| 00:01.47 | delx | citywok, [sipgate_in] exten => 4921129,1,AGI(/etc/asterisk/forward_call.php) |
| 00:01.48 | delx | exten => 4921129,n,Hangup |
| 00:01.50 | delx | is that correct? |
| 00:10.00 | leifmadsen | first priority needs ot be on its own line |
| 00:11.26 | delx | leifmadsen, do you talk to me? |
| 00:11.58 | carrar | do you, YOU!!!, feel like I do!! |
| 00:12.28 | delx | no, seriously.. |
| 00:39.09 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 00:39.09 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 00:40.06 | delx | -- AGI Script Executing Application: (Dial) Options: (SIP/0174xxxx@1141382t0,30,trg) |
| 00:40.08 | delx | WARNING[8987]: chan_sip.c:2921 create_addr: No such host: 1141382t0,30,trg |
| 00:40.25 | p3nguin | You should have taken my advice. |
| 00:40.37 | delx | p3nguin, I just replaced ${EXTEN} with the number! |
| 00:42.13 | delx | I try Dial(SIP/sipgate/${EXTEN},30) now |
| 00:42.35 | *** join/#asterisk korcan (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 00:43.06 | p3nguin | You have to have a peer called sipgate, and it needs to correctly be configured. |
| 00:44.03 | leifmadsen | most of these questions you're asking would be answered by reading documentation :) |
| 00:44.32 | pabelanger | leifmadsen: oh snap |
| 00:45.34 | citywok | russellb will be here soon to tell you to play nice! |
| 00:47.03 | diemos | leifmadsen: loldocumentation! |
| 00:47.20 | diemos | Installing it and just throwing commands out is much more fun! |
| 00:51.54 | delx | haha, it works! :) |
| 00:52.07 | delx | p3nguin, thanks - I called it sipgate now :) |
| 00:52.11 | delx | and remove ,30 |
| 00:52.43 | p3nguin | The ,30 is just a timeout on the dialing. |
| 00:53.08 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.140.172) |
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| 01:05.28 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:05.57 | cmnky | evening leif |
| 01:07.36 | diemos | wb leifmadsen |
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| 01:40.57 | *** join/#asterisk DeHackEd (~dehacked@dhe.execulink.com) |
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| 02:00.19 | greezmunkey | I ran into a situation to day that puzzles me. I changed my extensions.conf a few days ago to "allow" two digit numbers to hit my gateway. All that worked fine. Once I was done with it, I shut the system down (as root, shutdown -h 0). I powered everything up today, and modified the dialplan from two digits (_XX...) to four digits starting with a 7 (_7XXX). The digits arrived at the gateway, but for some reason the first two digits seem to have been |
| 02:03.33 | p3nguin | You truncated at "...seem to have been" |
| 02:04.03 | diemos | in more ways than one! |
| 02:04.09 | greezmunkey | seem to have been stripped. I did dialplan reload several times, restarted the service, but got the same results. Does asterisk store dial patterns anywhere other than extensions.conf? Signed, confused munkey... |
| 02:04.47 | p3nguin | If two digits are stripped, you probably used ${EXTEN:2} in your command. |
| 02:05.04 | greezmunkey | nope ${EXTEN} only |
| 02:05.27 | p3nguin | Then show me some verbose output of the failed call. |
| 02:06.37 | greezmunkey | p3nguin: I'll be able to do that in about 14 hours, the system is in the lab ot my office. |
| 02:07.35 | DeHackEd | I have an ATA I'm trying to register with asterisk. but I get "No matching peer found". asterisk 1.6.2.12 from fedora13 |
| 02:07.41 | greezmunkey | I honnestly think the issue is in the gateway. |
| 02:08.11 | DeHackEd | (over SIP) |
| 02:14.00 | greezmunkey | OK, I can't recreate the issue I'm having right now... I get that. I have seen how asterisk reacts to dialplan changes, they seem to take effect immediately. Is there any reason that it may not? |
| 02:14.43 | nny | greezmunkey: you have to reload dialplan or restart asterisk for them to take affect |
| 02:15.46 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 02:15.46 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 02:17.11 | greezmunkey | nny: exactly. What I'm looking for is conformation that dialplan elements (if modified in the conf file) are replaced upon running 'dialplan reload' |
| 02:17.19 | *** join/#asterisk [netman] (~netman@131.Red-83-39-224.dynamicIP.rima-tde.net) |
| 02:17.28 | greezmunkey | confirmation... |
| 02:17.30 | p3nguin | dialplan show |
| 02:17.59 | ectospasm | or, "dialplan show <context>" |
| 02:18.06 | p3nguin | If dialplan show doesn't show it right, it's not right. |
| 02:18.14 | *** join/#asterisk coppice (~chatzilla@m121-203-196-233.smartone-vodafone.com) |
| 02:18.41 | greezmunkey | p3nguin: I checked that as well, and what it displayed reflected the changes I made, so it can't be asterisk... |
| 02:19.36 | p3nguin | After I see the debug, then I'll see if I agree. |
| 02:21.45 | greezmunkey | I'll run syslog on the gateway in the morning to see what the gateway is doing. There isn't any reason for it to strip digits, as I am only using wildcards in its digit routing tables. But, I've worked with AudioCodes gates in the past, and had them do wierd things. |
| 02:24.17 | diemos | goodnight all, off to home and some Black Ops |
| 02:30.28 | DeHackEd | n/m it's working. sweet. |
| 02:36.10 | *** join/#asterisk cnu (cnu@2001:470:1f0b:ea::10) |
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| 03:43.40 | tish | hi, my name is danna. cr |
| 03:47.23 | ChannelZ | Can you prove it? |
| 03:49.16 | tish | hi, question about jingle in asterisk: what do you need to setup in order to make a call between two xmpp clients work, say Dial(JINGLE/abc@xxx.com/def@xxx.com) |
| 03:49.54 | tish | i have connected my asterisk server as a component to my xmpp server (ejabberd) |
| 03:50.01 | ChannelZ | well the typical syntax is to use & to dial multiple channels. Like Dial(SIP/foo&SIP/bar) |
| 03:50.51 | tish | i'm trying to implement it using jingle. |
| 03:51.00 | tish | how does the dialplan work? |
| 03:51.34 | ChannelZ | sooooo by extrapolation thenL JINGLE/abc@xxx.com&JINGLE/def@xxx.com |
| 03:51.45 | ChannelZ | s/thenL/then:/ |
| 03:52.24 | tish | ok, i'll try that, thanks |
| 03:52.38 | ectospasm | that won't bridge abc@xxx.com and def@xxx.com |
| 03:53.04 | ectospasm | that will place a call to both at the same time (ringroup), and whichever one answers first gets the call |
| 03:53.45 | ChannelZ | Yah. If you want to make a conference you'll have to MeetMe and dump them in there |
| 03:54.26 | ChannelZ | (or ConfBridge if you're newer, etc) |
| 03:54.41 | ectospasm | You could, however, create a call file, with one side in the Channel use JINGLE/abc@xxx.com, and for the Application use Dial, and Data use JINGLE/def@xxx.com |
| 03:55.49 | ChannelZ | except why involve Asterisk at all... |
| 03:56.08 | ectospasm | ChannelZ: accounting |
| 03:56.21 | ChannelZ | Jingle Roulette |
| 03:57.18 | tish | actually, it's not for a conference call but rather a peer to peer conversation. |
| 03:57.42 | tish | i'm trying to connect ejabberd to other sip servers using asterisk as it's main gateway component |
| 04:00.14 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 04:03.47 | tish | i'm trying to use this sample line in the extensions.conf: exten => 6390,1,Dial(JINGLE/callee/caller) ; Dial via jingle using labels |
