 | tzafrir | Strom_C, why do you need phon es? that's what call files are for? |
| 15:11.05 | tarmot | Strom_M: I only have FXOs, so the dial needs to originate from asterisk |
| 15:11.09 | Strom_M | tzafrir, i'm old-fashioned |
| 15:11.15 | Strom_M | tarmot, or you could use a softphone |
| 15:12.21 | tarmot | I think this would help me: What is the syntax from CLI to dial to number XX using channel 1? |
| 15:13.10 | Strom_M | beats me. I never dial from the CLI |
| 15:13.49 | tarmot | yes, I've figured it's not something people often need, since it quite well undocumented. :) |
| 15:14.55 | Strom_M | though since mine seems to be assuming a context, i would assume you assign the CLI a context in a config file and then dial extension numbers as if you were on a regular phone |
| 15:15.20 | Strom_M | ah, im right |
| 15:15.23 | Strom_M | gladstone*CLI> help dial |
| 15:15.23 | Strom_M | Usage: dial [extension[@context]] |
| 15:15.23 | Strom_M | <PROTECTED> |
| 15:16.17 | tarmot | well, the zap channel is in context "default". but giving "dial 12345@default" just complains that "No such extension '12345' in context 'default'" |
| 15:16.26 | Strom_M | well obviously |
| 15:16.37 | Strom_M | the CLI command works like a telephone set |
| 15:16.46 | Strom_M | not like the Dial() application in extensions.conf |
| 15:17.10 | tarmot | not so obvious to me... :-o |
| 15:17.32 | Strom_M | so if you want to test that, go to extensions.conf and add an extension that will dial things out over the zaptel channel |
| 15:17.35 | Strom_M | so, for example |
| 15:17.39 | Strom_M | [zap-out] |
| 15:17.55 | Strom_M | exten => _X.,1,Dial(ZAP/1/${EXTEN}) |
| 15:17.58 | Strom_M | or something |
| 15:18.15 | tarmot | oh, so I need to define each extension separately? that's just weird... |
| 15:18.33 | Strom_M | what do you mean |
| 15:18.52 | mercestes | _X. matches just about everything last time I checked. |
| 15:18.56 | *** join/#asterisk joda1234 (n=kvirc@p54AFC378.dip0.t-ipconnect.de) |
| 15:19.00 | Strom_M | yes |
| 15:19.10 | tarmot | Strom_M: ah, ok. sorry for being a newbie |
| 15:19.18 | Strom_M | no worries man |
| 15:19.25 | mercestes | We forgive you, but be certain to remedy it as soon as possible..;) |
| 15:19.55 | mercestes | Strom's statement wil match anything (2 chars or more) assign it to ${EXTEN} (a common variable) and outpulse it to zap trunk 1. |
| 15:19.57 | mut | wonder if i should upgrade my asterisk |
| 15:20.05 | mut | show version |
| 15:20.05 | mut | Asterisk CVS-v1-0-09/05/05-04:32:29 |
| 15:20.06 | tarmot | mercestes: well, the whole thingy is part of an open-source product, so I'll contribute... :) |
| 15:20.19 | mut | show uptime |
| 15:20.19 | mercestes | :) most excellent. |
| 15:20.19 | mut | System uptime: 26 weeks, 3 days, |
| 15:20.41 | Strom_M | mut, thats a really really outdated version of asterisk |
| 15:20.45 | mut | heh |
| 15:20.55 | mut | it's been running for 8 months straight w/o reboot or reload |
| 15:21.13 | mercestes | 1.0.9, I've heard, is the most stable, but has none of the cool features of 1.2 and some of the DTMF issues and transfer issues. |
| 15:21.22 | mercestes | I've heard 1.0.9 referred to as a "favorite." |
| 15:21.42 | Strom_M | and all of the security holes! :D |
| 15:21.44 | mut | 10 more days and the box uptime is 1year |
| 15:21.46 | Strom_M | i mean, uh |
| 15:21.57 | tarmot | Strom_M: thanks a bunch - I actually got a call out! |
| 15:22.02 | Strom_M | woot! |
| 15:22.19 | MGSsancho | mut wait 10 more days lol |
| 15:22.24 | mut | :P |
| 15:22.39 | QbY | Can anyone tell me if the QUEUE_PRIO would have an effect on multiple queues, ie. A call going into QueueA with a QUEUE_PRIO=10 would be answered faster than a call going into QueueB with a QUEUE_PRIO=1? Or is QUEUE_PRIO only for that specific queue? |
| 15:22.57 | [TK]D-Fender | DTMF transfers = BAD |
| 15:23.02 | [TK]D-Fender | get real phones! |
| 15:23.09 | MGSsancho | hahahaha |
| 15:23.24 | mercestes | lol |
| 15:23.26 | file | [TK]D-Fender is crazy, don't listen to him! use fake phones |
| 15:23.36 | mut | it's for phones in an lil subdivision/apartments |
| 15:23.54 | MGSsancho | oh cool so not a hobby setup |
| 15:23.57 | *** join/#asterisk ssokol (n=ssokol@dsl017-122-217.mci1.dsl.speakeasy.net) |
| 15:24.12 | *** join/#asterisk nounoursfr (n=nounours@core1.mesbox.net) |
| 15:24.14 | nounoursfr | hello |
| 15:24.15 | mercestes | I'm going to start a company that directly markets to massive telemarketing companies that uses only * and softphones and $200 dell machines with $800 fxs cards in them. |
| 15:24.29 | MGSsancho | pffttt |
| 15:24.31 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:24.33 | mercestes | open source soft phones... |
| 15:24.44 | MGSsancho | hi |
| 15:24.44 | mut | i went through 3 pc's before i found one that wouldn't crash with the te110 |
| 15:24.46 | sxpert | mercestes: 800$ fxs cards ? wtf |
| 15:24.53 | nounoursfr | do you have change GID in the voicemail.c ? |
| 15:24.53 | mut | 3rd pc ended up being my work pc |
| 15:24.54 | mut | heh |
| 15:25.00 | mut | which has been stable as fuk |
| 15:25.05 | [TK]D-Fender | mercestes : How do you use only softphones AND FXS? Whats the FXS for if not analog phones? |
| 15:25.09 | mut | i got a new pc out of it too |
| 15:25.21 | mercestes | To route it into a mitel phone system, of course. |
| 15:25.29 | syzygyBSD | of course |
| 15:25.55 | mut | AND |
| 15:26.05 | mut | this has been running boinc/seti the entire time |
| 15:26.13 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:26.22 | mut | never found any ET tho |
| 15:26.30 | *** join/#asterisk mmcgrath (n=mmcgrath@74.136.14.24) |
| 15:27.17 | *** part/#asterisk mmcgrath (n=mmcgrath@74.136.14.24) |
| 15:28.22 | EmleyMoor | If I have both SIP and IAX connections to FWD enabled at the same time, what happens when I receive a call? |
| 15:28.35 | Strom_M | EmleyMoor, the universe ceases to exist |
| 15:28.46 | *** join/#asterisk Dimik_ (n=laledimi@unaffiliated/dimik) |
| 15:29.05 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:30.13 | tzafrir | nounoursfr, why change the GID? |
| 15:30.31 | EmleyMoor | I doubt that - strange behaviour though |
| 15:31.25 | EmleyMoor | Comes in as two calls... |
| 15:31.47 | nounoursfr | change thr GID for the check voicemail other application |
| 15:32.21 | EmleyMoor | Won't do any - er -harm, will it? |
| 15:33.26 | EmleyMoor | Still getting the unexpected busy indications over IAX |
| 15:34.54 | file | the software that FWD is based on (SER) can handle multiple registrations per account, so your IAX2 one counts as one - and your SIP one counts as one |
| 15:35.00 | file | when you're called, it simply dials both |
| 15:35.30 | EmleyMoor | So, as I thought, it's not harmful :-) |
| 15:35.34 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 15:35.35 | SpaceBass | hey folks |
| 15:35.45 | *** join/#asterisk detach- (n=detach@cm181.delta76.maxonline.com.sg) |
| 15:36.09 | SpaceBass | I'm having an RTP problem where two different phones want to use two different ranges. Rather than chaning all the ranges, can I keep asterisk in the middle of the call? |
| 15:36.10 | EmleyMoor | As I say, I still can't actually make most calls on FWD using IAX, but they do work using SIP |
| 15:37.00 | *** join/#asterisk jhaig (n=jhaig@host-212-18-228-50.static.mailbox.co.uk) |
| 15:37.16 | nounoursfr | do you have a idea |
| 15:38.42 | EmleyMoor | file: and the fact that they are both eventually coming through to the same moziax matters not, as well... |
| 15:40.01 | file | it matters as much as... the fact it's a Monday |
| 15:40.16 | EmleyMoor | Indeed - and a Bank Holiday Monday at that :-) |
| 15:40.25 | sxpert | that quebec voice on the .fr prompts is too funny to listen to ;D |
| 15:41.05 | Strom_M | doo sees uh uh |
| 15:41.11 | Strom_M | juu repet |
| 15:41.23 | file | Strom_M: In'N'Out Burger! |
| 15:41.25 | Strom_M | i can't spell anything in french... |
| 15:41.30 | Strom_M | file, mmmmmmmm |
| 15:41.46 | sxpert | Strom_C: we really need to have fr_FR and fr_QC... |
| 15:42.08 | Strom_M | I know that quebecois is different from mainland french |
| 15:42.18 | Strom_M | but i dont know french at all, so i dont know the differences |
| 15:42.31 | Strom_M | all the french I know comes from disconnected number recordings |
| 15:42.47 | [TK]D-Fender | sxpert: We kinda do now.. The ones by June Wallack are infact CA-FR |
| 15:43.08 | [TK]D-Fender | sxpert: And the older packs done more FR-FR |
| 15:43.13 | *** join/#asterisk mmcgrath (n=mmcgrath@74.136.14.24) |
| 15:43.30 | mercestes | Spacebass: Does canreinvite = no fix it? |
| 15:43.34 | SpaceBass | alors! |
| 15:43.55 | SpaceBass | mercestes, thats a good question, is that, in fact what canreinvite does? would I set both devices to no? |
| 15:44.04 | sxpert | [TK]D-Fender: yeah. I mean, we should have both packs. because june's voice in .fr sounds hilarious |
| 15:44.13 | SpaceBass | basically I have a wifi sip phone that insists on using 9000 as the rtp port and I cannot change it |
| 15:44.19 | mercestes | Spacebass: It basically tells Asterisk to force itself to stay in the media path of two devices instead of removing itself from the media path. |
| 15:44.28 | *** part/#asterisk mmcgrath (n=mmcgrath@74.136.14.24) |
| 15:44.39 | file | SpaceBass: the RTP Port for itself? |
| 15:44.57 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 15:45.01 | sxpert | [TK]D-Fender: the first prompt I heard was "vous etes presentement la seule personne dans cette conference". we don't use "presentement" over here ;D |
| 15:45.12 | SpaceBass | file, the wifi phone's rtp port |
| 15:45.20 | file | SpaceBass: well that's fine... |
| 15:45.35 | file | sxpert: I suppose that's close enough to the english one... |
| 15:45.48 | SpaceBass | file, well I have one way issues...IE if I call from the LAN to the wifi network, i get audio...if I call from the wifi network to the lan, I get nothiong |
| 15:46.40 | file | SpaceBass: interesting |
| 15:46.49 | mercestes | VoIP + Wifi = bad. Have you tried doing nat=yes stuff? |
| 15:47.02 | SpaceBass | file, yeah, I'm at the end of my rope |
| 15:47.03 | mercestes | SpaceBass: or are they all the same subnet? |
| 15:47.19 | SpaceBass | well, this all worked fine for a long time... |
| 15:47.26 | SpaceBass | mercestes, different subnets, no NAT |
| 15:47.39 | SpaceBass | mercestes, there is a firewall b/t them, but its passing everything |
| 15:47.39 | sxpert | file: well, true, but those prompts would probably not really work over here ;( |
| 15:47.39 | mercestes | Try nat=yes anyways then..Sometimes that helps. |
| 15:47.45 | SpaceBass | i have nat=yes |
| 15:47.52 | file | try nat=no |
| 15:47.53 | file | :D |
| 15:48.03 | mercestes | Yea, basically. |
| 15:48.12 | file | if it works, it's an evil evil phone |
| 15:48.24 | SpaceBass | ive played with the nat settings to no avail |
| 15:48.39 | file | then do canreinvite=no, and rtp debug to see RTP traffic going |
| 15:48.45 | SpaceBass | all I can tell is when I do a rtp debug that the phone tries to use 9,000 and the cisco phones are set to use 10000-20000 |
| 15:48.46 | file | and you can confirm the IP/port it's going to |
| 15:48.47 | backblue | voip+wifi = bad? :O |
| 15:48.55 | SpaceBass | im going to try the canreinvite |
| 15:48.55 | file | a phone can use whatever it wants for RTP |
| 15:49.01 | file | it doesn't matter |
| 15:49.11 | file | there's a set range you're supposed to use, but some phones don't use it |
| 15:49.26 | backblue | seems like ppl tend to say allways bad, when they dont know what they are doing. |
| 15:49.42 | *** join/#asterisk pagec (n=pagec@64-252-103-215.adsl.snet.net) |
| 15:49.45 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 15:50.17 | SpaceBass | file, interesting |
| 15:50.17 | SpaceBass | the plot thickens |
| 15:50.17 | pagec | hey, where would i get a list of B2B VOIP providers and how good they are? |
| 15:50.22 | file | the port is parsed from the SDP (thing that contains IP/port for audio & codecs) and the phone should then send audio to that port |
| 15:51.43 | *** join/#asterisk jbroome (n=jbroome@unaffiliated/jbroome) |
| 15:51.52 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 15:51.57 | rene1 | hello |
| 15:52.23 | rene1 | i wonder if digium does support over IAX or if everything must go tru the PSTN |
| 15:52.39 | SpaceBass | file, thanks for the info! |
| 15:52.41 | Strom_M | no, you can call digium via iax |
| 15:52.50 | Strom_M | IAX2/guest@misery.digium.com/s |
| 15:52.58 | rene1 | thanks man |
| 15:53.44 | backblue | why dont use sip? |
| 15:53.48 | rene1 | eew |
| 15:54.19 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 15:54.34 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:56.03 | [TK]D-Fender | sxpert: How would you say it? |
| 15:56.45 | sxpert | "vous etes actuellement seul dans cette conf |
| 15:56.51 | sxpert | (...)érence" |
| 15:57.53 | sxpert | [TK]D-Fender: or "Vous etes pour l'instant seul dans cette conférence" |
| 16:01.16 | Nebukadneza | rehi |
| 16:01.22 | *** join/#asterisk bmg505 (n=leon@dsl-146-29-54.telkomadsl.co.za) |
| 16:01.45 | [TK]D-Fender | sxpert: or "vous etes maintenant la seul membre du conference" ? |
| 16:02.32 | benjk_ | membre should be masculin |
| 16:02.56 | TheCops | Vous petes presentement le seul membre de cette conference. |
| 16:02.56 | benjk_ | also it sounds more natural if you had maintenant at the beginning of the sentence |
| 16:02.59 | TheCops | -p |
| 16:04.18 | benjk_ | "maintenant, vous etes seul dans cette conference" is probably the most natural way |
| 16:04.20 | *** join/#asterisk Givemelove (n=non@208.57.229.162) |
| 16:04.31 | TheCops | benjk_, are you french ? |
| 16:04.48 | coppice | no, he justs plays one on IRC |
| 16:04.48 | Givemelove | I am |
| 16:04.50 | benjk_ | no, but I used to work and live in france |
| 16:05.03 | TheCops | Maintenant in front of the sentence is weird |
| 16:05.05 | TheCops | hehe |
| 16:05.15 | Givemelove | :D |
| 16:05.36 | SpaceBass | now you are in the conference |
| 16:05.51 | SpaceBass | makes sense to me |
| 16:06.00 | benjk_ | you can't just compare the order of words between two languages |
| 16:06.04 | TheCops | yeah, for english people :) |
| 16:06.34 | TheCops | lunch time |
| 16:06.35 | benjk_ | in ENglish its more natural to say "you are now the only ...." |
| 16:06.35 | TheCops | brb |
| 16:06.46 | SpaceBass | oui, c'est vrai |
| 16:07.09 | benjk_ | but in French that doesn't sound like people talk to you in daily life |
| 16:07.24 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 16:07.45 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 16:07.54 | Givemelove | ? |
| 16:07.57 | coppice | in lots of languages the "now" comes first. it does in chinese, for example |
| 16:08.04 | benjk_ | its not incorrect, but it sounds like the sentence has no flow, it sounds interrupted, French is all about flow |
| 16:08.13 | sxpert | [TK]D-Fender: no, that wouldn't work well |
| 16:08.25 | Givemelove | benjk: you got a point |
| 16:08.50 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 16:08.57 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.5) |
| 16:09.06 | benjk_ | if I here somebody say "vous etes ..." i would expect something to follow that doesn't interrupt the sentence |
| 16:09.15 | SpaceBass | speaking of France...I need to get my voxbone paris number working |
| 16:09.20 | [TK]D-Fender | benjk_: "Va-t'en mon ostie!" There's flow for you! ;) |
| 16:09.33 | benjk_ | you are green/blue/red, you are in Paris, you are a bastard ... |
| 16:09.45 | Givemelove | nonsense |
| 16:09.57 | benjk_ | but "you are now ..." kind of interrupts the flow |
| 16:10.18 | Givemelove | Benjk: where are U from in France? |
| 16:10.20 | SpaceBass | voxbone may be the greatest thing to happen to international travelers...I was able to get a french and spanish DID for $20 total |
| 16:10.23 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 16:10.31 | benjk_ | I am not from France, I used to work and live in France |
| 16:10.34 | [TK]D-Fender | benjk_: Tell that to Allison while you're at it .... flows fine for me, but then again I exist between both worlds... |
| 16:10.37 | Givemelove | but where |
| 16:11.03 | benjk_ | D-Fender, I was talking about the French translations presented earlier |
| 16:11.12 | benjk_ | Alsace |
| 16:11.16 | coppice | benjk: I live and work in hong kong, but I doubt anyone should trust my opinion on correct cantonese :-) |
| 16:11.26 | Strom_M | haha |
| 16:11.35 | sxpert | lol |
| 16:11.49 | sxpert | and I am a native french |
| 16:12.09 | Givemelove | SpaceBass: how much a DID costs @ VoxBone? |