| 04:04.27 | tish | and i have put entries for [callee] and [caller] in the jingle.conf, but i get the error chan_jingle.c:1490 jingle_request: Could not find recipient. |
| 04:05.50 | tish | i'm wondering which config files and modules i should check in order to get it work |
| 04:10.58 | *** part/#asterisk jazzy-J (~j@92.249.107.65) |
| 04:14.03 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 04:20.20 | ectospasm | tish: that's not the way the Dial() application works |
| 04:21.25 | ectospasm | Caller needs to find itself in that part of the dialplan, and execute 6390,1,Dial(JINGLE/callee) |
| 04:22.45 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:38.23 | *** join/#asterisk mlsmith9999 (~mlsmith@75-42-14-219.lightspeed.ltrkar.sbcglobal.net) |
| 04:42.04 | mlsmith9999 | evening, quick question. With AsteriskNOW 1.7 FreePBX GUI, where are the IVR's and Queue's? |
| 04:44.02 | ectospasm | mlsmith9999: I haven't looked in a while, but I don't think those modules are available in stock AsteriskNOW. |
| 04:44.16 | mlsmith9999 | Yeah, was afraid of that... |
| 04:47.53 | mlsmith9999 | I have to say as far as features and GUI goes, Trixbox has it. Even though they hack up the conf files all to hell. I guess that's what it takes to put a decent GUI on it. |
| 04:48.57 | ectospasm | who needs a GUI anyway? (-; |
| 04:50.12 | mlsmith9999 | that'd be me.. :( Love to have enough time to fully learn the system but I don't... |
| 04:51.48 | ectospasm | Depends on what you want to do with the system. If the GUI wholly meets your needs, there's always Switchvox |
| 04:52.19 | ectospasm | being able to call competent support is important |
| 04:55.58 | tvc123 | ectospasm: although I have had mixed results with Switchvox's support :( |
| 04:56.22 | ectospasm | tvc123: I'm biased, I'll admit. |
| 04:56.44 | tvc123 | ectospasm: what do you do for Digium? |
| 04:57.15 | ectospasm | Custom Telephony Solutions (non-Switchvox) Support |
| 04:57.52 | ectospasm | supporting the hardware, and our software products (including open source subscriptions for Asterisk) |
| 04:57.59 | tvc123 | ectospasm: nice I may end up going that route ... I'm getting frusterated with not being able to fix things and switchvox support taking WAY too long to fix thenm |
| 04:58.02 | tvc123 | err them |
| 04:58.22 | ectospasm | Remember that you sacrifice flexibility with any GUI system |
| 04:58.54 | ectospasm | if something's broken in open source Asterisk, you can either fix it yourself, or devise a workaround |
| 04:58.56 | zxvff | yes also it's harder to appear busy if someone walks by and you are looking at a static GUI |
| 04:59.07 | ectospasm | heheheh |
| 04:59.09 | zxvff | however with lots of console and scrolling texts and thigns you are always hard at work! |
| 04:59.26 | tvc123 | ectospasm: my problem is right now I have backup jobs just hanging and no way to fix it on a switchvox |
| 04:59.40 | mlsmith9999 | Well, I'm my boss, so I suppose I can fool myself... |
| 05:00.36 | ectospasm | tvc123: I dunno about how to fix that. All I can suggest is upgrade to the latest release. If you've done that, there's not much else to do than wait. |
| 05:01.21 | tvc123 | ectospasm: always fun to wait for an easy fix from support while your production goes to hell </rant> |
| 05:01.47 | mlsmith9999 | hmmm Backups hanging or breaking seem to be a thing with Asterisk... Had the same problem with my Trixbox Install. |
| 05:01.48 | ectospasm | you could say the same for any system, really. |
| 05:02.11 | tvc123 | ectospasm: any closed system that doens't have stellar support |
| 05:03.19 | tvc123 | for me the difference is I use things like a barracuda appliance .. but the second I am in touch with support they are beating down my door to fix the problem so its a non issue they fix it faster than I could |
| 05:03.43 | tvc123 | other products are nice because they are open .. I can fix them or hire anyone to fix them |
| 05:03.59 | tvc123 | when you have a closed box and the support isn't stellar that's where I have problems |
| 05:04.25 | ectospasm | so express that to them, we can't improve unless we hear about it. |
| 05:04.35 | tvc123 | ectospasm: I have |
| 05:04.52 | tvc123 | in much stonger terms |
| 05:05.31 | ectospasm | I never submit bugs for Switchvox, so I dunno how responsive that development team is. |
| 05:05.47 | tish | ectospasm: i'm trying to make the SIP-XMPP part work first, so i'm trying to place a call from a softphone to the extension 6390. my goal is for the XMPP user (callee defined in jingle.conf) to then receive an XMPP session-initiate packet. would that work? |
| 05:06.38 | tish | sorry, i'm a bit confused as to how to connect xmpp to sip and place calls between two xmpp users |
| 05:07.23 | ectospasm | SIP phone picks up, they dial 6390, which executes Dial(JINGLE/xmpp_endpoint) |
| 05:07.43 | ectospasm | I'm not sure exactly how you need to format the dial string for Jingle |
| 05:13.33 | tish | alright, thank you! |
| 05:14.08 | mlsmith9999 | Anybody get Shared Line appearance working with the new Cisco 50x Line yet? |
| 05:32.26 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
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| 06:33.43 | joeyjones | i wonder how difficult it would eb to have an extension prompt for a long distance number, dial out, and then connect... |
| 06:33.46 | joeyjones | *be |
| 06:34.04 | ChannelZ | trivial |
| 06:34.16 | joeyjones | i figured so |
| 06:34.26 | ChannelZ | DISA() will even help you |
| 06:35.29 | *** part/#asterisk Zipper_32 (~Zipper_32@CABLE-72-53-35-89.cia.com) |
| 06:36.29 | joeyjones | ChannelZ: i may be wrong, but isn't DISA for allowing external access to the internal network? |
| 06:36.33 | *** part/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 06:37.24 | ChannelZ | isn't that the point? And "external" and "internal" don't really have any particular meaning |
| 06:37.47 | ChannelZ | But you can build it yourself either way |
| 06:38.20 | joeyjones | ChannelZ: i meant allowing someone on the inside of the network to call a particular extension, then allow them to place a long distance call through there |
| 06:39.17 | ChannelZ | yeah... that's just an extension really. |
| 06:40.21 | joeyjones | i think i really need to get an asterisk box up to play with before i ask any more questions... |
| 06:42.59 | joeyjones | if only i had a spare box... |
| 06:44.15 | ChannelZ | VM one |
| 06:44.39 | joeyjones | ChannelZ: i didn't know asterisk played well with VM... |
| 06:45.13 | ChannelZ | It should be mostly fine, particularly for playing around |
| 06:45.35 | ChannelZ | I'd be dubious of using it with hardware but I think some people are even doing that |
| 06:45.45 | joeyjones | ChannelZ: cool |
| 06:46.03 | joeyjones | i would have expected some issues, some things jsut don't agree with VM |
| 06:47.08 | joeyjones | ChannelZ: wow, there's 2 amazon EC2 AMIs for asterisk |
| 06:48.25 | greezmunkey | I've got my sleeves rolled up, and I'm digging around in /etc/asterisk. If I have no need for anything a particular .conf file provides, can I simply mv the file to example.conf.orig, ensuring that it won't be read when asterisk is started? |
| 06:48.31 | ChannelZ | I'm sure it depends on the VM. It needs good stable timing if it's involved with the audio |