| 16:12.09 | benjk_ | I wouldn't say that, if you speak Cantonese and you live in HK, you'll get a feeling for how people say things |
| 16:12.43 | Givemelove | Benjk_: Alsace sucks :) |
| 16:13.02 | benjk_ | you may not know the grammar, and you may not be aware of all cases, especially not stuff that isn't spoken in daily life, but you nevertheless get a feel for how people talk |
| 16:13.05 | CunningPike | And I certainly wouldn't trust the opinions of most Americans on proper English ;) |
| 16:13.17 | angler | benjk_, hello |
| 16:13.25 | benjk_ | Givemelove_ why is that? |
| 16:13.37 | benjk_ | excellent food and wine there ;) |
| 16:13.44 | benjk_ | hi angler |
| 16:14.04 | Givemelove | freezing, old fashioned, too calm people |
| 16:14.18 | benjk_ | freezing only in winter |
| 16:14.33 | benjk_ | in Summer its one of the hottest places north of the Alps |
| 16:14.41 | Givemelove | :D |
| 16:14.45 | Givemelove | north of the alps |
| 16:14.54 | syzygyBSD | how many script kiddies use their own computers for brute force attacks? |
| 16:15.05 | EmleyMoor | Alsace is very nice |
| 16:15.12 | Givemelove | come on dude, it's freezing all year long |
| 16:15.15 | benjk_ | because the Vosges and Foret Noire locks in the climate, its like a greenhouse |
| 16:15.18 | Givemelove | yeah, Alsace is lovely |
| 16:15.21 | sxpert | I have one question related to the manager stuff... |
| 16:15.24 | Givemelove | there's a lot of history over there |
| 16:15.35 | syzygyBSD | sxpert: what is it? |
| 16:15.47 | sxpert | is there a way to get a list of installed zap channels ? |
| 16:15.54 | Givemelove | it's always a pleasure to go there, just hang out in Strasbourg, Mulhouse, but that's it |
| 16:16.01 | benjk_ | not even on Mont St.Odile is it freezing in Summer |
| 16:16.07 | sxpert | (and their type (fxs / fxo / bri / pri) |
| 16:16.31 | sxpert | benjk_: of course not, they have planes crashing there that give off some heat ;D |
| 16:16.39 | benjk_ | haha |
| 16:16.42 | SpaceBass | Givemelove, ranges....b/t $7 and $15 |
| 16:16.54 | syzygyBSD | sxpert: ZapShowChannels ? |
| 16:16.57 | hmmhesays | heh |
| 16:17.19 | sxpert | syzygyBSD: that's a manager Action ? |
| 16:17.26 | syzygyBSD | yes |
| 16:17.30 | benjk_ | no planes crashing on the Grand Ballon though and not freezing in Summer there either |
| 16:17.34 | SpaceBass | Alsace is great |
| 16:17.42 | syzygyBSD | http://www.voip-info.org/wiki-Asterisk+manager+API |
| 16:17.43 | SpaceBass | I'll be in Bordeaux in a month....looking forward to that |
| 16:17.47 | benjk_ | you gotta go a little further into Switzerland |
| 16:18.25 | Givemelove | SpaceBass: Lyon is great! |
| 16:18.41 | benjk_ | yes, Lyon is very nice too |
| 16:18.42 | SpaceBass | I still haven't been to Lyon |
| 16:18.50 | Givemelove | lovely place |
| 16:18.57 | SpaceBass | I lived in Paris for a year and have hardly seen the rest of france |
| 16:18.59 | SpaceBass | how bad is that |
| 16:19.03 | Givemelove | and it's in the middle of everywhere in France |
| 16:19.17 | Givemelove | Paris, Marseille, Alps : 2hours in train |
| 16:19.32 | benjk_ | a little far from Brest though :) |
| 16:19.37 | Givemelove | SpaceBass: the absolute contrary |
| 16:19.47 | Givemelove | Paris sucks for living |
| 16:19.51 | benjk_ | now, there's a place where it is freezing all year round |
| 16:20.02 | Givemelove | North Pole? |
| 16:20.09 | benjk_ | Brest |
| 16:20.12 | Givemelove | ^^ |
| 16:20.17 | Givemelove | freezing AND raining |
| 16:20.22 | benjk_ | yeah |
| 16:21.09 | coppice | It makes me sad if i'm too far from.... oh, Brest |
| 16:21.36 | benjk_ | coppice I think you were thinking of some other kind of brest, er breast |
| 16:21.52 | Givemelove | ^^ |
| 16:22.03 | sxpert | syzygyBSD: of course, I need proper zap channels configured to test it ;( |
| 16:22.07 | caio1982 | what makes me sad is the 2hours in train to go everywhere in the country... pretty sad when you live in a continent-sized place like brazil |
| 16:22.11 | SpaceBass | paris was great |
| 16:22.21 | QbY | What causes this behavior at the CLI? echo*CLI> reload |
| 16:22.22 | QbY | echo*CLI> show channels |
| 16:22.22 | QbY | echo*CLI> show agents |
| 16:22.25 | QbY | it won't respond at all |
| 16:22.41 | benjk_ | asterisk's narchitecture |
| 16:22.49 | benjk_ | it does that from time to time |
| 16:22.57 | benjk_ | having trouble with locks and all that |
| 16:23.07 | QbY | is there a fix? |
| 16:23.14 | benjk_ | just wait, and it may come back |
| 16:23.16 | QbY | because it flat out stops taking calls etc |
| 16:23.29 | sxpert | that's erh... |
| 16:23.32 | benjk_ | if it doesn't, kill it and restart |
| 16:23.48 | benjk_ | and don't do too many reloads next time |
| 16:23.52 | Qwell | agents are part of your problem |
| 16:23.59 | QbY | agents? |
| 16:24.01 | Qwell | I'd be willing to bet |
| 16:24.13 | QbY | i was reading a weight problem in the queue.conf -- i removed it |
| 16:24.23 | sxpert | syzygyBSD: is there a doc with a list of manager commands somewhere ? |
| 16:24.32 | Qwell | QbY: no, agents, not queues |
| 16:25.04 | QbY | i know, but i read a problem that was probably causing another problem.. we had weight= in queues.conf -- so i thought that may be ccausing the problem |
| 16:25.52 | *** join/#asterisk jtodd (n=jtodd@adsl-75-24-91-221.dsl.pltn13.sbcglobal.net) |
| 16:26.49 | EmleyMoor | Am I right in thinking it's unwise to put more than one TDM card in the same box? |
| 16:27.47 | *** join/#asterisk anthonyl (n=rmerz@c-24-12-179-217.hsd1.il.comcast.net) |
| 16:28.01 | sxpert | EmleyMoor: at most 2, if your motherboard is good and stars are aligned properly |
| 16:28.09 | coppice | some would say its unwise to use even one in a box :-) |
| 16:28.49 | sxpert | coppice: those that have other brands to sell ? |
| 16:28.54 | EmleyMoor | coppice: Ask them how to use one out of a box <g> |
| 16:28.56 | sxpert | EmleyMoor: rotfl |
| 16:28.59 | QbY | Qwell.. What do you suggest in place of angents? |
| 16:29.00 | Strom_M | EmleyMoor, it depends on the box |
| 16:29.22 | Qwell | QbY: dynamic queue members |
| 16:29.34 | QbY | makes reporting difficult though |
| 16:30.19 | Strom_M | EmleyMoor, if it's an older, slower machine, then yes, one TDM is generally an upper limit...if the machine is reasonably fast, you can easily get away with multiple TDM cards |
| 16:31.09 | EmleyMoor | So not worth seeking a TDM2400 and all the extra guff it needs just because I might end up needing 3S+3O at some point |
| 16:31.20 | sxpert | it's mostly a matter of interrupt latency anyways |
| 16:31.20 | EmleyMoor | 3S+1O will be what I need to start with |
| 16:31.42 | Strom_M | EmleyMoor, 35+10 what |
| 16:31.58 | benjk_ | 35+10 = 42 |
| 16:32.03 | Strom_M | haha |
| 16:32.07 | EmleyMoor | 3 Sierra and 1 Oscar, not thirty-five and ten |
| 16:32.17 | *** join/#asterisk marta (n=a@host184-5.pool8711.interbusiness.it) |
| 16:32.25 | marta | hello |
| 16:32.30 | sxpert | benjk_: there comes 42 again... it's the answer... ;D |
| 16:32.39 | benjk_ | 3 sierra + 1 oscare = 42 |
| 16:32.50 | benjk_ | its always 42 |
| 16:33.05 | Strom_M | EmleyMoor, yeah, you'll probably be fine with two TDM cards if you need 3 FXS and 3 FXO |
| 16:33.09 | Strom_M | depending on the machine |
| 16:33.18 | Strom_M | and depending on how you want to wire everything |
| 16:33.27 | syzygyBSD | sxpert: yes, I gave you the link http://www.voip-info.org/wiki-Asterisk+manager+API |
| 16:33.30 | Nebukadneza | i got some problems with iax2 and kiax ... i set my iax.conf up so i can register with my softphone. the phone also gets authenticated correctly. the section for the softphone has context 'iax'. in iax i do a _0.,1,Dial(somesipnumber) |
| 16:33.46 | sxpert | syzygyBSD: my bad, I zapped the link ;D |
| 16:33.55 | EmleyMoor | Am I right in thinking 1 FXS supports up to 5 REN? |
| 16:34.03 | Nebukadneza | but when i try to dial with kiax i get "rejected connect attempt from 127.0.,0.1, request 'somenumber@default' does not exist" |
| 16:34.05 | syzygyBSD | sorry, i was off tring to find mounting screws for my rack so I can go install this server |
| 16:34.06 | Nebukadneza | what can i do |
| 16:34.26 | syzygyBSD | I found 4 of the 8 needed |
| 16:34.39 | Strom_M | Nebukadneza, you need to Dial(SIP/whatever@whatever) |
| 16:34.47 | benjk_ | they screwed you on 4 screws then |
| 16:34.50 | Strom_M | EmleyMoor, i'm not sure |
| 16:35.09 | Nebukadneza | Strom_M: Dial(SIP/sip1/${EXTEN}) is the real command i use |
| 16:35.28 | Strom_M | well then you've probably misconfigured the sip1 entry in sip.conf |
| 16:35.32 | Nebukadneza | hm ... |
| 16:36.31 | Nebukadneza | oooh fuck |
| 16:36.33 | Nebukadneza | was a typo |
| 16:36.34 | Nebukadneza | :P |
| 16:36.39 | Nebukadneza | sorry to have bothered you |
| 16:36.44 | Strom_M | haha |
| 16:36.46 | Strom_M | no worries man |
| 16:37.05 | *** join/#asterisk |Materia| (n=Materia@bb220-255-226-24.singnet.com.sg) |
| 16:38.05 | Nebukadneza | hm ... next problem is already knocking the door. no application 'Dial' for extension (iax, 10000, 1) |
| 16:38.19 | EmleyMoor | If I'm right about the FXS REN support, I definitely only need 3 FXS and initially at least just 1 FXO |
| 16:38.30 | Nebukadneza | did the extensions.conf format change from 1.0 to 1.2? |
| 16:38.37 | eKo1 | Nebukadneza: what does 'show application dial' say? |
| 16:38.44 | Strom_M | Nebukadneza, show me the line in extensions.conf |
| 16:38.51 | *** join/#asterisk Defraz (n=t0tal@12.168.101.227) |
| 16:39.03 | Nebukadneza | eKo1: its not registered |
| 16:39.04 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 16:39.18 | Nebukadneza | Strom_M: exten => _X.,1,Dial(SIP/sip1/${EXTEN},,) |
| 16:39.27 | Strom_M | that looks right |
| 16:39.41 | Strom_M | you dont need those commas though |
| 16:39.52 | Nebukadneza | hm- kay |
| 16:40.53 | Nebukadneza | eKo1: how do i get my dial back? :P |
| 16:41.34 | Qwell | Nebukadneza: load app_dial.so |
| 16:42.04 | Qwell | You shouldn't be mucking with modules.conf, if you don't know how to load a module :p |
| 16:42.22 | benjk_ | who needs app_dial anyway |
| 16:42.34 | benjk_ | gimmicks |
| 16:42.45 | Strom_M | yeah, that whole dialing thing |
| 16:42.50 | Strom_M | we should all just be on ringdowns |
| 16:43.00 | benjk_ | whats the obsession with phone calls and all that |
| 16:43.22 | brookshire | asterisk makes phone calls? |
| 16:43.32 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 16:43.34 | Strom_M | no |
| 16:43.37 | Strom_M | asterisk makes cake |
| 16:43.41 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 16:43.42 | brookshire | oh.. that's right |
| 16:45.47 | eKo1 | Nebukadneza: load app_dial.so |
| 16:47.20 | RoyK | eKo1: erm. iirc that really should be loaded in most circumstancesz |
| 16:47.25 | RoyK | s/z$// |
| 16:47.32 | Nebukadneza | eKo1: aah - works |
| 16:47.37 | Nebukadneza | and Qwell :P |
| 16:47.52 | RoyK | ouch |
| 16:47.52 | RoyK | round-trip min/avg/max/stddev = 257.755/2280.611/4270.379/928.951 ms |
| 16:47.57 | eKo1 | RoyK: tell that to Nebukadneza |
| 16:48.09 | RoyK | heh |
| 16:48.19 | *** join/#asterisk joda1234 (n=kvirc@p54AFC96B.dip0.t-ipconnect.de) |
| 16:48.21 | RoyK | Nebukadneza: stripping down asterisk installation, are we? |
| 16:49.10 | joda1234 | hi, anyone here who has bring up a cisco 7970 with sip device? |
| 16:50.20 | RoyK | joda1234: probably lots, but i doubt there are very many of them in here, this being an asterisk channel :P |
| 16:54.30 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 16:54.36 | jbalcomb | Is anyone NOT using roundrobin for their call queue rotation? |
| 16:54.53 | joda1234 | RoyK: i know but all information about configure the phone was i found was in correlation with asterix |
| 16:54.56 | Strom_M | jbalcomb, I've used ringall before |
| 16:55.16 | *** join/#asterisk sdh (n=steve@65.110.40.20) |
| 16:55.24 | sdh | what's the default number to dial to collect voicemail please? :) |
| 16:55.36 | Strom_M | sdn: uh, whatever you configure it on |
| 16:55.38 | Qwell | sdh: there is no "default". You setup an extension |
| 16:55.38 | jbalcomb | Strom_M: was it more useful in general or was there a special case? |
| 16:55.48 | sdh | ah, ok - thanks. flatmate configured it, i'll grep :) |
| 16:55.52 | Strom_M | jbalcomb, the client wanted all the phones to ring |
| 16:56.03 | Qwell | sdh: just look for voicemailmain (grep -i) |
| 16:56.57 | jbalcomb | I'm being told by the reps that one person is getting most of the calls in a given day. |
| 16:57.03 | sdh | Qwell: awesome, it was 10 - thanks :) |
| 16:57.11 | Strom_M | jbalcomb, try rrmemory :) |
| 16:57.12 | Qwell | extension 10? silly roommates |
| 16:57.26 | Strom_M | what nub puts voicemail on extension 10 |
| 16:57.27 | jbalcomb | Is there anyway to see what asterisk thinks about who it sends the call to? |
| 16:57.33 | *** join/#asterisk GlobbeTotssz (i=GlobbeTo@200.122.149.57) |
| 16:57.38 | *** join/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 16:57.52 | Strom_M | jbalcomb, try leastrecent |
| 16:58.16 | Strom_M | that way, the longest-idle agent gets the call |
| 17:00.29 | jbalcomb | Strom_M: yeah, I like the sound of that. thanks. |
| 17:00.38 | sdh | Strom_M: why not? :) |
| 17:00.46 | *** join/#asterisk rnovotny22 (n=Bob@198.57.19.126) |
| 17:01.03 | Strom_M | sdb: potential numbering plan conflicts? |
| 17:01.18 | sdh | sdh: hm. there's only two of us... |
| 17:01.30 | *** join/#asterisk MACscr (n=MACscr@66.73.154.70) |
| 17:01.30 | sdh | <binary joke/> |
| 17:01.33 | CunningPike | Nebukadneza: Shouldn't it be Nebukadnezar? |
| 17:01.37 | MACscr | hello everony |
| 17:01.38 | MACscr | everyone |
| 17:01.40 | MACscr | =P |
| 17:02.06 | MACscr | CunningPike: he is the gangsta version |
| 17:02.07 | MACscr | =P |
| 17:02.12 | CunningPike | lol |
| 17:02.24 | MACscr | or there is a character limit =P |
| 17:02.28 | *** join/#asterisk Blake0PS (n=blake@c-66-41-195-142.hsd1.mn.comcast.net) |
| 17:02.34 | Strom_M | Blake0PS!! |
| 17:02.40 | CunningPike | Nebukadnizzle fo shizzle |
| 17:02.43 | Blake0PS | hey strom, how are you |
| 17:02.51 | Strom_M | doing well, and you? |
| 17:03.05 | Blake0PS | working at my little computer shop in minneapolis |
| 17:03.11 | Strom_M | sweet |
| 17:03.12 | Blake0PS | kinda wet, otherwise doing well :) |
| 17:03.36 | rnovotny22 | BlakeOPS: What's the name of the shop? |
| 17:03.47 | *** part/#asterisk javar (n=javar@Dynamic-IP-cr20011859233.cable.net.co) |
| 17:03.53 | MACscr | Can I add my cell phone number to a ring group? If so, will it show on my cells caller id, the number that was called? |
| 17:04.14 | Strom_M | MACscr, you're using freepbx, arent yoy |
| 17:04.15 | Strom_M | er, you |
| 17:04.27 | Qwell | Strom_M: what ever gave you that idea? |
| 17:04.40 | Strom_M | oh, my cupcake told me |
| 17:04.43 | MACscr | yes....... |
| 17:04.45 | Strom_M | it tells me everything |
| 17:04.51 | Strom_M | MACscr, read the topic |
| 17:05.17 | Nebukadneza | rehi |
| 17:05.34 | Blake0PS | i came here to ask if the FXS module |
| 17:05.34 | Blake0PS | brb |
| 17:05.37 | Nebukadneza | RoyK CunningPike ... sorry - didnt read. was trying to get * running :P |
| 17:05.37 | Blake0PS | customer |
| 17:05.46 | MACscr | So if im using Redhat, would it be wrong to ask a question in ##linux? |
| 17:06.00 | Nebukadneza | succeed in that one, but kiax seems to dislike /dev/dsp |
| 17:06.24 | Strom_M | MACscr, the difference is that freepbx adds countless layers of cruft and stupidity that the majority of us have no interest whatsoever in dealing with |
| 17:06.52 | Strom_M | configuring something in freepbx and configuring it in vanilla asterisk are two entirely different processes |
| 17:07.00 | Strom_M | debugging, also, is completely different |
| 17:07.04 | Nebukadneza | it says that opening /dev/dsp for O_RDONLY results in -10000 |
| 17:07.05 | CunningPike | MACscr: It's more like asking a question about Ubuntu in ##linux-kernel |
| 17:07.44 | MACscr | I understand, i wasnt asking anyone to step me through something. Was just lookign for the overall generic process. Heck, the plain vanilla proccess would be a great answer |
| 17:08.25 | Strom_M | MACscr, "ring group" is a freepbx thing |
| 17:08.39 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:08.45 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 17:09.36 | MACscr | Ok, guess I have been using the term "Ring Group" incorrectly when i was working with avaya and toshiba pbx's before |
| 17:09.42 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:10.09 | *** join/#asterisk Givemelove2k (n=bob@208.57.229.162) |
| 17:10.15 | DaveCanoe | I have an odd problem. Call path is phone->asteriskA->asteriskB and asteriskB has voicemail boxes. VoiceMailmain() is being called on asteriskB and I see asteriskB claiming to playback the voicemail password prompt --- but on random calls I get only silence. Protocol is GSM for all legs of audio. |
| 17:10.42 | Strom_M | DaveCanoe, what protocol are you using for trunking? |
| 17:10.46 | DaveCanoe | (the phones are granstream 2000's which speak gsm) |
| 17:10.52 | DaveCanoe | GSM for trunking. |
| 17:10.58 | Strom_M | protocol, not codec |
| 17:11.01 | DaveCanoe | GSM for SIP to the grandstream, too. |
| 17:11.06 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-96-211.red.bezeqint.net) |
| 17:11.12 | DaveCanoe | sorry interasterisk is IAX2. |
| 17:11.33 | Strom_M | you're passing an Answer() before you execute voicemailmain, right? |
| 17:11.48 | DaveCanoe | I tried that. It didn't seem to change the problem. |
| 17:12.01 | Strom_M | same network segment? |
| 17:12.14 | DaveCanoe | I've never had to pass answer before. Should I try Answer() followed by Wait(1) or the other way around? |
| 17:12.22 | Strom_M | show me iax.conf |
| 17:12.35 | Strom_M | also, what version of asterisk are you running? |
| 17:12.38 | DaveCanoe | No. the phone is local ethernet to asteriskA, the interasterisk link is dedicated DSL |
| 17:12.42 | DaveCanoe | 1.2.9.1 |
| 17:14.51 | Strom_M | ~pastebin |
| 17:14.53 | jbot | pastebin is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 17:15.58 | Strom_M | DaveCanoe, use that please |
| 17:17.35 | DaveCanoe | I just did http://paste.lisp.org/display/24890 |
| 17:18.23 | MACscr | Strom_M: this is an asterisk question. Does asterisk support the ability to have "simultaneus ring" on a particular extension? If so, can that second phone be a cell phone? |
| 17:18.31 | Strom_M | DaveCanoe, what happens if you turn jitterbuffer off? |
| 17:18.34 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 17:18.53 | Strom_M | MACscr, yes, you can pass multiple arguments to the Dial() application |
| 17:19.06 | Strom_M | one of those can be an IAX2 or SIP trunk to an ITSP |
| 17:19.18 | MACscr | ah, ok. Thanks |
| 17:19.25 | Strom_M | however, you may run into trouble if the cellphone voicemail answers immediately |
| 17:19.35 | Strom_M | i.e. your phone is off or the network can't find your phone |
| 17:19.39 | MACscr | yeah, i could see where that would be a problem |
| 17:19.58 | juanjoc | Has anybody been using the T.38 passthrough code with Asterisk 1.2? |
| 17:21.14 | MACscr | Right now when someoen calls, all the extensions ring, would just like to have my cell phone added to that. Plus when my extension is called, would ilke my cell to ring then too. Just trying to figure out the best way to go about it |
| 17:21.16 | DaveCanoe | so far so good on no jitter buffer. |
| 17:22.00 | Strom_M | DaveCanoe, don't use the jitterbuffer if you don't have jitter problems |
| 17:23.27 | DaveCanoe | OK. Certainly I don't need it. DSL is a 15 ms +- 2 ms, generally. |
| 17:23.36 | Strom_M | haha yes |
| 17:23.44 | Strom_M | no need for jitterbuffer :D |
| 17:24.03 | [TK]D-Fender | MACscr: You'll also run into trouble on the analog circuits where the call is considered answered the moment the number is DIALED. PRI based connections can help with this in most cases |
| 17:24.56 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 17:25.16 | *** join/#asterisk Tux889 (i=Tux889@mctnnbsa24w-142167039101.pppoe-dynamic.nb.aliant.net) |
| 17:25.28 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:25.55 | DaveCanoe | Now here's an advanced question: can I somehow pass the message waiting indicator through this trunk. Ie: can the phones attached to asteriskA (whose job is just trunking) be given message-waiting-indictor from the voicmail residing no asteriskB |
| 17:26.40 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 17:27.10 | MACscr | <PROTECTED> |
| 17:27.16 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 17:30.50 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 17:31.13 | MRH2 | hi anyone having issues with the latest zaptel 1.2? |
| 17:31.27 | Strom_M | what kind of issues |
| 17:32.05 | RoyK | itdoesntworkanditsallyourfault issues :) |
| 17:32.12 | MRH2 | getting what looks like a pci error |
| 17:32.28 | MRH2 | but only on versions about the 2nd July |
| 17:32.48 | MRH2 | can i pastebin what i have? |
| 17:32.58 | *** join/#asterisk Bullseye_Network (n=info@72-166-37-114.dia.static.qwest.net) |
| 17:32.59 | RoyK | hm... digium hardware? |
| 17:32.59 | RoyK | sure |
| 17:33.31 | MRH2 | http://pastebin.ca/152238 |
| 17:34.19 | MRH2 | yep digium 411P |
| 17:35.19 | [TK]D-Fender | MACscr: If you want to ring both at the same time you have to make sure that your ring time is less than your VM timeout on the cell and that you are using a channel taht supports proper call progress (BRI/PRI) |
| 17:35.59 | RoyK | MRH2: then i'd say don't file a bug at bugs.digium.com, since it'll probably be refused. those things are (ironically) called Digium support issues and not related to asterisk, somehow |
| 17:36.29 | MRH2 | if i downgrade to the 2nd July version of zaptel all works fine |
| 17:36.36 | *** join/#asterisk svenna_ (n=svenna@p548D3184.dip0.t-ipconnect.de) |
| 17:36.43 | RoyK | MRH2: eh |
| 17:37.14 | MACscr | <PROTECTED> |
| 17:37.20 | MRH2 | yeah weird |
| 17:37.23 | MACscr | woops, nothing but |
| 17:37.39 | Blake0PS | the FXS module on the TDM400P, does that only allow connections from the middle two pins on a RJ11 connector? |
| 17:37.50 | novafirst | Asterisk Version 1.2.11 -> can anyone confirm if latest version is using built-in mp3 player and it doesn't require external mpg123 |
| 17:37.54 | RoyK | MRH2: have you tried to shut down your computer completely and letting it stay without power for ten minutes or so (unplug power) and then trying again? there was some bugs in the old days that if zaptel drivers weren't loaded, zaptel hardware (my TE410Ps) didn't work until I did a cold restart |
| 17:37.59 | RoyK | really cold |
| 17:38.03 | RoyK | freezing damnit |
| 17:38.18 | RoyK | erm |
| 17:38.26 | RoyK | but that's not your error. sorry. i can see it generating interrupts |
| 17:38.40 | MRH2 | kind of - i left it for 2 mins |
| 17:38.58 | RoyK | but why is wct4xxp listed in interrupts before it's loaded? |
| 17:39.08 | RoyK | sounds a little wierd |
| 17:39.28 | RoyK | s/a little wierd/impossible |
| 17:39.54 | RoyK | jbot: ping |
| 17:39.56 | jbot | pong |
| 17:40.13 | RoyK | s/jbot// |
| 17:40.27 | novafirst | [TK]D-Fender: Asterisk Version 1.2.11 -> can you confirm if this version is using built-in mp3 player and if doesn't require external mpg123. |
| 17:40.42 | MRH2 | dunno |
| 17:40.56 | MRH2 | what is an NMI error? |
| 17:40.59 | RoyK | novafirst: there's an mp3 decoder in -addons |
| 17:41.07 | RoyK | non maskable interrupt |
| 17:41.10 | RoyK | pretty bad stuff |
| 17:41.53 | RoyK | most hardware generates that if they're shat on or something, but not before |
| 17:42.01 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 17:42.10 | novafirst | RoyK: format_mp3 ? |
| 17:42.15 | [TK]D-Fender | novafirst: You haven't needed MPG since 1.2.0. You jsut need to compile asterisk-addons to add MP3 SUPPORT to it. |
| 17:42.24 | RoyK | novafirst: iirc, yeah |
| 17:42.55 | MRH2 | the card actually seems to work though despite it |
| 17:43.28 | RoyK | [TK]D-Fender: although nativ moh somehow sucks, since it can't keep on playing a playlist round and round, but starts off playing the same song every time someone connects, which also means one decoder per callee |
| 17:43.34 | MRH2 | except i have this dell orange warning light sating PCI PArity error |
| 17:43.43 | RoyK | s/the\ same/a/ |
| 17:43.46 | novafirst | [TK]D-Fender: thanks. I am having problem with musiconhold. No sound! |
| 17:44.00 | MRH2 | do u think i should email digum then? |
| 17:44.10 | [TK]D-Fender | novafirst: pastebin your config for it, and did you compile the SEPERATE package I mentioned? |
| 17:44.11 | RoyK | MRH2: i'd try that |
| 17:44.23 | jbroome | I fell dirty. I just delivered a working 1.2.10 * install to a client, now they tell me they want a gui so the office admin can change extensions, and other minor tweaks. Any suggestions for one that won't involve a complete reinstall? |
| 17:44.25 | RoyK | novafirst: do not use mpg321 for one |
| 17:44.57 | RoyK | jbroome: just write a little GUI :D |
| 17:45.09 | RoyK | "Menus belong in restaurants" |
| 17:45.12 | MRH2 | I use agent call back login if they are changing extensions alot? |
| 17:45.37 | novafirst | [TK]D-Fender: RoyK: I am not using mpg123 instead I compiled one form addons. |
| 17:45.55 | jbroome | RoyK: not really an option. damn, i don't want to have to use trixbox |
| 17:46.04 | MACscr | ha ha |
| 17:46.06 | MACscr | =P |
| 17:46.09 | [TK]D-Fender | novafirst: pastebin the output of "show formats" |
| 17:46.33 | RoyK | jbalcomb: what sort of things do they need? most of it can be done quite a bit quicker from vi |
| 17:47.05 | [TK]D-Fender | jbroome: Too many things to do to add an exten. SIP/other entries for the teh, VM, extensions.conf to dial and handle it, etc.... its GUI or bust. |
| 17:47.23 | RoyK | jbalcomb: offer them an admin course, and charge them for it, then when they fuck up, charge them to fix it... |
| 17:47.25 | RoyK | hihi |
| 17:47.47 | MRH2 | RoyK do u think it worth narrowing it down to the exact commit that the error started? |
| 17:48.17 | RoyK | MRH2: indeed. with what version does it work? |
| 17:49.26 | MRH2 | svn 1.2 from 2nd July is ok - and it was sometime in the following few days it started |
| 17:49.26 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 17:49.26 | novafirst | [TK]D-Fender: http://pastebin.com/778094 |
| 17:49.43 | [TK]D-Fender | novafirst: The mode you have to use is "files" to use Native MoH. |
| 17:50.00 | [TK]D-Fender | novafirst: You clearly have not read the instructions. |
| 17:50.12 | rene1 | if lspci -vb says i am not sharing irqs but lspci -vv says i do, what gives? |
| 17:50.31 | RoyK | MRH2: then just svn up -D 'whenever it last worked', then try to upgrade to a few days later, then roll back to right in the middle of those two, then cut in half, again and again, and you're bound to find the exact time where it occured |
| 17:50.47 | RoyK | testing in between the upgrades, of course |
| 17:51.01 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 17:51.07 | MRH2 | ok will do and email digum |
| 17:51.08 | RoyK | it'll take some time, but it might be quicker than asking digium support to do it |
| 17:51.09 | novafirst | [TK]D-Fender: thanks. let me test. |
| 17:51.15 | RoyK | (just my 2c) |
| 17:51.55 | RoyK | MRH2: if you find where this occured, issue a report to bugs.digium.com as well, since then it's clearly related to zaptel/asterisk |
| 17:52.12 | MRH2 | ok |
| 17:52.17 | frenzy | whats that analog bank hardware called that connects to the TDM2400P |
| 17:52.30 | RoyK | frenzy: wot? |
| 17:52.47 | frenzy | the analog ports |
| 17:52.52 | rene1 | i am using APIC |
| 17:52.56 | RoyK | frenzy: a channel bank usually takes a T1 or an E1. the TDM2400 has lots of analog ports |
| 17:52.56 | frenzy | for using the TDM2400p |
| 17:52.59 | jbalcomb | multiplexor? |
| 17:53.02 | jbalcomb | channel bank? |
| 17:53.23 | RoyK | jbalcomb: not for a TDM2400p. that's a PCI channel bank somehow |
| 17:53.28 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 17:53.46 | frenzy | yes but how do I connect the analog lines? |
| 17:53.56 | RoyK | frenzy: that's what it's for.... |
| 17:54.31 | Blake0PS | strom are you around? |
| 17:54.35 | frenzy | huh |
| 17:55.50 | Strom_C | hi |
| 17:56.29 | Blake0PS | the fxs module on the tdm400p.. does it only support a phone on the middle 2 pins of an rj11? |
| 17:56.37 | Strom_C | yes |
| 17:56.45 | Strom_C | each module is a single line |
| 17:56.47 | frenzy | so does anyone know what i'm talking about.. |
| 17:56.51 | Strom_C | frenzy: yes |
| 17:57.09 | frenzy | so what is that socket thing called? |
| 17:57.10 | Strom_C | frenzy: you connect a 25-pair cable using an ampenol connector |
| 17:57.13 | Blake0PS | it doesnt support a single phone on the outside 2 pins then? |
| 17:57.23 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:57.25 | Strom_C | Blake0PS: no. the tdm400p only supports 4 channels |
| 17:57.33 | Blake0PS | ok thanks! |
| 17:57.42 | Strom_C | s/ampenol/amphenol/ |
| 17:57.48 | Strom_C | SOLDERING IRON?! |
| 17:58.05 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-232.dhcp.southerncoastalcable.net) |
| 17:59.08 | Blake0PS | i have a phone system that uses 2 channels per cable and i have to solder it up .. not the digium stuff just the wire |
| 17:59.08 | Sonderblade | can you call using a modem using asterisk? |
| 17:59.28 | QbY | Ok, my Asterisk is continuing to behave in a way that is not normal (compared to my other asterisk installations)... Yet, it isn't telling me anything.. How can I get it to tell me what is bothering it so much? |
| 18:00.23 | Strom_C | Blake0PS: why even bother with soldering? you can get modular diplexers, you know |
| 18:01.00 | jbalcomb | QbY: approach it with compassion and rub its shoulders a little? |
| 18:01.05 | *** join/#asterisk s1gny|wrk (n=s1gny@p54916A09.dip.t-dialin.net) |
| 18:01.24 | QbY | jbalcomb.. heh.. |
| 18:01.37 | vader-- | doy ou guys know how to create a poe injector? |
| 18:02.06 | jbalcomb | QbY: you might also try setting up another log file and sending EVERYTHING to it then stare at it with tail -f |
| 18:02.13 | [TK]D-Fender | vader--: : Just BUY one |
| 18:02.34 | vader-- | can i buy one from compusa? |
| 18:02.39 | [TK]D-Fender | vader--: : If you have to ask how, then the answer is "you're not qualified". :) |
| 18:02.42 | jbalcomb | QbY: if its crashing then set up remote logging via syslog (preferreably syslog-ng) |
| 18:02.46 | *** join/#asterisk mountainm2k (n=mountain@216.87.64.218) |
| 18:02.52 | *** part/#asterisk Bullseye_Network (n=info@72-166-37-114.dia.static.qwest.net) |
| 18:02.54 | [TK]D-Fender | vader--: MAYBE... they likely resell Linksys which makes one. |
| 18:03.09 | *** join/#asterisk jogy999 (n=stef@mail.innovate-it.ch) |
| 18:03.24 | [TK]D-Fender | vader--: And reason you can't order one through a normal compuer store? |
| 18:03.31 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 18:03.43 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 18:03.50 | MACscr | i know, who honestly picks up computer parts at a local retailer =P |
| 18:04.11 | mountainm2k | I have a Microcenter local to me -- so I don't count... :-P |
| 18:04.25 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 18:04.58 | jbalcomb | microcenter makes me sad for price tags |
| 18:05.19 | mountainm2k | ? they seem pretty cheap to me -- way better than compusa... |
| 18:05.48 | jbalcomb | last time i was there they had a ISA 56K modem marked down to $27USD |
| 18:05.56 | [TK]D-Fender | You can't even SEARCH for PoE... it translates as "pie" when you do. &^%#$ dumbasses |
| 18:06.16 | sx-wks | yeehah... |
| 18:06.21 | rene1 | an irq sharing question: i am confused lspci -v says the digium card is sharing IRQs with the IDE system (i.e. both using 169), and lspci-vb says IDE is using 255 and the digium card 12. I tested for interrupts and it seems i am getting 10002 every 10 seconds, the wiki says that it is something reasonable. how should i read the different measures from lspci ? |
| 18:06.53 | sx-wks | yeepee :D |
| 18:06.59 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 18:07.53 | jbalcomb | [TK]D-Fender: looks like i'll be switching our queue strategy to rrmemory |
| 18:08.20 | jbalcomb | [TK]D-Fender: i had to update all the periodic-announce times to 45 seconds today |
| 18:08.32 | *** join/#asterisk rpm (n=russell@S01060002b3d10d24.cg.shawcable.net) |
| 18:09.36 | vader-- | hmmm |
| 18:09.41 | vader-- | no one really seels just the injector |
| 18:09.44 | jbalcomb | now if i could only confirm and/or correct the wrapuptime issue |
| 18:10.07 | mountainm2k | [TK]D-Fender: you on here? quick Polycom config q? |
| 18:10.20 | novafirst | [TK]D-Fender: where can I read more about setting musiconhold but for latest AST version. |
| 18:10.47 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 18:10.54 | jbalcomb | novafirst http://www.voip-info.org/wiki/index.php?page=Asterisk |
| 18:10.58 | [TK]D-Fender | jbalcomb: Excellent, its the strategy I suggest the most. |
| 18:11.26 | [TK]D-Fender | novafirst: What I jsut suggested IS current and is what you should do in 99% of cases |
| 18:12.07 | novafirst | jbalcomb: most info that I found at voip-info is outdated |
| 18:12.22 | rene1 | asterisk moves quickly |
| 18:12.59 | jbalcomb | novafirst: musiconhold.conf.sample |
| 18:13.24 | jbalcomb | rene1: asterisk is developed in a poorly documented fashion |