| 06:48.36 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 06:48.51 | ChannelZ | But you could even run VirtualBox and fart around all day long, it'd be fine |
| 06:49.50 | ChannelZ | greezmunkey: pretty much yes |
| 06:50.10 | Docfxit | I would like to re-direct the output of dialplan show to a file. How can I do that? The output scrolls off the screen. I'm using ubuntu/asterisk |
| 06:50.38 | kaldemar | Docfxit: asterisk -rx 'dialplan show' > /path/to/file |
| 06:51.00 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-wciaanarfpmgvqpb) |
| 06:51.13 | Docfxit | Thank you. I'll try it. |
| 06:52.23 | joeyjones | ChannelZ: it seems that the EC2 AMIs in clude a 1000hz kernel timer |
| 06:52.57 | greezmunkey | ChannelZ: So, if I find functions in .conf file I know I don't need, it follows that I should noload the associated modules... |
| 06:52.59 | ChannelZ | Yah that'd help |
| 06:53.22 | ChannelZ | greezmunkey: sure |
| 06:54.18 | Docfxit | kaldemar It says no such command. |
| 06:54.27 | kaldemar | Docfxit: in shell, not in asterisk CLI. |
| 06:56.13 | Docfxit | kaldemar it says no such file or directory. Do I have to create an empty file first? |
| 06:57.07 | ChannelZ | /path/to/file should be something legit to you |
| 06:57.29 | ChannelZ | and the asterisk bin should presumably be in your path or you have to specify that too |
| 06:58.35 | ChannelZ | `which asterisk` -rx 'dialplan show' > /tmp/mydialplan.txt |
| 07:00.10 | *** join/#asterisk reber (~reber@212-198-99-56.rev.numericable.fr) |
| 07:05.39 | Docfxit | ChanServ I finally got it. Thanks. |
| 07:06.05 | greezmunkey | In the modules dir, some files are named res_ what does the res_ signify? |
| 07:07.12 | ChannelZ | resources. They're not apps, codecs, functions.. they're other things |
| 07:07.51 | greezmunkey | ty |
| 07:38.05 | *** join/#asterisk aeboi80 (~aeboi80@c-71-61-59-113.hsd1.pa.comcast.net) |
| 07:40.23 | p3nguin | I just had to hook up a DC-powered router. First time I've ever encountered one of those. |
| 07:42.28 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 07:43.12 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 07:43.19 | ChannelZ | what, like batteries? |
| 07:43.25 | p3nguin | yes |
| 07:43.55 | ChannelZ | Hmm. I'm sure that's exciting every couple of months |
| 07:44.11 | ChannelZ | Network's down! Someone go change the batteries! |
| 07:45.01 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 07:45.14 | ChannelZ | Is this for, like, Saddam's old hole in the ground? |
| 07:45.25 | p3nguin | It's at a cellular tower. |
| 07:45.45 | p3nguin | Everything runs on DC there, it seems. There's a huge battery with continuous charging and a generator backup. |
| 07:46.16 | p3nguin | I was just intrigued by the Cisco router being DC. |
| 07:47.10 | p3nguin | I had no idea such a thing existed. |
| 07:47.21 | *** join/#asterisk joobie (~joobie@CPE-124-176-177-112.vic.bigpond.net.au) |
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| 07:51.08 | joobie | guys im getting this "Found unknown media description format G.729 for ID 18" error - where asterisk is not recognizing rtpmap G.729 (it looks like most other providers use G729 for the identifier) |
| 07:51.34 | joobie | i was thinking i need to edit the code to fix this, but wanted to confirm this is definitely the problem before i go and do something like this |
| 07:52.03 | ectospasm | joobie: which G.729 codec are you using? |
| 07:52.31 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 07:52.48 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 07:52.54 | greezmunkey | p3nguin: most equipment in the central office is -48vDC as well. |
| 07:53.06 | joobie | ectospasm, im using the digium software one |
| 07:53.33 | joobie | but when i talk to a particular peer with G.729, it fails with that error |
| 07:53.43 | joobie | i can talk to other peers with g.729 OK |
| 07:53.59 | joobie | it looks like the rtpmap codec identifer in SDP is using G.729 and the others all use G729 |
| 07:54.09 | ectospasm | joobie: be sure your peer only speaks Annex A. If you've got Annex B, the Digium codec doesn't handle it |
| 07:54.20 | joobie | # |
| 07:54.20 | joobie | a=rtpmap:18 G.729/8000 |
| 07:54.20 | joobie | # |
| 07:54.20 | joobie | a=fmtp:18 annexb=no |
| 07:54.24 | joobie | that is part of SDP |
| 07:54.26 | joobie | in sip debug |
| 07:54.33 | joobie | presuming that means they are not using annex b |
| 07:54.44 | ectospasm | so it works for other peers? |
| 07:56.41 | joobie | ya |
| 07:56.56 | joobie | and when i debug sip for the other peers, it says "G729" not "G.729" |
| 07:57.16 | joobie | i looked at chan_sip.c and it gives me that error above if it cant make the codec to a known audio codec |
| 07:57.29 | joobie | so pretty sure it's just not liking the dot |
| 07:57.59 | joobie | but i also had a look at the code to try find where it specifies the rtpmap codec names and that is not too clear |
| 07:59.55 | ChannelZ | what version of * out of curiousity |
| 08:05.36 | ChannelZ | I see a table for mapping 'G729', 'G729A' and 'G.729' all to the same codec.. |
| 08:07.58 | p3nguin | Someone else just had this same issue a few days ago. |
| 08:07.59 | greezmunkey | Is there a way to show loaded modules? |
| 08:08.07 | p3nguin | module show? |
| 08:08.50 | greezmunkey | facepalms |
| 08:09.03 | greezmunkey | ty |
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| 08:13.08 | asterisk-learner | hello |
| 08:13.22 | rlfx | hi |
| 08:13.44 | asterisk-learner | i am facing an issue and i was hoping someone can help me out... |
| 08:14.00 | p3nguin | ~ask |
| 08:14.00 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 08:14.03 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 08:14.30 | asterisk-learner | ok, I have a virtual machine running centos and asterisk running inside it. |
| 08:14.53 | asterisk-learner | I also have a softphone running on the host machine (windows 7) and a SIP hardphone |
| 08:15.19 | asterisk-learner | i have been using this seup for months as test environnement |
| 08:15.22 | asterisk-learner | and it was working finr |
| 08:15.25 | asterisk-learner | fine* |
| 08:15.53 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:16.11 | asterisk-learner | even yesterday morning, all was fine but in the afternoon, suddenly I couldnt make any IAX2 calls |
| 08:16.17 | asterisk-learner | i am getting no route to destination |
| 08:16.26 | asterisk-learner | like there is a network problem or somehitng |
| 08:17.05 | ChannelZ | sounds like it |
| 08:17.07 | asterisk-learner | I restart my hard phone , and i can make SIP calls but no IAX2 calls |
| 08:17.29 | kaldemar | be more elaborate on "make calls". |
| 08:17.30 | asterisk-learner | i also restarted my virtual machine, asterisk, network services but no result |
| 08:18.02 | asterisk-learner | ok when dialing an iax2 extension i am getting : Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 08:18.03 | kaldemar | what is calling where when it fails? what do you see in asterisk when a call fails? |
| 08:18.26 | asterisk-learner | however Dialing a SIP extension works fine |
| 08:18.40 | kaldemar | so asterisk doesn't know any reachable address for the IAX2 peer. is it dynamic? |
| 08:18.49 | asterisk-learner | i also restarted my host machine today and cleared dns cahce under centos |
| 08:19.47 | asterisk-learner | kaldemar: yes, dynamic |