| 18:14.18 | rene1 | well.. yes, the only way to really be upto date is the bug tracker |
| 18:14.33 | novafirst | [TK]D-Fender: I made change change to musiconhold.conf and restarted AST. Still no sound. Log file shows -- Started music on hold, class 'default', on SIP/xlite01-08d02a38 |
| 18:15.17 | [TK]D-Fender | novafirst: Pastebin your musiconhold.conf and the CLI output of "show formats". Also make sure your MP3's have NO ID3 tags. |
| 18:15.33 | [TK]D-Fender | novafirst: And avoid VBR's as well |
| 18:17.02 | Blake0PS | strom, i need it today, any retail chains carrry diplexers? |
| 18:17.16 | *** join/#asterisk Katty (n=Katty@64.82.232.54) |
| 18:17.18 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 18:17.22 | Katty | allo. |
| 18:17.29 | [TK]D-Fender | Blake0PS: CornerStoreCoreTelecom obviously! ;) |
| 18:17.34 | benjk_ | D-Fender, whats wrong with ID3 tags? |
| 18:17.36 | CunningPike | Hello, Katty |
| 18:17.37 | [TK]D-Fender | Katty: Salut ma chere ;) |
| 18:17.41 | Katty | CunningPike: hihi (= |
| 18:17.45 | Katty | [TK]D-Fender: mew. |
| 18:17.47 | [TK]D-Fender | Katty: translation : Mew. ;) |
| 18:17.52 | Katty | [TK]D-Fender: i'm in a bit of a pickle! |
| 18:17.52 | Strom_C | Blake0PS: radioshack? rite-aid? |
| 18:17.58 | Qwell | rite-aid? |
| 18:18.03 | Strom_C | yes |
| 18:18.05 | [TK]D-Fender | Katty: Just add Ritz! ;) |
| 18:18.05 | Strom_C | rite-aid |
| 18:18.08 | Katty | [TK]D-Fender: silly. |
| 18:18.08 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 18:18.13 | [TK]D-Fender | indeed |
| 18:18.14 | Strom_C | or whatever crazy drugstore chains you have in minnesota |
| 18:18.16 | Katty | [TK]D-Fender: just add call recording, you mean ;) |
| 18:18.24 | novafirst | [TK]D-Fender: show format -> this option not available. All mp3 are saved without ID3 tags and I use lame to convert to right format |
| 18:18.46 | *** join/#asterisk harlequin516 (n=sham@dsl01-ppp-4444.fastq.com) |
| 18:18.49 | Katty | [TK]D-Fender: i have call recording enabled. and it works just peachy... except for when i try to record an /outgoing/ call. |
| 18:19.16 | Katty | [TK]D-Fender: where do i start with this issue? |
| 18:19.17 | harlequin516 | ]I'm trying to get DISA to work. What I get is a dialtone for one or two seconds and then it hangs up. Is this normaL |
| 18:19.21 | harlequin516 | ? |
| 18:20.09 | [TK]D-Fender | novafirst: "show modules like format" |
| 18:20.25 | [TK]D-Fender | Katty: Ok, what happens when you try? |
| 18:20.36 | Katty | [TK]D-Fender: nada, zip, nilch. |
| 18:20.47 | Katty | [TK]D-Fender: not even an error. |
| 18:20.49 | harlequin516 | vOIP SITE IS scant on the details of this function. Also does the password need to be quoted, when specifying no-password, what is supposed to happen? |
| 18:20.51 | novafirst | [TK]D-Fender: http://pastebin.com/778131 |
| 18:20.53 | Katty | [TK]D-Fender: i've my verbose set at 7 |
| 18:20.54 | [TK]D-Fender | Katty: Nein? Nyet? Non? Rien? |
| 18:21.03 | Katty | [TK]D-Fender: yubyub, absolutely nothing at console. |
| 18:21.17 | Katty | [TK]D-Fender: unless, of course, i call an inside extension...in which case it works just peachy. |
| 18:21.24 | [TK]D-Fender | Katty: Show me some CLI.... |
| 18:21.24 | *** part/#asterisk jogy999 (n=stef@mail.innovate-it.ch) |
| 18:21.33 | Katty | [TK]D-Fender: okay. |
| 18:22.00 | [TK]D-Fender | novafirst: File looks fine, no the CLI output I just asked for please... |
| 18:22.22 | *** join/#asterisk schuylerdigium (n=bosco@gateway.digium.com) |
| 18:22.44 | *** join/#asterisk sx-wks (n=sxpert@navsys.org) |
| 18:22.56 | Katty | [TK]D-Fender: someone started listening to their voicemail in the middle of this >.< |
| 18:23.04 | Katty | how utterly annoying! |
| 18:23.09 | Katty | silly co-workers. |
| 18:23.16 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 18:23.25 | [TK]D-Fender | You go kill them for their insolence, right now! |
| 18:23.59 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 18:24.35 | Katty | [TK]D-Fender: this is the pastebin of a recorded call... |
| 18:24.36 | [TK]D-Fender | novafirst: s/no/now/ |
| 18:24.48 | Katty | [TK]D-Fender: http://pastebin.ca/152309 |
| 18:25.00 | Katty | [TK]D-Fender: it is in the [from-sip] context |
| 18:25.33 | Katty | [TK]D-Fender: from sip also includes [to-phone-company] |
| 18:25.52 | Katty | [TK]D-Fender: to phone company includes all my zaptel lines. |
| 18:25.53 | [TK]D-Fender | Katty: They recorded for just a few seconds before cancelling? |
| 18:26.03 | Katty | [TK]D-Fender: nono, it records in /full/ |
| 18:26.08 | [TK]D-Fender | Katty: And did dso by DTMF/features.conf? |
| 18:26.11 | Katty | [TK]D-Fender: but only on in house extensions. |
| 18:26.16 | [TK]D-Fender | User hit '#9' to record call. |
| 18:26.26 | Katty | [TK]D-Fender: yes, as you can see it works. |
| 18:26.30 | Katty | [TK]D-Fender: works beautifully (= |
| 18:26.37 | Katty | [TK]D-Fender: but it won't work when i call a /real/ number. |
| 18:27.01 | [TK]D-Fender | Katty: Show me a REAL one then please :) |
| 18:27.08 | Katty | [TK]D-Fender: i can't. |
| 18:27.15 | [TK]D-Fender | Katty: ...because? |
| 18:27.19 | Katty | [TK]D-Fender: it would be pointless, because nothing comes up when I hit #9 |
| 18:27.23 | _DAW | Can someone help me with a question about * and hold? |
| 18:27.28 | [TK]D-Fender | Katty: Please show me ANYWAYS. |
| 18:27.32 | Katty | okie dokie. |
| 18:27.42 | novafirst | [TK]D-Fender: http://pastebin.com/778137 |
| 18:27.44 | [TK]D-Fender | _DAW: Just ask |
| 18:28.50 | Katty | [TK]D-Fender: http://pastebin.ca/152314 |
| 18:29.28 | [TK]D-Fender | nova, no do an "ls" for your mp3 folder to double check them.... |
| 18:30.03 | sb_mx | Katty, i think it has to do with the 'wW' options in your outgoing dialstring |
| 18:30.09 | _DAW | OK.. I have asterisk between my phone and sip gateway. When I press hold the rtp stream stops between the phone and * however no reinvite is sent between * and the gateway and the rtp stream stays up in one direction. Am I missing something in the config? |
| 18:30.44 | novafirst | [TK]D-Fender: inside mp3 folder it shows one mp3 file. |
| 18:30.52 | [TK]D-Fender | Katty: You didn't add the "W" parameter to your outgoing dial. That's why |
| 18:31.04 | [TK]D-Fender | novafirst: SHOW |
| 18:31.53 | Katty | [TK]D-Fender: so it needs to be asdjfalsdf/g1/wwW${EXTEN}? |
| 18:32.06 | *** part/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 18:32.16 | Un1x | root 6242 0.0 0.0 0 0 ? Z 13:41 0:00 [asterisk] <defunct> |
| 18:32.16 | *** join/#asterisk sangee (n=rkuru@206.191.114.66) |
| 18:32.18 | Un1x | ???? |
| 18:32.27 | novafirst | [TK]D-Fender: there is one mp3 only inside /var/lib/asterisk/mohmp3 |
| 18:32.32 | Un1x | whats <defunct> supposed to mean |
| 18:32.38 | Katty | [TK]D-Fender: or asdljfalsdf/g1/ww${EXTEN,,wW} |
| 18:32.39 | gmfm | Katty: Dial(Zap/g1/${EXTEN}||wW) |
| 18:32.51 | Katty | gmfm: Ooo. |
| 18:32.55 | florz | Un1x: That the process has exited |
| 18:33.02 | Un1x | oh ok |
| 18:33.09 | Katty | ah ha! |
| 18:33.24 | novafirst | [TK]D-Fender: do I need to define musiconhold=default anywhere? |
| 18:33.25 | [TK]D-Fender | Katty: Dial(ZAP/g1/www123456,,W). |
| 18:33.26 | Katty | sure enough, my in house extensions have the wW on the end :> |
| 18:33.42 | [TK]D-Fender | Katty: Only Uppercase.. you don't the person you are calling to be able to initiate recording |
| 18:34.30 | Katty | [TK]D-Fender: okay, i need help formatting this since I've got two different people telling me the format. |
| 18:34.38 | [TK]D-Fender | Katty: And you may want to revisit your macro/exten strategy for optimal mix of recording controls because of this revelation. |
| 18:34.40 | Katty | exten => _1xxxxxxxxxx,1,Dial,Zap/g1/ww${EXTEN} <- that's my exact line. |
| 18:34.53 | Katty | do I put my () around the Zap? |
| 18:35.01 | [TK]D-Fender | Katty: exten => _1xxxxxxxxxx,1,Dial,Zap/g1/ww${EXTEN},,W |
| 18:35.06 | Katty | okay |
| 18:35.26 | gmfm | Katty: why do you precede your Zap dialstring with ww? |
| 18:35.36 | [TK]D-Fender | Katty: (to maintain your format, which I would suggest changing, but for this 1 line.... what the heck. Leav cleanup for later. |
| 18:35.47 | Katty | gmfm: because we used plain ole phone lines, and we had to wait for a dialtone first before dialing |
| 18:35.55 | [TK]D-Fender | gmfm: Clearly a delay is required before dialing. |
| 18:35.57 | frenzy | LOL |
| 18:36.13 | gmfm | Katty: makes sense, i guess... slow dialtone eh? |
| 18:36.27 | [TK]D-Fender | novafirst: Please should me the "ls-L" of your MP3 folder |
| 18:36.37 | Katty | gmfm: i suppose, all i know is if i don't ww, the phone company thinks i want to do a call trace |
| 18:36.48 | jbalcomb | Un1x: nothing really, thats a 'man ps' or google.com/linux question really. not much to do with asterisk really. |
| 18:36.49 | [TK]D-Fender | Katty: Don't scare the general populace with stories of your channel-banked setup ;) |
| 18:36.51 | frenzy | never heard of a phone line that tequires you to waif for a dial tone before dialing |
| 18:36.57 | Katty | [TK]D-Fender: hehe okay |
| 18:37.07 | Katty | [TK]D-Fender: SHH, don't tell! we really have a t1 ;) |
| 18:37.28 | [TK]D-Fender | Katty: Buied so deep AT&T would have trouble proving it? ;) |
| 18:37.32 | Katty | aww. it's still not recording. |
| 18:37.51 | Katty | let me make sure i got the format right |
| 18:37.52 | [TK]D-Fender | Katty: we're almost there... show... |
| 18:38.26 | novafirst | [TK]D-Fender: -rw-rw---- 1 asterisk asterisk 402048 Aug 28 13:59 clasic_01.mp3 |
| 18:38.51 | Katty | [TK]D-Fender: http://pastebin.ca/152324 |
| 18:38.57 | [TK]D-Fender | novafirst: Looks kinda small.... |
| 18:39.13 | Givemelove2k | Guys |
| 18:39.21 | Givemelove2k | Anybody know how to change the ptime? |
| 18:39.24 | jbalcomb | [TK]D-Fender: could be about 40 seconds long |
| 18:39.31 | gmfm | Katty: you need another comma |
| 18:39.32 | novafirst | [TK]D-Fender: let me test mp3 file |
| 18:39.36 | Katty | [TK]D-Fender: when i dial an in house one... it shows ||wW on the end |
| 18:39.41 | [TK]D-Fender | jbalcomb: My trust begins with larger increments :) |
| 18:39.48 | [TK]D-Fender | novafirst: BRILLIANT idea :) |
| 18:39.51 | jbalcomb | [TK]D-Fender: ;) as it should |
| 18:39.56 | Katty | gmfm: ah ha! |
| 18:40.07 | [TK]D-Fender | Katty: Missed the double comma I put in there ;) |
| 18:40.16 | Katty | [TK]D-Fender: sorry, had a blonde moment. |
| 18:40.21 | [TK]D-Fender | Katty: You need to pybass the timeout :) |
| 18:40.24 | gmfm | Katty: check out 'show application Dial' and it'll tell you all about it :) |
| 18:40.27 | sx-wks | hah! of course, mpg123-0.59-r doesn't compile on this amd64 box :D |
| 18:40.43 | Katty | [TK]D-Fender: mew? |
| 18:40.47 | Katty | [TK]D-Fender: your comment did not parse. |
| 18:40.56 | jbalcomb | sx-wks: mpg123 is defunt, use madplay |
| 18:41.00 | [TK]D-Fender | s/pybass/bypass/ |
| 18:41.12 | Qwell | mmm...pie pass |
| 18:41.16 | [TK]D-Fender | Katty: Pig-IRC speak! |
| 18:41.17 | Katty | [TK]D-Fender: that still doesn't parse )= |
| 18:41.46 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 18:41.46 | Katty | [TK]D-Fender: and! it still doesn't record )= |
| 18:41.54 | [TK]D-Fender | Katty: exten => _1xxxxxxxxxx,1,Dial,Zap/g1/ww${EXTEN},,W <- 2 consecutive commas to skip the timeout parm. |
| 18:41.54 | sx-wks | jbalcomb: how do I configure this in asterisk 1.2.11 ? |
| 18:42.12 | jbalcomb | sx-wks: |
| 18:42.14 | Katty | [TK]D-Fender: http://pastebin.ca/152334 |
| 18:42.23 | Katty | this will work if it's the last thing i do today! |
| 18:42.27 | novafirst | [TK]D-Fender: mp3 plays fine |
| 18:42.41 | jbalcomb | sx-wks: Go here: http://www.voip-info.org/wiki/index.php?page=Asterisk and search for madplay |
| 18:42.44 | Qwell | Katty: Are you also hitting the key combo very quickly? |
| 18:42.48 | Qwell | By default, you've got 500ms |
| 18:42.49 | sx-wks | jbalcomb: anyhow, I don't really care, I have a nice 160G empty drive for MOH :D |
| 18:42.54 | Katty | Qwell: yes, i know how to record a call. |
| 18:43.03 | Katty | Qwell: the internal 4 digit catch all works fine. |
| 18:43.09 | Qwell | oh |
| 18:43.10 | novafirst | [TK]D-Fender: do I need to define musiconhold=default? |
| 18:43.13 | [TK]D-Fender | Katty: try "wW" |
| 18:43.34 | [TK]D-Fender | novafirst: You need to make sure ton context exists as its being shown in your CLI output. |
| 18:43.47 | Katty | [TK]D-Fender: but that's both ways. |
| 18:43.50 | Katty | [TK]D-Fender: look at this... |
| 18:44.00 | [TK]D-Fender | Katty: exten => _1xxxxxxxxxx,1,Dia(Zap/g1/ww${EXTEN},,W) |
| 18:44.01 | Katty | [TK]D-Fender: exten => _xxxx,1,Dial(SIP/${EXTEN},20,wW) |
| 18:44.10 | Katty | [TK]D-Fender: yeah, () |
| 18:44.15 | [TK]D-Fender | Katty: exten => _1xxxxxxxxxx,1,Dial(Zap/g1/ww${EXTEN},,W) |
| 18:44.23 | [TK]D-Fender | Katty: that last one really should work. |
| 18:44.29 | novafirst | [TK]D-Fender: please explain? |
| 18:45.01 | [TK]D-Fender | novafirst: Make sure the context you see being called in CLI when you press hold exists and is configured to match in musiconhold.conf |
| 18:45.10 | Katty | ah ha, now we have pipes |
| 18:45.30 | Katty | YAY |
| 18:45.40 | sx-wks | lol |
| 18:45.49 | Katty | [TK]D-Fender: all better, thank you (= |
| 18:46.01 | file | Katty: ! |
| 18:46.06 | Katty | file: ! |
| 18:46.25 | novafirst | [TK]D-Fender: Started music on hold, class 'default', on SIP/lite_01-095bc068 |
| 18:47.58 | *** join/#asterisk okdo (n=goldenol@26.sub-70-215-240.myvzw.com) |
| 18:47.59 | okdo | hi |
| 18:48.17 | okdo | should I be able to use a straight patch cable for a PRI or do I need a special cable? |
| 18:49.18 | mountainm2k | Having a hard time making Page() work with Polycom... |
| 18:49.30 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 18:49.41 | mountainm2k | Can't seem to get the phone to auto-answer, it rings instead... |
| 18:49.59 | mountainm2k | okdo: PRI == T1 |
| 18:50.10 | [TK]D-Fender | novafirst: please pastebin your nusiconhold.conf again, and please use .ca this time... |
| 18:50.14 | mountainm2k | pins 1, 2, 4 and 5 on an RJ45 |
| 18:50.20 | mountainm2k | ethernet cable works |
| 18:50.49 | novafirst | [TK]D-Fender: .ca ? |
| 18:51.03 | [TK]D-Fender | novafirst: pastebin.ca. pastbin.com is screwed up |
| 18:51.08 | novafirst | ok |
| 18:51.54 | sx-wks | there's also rafb.net/paster |
| 18:51.55 | sx-wks | there's also rafb.net/paste |
| 18:51.57 | sx-wks | even |
| 18:52.59 | novafirst | [TK]D-Fender: http://pastebin.ca/152345 |
| 18:53.36 | Katty | file: /me hugs on [TK]D-Fender |
| 18:53.39 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-32-121.phlapa.east.verizon.net) |
| 18:53.39 | Katty | oh. |
| 18:53.41 | Katty | that's just peachy. |
| 18:53.50 | [TK]D-Fender | novafirst: Ok, it all LOOKS kosher assuming priveledges aren't an issue and the folder is indeed correct and the fines in good shape.... |
| 18:53.59 | Katty | file: see what happens when i get called away from my desk! |
| 18:54.03 | Katty | file: i loose my train of thought! |
| 18:54.08 | file | indeed, it derailed |
| 18:54.10 | Qwell | lose |
| 18:54.21 | Katty | Qwell: thank you, mister english major. |
| 18:54.35 | novafirst | [TK]D-Fender: asterisk:asterisk rwX rwX |
| 18:54.37 | Katty | Qwell: that's really sad, actually, because i was going to be an english major ;) |
| 18:54.47 | Qwell | :P |
| 18:55.05 | [TK]D-Fender | I'm a cunning linguist ;) |
| 18:55.06 | sx-wks | Katty: now you're an asterisk major :D |
| 18:57.35 | novafirst | [TK]D-Fender: maybe x-lite is not capable of musiconhold ? |
| 18:57.57 | Katty | sx-wks: oh god )= |
| 18:59.30 | novafirst | [TK]D-Fender: since VoiceMail is working fine but it's using gsm codec and I think x-lite doesn't have mpr3 codec |
| 19:00.27 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 19:00.32 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
| 19:00.58 | kratzers | anybody patch 1.2 sources against AEL2 1.2 changes? |
| 19:01.40 | *** join/#asterisk isbo (n=isbo@csil.cs.ucsb.edu) |
| 19:02.05 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 19:02.08 | kratzers | can't get it to work for some reason |
| 19:02.13 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 19:02.20 | isbo | hi! has anybody worked with libiax2? |
| 19:03.11 | kratzers | patching and compiling go fine, but asterisk won't load the AEL2 module |
| 19:03.24 | kratzers | maybe I'm just doing something retarded |
| 19:03.31 | dhill | i have a PRI that feeds a Cisco As5300 which feeds sip to asterisk.. which feeds sip to sipura devices. Some Sipura users can hear themselves because of an echo.. but to the outsiders.. it is crystal clear. Am I looking at an asterisk setting or a Cisco setting to stop the echo for the sipura users? |
| 19:04.00 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 19:04.59 | dhill | hard to debug since it is not constant... |
| 19:08.38 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 19:12.28 | *** join/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 19:14.51 | *** join/#asterisk num000 (n=numerobi@e177186067.adsl.alicedsl.de) |
| 19:15.46 | num000 | when doing a call my asterisk crashes with a segmentation fault, -- Attempting native bridge of SIP/100-326f and SIP/4316090-a256 |
| 19:16.00 | num000 | Urgent handler, Segmentation fault. |
| 19:16.12 | num000 | has anyone got any ideas where to search for the problem? |
| 19:17.20 | mercestes | So someone tell me all about * Business Edition. |
| 19:18.15 | *** join/#asterisk |dennis| (n=dennis@etcdsl1.btl.net) |
| 19:21.51 | num000 | does anyone know what asterisk means with this message in his logfiles? Jan 1 05:05:36 WARNING[528]: Failed to write frame |
| 19:22.31 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:23.59 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:24.17 | *** join/#asterisk Telamon (i=telamon@blk-222-23-213.eastlink.ca) |
| 19:24.33 | Telamon | I'm having some problems with echo on incoming calls from a PRI. I've played around with different echo cancellation algorithms and training settings with no real luck. Anyone have some ideas as to what I can do to track the problem down? |
| 19:25.15 | *** join/#asterisk bethaud (n=bethaud@host-84-9-82-198.bulldogdsl.com) |
| 19:26.17 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:27.36 | *** join/#asterisk Lyfe (n=lyfe@69.8.146.78) |
| 19:27.42 | *** join/#asterisk new2voip (n=new2voip@secure.kayacorp.com) |
| 19:27.47 | murf | kratzers: I patched the branch to work. Try it again and see if it's OK. |
| 19:28.02 | kshyvy | anyone played with SPA942 locale settings? |
| 19:30.38 | murf | kratzers: I'm talking about the AEL2-1.2 branch...! |
| 19:30.57 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 19:31.26 | kratzers | trying now |
| 19:31.33 | _DAW | Hi.. I have asterisk between my phone and sip gateway. When I press hold the rtp stream stops between the phone and * however no reinvite is sent between * and the gateway and the rtp stream stays up in one direction. Am I missing something in the config? |
| 19:32.53 | bkw__ | chances are no |
| 19:34.51 | *** join/#asterisk adorah (n=Administ@84.94.147.85.cable.012.net.il) |
| 19:36.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-96-211.red.bezeqint.net) |
| 19:37.29 | *** join/#asterisk jooray (n=jooray@smtp.rebelium.net) |
| 19:37.35 | *** part/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 19:38.03 | _DAW | is there a way to tell * to send the reinvite? |
| 19:38.19 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 19:38.19 | jooray | hello. I would like to use group() to limit maximum number of concurrent calls for an IAX2 channel, which is dialed from queue. Anyone has precise idea how it works? |
| 19:38.41 | jooray | _DAW: set canreinvite=yes in your peer/friend's sip.conf |
| 19:38.45 | rene1 | hello, does digium boards work with APIC? |
| 19:39.41 | jooray | or is there a way to limit number of concurrent calls for iax users? there's call-limit in SIP, but I can't see or find anything like that in iax config. |
| 19:39.48 | jooray | ? |
| 19:41.09 | new2voip | How does * interface with legacy PBX uisng a T1? Do I need to buy a card for * or for my legacy pbx? |
| 19:41.42 | rene1 | you need to have one T1 port in asterisk and one in your legacy PBX |
| 19:41.49 | kratzers | murf, the following is what I do and the results that I get. I'm probably just doing something profoundly stupid. http://pastebin.com/778219 |
| 19:41.55 | rene1 | new2voip: |
| 19:42.03 | jooray | new2voip: depends on signalling. T1 is just a line. you may buy voip module for pbx or T1 (supporting your signalling) into * or even an external box, that will do the conversion for you and deliver it via sip |
| 19:42.27 | rene1 | that probably means you want a 2 port T1 card for asterisk and use your existing pbx T1 port to connecto to asterisk, the other asterisk T1 port goes to telco |
| 19:43.08 | jooray | i found that external T1 boxes are more reliable, cheaper |
| 19:43.25 | *** join/#asterisk |dennis| (n=dennis@etcdsl1.btl.net) |
| 19:43.26 | jooray | and if you have spare server, you can more easily do clustering |
| 19:43.40 | jooray | but that's just my opinion, you can do better magic with a card |
| 19:44.44 | new2voip | can you give mfg for ext t1 boxes and I do have a spare server. |
| 19:45.08 | jooray | i'm in europe, so we use e1, not t1 lines, so I won't help on that |
| 19:45.53 | new2voip | k, will do some searching... I am down the street from digium think they will let me test a card. hehe |
| 19:49.11 | file | sneak in... |
| 19:50.17 | *** part/#asterisk MACscr (n=MACscr@66.73.154.70) |
| 19:50.25 | jooray | oh, help me, how do I limit number of incoming calls to iax channel |
| 19:50.31 | jooray | ? |
| 19:51.21 | num000 | my asterisk crashes with a segmentation fault, wenn i make calls outside it gives the following error to the messages file: 05:26:51 WARNING[612]: Failed to write frame, does anyone know where to start solving the problem? |
| 19:51.56 | jooray | num000: what version of asterisk do you have? |
| 19:52.12 | Telamon | jooray: I do it in the dialplan with GROUPCOUNT, don't know if that will help you. Just in your "exten => #,1,Dial" block put in a priority 1 if ($(groupcount) > max_calls) then Set(DIALSTATUS=CHANUNAVAIL) |
| 19:52.35 | num000 | jooray it is 1.0.10 on an openwrt |
| 19:52.56 | ido | ooh |
| 19:53.25 | ido | is openwrt + asterisk + festival + sphinx possible on a linksys wrt54g? (heck, is openwrt+asterisk possible on a wrt54g?) |
| 19:53.51 | jooray | ido: it is, but it is very slow for festival and sphinx |
| 19:53.58 | num000 | ido thats what i'm trying all the time, echo tests do work well, calling inside aswell |
| 19:53.59 | *** join/#asterisk ComputerWarm (n=dan@h188.43.63.69.cable.ottr.cablerocket.net) |
| 19:54.01 | jooray | num000: try to upgrade, it is a very old version |
| 19:54.02 | ComputerWarm | hello all |
| 19:54.13 | num000 | jooray which version would you recommend? |
| 19:54.27 | jooray | num000: 1.2.11, but I don't know if it's been ported to wrt yet |
| 19:54.31 | num000 | jooray how much memory does asterisk need? |
| 19:54.36 | ido | num000: i think my router is the only box stable enough and with a low enough power profile to actually remain up during a power outage with the ups... |
| 19:54.46 | ido | i am tempted to openwrtize it and make it a server. |
| 19:54.48 | jooray | num000: I don't know, I ran it on openwrt for few days only |
| 19:54.51 | ComputerWarm | is there anyway to see a manager connection from cli? or even the attempt so i can figure out why its not connecting |
| 19:55.00 | ido | jooray: have you done it before? |
| 19:55.01 | num000 | jooray are you running it successfuly? |
| 19:55.09 | jooray | ido: yes |
| 19:55.13 | jooray | ido: but, heck, it's really slow |
| 19:55.24 | ido | jooray: explain slow |
| 19:55.29 | jooray | num000: it was running successfuly, I got rid of it, I didn't need it, I was just playing |
| 19:55.53 | jooray | num000: anyway, if you want to give 1.2.11 a try, google for debian install howto |
| 19:55.59 | num000 | jooray actually i'm running so far the echotest and also did calling from outside to the extension which did work very well |
| 19:56.03 | jooray | num000: i recommend installing debian on a usb flash disk |
| 19:56.07 | jooray | num000: mips version will run on wrt |
| 19:56.11 | jooray | num000: there are lots of howtos |
| 19:56.18 | num000 | jooray but i do not have usb flash disc |
| 19:56.18 | jooray | num000: then you can use 1.2.11 from unstable possibly |
| 19:56.20 | kratzers | ComputerWarm, show manager connected |
| 19:56.29 | ido | jooray: what do you mean by it is slow? is it responsive enough to route calls on the fly? |
| 19:56.30 | jooray | num000: or any usb storage device |
| 19:56.44 | jooray | ido: route yes, but I would not let it transcode anything |
| 19:56.47 | num000 | jooray the wrt54gl does not have usb at all |
| 19:56.52 | jooray | ah |
| 19:56.55 | jooray | sorry :) |
| 19:57.02 | num000 | no problem, |
| 19:57.03 | ido | jooray: i see...so no voip then? :-/ |
| 19:57.05 | jooray | I have asus wl500gd, it has two usb 2.0 ports :) |
| 19:57.13 | jooray | ido: install openwrt's ipkg |
| 19:57.22 | ido | jooray: ipkg? |
| 19:57.25 | num000 | jooray i suppose the asus is the better choice |
| 19:57.35 | ido | i see |
| 19:57.43 | num000 | ido ipkg is the paackage manager on wrt |
| 19:58.17 | ido | k |
| 19:58.56 | ido | wow thanks num000 and jooray. off i go! |
| 19:59.01 | ComputerWarm | kratzers ya but that doesn`t show the reason it failed does it? |
| 20:00.09 | kratzers | nope |
| 20:00.35 | ComputerWarm | kratzers is there anything that does? |
| 20:01.47 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 20:02.15 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 20:03.00 | *** join/#asterisk ComputerWarm (n=dan@h188.43.63.69.cable.ottr.cablerocket.net) |
| 20:03.06 | ComputerWarm | sorry lost connection |
| 20:03.17 | kratzers | did you try to connect locally? |
| 20:03.25 | ComputerWarm | ya but through a script |
| 20:03.30 | ComputerWarm | not sure how to do it through telnet |
| 20:03.30 | Nugget | telnet is eeeeeeevil! |
| 20:03.39 | kratzers | just try 'telnet localhost 5038' |
| 20:03.40 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 20:04.25 | ComputerWarm | kratzers ya i can connect that way |
| 20:04.44 | kratzers | well then you know the server side is functioning OK |
| 20:04.47 | *** part/#asterisk svenna_ (n=svenna@p548D3184.dip0.t-ipconnect.de) |
| 20:05.06 | ComputerWarm | kratzers how do i get out of that? |
| 20:05.13 | kratzers | ctrl + ] |
| 20:05.16 | kratzers | then type close |
| 20:05.29 | kratzers | bah |
| 20:05.43 | ComputerWarm | ty |
| 20:05.59 | new2voip | tell me if I understand this correctly... I setup and * box to use a voip pstn service then I woul not have to buy and cards for the * box because the calls are traveling over the data network to the voip provider? Is that accurate? |
| 20:06.10 | kratzers | ACTION: LOGOFF |
| 20:07.22 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 20:12.07 | Qwell | new2voip: yes |
| 20:12.47 | new2voip | thanks qwell |
| 20:16.18 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 20:17.07 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:17.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:17.10 | _Thor | hello everyone |
| 20:17.36 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:17.44 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 20:18.55 | murf | kratzers: SOrry for delay-- looked at pastebin, no, you aren't wrong. the svn.digium.com server wasn't updating, it appears, from the origsvn.digium.com server... just within the last few minutes, I can finally see the update (41233). |
| 20:19.35 | _Thor | Anybody can give me pointers on how to port a verizon number to voip?? |
| 20:19.59 | *** join/#asterisk pjz (n=pj@zachs.place.org) |
| 20:20.11 | murf | kratzers: Try again; do an svn update, and see if you pull in a new copy of pbx_ael2.c; if all goes well, let me know. |
| 20:20.30 | pjz | so anyone know of a debian stable version of asterisk mroe recent than 1.0.7 ? |
| 20:20.43 | murf | kratzers: So I can stop worrying! ;) |
| 20:20.52 | pjz | I'd like to be able to just add something to my sources.list that will stay moderately up to date |
| 20:26.42 | _Thor | Does anybody know how to port a number from verizon? |
| 20:27.53 | C6Vette | _Thor: you have to have another provider request it. |
| 20:28.36 | *** part/#asterisk new2voip (n=new2voip@secure.kayacorp.com) |
| 20:28.45 | C6Vette | Thor, Who is the new service provider you are moving it to? |
| 20:33.07 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 20:33.45 | Assid | .11 ?? |
| 20:34.00 | Assid | oh yeah |
| 20:34.59 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:37.06 | *** part/#asterisk schuylerdigium (n=bosco@gateway.digium.com) |
| 20:37.22 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 20:37.22 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:38.28 | *** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee) |
| 20:39.06 | sx-wks | haha. I have set up my freephonie sip account... this is fun... |
| 20:39.39 | sx-wks | guess I could redirect calls now :d |
| 20:39.53 | sx-wks | for instance, to work when I'm there ;D |
| 20:40.00 | sx-wks | yeehah |
| 20:42.46 | _Thor | C6Vette: actually, I would want to provide the service with asterisk, it is for a customer I am setting an asterisk pbx |
| 20:43.11 | _Thor | C6Vette: what are the steps? |
| 20:43.14 | C6Vette | You still need to have a provider. |
| 20:43.33 | _Thor | C6Vette: I would be provider |
| 20:43.34 | C6Vette | The number has to be routed to someone. by somebody |
| 20:43.43 | *** join/#asterisk ^EiEiOh^ (n=eieiohh@adsl-68-78-137-136.dsl.emhril.ameritech.net) |
| 20:43.58 | _Thor | C6Vette: I will route the number |
| 20:44.05 | *** join/#asterisk j0 (n=dan@CABLE-72-53-45-212.cia.com) |
| 20:44.10 | C6Vette | Your not a phone company right? |
| 20:44.30 | benjk_ | lol |
| 20:45.10 | KDan | _Thor is a switch. |
| 20:45.12 | benjk_ | sounds like a great idea |
| 20:45.23 | benjk_ | I am going to open my own restaurant |
| 20:45.25 | tzafrir_laptop | Unrelated question: is $(ARCH) supposed to be 'i386' on an amd64 system? (Sarge amd64. uname -a says x86_64) |
| 20:45.36 | ToTo | some listen pantera? |
| 20:45.37 | benjk_ | cause then I will not have to buy any food ever again |
| 20:45.41 | _Thor | C6Vette: what defines a phone company?, do you mean if I have the fcc214?, or if I am a CLEC? |
| 20:45.41 | ToTo | someone.. |
| 20:46.00 | tzafrir_laptop | benjk_, actually you'll have to buy tons of food |
| 20:46.10 | benjk_ | precisely my point |
| 20:46.10 | eKo1 | tzafrir_laptop: x86_64 is NOT i386 |
| 20:46.27 | ^EiEiOh^ | Hello... Would this be a forum to ask for help with an Asterisk question? |
| 20:46.42 | Strom_C | yes |
| 20:46.58 | tzafrir_laptop | eKo1, sadly, make sets ARCH to i386 for a reason I fail to understand |
| 20:47.05 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 20:47.51 | ^EiEiOh^ | thank you.. New to * and Linux. I finally got * running but, it takes +- 20 seconds for the caller to hear ringing.. Is this normal? |
| 20:48.02 | *** join/#asterisk Bladerunner05 (n=feelme@adsl-20-214.37-151.net24.it) |
| 20:48.34 | *** join/#asterisk si_alchemist (n=sialchem@pool-71-162-118-38.bstnma.fios.verizon.net) |
| 20:48.48 | tzafrir_laptop | ^EiEiOh^, what sort of call? from what device? |
| 20:49.13 | _Thor | C6Vette: look, I don't believe the "you have to be a phone company" crap, <g> <g>, does anybody knows where in verizon I have to call to get my number ported to my asterisk server?? |
| 20:49.22 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:49.30 | _Thor | btw, please |
| 20:50.15 | ^EiEiOh^ | I've configured SIP Trunks (VoiceEclipse)... When I try calling from a pots line, thats when I get the delay. My worry is that a caller might not be patient and hang up |
| 20:50.22 | gbodemantv | hey all |
| 20:50.26 | si_alchemist | any asterisk developers here? |
| 20:50.31 | gbodemantv | so I have a bizarre issue here |
| 20:50.45 | file | si_alchemist: ...maybe |
| 20:50.55 | mog | perhaps |
| 20:50.57 | si_alchemist | i am looking for some insight on the feasibility of writing a patch to sip.h such that Contact: is an over-ride option in sip.conf |
| 20:51.02 | gbodemantv | I am having many users complain that when they call from our custom app it never reaches the sip phone |
| 20:51.06 | si_alchemist | we are pretty much at our wit's end trying to do it with a proxy |
| 20:51.10 | gbodemantv | I am using Realtime Mysql |
| 20:51.34 | gbodemantv | all of these call attempts show up in the CDR with Disposition FAILED |
| 20:51.38 | C6Vette | Ok,_Thor, AT&T owns ALL 800 numbers and they arenot gonna rought to your asterisk. |
| 20:51.50 | gbodemantv | this seems to last for a few minutes |
| 20:51.54 | C6Vette | s/rought/route/ |
| 20:51.56 | *** join/#asterisk docelm0 (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 20:52.44 | Qwell | uhh |
| 20:52.56 | Qwell | AT&T does not own all 800 numbers |
| 20:53.05 | Strom_C | maybe if this were still 1977 |
| 20:53.09 | gbodemantv | but is very problematic in a call center environment |
| 20:53.09 | C6Vette | They own the patent |
| 20:53.12 | si_alchemist | Reach out and monopolize someone |