| 08:20.12 | kaldemar | and iax2 show peers doesn't show an address for it? |
| 08:20.13 | p3nguin | Does "iax2 show peers" list the peer you're trying to call? |
| 08:20.55 | ChannelZ | sounds like a dynamic peer that hasn't properly registered, or a bunk peer name |
| 08:21.16 | asterisk-learner | kaldemar: yes |
| 08:21.23 | asterisk-learner | it is showing all of them |
| 08:21.42 | joobie | ChannelZ, im running an old ver sec ill check |
| 08:21.43 | kaldemar | asterisk-learner: so no address. make the peer register to asterisk. |
| 08:21.45 | ChannelZ | yeah but does it know the right IP? |
| 08:22.00 | joobie | 1.4.22.1 |
| 08:22.04 | asterisk-learner | ChannelZ: but a complete system restart wouldnt make it work ? |
| 08:22.17 | ChannelZ | who knows? |
| 08:22.18 | p3nguin | The device has to register. |
| 08:22.21 | kaldemar | asterisk-learner: your system restart won't make any peers register to asterisk. |
| 08:22.21 | joobie | ChannelZ, where is the mapping table? |
| 08:23.01 | ChannelZ | rtp.c |
| 08:23.04 | ChannelZ | main/rtp.c |
| 08:23.29 | asterisk-learner | p3nguin: what should i do, to re-register it ? if it is already available in iax2.conf |
| 08:23.29 | ChannelZ | in your sip debug I assume you have an m=audio line? |
| 08:24.06 | kaldemar | asterisk-learner: you can't fix this by restarting your box or modifying configuration. make your device re-register. |
| 08:24.14 | p3nguin | I'd probably power cycle the device and see if that makes it register. I would also turn on qualify before I cycle the device so I can see what happens. |
| 08:24.17 | joobie | that's it chanelz |
| 08:24.20 | joobie | G.729 isnt in there |
| 08:24.20 | ChannelZ | you haven't answered, does the peer show up with the right IP in 'iax2 show peers'? |
| 08:24.29 | asterisk-learner | btw, I went into the code and added some print statemnets |
| 08:24.43 | kaldemar | ChannelZ: i assume he did answer to me. |
| 08:24.43 | asterisk-learner | it is failing in create_addr() of chan_iax2.c |
| 08:24.44 | ChannelZ | joobie: huh. It's in 1.4.27.1 anyway |
| 08:25.21 | kaldemar | asterisk-learner: the problem is obvious and you've been told how to fix it. |
| 08:25.58 | asterisk-learner | ChannelZ: no i didnt answer abt the ip , just if it is there, and i guess no , i am getting in the ip column : Unspecified |
| 08:26.00 | asterisk-learner | :-S |
| 08:26.06 | ChannelZ | joobie: I'm sure you could probably just patch your rtp.c (it's just an array) but maybe it's time for an upgrade? :) |
| 08:26.36 | p3nguin | Maybe if we say again a dozen times or so that the device has to register it will help you figure out what is needed. |
| 08:26.38 | asterisk-learner | kaldemar: ok i will try a power off and see |
| 08:26.42 | ChannelZ | so as everyone has said you need to make your device re-register |
| 08:26.53 | kaldemar | p3nguin: you take the first 11, i'll do #12. |
| 08:27.04 | p3nguin | We're already half way there. |
| 08:27.33 | kaldemar | ~cluebat asterisk-learner |
| 08:27.34 | infobot | ACTION pulls out a ClueBat (tm) and thwaps asterisk-learner. |
| 08:27.47 | ChannelZ | asterisk-learner: did your VM's IP change, such that your softphone has no idea who to register to? |
| 08:28.49 | asterisk-learner | ChannelZ: nope, i am using static IP, and when i faced the problem i tried disconnecting and reconneting the VM eth device |
| 08:29.21 | *** join/#asterisk evharten (~evharten@vpn.evertje.net) |
| 08:29.41 | *** part/#asterisk evharten (~evharten@vpn.evertje.net) |
| 08:30.56 | *** join/#asterisk evharten (~evharten@vpn.evertje.net) |
| 08:33.22 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 08:34.00 | joobie | ChannelZ, i want to upgrade but im scared ill break shit :P |
| 08:34.57 | ChannelZ | it's broken now :) |
| 08:41.30 | joobie | just this codec |
| 08:41.51 | joobie | like G729 install and ISDN install, i just cant remember crap about it |
| 08:42.05 | joobie | and it's a production system so if i upgrade and break it, i have a few hours to get it up |
| 08:42.22 | joobie | if i wanted to just patch rtp.c - how can i recompile this ? |
| 08:42.30 | ChannelZ | 'make' |
| 08:44.50 | rlfx | Hi, I wonder can I somehow make to not say anything when asterisk parks call? I want to call xxx and automatically park it. I use answer(), park(). And then server says parking lot numberÂ… I use v1.2. |
| 08:45.28 | ChannelZ | well there is the 's' flag to Park to make it not say the space number, but I have no idea if that's in 1.2.. |
| 08:46.06 | ChannelZ | core show application Park |
| 08:47.11 | rlfx | hm, I can't see any flats notation |
| 08:47.46 | ChannelZ | hmm.. too old then, dunno |
| 08:48.18 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:49.17 | greezmunkey | ChannelZ: checking like that on my system produces headings without any data, I'm assuming there is some documentation package I need to install? |
| 08:49.40 | ChannelZ | for doing what now? |
| 08:49.51 | greezmunkey | ChannelZ: core show app |
| 08:50.10 | greezmunkey | I'll have to see if there is any docs loaded. |
| 08:50.14 | ChannelZ | what do you mean it shows no headings |
| 08:50.52 | greezmunkey | ChannelZ: It shows the headings, but under each the is a msg saying it can't find the docs |
| 08:51.46 | ChannelZ | hmm.. did you build this from source or is it a package? |
| 08:52.20 | greezmunkey | ChannelZ: package from digium. I'm looking to see if there is a package that contains documentation. |
| 08:52.49 | ChannelZ | not sure then but maybe. There should be a "core-en_US.xml" or somesuch in /var/lib/asterisk/documentation |
| 08:53.12 | greezmunkey | ChannelZ: I'll check... |
| 08:55.15 | *** join/#asterisk Godfather_ (~godfather@89.131.93.52) |
| 08:55.21 | greezmunkey | ChannelZ: that explains that, /var/lib/asterisk/documentation isn't there! No problem, I'll fix that now. |
| 08:55.46 | *** join/#asterisk bjornts (~BTS@2001:700:200:10:c62c:3ff:fe16:a044) |
| 08:55.56 | ChannelZ | well it could be elsewhere depending on how the package was built, but it sounds like it could just be in a different package perhaps |
| 08:56.51 | greezmunkey | yeah, that file does not exist here. |
| 08:56.53 | ChannelZ | you can look at your asterisk.conf and see where 'astdatadir' or 'astvarlibdir' is pointed at to verify (not sure which it uses) |
| 08:57.46 | greezmunkey | ChannelZ: good point, I'll see. |
| 08:58.10 | ChannelZ | peeking at their YUM page it looks like there is a "-doc" package |
| 08:59.28 | greezmunkey | installing it now, looks pretty useful. |
| 08:59.47 | ChannelZ | if none of your 'core show' stuff is working, yes, massively |
| 09:01.11 | *** join/#asterisk raddy (~raddy@117.192.228.155) |
| 09:01.23 | raddy | Hello Everybody |
| 09:01.23 | greezmunkey | ChannelZ: that stuff seems to work just fine. |
| 09:01.41 | raddy | I have a Voxzone x100p fxo card. |
| 09:01.51 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:01.56 | raddy | In one system, irq conficts prevented the card from working. |
| 09:02.03 | greezmunkey | ChannelZ: Wierd part is that the package mgr indicates that I have it. Checking with yum. |
| 09:02.04 | ChannelZ | I meant 'core show application xxxx' and such that you said wasn't showing you much |
| 09:02.10 | raddy | But i installed the card in another system, |
| 09:02.35 | raddy | In this, there is no irq conflicts even in lspci |
| 09:03.19 | raddy | when i ran dahdi_hardware, the fxo card was shown. |
| 09:03.47 | raddy | But in all other dahdi related tools, the device is not being recognized. |
| 09:04.00 | ChannelZ | such as |
| 09:04.50 | ChannelZ | actually it doesn't matter.. they probably require the drivers to be running.. have you started them? |
| 09:05.12 | raddy | ChannelZ : I'll see. |
| 09:05.37 | *** join/#asterisk SeTTleR (~bernd@p5DDEFD58.dip.t-dialin.net) |
| 09:05.42 | SeTTleR | hi |
| 09:05.52 | ChannelZ | hEllO |
| 09:07.45 | *** join/#asterisk ccesario (~ccesario@187.10.57.94) |
| 09:08.20 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 09:14.23 | raddy | ChannelZ : I loaded the appropriate drive wcfxo, but it shows failed to initialize in spite of having no irq conflicts :( |
| 09:15.29 | petern_ | irq conflicts should've been gone with the days of ISA... |
| 09:15.37 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 09:15.39 | petern_ | pci's always had irq sharing, heh |
| 09:16.45 | *** join/#asterisk goddva (~glarsen@77.40.154.242) |
| 09:17.01 | raddy | petern_ : In my case, the device has it's own irq |
| 09:22.13 | *** join/#asterisk Jasnejac (kvirc@81.91.107.235) |
| 09:28.19 | ectospasm | real time devices, like PSTN adapters, have more serious interrupt requirements than your typical PCI device |
| 09:29.23 | *** join/#asterisk Tim_Toady (~moi@178.128.21.207.dsl.dyn.forthnet.gr) |
| 09:30.52 | petern_ | ectospasm, soundcards are pretty common real time devices |
| 09:31.49 | ectospasm | yeah, and interrupt problems will affect them, too |
| 09:31.58 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 09:32.11 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 09:32.15 | petern_ | yeah. irq sharing isn't a problem though, heh |
| 09:32.27 | ectospasm | not always, you're right |
| 09:32.57 | ectospasm | ...but folks trying to run a PBX on a desktop class system might be in for some trouble. |
| 09:33.29 | ectospasm | even if the PSTN adapter isn't sharing IRQs with anything else |
| 09:34.16 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 09:34.19 | ectospasm | I had a customer who managed to isolate their card on its own IRQ, and still the driver stopped processing due to too many missed interrupts |
| 09:36.00 | WIMPy | Yes, I geuss that's what causes all those HDLC errors I saw yesterday. |
| 09:37.03 | ectospasm | In Asterisk, HDLC aborts are notices, not errors or warnings. infrequent HDLC aborts are considered to be normal |
| 09:37.37 | WIMPy | I had them scoll by. |
| 09:38.14 | WIMPy | And someone doesn't seem to like them. I had Asterisk crash twice while doing that. |
| 09:38.22 | ectospasm | yeah, interrupt problems will usually cause bursty HDLC aborts |
| 09:38.24 | *** join/#asterisk sam_affable (sam@202.53.10.123) |
| 09:38.29 | sam_affable | hello there |
| 09:39.57 | sam_affable | my isp blocks port 5060, can i use 5061? |
| 09:40.31 | WIMPy | sam_affable: You can use whatever you want as long as you tell the other end. |
| 09:44.12 | WIMPy | Hmm. A million context switches in a second seems to be quite something. I guess that old Dial PIII sinply isn't up to it. |
| 09:45.04 | sam_affable | WIMPy: okie ty , the reason im asking this is i've setup iptel.org account in asterisk, i changed asterisk default port to 5061 and when i try to connect to my asterisk server using xlite ..it shows login timed out and xlite log shows its trying to connect to 5060 and outputs Discovered Blocked Firewall |
| 09:45.19 | *** part/#asterisk raddy (~raddy@117.192.228.155) |
| 09:45.53 | WIMPy | sam_affable: Yes, you have to tell you xlite to use the other port, off course. |
| 09:47.40 | sam_affable | WIMPy: oh okie i didnt find the the option to change SIP port in xlite ...using xlite for linux ...maybe i overlooked it ..will take a look at it again |
| 09:47.41 | sam_affable | ty |
| 09:50.23 | *** join/#asterisk raddy (~raddy@117.192.228.155) |
| 09:50.30 | raddy | Hello Everybody |
| 09:50.45 | raddy | What is the good fxo card. |
| 09:50.53 | raddy | Someone please suggest. |
| 09:51.15 | raddy | It should be not too costly and should asterisk compatible. |
| 09:51.44 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-khdfnfmgouujysng) |
| 09:52.24 | raddy | Please suggest some good fxo card. |
| 09:53.39 | tuxx- | please dont expect people to reply within 2 minutes. |
| 09:53.40 | tuxx- | :P |
| 09:55.03 | ectospasm | raddy: the TDM410 or AEX410 is good, and Digium will help you install it and get it working. |
| 09:55.09 | ectospasm | disclaimer, I work for Digium |
| 09:55.21 | tuxx- | :-D |
| 09:55.39 | raddy | ectospasm : I am from India, so i think Digium won't help me. |
| 09:55.54 | ectospasm | raddy: I help customers from India almost daily |
| 09:56.19 | raddy | ectospasm : But, via IRC ;) ? |
| 09:56.31 | ectospasm | raddy: if you can do IRC, you can do Skype. QED |
| 09:57.01 | raddy | ectospasm : But, would you install and configure the card in customer premises? |
| 09:57.30 | ectospasm | raddy: we do nothing physical, but we can do everything else up to but not including configuring Asterisk dialplan |
| 09:57.37 | sam_affable | WIMPy: ty that fixed my problem |
| 09:57.56 | ectospasm | raddy: we require SSH access for that |
| 09:58.17 | ectospasm | *root* SSH access, either direct or reverse |
| 09:58.36 | WIMPy | reverse? |
| 09:59.17 | ectospasm | WIMPy: yeah, they connect to our shell server, we tunnel back through that connection via SSH. Kind of a workaround for restrictive firewalls and systems behind NAT |
| 09:59.34 | ectospasm | WIMPy: uses the -R switch to ssh |
| 09:59.41 | WIMPy | aye |
| 10:00.02 | raddy | ectospasm : okkk, so you would do all sorts of things from remote ;) ah? |
| 10:00.50 | ectospasm | raddy: we do everything that proves the card functions by passing calls in both directions, the configuration of Asterisk dialplan is not included without an open source subscription |
| 10:01.35 | raddy | ectospasm : NP, that our duty. |
| 10:01.41 | raddy | ectospasm : I just want the card to work. |
| 10:01.58 | ectospasm | raddy: we will do everything we can remotely to make that happen |
| 10:02.14 | WIMPy | Yes, 6 pri + 8 bri was obviousely too much for my test box. No more errors with 4 pri + 8 bri. |
| 10:02.38 | ectospasm | raddy: and you get that service for the life (warranty) of the card, which currently is five years from the date of purchase. |
| 10:02.58 | ectospasm | WIMPy: sounds like a monster (-; |
| 10:03.34 | WIMPy | "Testing"... |
| 10:04.24 | ectospasm | Maybe this is naïve, but I always envision testing machines to be less beefy than their production counterparts. |
| 10:05.08 | WIMPy | A dual PIII-1266 is not exactely state of the art. |
| 10:05.24 | ectospasm | maybe for 2001 or so |
| 10:05.49 | WIMPy | yes |
| 10:07.10 | ectospasm | heh, my first PC that I'd call my own (not my family's machine) was a P75. I want to say that was a whopping 16MHz. May not be capable of carrying more than one call at a time (-; |
| 10:07.48 | WIMPy | I remember some time around '01/'02 whe I got such a thing the first time, I managed to compile the kernel for it in under a minute for the first time. |
| 10:07.57 | *** join/#asterisk mahiti-irc (~mahiti@122.166.127.74) |
| 10:08.21 | mahiti-irc | hi guys |