| 20:53.18 | Qwell | there is no patent on 800 numbers |
| 20:53.23 | benjk_ | haha |
| 20:53.26 | si_alchemist | file, mog, any comments on my intentions to write a patch for sip.h? |
| 20:53.29 | Strom_C | C6Vette: which bonkers alternate universe are you living in? |
| 20:53.33 | Qwell | and if there is, it was shot down 40 years ago |
| 20:53.38 | C6Vette | Roy Weber, a wry 52-year-old AT&T scientist, is the man responsible for 1-800 Nation. "The patent has my name on it, but AT&T owns it,". "Otherwise, Bill Gates would the second-richest person in America." |
| 20:53.45 | si_alchemist | i desperately need the ability to re-write contact: header without munging the To: header in the process |
| 20:53.47 | mog | why sip.h ? |
| 20:53.55 | si_alchemist | because that's where sip headers seem defined? |
| 20:54.06 | si_alchemist | i am probably betraying deep ignorance here |
| 20:54.12 | benjk_ | that doesn't mean there is a patent on the 800 numbers |
| 20:54.37 | Qwell | http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO2&Sect2=HITOFF&p=1&u=%2Fnetahtml%2FPTO%2Fsearch-bool.html&r=0&f=S&l=50&TERM1=Roy+Weber&FIELD1=&co1=AND&TERM2=&FIELD2=&d=PTXT |
| 20:54.40 | benjk_ | its a patent on some kind of process or apparatus needed in combination with providing those numbers or routing them |
| 20:54.43 | Qwell | Those are the patents with Roy Webers name on it |
| 20:55.14 | _Thor | C6Vette: I don't want to port an 800, I am porting a 212 DID |
| 20:55.33 | si_alchemist | 212 is in michigan or something |
| 20:55.35 | si_alchemist | isn't it? |
| 20:55.36 | C6Vette | Well the bottom line is he isnt gonna get the numbers directly routed to his asterisk box, by calling Verizon as he requested |
| 20:55.46 | Qwell | sure he can |
| 20:55.46 | si_alchemist | yeah that's for sure |
| 20:55.51 | _Thor | 212 is New York, NY |
| 20:56.00 | si_alchemist | we've found it easier to just get our own incoming PRIs through a CLEC |
| 20:56.05 | si_alchemist | and then route to asterisk from there |
| 20:56.06 | Qwell | call Verizon, and say "I want an 800 DID on my line" |
| 20:56.10 | si_alchemist | this obviously only makes sense for larger applications |
| 20:56.13 | Strom_C | 212 is Manhattan |
| 20:56.19 | _Thor | C6Vette: of course I can, it is the law |
| 20:57.19 | C6Vette | Yes you can have them port a 800 number to a DID. That makes sense |
| 20:57.59 | benjk_ | still, not even the biggest telco can route all the calls themselves |
| 20:58.12 | _Thor | C6Vette: Let me back up here: In mobile phones, if I want to change my celluar carrier, say from sprint to Verizon, they can do it... It is the same thing with landlines |
| 20:58.15 | benjk_ | there will always be calls that they need to pass on to other telcos |
| 20:58.23 | benjk_ | and that was the point |
| 20:58.56 | _Thor | C6Vette: they have to give me the DID if I request it |
| 20:58.59 | C6Vette | _thor, correct, but 800 is a little different |
| 20:59.08 | benjk_ | Thor wants to route everything himself as it would seem, never need any provider |
| 20:59.11 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 20:59.13 | benjk_ | that didn't make sense |
| 20:59.19 | gbodemantv | Anybody have experience with this issue |
| 20:59.24 | key2 | hey |
| 20:59.25 | gbodemantv | can't find much about it online |
| 20:59.29 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 20:59.30 | key2 | is there someone around that has a 3G phone ? |
| 20:59.44 | Assid | anyone know any free sms providers.. those 10 sms /day type credits |
| 21:00.08 | _Thor | benjk_: do you really understand what we are talking about? |
| 21:00.45 | Un1x | someone help me i wanna find some kind of externel case or something to puit my PCI card in |
| 21:00.51 | benjk_ | what I understand is that somebody said you will still need to interconnect with some provider and you said you won't |
| 21:00.55 | C6Vette | benjk_, that is what I was thinking he was meaning |
| 21:00.56 | benjk_ | that doesn't make sense |
| 21:01.01 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 21:01.32 | benjk_ | even Primus have interconnects with other telcos |
| 21:01.44 | benjk_ | they own most of the sea cable capacity on the planet |
| 21:02.46 | C6Vette | _Thor, what you CAN do is call ANY phone company, Verizon, Broadwing, who ever and have them pull the number from who ever has the number now and forward it to another DID number |
| 21:03.35 | _Thor | I want Verizon to give me the DID that they have on my landline, because I want to put it on my asterisk box |
| 21:04.16 | C6Vette | Verizon does not support asterisk. they wont port to asterisk, they CAN port to a DID number |
| 21:04.52 | docelm0 | thor if you want your DID ported to VOIP you will need to get with a carrier that does this |
| 21:05.05 | docelm0 | There are many but you need to check and see if they can do your number. |
| 21:05.20 | docelm0 | otherwise buy a TDM card for your asterisk box |
| 21:05.36 | docelm0 | C6Vette yes Verizon does.. Via TDM interface. |
| 21:05.45 | _Thor | C6Vette: Great, agree, who will know the phone number or web link I can go to, to ask verizon to forward the DID to another carrier? |
| 21:05.45 | C6Vette | well yea... |
| 21:05.58 | C6Vette | Just call customer service |
| 21:05.59 | docelm0 | You dont ask verizon |
| 21:06.34 | docelm0 | You need to call a carrier that can port your number.. They will send you a LOA for you to sign. Then they can begin the porting process.. Sometimes it takes a week but could be up to 1 month |
| 21:06.35 | C6Vette | You can ask them |
| 21:06.59 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:07.15 | C6Vette | They WONT port directly to asteirsk they will, as docelm0 says, to a tdm interface card. |
| 21:07.36 | benjk_ | they will also port to an SS7 node if you have it |
| 21:07.49 | benjk_ | which could be an * box |
| 21:08.08 | docelmo | benjk_ if you wanna be technical.. |
| 21:08.13 | _Thor | C6Vette: asking asterisk customer service will be like asking a wall, they are morons... meaning no disrespect to verizon people <g> |
| 21:08.31 | docelmo | but this guy seems like he doesnt know what the underlying foundation of the phone network is |
| 21:08.41 | _Thor | C6Vette: sorry, I meant: asking verizon customer service is useless |
| 21:08.52 | syzygyBSD | or 5 months if you are trying to get it from cingular |
| 21:08.55 | docelmo | sigh.. have you not been paying attention to what I told you? |
| 21:09.31 | _Thor | docelmo: for your information, I am a voip consultant with 14 years in the business |
| 21:09.44 | docelmo | Then you should fucking know how to do this then |
| 21:09.46 | syzygyBSD | ohh... so when i get SS7 I won't have to deal with my current PRI provider? |
| 21:09.52 | C6Vette | I have 50+ 800 numbers going to my asterisk boxes. |
| 21:10.00 | Un1x | hey People i need help please!, i'm trying to get TDM400P to work with my laptop any suggestions for some type of externel closure or something |
| 21:10.05 | syzygyBSD | _Thor: voip has been around for 14 years? |
| 21:10.06 | _Thor | docelmo: I will like to know how much you fucking know |
| 21:10.19 | docelmo | and thor.. 14 years? MAYBE 5 or 10 if you strech it |
| 21:10.46 | docelmo | What do you want to know? I have 6 years of hard core telecom 4 years VOIP and over 10 years network engineering.. |
| 21:11.16 | docelmo | I have worked with SS7, IMT, PRI, SIP, IAX, H323, etc |
| 21:11.23 | Un1x | docelmo to be honest years dont matter |
| 21:11.32 | benjk_ | 14 years of VoIP? hmmm, let me see, 2006-14=1992 |
| 21:11.34 | docelmo | I have administered DMS100's, 5ess's etc |
| 21:11.44 | benjk_ | not sure there was any VoIP back then |
| 21:11.56 | Un1x | people have been using windows for the last what 10 years and yet still till this day only certain percentage of them still learn any indephth stuff about windows |
| 21:12.01 | docelmo | There wasnt.. Cisco's first release of a voip product was the AS5300 |
| 21:12.16 | docelmo | Un1x you are correct |
| 21:12.27 | Doctor | hey for an FXO on linux, would the Digium X100p be the best value? |
| 21:12.36 | Un1x | i need help man |
| 21:12.40 | Un1x | i need some kind of encloser |
| 21:12.41 | docelmo | cheapest.. yes |
| 21:12.43 | Un1x | for the tdm400p |
| 21:12.49 | benjk_ | Doctor, no |
| 21:12.58 | Un1x | benjk hey you're around help lol |
| 21:13.13 | _Thor | Look docelmo, can you contribute to my question?, otherwise you should not get involved |
| 21:13.15 | docelmo | Un1x what do you mean elclosure? |
| 21:13.21 | benjk_ | Un1x, how did your thing go with asteria? |
| 21:13.28 | Un1x | not well |
| 21:13.30 | sx-wks | Doctor: x100p are deprecated. |
| 21:13.32 | docelmo | SIGH look ass scroll back I told you what to do |
| 21:13.37 | Un1x | docelmo i need to be able to plug in my tdm400p to it |
| 21:13.43 | Un1x | so i can use it with asterisk on my laptop :D |
| 21:13.57 | docelmo | Un1x I dont think there is such a device |
| 21:14.00 | benjk_ | Un1x, what do you mean "not well" ? |
| 21:14.07 | _Thor | and for anyone interested, 14 years ago people were doing voice over networks |
| 21:14.07 | Doctor | oic so thats why the toms hardware said $10-$20 and I checked ebay and saw more than that |
| 21:14.21 | docelmo | Ya. I was doing Voice over PRI's too |
| 21:14.31 | benjk_ | Doctor, forget about those |
| 21:14.42 | sx-wks | Un1x: hmmm. get yourself one of those astribank USB channel banks |
| 21:14.45 | Un1x | well i got a number from swk |
| 21:14.47 | Un1x | but i lost it ;( |
| 21:14.55 | Un1x | sx-wks link? |
| 21:14.55 | eKo1 | usb channel banks?!ª |
| 21:15.00 | benjk_ | I was doing Voice over IP in the early 1980s |
| 21:15.01 | _Thor | and besides, I meant 14 years overall telecomo experience |
| 21:15.04 | Un1x | look |
| 21:15.08 | docelmo | Doctor the best bang for your buck would be a A400 from Openvox or Digium TDM400P |
| 21:15.09 | Un1x | docelmo http://www.mobilityelectronics.com/expansion/products/cardbus_expansion/2slot/index.html# |
| 21:15.11 | benjk_ | Voice over International Phonelines |
| 21:15.17 | Un1x | its kinda big and expensive tho 1200$ |
| 21:15.18 | syzygyBSD | udev/modprobe isnt' creating /dev/zap/* fast enough so asterisk is dying on reboot. does anyone know how to fix this? |
| 21:15.19 | Un1x | :( |
| 21:15.34 | docelmo | Un1x no shit.. |
| 21:15.36 | sx-wks | Un1x: http://www.voip-info.org/wiki/view/Xorcom+Astribank |
| 21:15.42 | _Thor | I was doing VSATs in the early 1980s |
| 21:15.45 | Doctor | okay I'll look into those two, thanks docelmo |
| 21:16.04 | Un1x | lol |
| 21:16.19 | Un1x | man i want a frigen way to use tdm400p with my laptop |
| 21:16.27 | Un1x | or a PCMCIA card that has 1fxo and 1 fxs |
| 21:16.30 | benjk_ | why? |
| 21:16.57 | benjk_ | you can always get a Macbook and use the Apple USB modem as an FXO device |
| 21:17.08 | benjk_ | but no FXS, although you can use an ATA |
| 21:17.13 | Un1x | Nah but i still need a FXS. |
| 21:17.28 | benjk_ | ATA with a battery pack |
| 21:17.36 | benjk_ | in fact, you could use a Sipura 3000 |
| 21:17.47 | sx-wks | was gonna say |
| 21:17.47 | benjk_ | build a battery pack for it |
| 21:17.48 | Un1x | how would i connect SPA-3000? CrossoveR? |
| 21:17.54 | benjk_ | yes |
| 21:18.00 | sx-wks | ethernet crossover |
| 21:18.11 | Un1x | hmm you're right i just need to get a crossover cable but what about the battery pack |
| 21:18.17 | hads | syzygyBSD: sleep |
| 21:18.17 | benjk_ | if you don't have mains, you'll need to build a battery pack |
| 21:18.19 | Un1x | and would asterisk still be my main dailplan and stuff? |
| 21:18.27 | benjk_ | of course |
| 21:18.39 | syzygyBSD | :( ya, figures, that just seems like a hack for it |
| 21:18.47 | sx-wks | Un1x: you get the sipura to connect to the asterisk as a sip device |
| 21:19.00 | sx-wks | that was shown on the systm video |
| 21:19.01 | Un1x | i've never done that in my life lol :( |
| 21:19.09 | Un1x | There is a video on how to do it? |
| 21:19.12 | sx-wks | yeah |
| 21:19.27 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 21:20.04 | sx-wks | Un1x: over here : http://revision3.com/systm/asterisk/ |
| 21:20.28 | Un1x | but isn't there something already built i can use something like a PCMCIA card or those extension slots i showed you? |
| 21:20.40 | benjk_ | no |
| 21:21.00 | benjk_ | there may be some such hardware, but no drivers for voice and asterisk |
| 21:21.01 | sx-wks | Un1x: those extension slots are like the atomic bomb to kill flies |
| 21:21.05 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 21:21.16 | Un1x | meh IMHO digium should build a card for laptops i'd pay upto 1000$ for it seeing how i need one bad |
| 21:21.19 | key2 | http://fr.news.yahoo.com/28082006/202/chine-chien-au-volant-accident-au-tournant.html |
| 21:21.24 | key2 | sorry |
| 21:21.25 | Un1x | and going other routes is going to cost me pretty much the same ammount |
| 21:21.37 | benjk_ | you can always buy a BRI PCMCIA card from Odin |
| 21:21.38 | Un1x | sx-wks i seen lol there huge |
| 21:21.42 | benjk_ | its about 600 bucks |
| 21:21.49 | Un1x | i told the lady at customer service if i was going to buy that for 1200$ and it's that big |
| 21:21.54 | Un1x | then whats the point of me buying it period |
| 21:21.57 | Un1x | it's big as my desktop |
| 21:22.06 | Un1x | midas well take that walking around with a batterypack |
| 21:22.08 | sx-wks | the thing appears to be a pc case anyhow :D |
| 21:22.17 | Un1x | yea it's what i said ^^ |
| 21:22.30 | benjk_ | try the Odin PCMCIA card then |
| 21:22.42 | sx-wks | get yourself a sipura and a 12v battery pack :D |
| 21:23.00 | Un1x | benjk_ i dont know anyone or any company that manufactures one |
| 21:23.07 | Un1x | imma try convince Mark to get digium to make :D |
| 21:23.11 | Un1x | it'd be awesome man |
| 21:23.17 | benjk_ | http://www.odints.com/pages/prod/t1e1j1/t2ciacst/t2ciacstfs.htm |
| 21:23.34 | benjk_ | Digium won't be so foolish |
| 21:23.39 | mog | we have one |
| 21:23.42 | mog | ^_^ |
| 21:23.45 | mog | its called the wiaxy |
| 21:23.57 | mog | we have showed it off at shows every now and then |
| 21:24.09 | Qwell | wiaxy? |
| 21:24.14 | sx-wks | benjk_: problem is, you still need a channel bank at the back of that thing |
| 21:24.16 | mog | its just an iaxy in an old at2500 set |
| 21:24.22 | benjk_ | the PCMCIA standard is about to make space for a new one |
| 21:24.24 | Qwell | at2500? :D |
| 21:24.42 | benjk_ | if you are not already in the market with PCMICA products now is not the time to enter it |
| 21:24.45 | Un1x | :| |
| 21:25.08 | Un1x | benjk why not |
| 21:25.10 | mog | http://www.sxc.hu/pic/m/m/mi/mindkillaz/130023_old_telephone.jpg |
| 21:25.12 | Un1x | no ones buying PCMCIA anymore? |
| 21:25.31 | Qwell | oh, that |
| 21:25.34 | benjk_ | sx-wks, they also have a BRI card |
| 21:51.56 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 21:51.56 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.11, Asterisk-addons 1.2.4, and Zaptel 1.2.8 released! (August 22, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 21:51.57 | Un1x | or else research and dev costs wouldn't be there |
| 21:52.04 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 21:52.04 | Un1x | nah not 2k |
| 21:52.05 | Un1x | more ;) |
| 21:52.09 | Un1x | i know it will cost alot more |
| 21:52.12 | rpm | is there an asterisk application to dump all of the current environment variables inside of asterisk to the CLI? such as CALLERIDNUM, EXTEN.. etc. without doing a NoOp() on each seperately? |
| 21:52.15 | Un1x | but immma get some qoutes tomorow |
| 21:52.27 | smackus | dev-asterisk*CLI> sip debug peer 120 |
| 21:52.27 | smackus | Unable to get IP address of peer '120' |
| 21:52.47 | smackus | but isnt this supposed to be debugging 120 on the box i ran sip debug from? |
| 21:52.51 | benjk_ | in any event patents are not what you seem to think |
| 21:52.52 | Mattwj2005 | I prefer to open source it :) |
| 21:52.52 | TheCops | Someone know where can I get reference for Asterisk ?! I mean, some compagny name who is using it stuff like that.. |
| 21:53.49 | Qwell | TheCops: Digium uses it |
| 21:53.54 | benjk_ | lol |
| 21:53.56 | TheCops | I guess :P |
| 21:53.57 | shodan | uh ? a pcmcia fxo ? that's called a pcmcia voice modem duh ! |
| 21:54.05 | TheCops | But my client dont know what's digium |
| 21:54.14 | Mattwj2005 | are you just looking for more information about Asterisk? |
| 21:54.16 | TheCops | I mean, no governement, big compagny or something? |
| 21:54.21 | benjk_ | yeah, but you will have to write drivers for a pcmcia voice modem |
| 21:54.37 | file | Sam Houston University |