| 10:08.56 | mahiti-irc | i have configure my asterisk 1.4 and Hypermedia GSM gateway using this link http://www.tek-tips.com/viewthread.cfm?qid=1254720&page=521 |
| 10:09.14 | mahiti-irc | now can anyone help that how do i test my config? |
| 10:09.19 | WIMPy | I used other stuff before. Came late to the PC world. Was a 486-SX33. Start the compiler in the evening, and maybe it's finished by lunch time. |
| 10:09.49 | WIMPy | mahiti-irc: Like making calls? |
| 10:10.46 | *** join/#asterisk sam_affable (sam@202.53.10.123) |
| 10:11.34 | sam_affable | hey i was reading that dialing *65 on softphone will read back to asterisk extensions right ? i followed this guide registered my iptel account with asterisk and added extensions accordingly but when i dial *65 from x-lite , nothing happens and says call not approved , i chnged my asterisk sip settings , trunk and extension and x-lite to listen to port 5061 coz 5060 is blocked |
| 10:12.33 | ectospasm | sam_affable: the port and your feature extensions are independent |
| 10:13.08 | ectospasm | typically features.conf governs features dialed from analog extensions. You can define in dialplan *65 to do anything you want. |
| 10:13.09 | sam_affable | ectospasm: http://sysadminman.net/blog/2009/getting-started-with-freepbx-part-2-setting-up-an-extension-353 |
| 10:13.20 | sam_affable | oh okie |
| 10:13.29 | ectospasm | sam_affable: we don't support FreePBX here, try #freepbx |
| 10:13.57 | sam_affable | oh okie sorry ...just didnt know what is it releated asterisk or freepbx |
| 10:14.02 | WIMPy | ectospasm: Can yo tell me if I'm right on the HWEC things: If you have an VPM400M on a 4PRI card, it will only do EC for half the length, right? |
| 10:14.38 | ectospasm | WIMPy: IIRC, the VPM400 will echo cancel up to 120 bearer channels |
| 10:15.00 | ectospasm | WIMPy: I must admit I'm a bit foggy on the VPM400, we haven't sold that EC in a while |
| 10:15.37 | WIMPy | If I understand correctely it only has half the RAM of the bigger one, does it? |
| 10:15.50 | ectospasm | that I don't know |
| 10:16.28 | ectospasm | its algorithm for EC is deficient, IIRC, hence the switch to the VPMOCT128 |
| 10:16.42 | WIMPy | There doesn't seem to be too much info around on the HWEC modules. |
| 10:17.05 | ectospasm | WIMPy: yeah, their firmwares tend to be proprietary blobs |
| 10:17.30 | ectospasm | and the firmware is where most of the magic happens |
| 10:18.11 | WIMPy | What I think to understand so far is that VPM400 and VPMOCT64 have half the memory of VPM450 and VPMOCT128 and that teh OCT versions use a more advanced algorithm. |
| 10:18.51 | ectospasm | That actually makes sense, since the VPM400 and the VPMOCT64 are designed for dual-span, not quad-span cards |
| 10:19.11 | WIMPy | So my conclusion is that VPM400 and VPMOCT64 will only cancel echo of half the duration wen used on a 4pri card. |
| 10:19.53 | ectospasm | yep. THis seems to be a roundabout way to arriving at the answer to your question |
| 10:20.40 | WIMPy | Now the thing that I find interesting: Would it be possible to get them to doo the full duration on only two of the four ports? |
| 10:20.53 | mahiti-irc | WIMPy, sry for the missing detail, how do i do a call out? |
| 10:21.27 | mahiti-irc | using ooh323? |
| 10:22.17 | WIMPy | mahiti-irc: Not sure, what you're asking for. You have the dial statements on that page you shown us. |
| 10:22.42 | mahiti-irc | ya right WIMPy , but how to initiate the dial? |
| 10:22.44 | ectospasm | WIMPy: yes, that would be possible, but you'd essentially have to disable the other ports |
| 10:22.48 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 10:23.36 | WIMPy | ectospasm: I don't have the impression there's any option to do so. |
| 10:23.57 | ectospasm | WIMPy: sure there is, disable the other two spans (-; |
| 10:24.40 | WIMPy | I think that's too late. |
| 10:24.46 | greezmunkey | ChannelZ: woohoo! The only good thing about the last hour is that I ripped * out, fixed the broken repo definitions, and reinstalled - oh, and that it only took an hour. |
| 10:25.01 | WIMPy | it looks to me the EC is initialised before the spans are. |
| 10:27.12 | jsjc | is it possible to have PSTN==>phone==>asterisk==>user so if a PSTN call gets in phone will ring, asterisk will not Answer but Dial user and then when eithe both pick up the call the other phone stops ringing?? |
| 10:27.20 | jsjc | I dont know how to approach this issue |
| 10:28.07 | greezmunkey | jsjc: pstn call gets in where, a single phone? |
| 10:28.35 | WIMPy | jsjc: Not exactely sure what you're trying to do, but that sounds like what would be the normal behaviour. |
| 10:28.53 | kaldemar | jsjc: do you want two phones to ring when a call comes in and either one of them to be able to answer the call? |
| 10:30.27 | mahiti-irc | guys for the ooh323, do i have to place a .call file in /var/spool/asterisk to make a call out? |
| 10:30.50 | kaldemar | mahiti-irc: how about using a phone? |
| 10:31.08 | jsjc | kaldemar yes |
| 10:31.09 | ectospasm | mahiti-irc: call files are technology agnostic, which means you can use SIP, IAX2, ooh323, etc. |
| 10:31.16 | *** join/#asterisk sbszulu (~dundubala@41.16.58.94) |
| 10:31.21 | WIMPy | mahiti-irc: What's the purpose of your setup? You surely must have something else connected to place a call? |
| 10:31.41 | mahiti-irc | ya i have a Hypermedia GSM gateway connected to asterisk |
| 10:31.57 | WIMPy | And what else? |
| 10:32.01 | kaldemar | jsjc: Dial(Tech/resource&Tech/resource), for example Dial(SIP/100&SIP/200) |
| 10:32.12 | mahiti-irc | but that asterisk on a seperate machine. |
| 10:32.21 | mahiti-irc | both connected via LAN |
| 10:32.40 | WIMPy | mahiti-irc: What's the purpose of your setup? |
| 10:33.27 | mahiti-irc | we are trying to automate calls to certain groups |
| 10:33.53 | mahiti-irc | PHPAGI takes care finding the right number to dial |
| 10:34.00 | kaldemar | use call files or AMI originate. |
| 10:34.01 | WIMPy | So you don't plan to make normal phone calls with it? |
| 10:34.03 | *** join/#asterisk Weazel (~kvirc@213.8.83.6) |
| 10:34.14 | jsjc | WIMPY the purpose is. I have a big house asterisk machine is upstairs and there is just 2 phones in the whole house (1 up 1 down). Asterisk does some voip in the top floor and the bottom has pstn line. I want to be able to get pstn phonecalls upstairs and downstairs. |
| 10:34.20 | mahiti-irc | i used the same setup with PRI card ,asterisk 1.4 |
| 10:34.21 | WIMPy | Or channel originate to test. |
| 10:34.33 | mahiti-irc | that worked by using a .call file |
| 10:34.53 | mahiti-irc | i am replacing the PRI card with the asterisk H/W |
| 10:35.06 | WIMPy | mahiti-irc: It will work just the same as before then. Just put the other channel name in your call files. |
| 10:35.19 | mahiti-irc | ok |
| 10:35.27 | jsjc | WIMPy forgot to mention that there is pstn line all around house so asterisk can be connected to pstn but SIP phoneupstairs cannot be connected to pstn. |
| 10:37.10 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e4.bcn.adamo.es) |
| 10:37.50 | mahiti-irc | WIMPy, ok then i shud replace trunk which was 'DAHDI/g0/98xxxxx' to 'OOH323/g0/98xxxx' |
| 10:37.54 | mahiti-irc | right? |
| 10:39.00 | WIMPy | You need to replace g0 with the peer name as configured in the channel. Or name the peer g0. |
| 10:39.23 | WIMPy | ooh323.conf that is. |
| 10:39.23 | mahiti-irc | ok the peer name !! |
| 10:39.26 | mahiti-irc | ya |
| 10:39.31 | mahiti-irc | ok ok |
| 10:51.32 | *** join/#asterisk coppice (~chatzilla@m121-203-196-233.smartone-vodafone.com) |
| 10:55.24 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:58.03 | greezmunkey | thanks again everyone, g'night. |
| 11:01.44 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:02.08 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 11:04.34 | *** join/#asterisk slackytude (~slacky@2001:41b8:9bf:fe75:213:ceff:fef0:2369) |
| 11:06.05 | FlashDeluxe | Hi everyone! i am using Asterisk 1.4.21.2-BRIstuffed-0.4.0-RC3c and i got a problem. I have to say that the server is running since a few months without any problems. But now it happens quite often, that i cannot hear my calling partner for 5-10 seconds (but he can hear me). After these seconds, we can call furhter on, normally. Does anybody has got any suggestions? |
| 11:07.40 | ectospasm | FlashDeluxe: is there a reason you're using the BRIstuffed build? Asterisk 1.4.21.2 is really, really old |
| 11:09.47 | FlashDeluxe | this server was running since a few years and i dont want to write a new dialplan :( |
| 11:10.14 | WIMPy | FlashDeluxe: Is it only an audio problem od does perhaps the whose machine hang for a few moments? |
| 11:10.50 | ectospasm | FlashDeluxe: if you stay in the 1.4 branch, you shouldn't have to change your dialplan |
| 11:10.50 | WIMPy | FlashDeluxe: The dialplan shouldn't need much changing, if at all. |
| 11:10.58 | FlashDeluxe | i guess its just an audio problem |
| 11:11.07 | WIMPy | Unless it came from a 1.2 version. |
| 11:12.35 | tzafrir_laptop | FlashDeluxe, is this on a DAHDI channel? |
| 11:13.02 | FlashDeluxe | it is zap |
| 11:13.30 | ectospasm | that follows from the fact that you're running 1.4.21.2 |
| 11:14.01 | ectospasm | what version of Zaptel? |
| 11:15.51 | FlashDeluxe | zaptel-1.4.12.1 |
| 11:16.54 | FlashDeluxe | the customer is a bit complicated, so it is not so easy to update the server :/ |
| 11:20.32 | ectospasm | which PSTN adapter are you using, FlashDeluxe |
| 11:20.36 | ectospasm | ? |
| 11:21.03 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 11:21.12 | FlashDeluxe | i am using junghanns quadbri |
| 11:28.28 | *** join/#asterisk ccesario (~ccesario@189.20.219.10) |
| 11:31.00 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:32.32 | *** join/#asterisk dimm (~miniadmin@unaffiliated/dimm) |
| 11:32.46 | dimm | how i can look current settings in CLI ? |
| 11:32.54 | dimm | something like " |
| 11:32.59 | dimm | core show settings |
| 11:33.04 | dimm | or maybe another way? |
| 11:35.10 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
| 11:38.41 | DeHackEd | is there a way to listen for a fax machine's tones? I'm using AGI, and wait for digit isn't returning anything... |
| 11:45.28 | ectospasm | DeHackEd: what technology? DAHDI, SIP, etc. |
| 11:53.01 | cmnky | gm pbx'rs |
| 11:54.39 | ectospasm | mornin' |
| 11:55.20 | *** join/#asterisk raddy (~raddy@117.192.228.155) |
| 11:55.28 | raddy | Hello Everybody |
| 11:55.43 | raddy | Shall i buy openvox based cards? |
| 11:56.22 | WIMPy | based? |
| 11:56.24 | cmnky | dunno raddy ... my phone doesnt even work yet ... too noobish to answer |
| 11:56.27 | WIMPy | What kind of interface? |
| 11:56.32 | raddy | they are promising free support for customers who are using asterisk and many other opensource telephony systems. |
| 11:56.46 | ectospasm | raddy: you get free support with Digium adapters. |
| 11:56.57 | raddy | WIMPy : Interface means? PCI or PCI-Express? |
| 11:57.17 | cmnky | doesnt digium support asterisk development ? |
| 11:57.29 | WIMPy | raddy: The other end of the card. |
| 11:57.31 | ectospasm | cmnky: Digium owns the copyrights to Asterisk (-; |
| 11:57.49 | raddy | ectospasm : I don't know what is Digium Adapters, but a barebone Digium FXO card costs $300 in our currency. |
| 11:58.05 | cmnky | ectospasm, legal issues aside ... buying digium = revenue for further development of asterisk ? or is that just someones marketing dept fantasy ? |
| 11:58.25 | raddy | WIMPy: Please give some examples, then i'll tell you. |
| 11:58.32 | ectospasm | cmnky: Digium hires most of the core developers of Asterisk |
| 11:59.05 | cmnky | ectospasm, because sometimes its worth paying a little more ... to keep the community alive |
| 11:59.10 | ectospasm | raddy: I can't vouch for the OpenVox support quality, but Digium is awesome. |
| 11:59.16 | WIMPy | raddy: FXO was what I was looking for. |
| 11:59.30 | ectospasm | cmnky: it's a valid reason as any for choosing Digium over some other vendor (-; |
| 11:59.35 | cmnky | raddy, see my last comment |
| 11:59.51 | ectospasm | again, I work for Digium, so I'm biased |
| 12:00.18 | raddy | WIMPy : FXO Card. |
| 12:00.41 | cmnky | ectospasm, awesome ... can i call your support hotline to get my hacked ATA to make phone calls ? ;) |
| 12:01.05 | ectospasm | cmnky: no (-; |
| 12:01.20 | cmnky | ectospasm, its a linksys ... wait ... no ... its a "digium SPA2102" |
| 12:01.20 | raddy | cmnky : But, we want experiment with this, we are not huge corporate guys who spends millions of $$$ |
| 12:01.22 | cmnky | ;) |
| 12:01.35 | ectospasm | cmnky: that's not a supported model (-; |
| 12:01.44 | ectospasm | cmnky: we don't currently produce an ATA |
| 12:01.44 | cmnky | raddy, neither am I ... but you get what you pay for ... it is unavoidable |
| 12:01.45 | DeHackEd | acidchild: sorry. SIP |
| 12:02.10 | ectospasm | raddy: what's the warranty period of the OpenVox adapters? |
| 12:02.18 | raddy | WIMPy : 400p |
| 12:02.29 | cmnky | raddy, so lets say you spend $50 more on a digium ... that might very well fund some enhancement to asterisk you want, that you aren't even thinking about yet, and it'll be ready when you think about it |
| 12:02.31 | WIMPy | Well, I'd say there a lots of companies out there who manage very well to avoid giving you what you paid for. |
| 12:02.51 | raddy | ectospasm : Five year warranty. |
| 12:03.05 | ectospasm | raddy: and they have support during your business hours? |
| 12:03.27 | cmnky | raddy, where is the openvox guy ? ectospasm is right here, ready to help ... i dont see the openvox guy anywhere |
| 12:03.29 | WIMPy | raddy: I have no idea about analogue stuff beyond using modems for that a loooong loooong time ago. |
| 12:03.57 | raddy | cmnky : It is not $50 more, we bought a x100p for $50, the starting price of digium card is $300 usd. |
| 12:04.23 | cmnky | why not get a magickjack for $15 ? |
| 12:04.24 | ectospasm | X100P doesn't come with HWEC |
| 12:04.33 | ectospasm | cmnky: $15... A YEAR! |
| 12:04.50 | ectospasm | wait, I thought it was $19.99... A YEAR! |
| 12:04.53 | WIMPy | Use some resistors and your sound card :-) |
| 12:05.20 | ectospasm | I hear MagicJack uses Asterisk on the backend |
| 12:05.24 | cmnky | raddy, i completely understand being on a budget ... im a broke, 1 man operation myself .. but i also know $50 for a card that you never use ... is 100% wasted |
| 12:05.34 | cmnky | ectospasm, i wouldn't doubt it |
| 12:05.52 | DeHackEd | ectospasm: wait, that goes to you. it's SIP |
| 12:06.19 | ectospasm | SIP does not necessarily means digium|Asterisk |
| 12:06.28 | ectospasm | s/ns/n |
| 12:06.39 | ectospasm | s/ns/n/ |
| 12:06.42 | ectospasm | oops |
| 12:06.44 | ectospasm | fail |