| 21:54.39 | lesouvage | TheCops: Free World Dialup is running on Asterisk and Voop can be a IP telephony provider in Europe because of Asterisk. |
| 21:54.41 | benjk_ | Un1x wants a PCMCIA modem that has zaptel drivers |
| 21:54.48 | benjk_ | and there is no such thing |
| 21:54.56 | mog | lots of people use it |
| 21:54.58 | Un1x | no |
| 21:54.59 | Un1x | not zaptel |
| 21:55.01 | C6Vette | cvctermination.com uses asterisk |
| 21:55.01 | shodan | benjk_, yep , but the hardware already exists, so you can't patent it |
| 21:55.02 | Un1x | i need a FXS CARD |
| 21:55.10 | mog | even governments |
| 21:55.20 | benjk_ | fxs will be very tricky on a notebook anyway |
| 21:55.21 | Mattwj2005 | www.voip-info.org has a lot of good information |
| 21:55.25 | lesouvage | TheCops: where are you |
| 21:55.26 | sx-wks | file: that sounds like "WTF university" from thedailywtf.com :D |
| 21:55.30 | Un1x | so there is already a FXO? |
| 21:55.31 | benjk_ | you need to provide a bit of power to the phone |
| 21:55.36 | TheCops | lesouvage, Quebec, Canada. |
| 21:55.44 | benjk_ | your notebook may burst in flames |
| 21:56.01 | benjk_ | Un1x, FXO == voicemodem |
| 21:56.14 | shodan | hehehe fitting a fxs into a pcmcia , good luck ! |
| 21:56.17 | Mattwj2005 | TheCops: what are you looking to do? |
| 21:56.24 | sx-wks | TheCops: http://www.shsu.edu/administrative/training/guides/pdfs/quicktips_asterisk.pdf#search=%22asterisk%22 |
| 21:56.31 | TheCops | mog, someone from Bell Canada told to my client that asterisk is very unstable and I loose a lot of money on this shot, my client is very unsure hehe |
| 21:56.37 | TheCops | I hate bell guys |
| 21:56.54 | mog | that is very odd |
| 21:57.00 | mog | comming from bell canada |
| 21:57.02 | bkw__ | TheCops, well compaired to what bell runs Asterisk is unstable. |
| 21:57.05 | mog | as they tend to like asterisk |
| 21:57.08 | lesouvage | TheCops: There must be an intalled base of Asterisk in Canada. I'm sure if you google Canada asterisk VOIP provider some company will show up. |
| 21:57.29 | benjk_ | nah, nobody is using asterisk in Canada |
| 21:57.31 | TheCops | mog, yup yup, my father is working for bell since 30 years ago and love Asterisk |
| 21:57.43 | justinu|laptop | cops: comment ca va? |
| 21:57.45 | smackus | sx-wks: http://pastebin.ca/152563 |
| 21:57.48 | TheCops | justinu, hey ;) |
| 21:57.54 | TheCops | justinu, pretty well thanks, what about you |
| 21:57.57 | benjk_ | Canada is an asterisk free zone |
| 21:58.05 | sx-wks | benjk_: lol |
| 21:58.07 | file | does that mean someone is going to come kill me? :( |
| 21:58.13 | shodan | we just got electricity , it'll be a while before we get the phone EH ? |
| 21:58.14 | Mattwj2005 | you know it is funny you say that....some telephone companies are even selling VoIP service :-P |
| 21:58.15 | justinu|laptop | silly ppl at work broke the voip platform somehow |
| 21:58.21 | bkw__ | I think asterisk is more unpredictiable than unstable |
| 21:58.43 | sx-wks | smackus: ok, that's what I said... you can't call a number wiht only 3 digits on that other system. = |
| 21:58.54 | *** join/#asterisk okdo (n=goldenol@26.sub-70-215-240.myvzw.com) |
| 21:59.05 | okdo | anyone had any luck buying unlocked atas in best buy, circuit city, frys, etc. ? |
| 21:59.06 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 21:59.11 | okdo | and if so, which store(s)...? |
| 21:59.22 | Mattwj2005 | does ebay count? |
| 21:59.29 | benjk_ | I can give you a reference for a company in Botswana who use ASterisk though :D |
| 21:59.34 | mog | i think compusa sells em now |
| 21:59.50 | sx-wks | smackus: you're trying to place a call to 120. I suppose this is NANP, so should be a long distance call, and you're missing 8 digits |
| 21:59.51 | benjk_ | or in Lagos Nigera |
| 22:00.04 | benjk_ | you can get any kind of reference from Lagos Nigeria |
| 22:00.27 | benjk_ | just tell them what you need confirmed, certified, authorised, they will print it for ya |
| 22:00.29 | sx-wks | benjk_: only to contact some minister with a 50M USD account to shove out the country |
| 22:00.32 | smackus | ok, so in the case of: http://pastebin.ca/152564 |
| 22:01.08 | benjk_ | Lagos has the highest concentration of Heidelberg printing presses |
| 22:01.14 | sx-wks | smackus: you're missing the 1 in the front for the long distance call |
| 22:01.29 | benjk_ | Heidelberg printing presses are the Rolls Royces of printing machinery |
| 22:01.38 | benjk_ | the sort of thing governments use to print money |
| 22:01.44 | smackus | it is stripped as per requirements of the provider |
| 22:01.45 | jbalcomb | benjk_: we have three of them |
| 22:01.52 | smackus | exten => _1XXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1}) |
| 22:01.56 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 22:01.59 | sx-wks | smackus: well... apparenly this fscks up |
| 22:02.05 | okdo | anyone have any suggestions for buying a T1 router at retail... like where to find one |
| 22:02.16 | benjk_ | yet, Lagos has the highest density of Heidelbergs |
| 22:02.21 | sx-wks | btw, that should be _1NXXNXXXXXX |
| 22:02.23 | benjk_ | and they are not cheap either |
| 22:02.30 | shodan | I need a business-looking phone that works good with *, and is kinda comfortable to use for a receptionnist must have a transfert and flash button price between 150$usd and 215$usd what should I look for ? |
| 22:03.06 | jbalcomb | shodan: polycom |
| 22:03.18 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 22:03.41 | smackus | ok, still same thing... address incomplete. |
| 22:03.48 | C6Vette | smackus, 8018285230 wont match to _1XXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1}) |
| 22:03.50 | sx-wks | shodan: get one with a headset instead of a handset |
| 22:03.52 | C6Vette | need the 1 |
| 22:03.57 | smackus | yeah, i changed it... |
| 22:04.00 | lesouvage | shodan: swissphone is also a nice, good looking phone. |
| 22:04.01 | smackus | thanks for the catch, |
| 22:04.05 | jbalcomb | shodan: http://www.voipsupply.com/manufacturers/Polycom.html |
| 22:04.08 | smackus | but it still gives the same response. |
| 22:04.11 | smackus | address incomplete |
| 22:04.22 | C6Vette | smackus, what error are you getting on the other * server? |
| 22:04.29 | shodan | sx-wks, I tried to talke the receptionnist into using a headset but she hates it |
| 22:04.36 | sx-wks | hah. |
| 22:04.43 | shodan | jbalcomb, thans I'll take a look |
| 22:04.47 | C6Vette | smackus, looks like its not going to the correct context maybe. |
| 22:04.58 | smackus | nothing... not getting to the other server is what it looks like |
| 22:05.04 | smackus | i get nothing in the CLI |
| 22:05.16 | smackus | should be going to the default context. |
| 22:05.19 | benjk_ | okdo, check out Soekris |
| 22:05.33 | smackus | how can i specify the context? |
| 22:05.39 | smackus | i have not been able to figure that one out. |
| 22:05.44 | lesouvage | smackus: does the context the call is routed to give access to the extension you are dialing? |
| 22:05.49 | C6Vette | smackus, is the ip address on the local * allowed on the remote? |
| 22:06.18 | smackus | lesouvage: yes |
| 22:06.34 | smackus | C6Vette: is this an asterisk setting? if so, i dont think i have set that up |
| 22:06.56 | C6Vette | smackus, how are you authenticating? |
| 22:07.27 | smackus | all i am using is the sip.conf. but if i am dialing an outbound number, how do I do that? |
| 22:07.32 | *** join/#asterisk clyrrad (n=ddd@Quebec-HSE-ppp227879.qc.sympatico.ca) |
| 22:07.50 | rpm | can the Goto statement not jump to labels? I know that GotoIf can |
| 22:08.06 | clyrrad | far as i know it can |
| 22:09.47 | lesouvage | smackus: maybe this helps you. http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers |
| 22:10.03 | C6Vette | there ya go |
| 22:10.27 | smackus | lesouvage: reading... thanks |
| 22:10.57 | shodan | is that UTstarcom F300 wifi voip phone any good ? |
| 22:11.16 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 22:12.21 | flujan | hi guys... I hear about a new digium board which do not use the cpu to convert voice. Is it true? |
| 22:12.27 | *** part/#asterisk bpiper (n=bpiper@70.159.49.40) |
| 22:13.46 | lesouvage | shodan: all wifi voip phones have short up time. You have to keep them on the powersupply using them during the day. A DEC with an ATA is much more convinient. |
| 22:14.08 | *** join/#asterisk s1gny|wrk (n=s1gny@p54916A09.dip.t-dialin.net) |
| 22:14.37 | shodan | short as in 10 hours standby/1-2 hour talk or a lot less than that ? |
| 22:15.38 | lesouvage | shodan: I haven't tried them all but I drain the batteries of my zyxel w2000 and and my Nokia E60 while using them. |
| 22:16.44 | lesouvage | shodan: If you ae not making phonecalls all the time it's no problem, but if you're calling all the time you can't rely on this phones. |
| 22:17.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-96-211.red.bezeqint.net) |
| 22:17.18 | okdo | somebody must have a suggestion on where to find a cisco router in person? |
| 22:17.35 | lesouvage | shodan: although the nokia E60 is a nice gadget :) |
| 22:17.56 | shodan | k, I'll keep that in mind, anyway all the offices will have wired phones too so worst case I could just transfert the call to the nearest wired phone |
| 22:18.00 | benjk_ | yay, we got opbx working with coroutines, experimentally, but still |
| 22:18.28 | *** join/#asterisk kronos003 (n=kronos00@c-68-43-231-70.hsd1.mi.comcast.net) |
| 22:18.35 | shodan | yes the E60 looks great ! but I could buy a small computer for the price of one of those ;) |
| 22:19.09 | lesouvage | shodann: the idea of the E60 is to get rid of the wired phones and have just one phone with sim card and sip account(s). |
| 22:19.59 | shodan | they're a lot less expensive than I thought |
| 22:20.08 | lesouvage | shodan: the E60 works great with Asterisk |
| 22:20.27 | kronos003 | link? |
| 22:21.06 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 22:21.47 | kronos003 | what is the E60? |
| 22:22.09 | kronos003 | is that the nokia E60? |
| 22:23.42 | shodan | kronos003, it's a gsm+wifi/voip phone |
| 22:23.57 | kronos003 | so it IS the nokia E60 - Nice |
| 22:24.26 | kronos003 | that will talk to my asterisk installation when i get one? wow |
| 22:24.49 | kronos003 | btw have you guys heard of shoretel? |
| 22:24.50 | mog | if you are state side my answer would point to no |
| 22:24.56 | mog | i hve |
| 22:25.24 | kronos003 | i was offered a setup fro them - the installer wanted an arm and a leg |
| 22:25.36 | kronos003 | plus the licence sceme sux |
| 22:25.56 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 22:26.00 | kronos003 | can asterisk make use of or control any of their hardware? |
| 22:26.33 | kronos003 | Probably the phones for sure but how about the switches? |
| 22:27.14 | mog | i thought the shoretel stuff was all propitery |
| 22:27.15 | mog | and not even sip |
| 22:28.10 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 22:29.00 | kronos003 | they have "sip expandibility" for a price - acctually i dont much care about their gear really - the one thing i need though is to put a phone on my desk that runs on ethernet |
| 22:29.12 | kronos003 | i guess that would be a sip phone? |
| 22:29.16 | Strom_C | there's no way to tell a sip peer or user "turn off g729 VAD" in sip.conf, is there? |
| 22:29.47 | kronos003 | for an office phone with some cool features what should i get? |
| 22:29.57 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 22:30.07 | mog | no Strom_C its on the end point |
| 22:30.12 | mog | polycom |
| 22:30.13 | mog | 601 |
| 22:30.18 | Strom_C | kronos003: Cisco 7960/7961 or Polycom SPIP601 |
| 22:30.23 | Strom_C | mog: thats what I thought |
| 22:30.40 | lesouvage | Strom_C: disallow=all (next line) allow=<whatever codec you want to allow> The other side must of course also support the codec. |
| 22:30.53 | Strom_C | the question was about VAD, not about codecs, lesouvage |
| 22:31.03 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 22:31.18 | kronos003 | where can i get one |
| 22:32.15 | mog | google |
| 22:32.25 | [TK]D-Fender | kronos003 : Depends where you are... |
| 22:32.32 | kronos003 | US here |
| 22:32.43 | [TK]D-Fender | kronos003 : www.telephonydepot.com |
| 22:32.57 | kronos003 | checked ebay - those cisco phones are 189 first pass |
| 22:33.09 | [TK]D-Fender | kronos003 : What do you NEED in an IP phone? |
| 22:33.47 | Assid | a dialpad :P |
| 22:33.53 | kronos003 | just something to put on my desk to talk to my asterisk install - gonna have like 4 lines to start |
| 22:34.08 | kronos003 | need all the usual features |
| 22:34.09 | [TK]D-Fender | Assid : No, I say "immediate mode" + Sphinx ;) |
| 22:34.27 | [TK]D-Fender | kronos003 : 4 lines is a lot actually... I'd suggest an IP 501 in its place. |
| 22:34.43 | Strom_C | IP501 is only three appearances though |
| 22:34.57 | [TK]D-Fender | kronos003 : Strom_C : And how many people even use 3? |
| 22:35.11 | shodan | I'm thinking of getting a polycom 301 , but is there any reason to get a 501 except for the 3rd line ? |
| 22:35.18 | Strom_C | but then again, I'm kind of bonkers and have at least fifteen line appearances on my desk alone |
| 22:35.21 | Assid | i barely managed to use my 301 for full potential |
| 22:35.35 | Assid | shodan: and a few extra buttons |
| 22:35.36 | [TK]D-Fender | shodan : 501 = bigger display, extra line-key, full-duplex speakerphone |
| 22:35.52 | [TK]D-Fender | Strom_C : At least you're not in denial! ;) |
| 22:35.55 | kronos003 | right now i have a polycom soundpoint pro 2 line skr phone - its pretty nice - am looking for something that sounds as good or better |
| 22:36.08 | Assid | actually the full duplex speakerphone makes a difference |
| 22:36.15 | [TK]D-Fender | kronos003 : All of Polycom is great on audio quality. |
| 22:36.23 | Assid | yeah.. i second that |
| 22:36.32 | kronos003 | i guess ill stick with polycom then |
| 22:36.32 | Strom_C | [TK]D-Fender: you should see my apartment. There's pretty much a telephone within five steps of any given location |
| 22:36.40 | kronos003 | LOL |
| 22:36.53 | [TK]D-Fender | Strom_C : Big deal.. my bedroom here alone has an IP 301, IP 430, and IP 501 ;) |
| 22:36.59 | shodan | does the 301 (or any polycom phone) have a flash button ? |
| 22:37.09 | Assid | flash ?????????? |
| 22:37.18 | Strom_C | "flash" is an analog phone thing |
| 22:37.22 | Strom_C | doesn't exist on SIP phones |
| 22:37.35 | kronos003 | mmm IP501 definitly in budget - at least more so than 7.5kUSD |
| 22:37.39 | shodan | it's for call waiting , it tells the fxo to flash the line |
| 22:37.45 | [TK]D-Fender | shodan : lol.... |
| 22:37.57 | *** join/#asterisk Samoied (n=Samoied@201.21.228.97) |
| 22:38.03 | Strom_C | call waiting on FXO trunks == bad idea |
| 22:38.09 | Strom_C | or, actually |
| 22:38.13 | Assid | shodan: ip phones wwayyyyyy differently |
| 22:38.16 | [TK]D-Fender | kronos003 : Everything depends on your deployment strategy & budget. How many, what kind of wiring scenario, etc... |
| 22:38.17 | Strom_C | bad system engineering |
| 22:38.46 | kronos003 | only one user |
| 22:38.49 | shodan | problem is one of my incoming line is not stackable and not forwardable , it's call waiting or 2nd called gets busy signal |
| 22:38.49 | kronos003 | me |
| 22:38.50 | [TK]D-Fender | Analog lines = bad engineering ;) |
| 22:38.53 | kronos003 | LOL |
| 22:39.04 | shodan | and I can't port the number too |
| 22:39.10 | [TK]D-Fender | shodan : As we said you problem is your incoming LINE ;) |
| 22:39.12 | kronos003 | i just need to be sure the customer never heres a busy signal |
| 22:39.33 | [TK]D-Fender | kronos003 : Pay a lot ofr unlimited channel VoIP account on a T3 then ;) |
| 22:39.33 | Assid | alrite gnight folks |
| 22:39.42 | kronos003 | i like the looks of that IP601 - is it anygood? |
| 22:39.49 | mountainm2k | Trying to set up Page() -- phones that are in use when Page() is called act really weird... |
| 22:39.54 | mountainm2k | (Polycom) |
| 22:39.59 | eKo1 | kronos003: check for busy and play a nice song when that happens |
| 22:40.02 | [TK]D-Fender | kronos003 : Great, but I'm betting overkill |
| 22:40.16 | [TK]D-Fender | kronos003 : I own every desk model they produced |
| 22:40.32 | kronos003 | [TK]D-Fender: that 7.5kusd was for that damn shortel rig - they wanted me to have a win2k3 server on my rack ugh |
| 22:40.33 | [TK]D-Fender | kronos003 : 3/4 of them in hands reach right now ;) |
| 22:40.48 | [TK]D-Fender | kronos003 : Screw Shoretel... * + Polycom = win |
| 22:41.32 | kronos003 | ya - i just found asterisk - im VERY partial to OSS so as long as it works im there |
| 22:41.35 | [TK]D-Fender | kronos003 : So how many phones, what kind of wiring etc for your project? |
| 22:41.54 | kronos003 | one phone in office and one cell phone |
| 22:42.10 | [TK]D-Fender | kronos003 : as in 2 phones period? |
| 22:42.24 | kronos003 | i need reliability so im thinking 4 analog lines into a digium card |
| 22:42.24 | [TK]D-Fender | kronos003 : One of which clearly having nothing to do with your PBX... |
| 22:43.00 | [TK]D-Fender | 4 lines for 1 phone ? Seems kinda wierd.... |
| 22:43.02 | kronos003 | [TK]D-Fender: i will have the system connect out to the cell phone whenever a call is received |
| 22:44.20 | kronos003 | [TK]D-Fender: I havent committed to anything yet - currently i have a vonage setup and many of my customers are complaining of busy signals when they call me and im not sure what i need to do to stop that |
| 22:44.40 | [TK]D-Fender | kronos003 : How many minutes / month do you really expect to need? |
| 22:44.54 | kronos003 | 3000/m easily |
| 22:45.14 | kronos003 | inbound at least |
| 22:45.20 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:45.32 | *** join/#asterisk tessier__ (n=treed@gw.drjays.com) |
| 22:45.38 | tessier__ | Whut up dawgz |
| 22:45.51 | lesouvage | kronos: If you fined yourself a proper IP telephony provider you are much more flexible on the incoming lines and an outgoing line doesn't decrease the number of incoming lines. You also don't need expensive hardware to connect landlines to your PBX. You just enoughh bandwidth. And you can use queues in Asterisk. |
| 22:45.56 | [TK]D-Fender | kronos003 : get a DID, and a by-the-minute VoIP PSTN plan. You should be able to get .02$ a min EASY if not less. that = 60$ month, and NO card to buy |
| 22:46.26 | kronos003 | hmmm |
| 22:46.40 | kronos003 | who do you suggest for voip |
| 22:47.08 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 22:47.14 | [TK]D-Fender | kronos003 : Broadvoice & VoicePulse (later seems most preferred) |
| 22:47.21 | kronos003 | hmm |
| 22:47.27 | kronos003 | are they US based? |
| 22:47.34 | [TK]D-Fender | VP's plans are dirt cheap |
| 22:47.44 | [TK]D-Fender | kronos003 : Clearly for my suggesting them to you :) |
| 22:47.53 | kronos003 | I see |
| 22:48.02 | kronos003 | im new on this channel |
| 22:48.06 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 22:49.46 | smackus | can SetCIDName be used to change an outside callers caller id name to something based of of the number they dialed? for example if I dialed an 800 number located on my asterisk server from my cell phone, could I change the number displayed from my cell phone number to say something like: Customer Server Main Line? |
| 22:49.51 | kronos003 | heres the sucky part - my number on vonage is heavily advertised and so far as i can see NON F!@## portable - and ive tried |
| 22:50.11 | smackus | the idea is to allow agents on the floor to answer the phone based on the number the customer called. they would answer differently based on what 800 number they called into. |
| 22:50.16 | [TK]D-Fender | smackus : Typically you can't set the name, only the number |
| 22:51.17 | [TK]D-Fender | smackus : You can't receive a name on your cell anyways, jsut the number. Whci if you rig you CAN synch to an address boook entry in your phone |
| 22:51.40 | [TK]D-Fender | kronos003 : Well, thats a "damned if you do, damned if you don't" statement anyways... |
| 22:51.50 | kronos003 | ya |
| 22:51.51 | smackus | <PROTECTED> |
| 22:51.55 | smackus | backwards |
| 22:52.09 | *** join/#asterisk grantm (n=grantm@207.88.78.2) |
| 22:52.12 | lesouvage | smackus: you can route incoming calls to agents based on numbermatching. |
| 22:52.14 | smackus | based on the number dialed by a customer, I want my agents phone to display the name of the 800 number that customer dialed. |
| 22:52.24 | smackus | dont want to route... only change display |
| 22:52.54 | smackus | all agents take all calls, but they answer differently based on what number the customer dials. |
| 22:53.06 | smackus | rather than have the agents learn all of the 800 numbers, I would like to display on their phones the "name" of the 800 number they dialed. |
| 22:53.30 | [TK]D-Fender | smackus : You mean rig the name for INSIDE phones on SIP? |
| 22:53.34 | _DAW | I am confused.. when I call my sip gateway through * and place the call on hold, should * suspend rtp between itself and the gateway? |
| 22:53.38 | smackus | correct |
| 22:53.44 | Strom_C | smackus: use different queues for each 800 number and play a recording to the agent before the call is connected telling them which queue they're being connected to |
| 22:53.54 | [TK]D-Fender | smackus : Oh that... yeah easy |
| 22:54.04 | sx-wks | so there. when local phones don't answer, I call at work, just in case... |
| 22:54.22 | [TK]D-Fender | smackus : I jsut prefix my callerID like "Q:Original Name here" and so on |
| 22:54.24 | sx-wks | should use the gotoiftime thingie for that, don't want to call there at 3 am :D |
| 22:55.14 | lesouvage | smackus: it's jut an idea, but you can route calls to different queues based on the number used and let all the agents handle all the quess. There is an option that inform the agents about the queue the call comes from. |
| 22:55.41 | kronos003 | for now i think im gonna start by migrating into asterisk and see what i can do to have asterisk controll the vonage connection directly |
| 22:56.07 | lesouvage | smackus: f.i. "here comes a call from Amsterdam" |
| 22:56.18 | [TK]D-Fender | kronos003 : Let me just say the experience is going to supremely SUCK. |
| 22:56.38 | *** join/#asterisk SkoZombie (n=awhalan@203-217-86-249.dyn.iinet.net.au) |
| 22:56.39 | smackus | lesouvage: correct... |
| 22:56.44 | [TK]D-Fender | kronos003 : Only way to take Vonage IN is remarkably kludgy and you lose call-waiting, etc... |
| 22:56.52 | smackus | but I want it displayed, not played. |
| 22:57.04 | kronos003 | hmm |
| 22:57.20 | [TK]D-Fender | smackus : Set(CALLERID(name)=Q1:${CALLERID(name)}) |
| 22:57.24 | kronos003 | [TK]D-Fender: is there a writup on that anywhere |
| 22:57.27 | *** part/#asterisk jarg (n=jarg@200.56.225.61) |
| 22:58.17 | smackus | [TK]D-Fender: that is what I was thinking... so that will work on inbound calls to agents? |
| 22:58.17 | kronos003 | [TK]D-Fender: also would Voicepulse conect to asterisk better than vonage |
| 22:58.22 | [TK]D-Fender | kronos003 : No. Its jsut a know fact. Vonage won't let * log into your account, and the only way to take it in means getting an FXO port into * and plugging the Vonage ATA into it. That means double-encoding on the line involving extra time delay before being answered and no ability to take in the 2nd call. |
| 22:58.44 | tessier__ | Aug 28 15:58:23 WARNING[4370]: channel.c:2564 ast_request: No channel type registered for 'Zap' |
| 22:58.45 | kronos003 | tash! |
| 22:58.48 | kronos003 | trash! |
| 22:58.48 | [TK]D-Fender | smackus : Yes, use chan local and rig it there, or do it before the call enters the Queue |
| 22:58.50 | tessier__ | Anyone know what the deal with that is? |
| 22:59.11 | [TK]D-Fender | kronos003 : not to mention quality reduction as well. |
| 22:59.24 | kronos003 | so none of these issues with Voicepulse? |
| 22:59.48 | [TK]D-Fender | kronos003 : Nop, none of those. VP has * specific services.... |
| 23:00.16 | kronos003 | hopefully VP can swipe the number from von |
| 23:00.32 | SkoZombie | I've got our asterisk system rocking, but there's about a 3 ring delay before the call rings on the reception phone |
| 23:00.35 | SkoZombie | any suggestions? |
| 23:00.49 | SkoZombie | i'm using a TDM400P |
| 23:01.15 | [TK]D-Fender | SkoZombie : * waits for CallerID. Thats why |
| 23:01.25 | SkoZombie | ah ok |
| 23:01.29 | [TK]D-Fender | SkoZombie : You CAN turn it off if you want, but well... you know the consequence |
| 23:01.48 | SkoZombie | it never seems to get it, and for our purposes i think callerID isn't much use |
| 23:03.44 | mountainm2k | Anybody use Page() with Polycom phones? |
| 23:04.22 | Strom_C | is there any way to debug exactly where SIP frames containing VAD are coming from? |
| 23:05.41 | kronos003 | does * have a Yum repo (as in fedora core 3) |
| 23:06.00 | Qwell | kronos003: it's in most distros, but asterisk packages generally suck. |
| 23:06.14 | Qwell | For most people, we recommend building the source yourself |
| 23:06.16 | kronos003 | whats the best way then? |
| 23:06.21 | kronos003 | i see |
| 23:06.56 | kronos003 | is that as simple as configure && make && make install ? |
| 23:07.01 | Qwell | exactly |
| 23:07.11 | Qwell | in 1.2, you don't even need the configure |
| 23:07.12 | kronos003 | cool |
| 23:07.43 | kronos003 | i suppose i should install it and try it out |
| 23:07.46 | *** join/#asterisk fiber0pti (n=John@207.114.199.107) |
| 23:08.12 | fiber0pti | is there a way to turn off the message envelope when first listening to voicemail |
| 23:08.40 | kronos003 | does * have a web interface - is there a demo of this anywherew |
| 23:08.46 | *** join/#asterisk [koss] (i=koss@adsl-75-36-15-21.dsl.bcvloh.sbcglobal.net) |
| 23:09.11 | carrar | click START, then click on RUNASTERISK |
| 23:10.10 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:11.00 | SkoZombie | How do i change the text on the internal SIP fone from "asterisk" to "PSTN Call" for pstn calls |
| 23:11.14 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:11.53 | [TK]D-Fender | kronos003 : Forget packaging. Compile from source |
| 23:11.53 | kronos003 | do i need a PoE module to run these IP phone on power over ethernet or is that already built into the phones |
| 23:12.06 | kronos003 | er um switches |
| 23:12.53 | kronos003 | like is PoE already a function of my bargain basement 8 port ethernet switch |
| 23:13.09 | [TK]D-Fender | kronos003 : the IP 430 & IP 601 have PoE built in meaning if your switch provides PoE you're set. If you have a PoE switch & wish to use a 301 / 501 you need a special cable (no you can't make it yourself) to use them with PoE. |
| 23:13.20 | [TK]D-Fender | kronos003 : Netgear? |
| 23:13.40 | kronos003 | nope : ABS |
| 23:14.05 | kronos003 | im using a 10/100/1000 switch |
| 23:14.20 | kronos003 | for gb ethernet its supposed to be pretty good |
| 23:14.21 | shmaltz | anybody here know anything about Avaya Partner Voice Messaging System (PC Card)? |
| 23:14.24 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 23:19.41 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 23:19.47 | *** part/#asterisk Samoied (n=Samoied@201.21.228.97) |
| 23:20.11 | hads | SkoZombie: Do a Set(CALLERID(name)="PSTN Call") before your dial. |
| 23:20.13 | novafirst | [TK]D-Fender: I am still having problem with musiconhold. |
| 23:20.32 | SkoZombie | hads: sweet |
| 23:20.35 | SkoZombie | and my final question |
| 23:21.05 | SkoZombie | we call the reception SIP phone first, any way someone else can answer that call? |
| 23:21.16 | novafirst | do I need any settings within extensions.conf ? to make musiconhold to work? |
| 23:21.23 | SkoZombie | ie. "Oh she's not here *key combo & answers call*" |
| 23:21.36 | hads | Look for callgroup and pickupgroup |
| 23:21.43 | *** join/#asterisk hoytbowUE (i=webspinn@203.11.107.242) |
| 23:21.49 | novafirst | or maybe x-lite is not compatible with mp3 musiconhold |
| 23:22.13 | hads | SkoZombie: and the key sequence is set in features.conf (normally *8 |
| 23:22.33 | SkoZombie | ah ok |
| 23:22.39 | SkoZombie | so add the incoming call to a callgroup? |
| 23:22.58 | hads | add a callgroup/pickupgroup to your friend in sip.conf |
| 23:23.43 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 23:23.47 | hads | s/friend/friends |
| 23:23.53 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 23:23.55 | SkoZombie | k |
| 23:24.16 | SkoZombie | so just basically have everyone on a call group so they can answer each other's calls if need be (it's a small office) |
| 23:24.17 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 23:24.25 | SkoZombie | we only have 5 extensions |
| 23:24.46 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:25.07 | hads | pickupgroup is the one that will let you do the pickup (funnily enough :) |
| 23:25.28 | SkoZombie | yeah :) |
| 23:26.20 | hmmhesays | so my employer completely farked me over |
| 23:26.38 | file | hmmhesays: ? |
| 23:26.56 | hads | SkoZombie: http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
| 23:27.11 | hmmhesays | file: is trying to screw me out of thousands of dollars |
| 23:27.23 | hmmhesays | by retracting a verbal agreement we had |
| 23:27.29 | file | ah |
| 23:29.39 | SkoZombie | hads: thank you very much for your help. I won't get a chance to go into the office til next week but I'm sure that we'll get a lot closer to where we want to be when i do :) |
| 23:29.56 | hads | No probs :) |
| 23:30.29 | SkoZombie | where in NZ are you ? |
| 23:30.32 | SkoZombie | I'm in newcastle, AU |
| 23:31.24 | hads | <PROTECTED> |
| 23:31.30 | SkoZombie | ah cool :) |
| 23:31.31 | novafirst | is x-lite compatible with asterisk music-on-hold using mp3 files as moh source ? |
| 23:31.41 | SkoZombie | I'm planning on coming to NZ for a sking holiday next year |
| 23:31.58 | hmmhesays | novafirst to play local mp3's? |
| 23:32.21 | hads | SkoZombie: Cool, I'd go to the South Island myself rather than the North. |
| 23:32.34 | SkoZombie | I'll keep that in mind :) |
| 23:33.09 | SkoZombie | oh one last problem, I've noticed problems trying to hang up and disconnect calls |
| 23:33.14 | novafirst | I am testing my asterisk box but there is no sound ?? I am using x-lite as sip client |
| 23:33.23 | SkoZombie | and busy tones etc |
| 23:33.37 | hads | SkoZombie: You're using analog lines with a TDM400 right? |
| 23:33.45 | SkoZombie | yeah |
| 23:34.11 | hads | So you are having problems with Asterisk not hanging up the line when the person calling in hangs up the phone? |
| 23:37.25 | SkoZombie | I'm struggling to get it to do Playtones(busy) ... I can't work out what module that's defined in, and Hangup(Zap/1) says the line is in a wierd state 7 |
| 23:37.40 | SkoZombie | the other admin (who is hopeless) still hasn't got me remote access to the system |
| 23:37.42 | fafnir | Asssterisk ASSSSTERISK assssssTeeeeerrrrrIIIIISSSSSSK |
| 23:40.47 | SkoZombie | hads: don't worry about it, if they can't forward a port for me, I'm not going to worry about it any further until I can get in |
| 23:40.47 | hads | SkoZombie: Playtones is in res_indications |
| 23:40.55 | SkoZombie | awesome :) |
| 23:42.05 | hads | Dunno about the weird state, probably be a lot easier if you were in front of a console like you say. |
| 23:43.48 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:44.12 | tessier__ | Anyone know how to set a PRI to use timing from the other end instead of internal? |
| 23:44.51 | [TK]D-Fender | tessier : 2nd parm on your span line. |
| 23:45.13 | [TK]D-Fender | bbiab |
| 23:45.15 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.160.146) |
| 23:46.22 | tessier__ | odd |
| 23:46.31 | tessier__ | I have the second param set to 1 which means use them for timing |
| 23:46.46 | tessier__ | But the zttool says internally clocked |
| 23:46.57 | CunningPike | tessier__: There may not be a timing signal then |
| 23:47.19 | CunningPike | tessier__: What is your span = line |
| 23:47.38 | tessier__ | span=1,1,0,esf,b8zs |
| 23:47.57 | tessier__ | span 1, their timing, I forget what 0 is db or something?, esf, b8zs standard stuff |
| 23:48.33 | tessier__ | CunningPike: But zttool says internally clocked. Does that mean something else? |
| 23:48.40 | tessier__ | The d channel on this pri is not comingup |
| 23:48.45 | *** join/#asterisk FullService (n=jiggaman@CPE0002724fc55e-CM000f211fd29c.cpe.net.cable.rogers.com) |
| 23:48.46 | tessier__ | according to the guy at the switch |
| 23:48.59 | CunningPike | tessier__: No d-channel => no timing |
| 23:49.34 | CunningPike | tessier__: You need to fix whatever is preventing your PRI from coming up before you can expect to have a valid timing signal |
| 23:49.46 | tessier__ | CunningPike: We get a green light and no alarms |
| 23:50.04 | tessier__ | so both ends agree on carrier |
| 23:50.18 | tessier__ | So the PRI is "up" but no d channel |
| 23:50.46 | tessier__ | hmm |
| 23:52.25 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 23:53.29 | QbY | Asterisk Realtime with Postgres Question--Per voi-info.org I need to make sure that I have ODBC already installed and running.. Would that be done already considering Postgres is installed and running? Or is there something additional? |
| 23:58.24 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:59.00 | blitzrage | QbY: no -- you need to install the odbc packages |
| 23:59.16 | blitzrage | you use ODBC to connect to Postgres |