| 12:09.37 | cmnky | i was looking at free ssl certs yesterday, but they want my personal info, and they are only 256 bit .... would it be better, to secure comms between 2 private parties, to just use self-signed certs ? |
| 12:10.25 | WIMPy | cmnky: Why would you need certificates on a point-to-point connection? |
| 12:10.35 | DeHackEd | ssl cert encryption strength would be either the hash (sha256?) or the public key (rsa 256?). RSA 256 shall not be considered under any circumstances |
| 12:10.58 | raddy | I am not looking for a world leading support. |
| 12:11.09 | cmnky | WIMPy, i want secure comms for voip and https ... among a few private parties |
| 12:11.28 | raddy | But i just want a working card that i can rely on even without support. |
| 12:11.44 | WIMPy | cmnky: So it's more than 2 parties. |
| 12:11.45 | cmnky | WIMPy, without dealing with vpn installation issues on clients |
| 12:12.08 | cmnky | WIMPy, its 2 right now ... but yes, its just a handful of peeps |
| 12:12.17 | jsjc | is it possible to check if a dahdi channel line is being used besides asterisk connected clients? |
| 12:12.59 | raddy | But that card is cerifified only on trixbox CE and Elastix |
| 12:13.09 | dimm | what is "Internal IP" in 'sip show settings' output ? |
| 12:13.53 | raddy | openvox has forum, wiki, etc. |
| 12:14.07 | cmnky | raddy, you want the cheap card, to work, without support, and be certified for asterisk ? |
| 12:14.20 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 12:14.31 | cmnky | raddy, shouldn't you be addressing that question to the cheap card maker ? |
| 12:15.04 | raddy | cmnky : I'll |
| 12:15.28 | DeHackEd | is away |
| 12:16.33 | ectospasm | raddy: remember, you get what you paid for. When you need to rely on your PSTN adapter, if you don't have reliable support, your system will be out of commission, and you may be hung out to dry. |
| 12:16.39 | cmnky | WIMPy, is there a better option ... like private keys that will also auth each user ? |
| 12:16.50 | ectospasm | raddy: if you're not intending to put the thing into live production, then it doesn't matter |
| 12:17.12 | cmnky | ectospasm, amen ... hardware that doesnt work is 100% wasted time and money ... nothing bugs me more |
| 12:17.31 | WIMPy | cmnky: There are endless possibilities, but if you want https, you need a cert. |
| 12:18.31 | cmnky | WIMPy, i know the possibilities are endless ... but the users are, users ... you know how that goes ... cant exactly explain to them how to get port knocking daemons and vpns working |
| 12:18.35 | ectospasm | raddy: Trixbox and Elastix are terrible, but I'm biased |
| 12:18.52 | cmnky | WIMPy, i could probably manage to get them a CERT for ffox tho |
| 12:19.11 | raddy | ectospasm : But they says, they are compatible with asterisk. |
| 12:19.57 | cmnky | WIMPy, could i create 2 certs for each users, 1 for them, 1 for the server ... with unique info ... such that .. it auths them as well as secures the channel ? |
| 12:20.09 | dimm | can i add some settings for some peer, in sip.conf ? |
| 12:20.36 | WIMPy | cmnky: I understand that is possible, but I have never seen anyone doing it. |
| 12:21.08 | *** join/#asterisk seabro (~sb@188-220-144-40.zone11.bethere.co.uk) |
| 12:21.16 | raddy | ectospasm : One more doubt, OpenVox recommends to buy the FXO100 with A400P to construct a workable system, he compares A400P to Digium TDM400p and FXO100 with X100M. |
| 12:21.26 | cmnky | WIMPy, cool .. i'll research it .. thnks |
| 12:21.29 | raddy | ectospasm : Why should i buy both? |
| 12:22.21 | cmnky | raddy, buy a digium card |
| 12:23.16 | raddy | cmnky : My question is, why should i buy two cards? |
| 12:24.19 | cmnky | raddy, because ... you are lost with the cheap card vendor ... you'll have to ebay it ... and buy a digium to get support and make it work ;) |
| 12:25.02 | raddy | cmnky : don't joke, why should i buy two cards even from Digium? |
| 12:25.11 | cmnky | raddy, when you start to value the time messing about ... $300 might be less expensive |
| 12:25.14 | cmnky | im not joking |
| 12:26.05 | cmnky | anyone who has ever tried to save a couple dollars, on hardware that isnt proved, only to find out trying to make it work was a complete waste of time and a headache, understands exactly what im saying |
| 12:26.31 | cmnky | im guilty of it ... newegg mobo special ... $39.95 ... absolutely useless ... doesn't work ... straight in the garbage |
| 12:26.33 | raddy | cmnky : It seems you still didn't understand what i am asking. |
| 12:26.47 | cmnky | right ... im lost ... sorry |
| 12:26.54 | raddy | cmnky : why should i buy Digium TDM400p and X100M combined? |
| 12:27.34 | cmnky | ectospasm can answer that probably ... assuming you provide him with your usage scenario |
| 12:28.45 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 12:29.03 | raddy | What other brands of cards i can buy |
| 12:29.04 | ectospasm | bbiab |
| 12:29.10 | cmnky | gm ukine_work |
| 12:29.20 | raddy | Please suugest a decent middle ground |
| 12:29.37 | ukine_work | mornin' cmnky |
| 12:30.32 | cmnky | raddy, i dont think you've explained to anyone what it is you are trying to achieve ... without that info ... recommendations can't possibly be made |
| 12:30.44 | cmnky | and im a real asterisk noob ... so i cant help even if you do explain it |
| 12:30.55 | raddy | I just need one port FXO card. |
| 12:30.58 | cmnky | ukine_work, hows tampa this morning ... chilly ? |
| 12:31.18 | raddy | What else information any one want |
| 12:31.55 | FlashDeluxe | hi at all! I still get this error, that i cant hear the person i called and now i got an output of the problem i think |
| 12:32.02 | FlashDeluxe | http://pastebin.ubuntu.com/538964/ |
| 12:32.11 | FlashDeluxe | maybe this could be helpful? |
| 12:32.44 | FlashDeluxe | this error occurs quite often |
| 12:35.36 | ukine_work | cmnky, 42F |
| 12:35.38 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 12:36.06 | cmnky | ukine_work, i saw your ip address ... im near tampa on rr as well |
| 12:36.48 | ukine_work | brrr :] |
| 12:38.18 | seabro | been trying to setup ISDN BRI on Ast 1.6.0.26 (Trixbox) and get Segmentation Fault, asterisk dies and restarts everytime a call comes in. Any ideas as to cuase? |
| 12:38.53 | *** part/#asterisk raddy (~raddy@117.192.228.155) |
| 12:39.19 | cmnky | poor raddy |
| 12:39.37 | dimm | how i can edit internal ip? http://pastebin.com/YTPeX0mZ |
| 12:40.19 | FlashDeluxe | maybe anybody can tell me what "Set option AUDIO MODE, value: OFF(0) on Zap/1-1" does portend? |
| 12:42.48 | WIMPy | seabro: That's a complex topic in itself, but you will probably have to go along with what you can find out in #trixbox. |
| 12:45.17 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:45.17 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:46.00 | *** join/#asterisk toe (~toe@static.78.209.40.188.clients.your-server.de) |
| 12:46.46 | rlfx | Can I somehow view parked calls list? |
| 12:47.43 | seabro | Thanks WIMPy |
| 12:49.39 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:50.13 | toe | hi. do i need additional ports to use multiple sip accounts for incoming calls? after a 'core restart', asterisk log tells me that the accounts are reachable, but within 20 seconds it starts logging error messages: log snippet: http://www.dpaste.org/by0z/ |
| 12:50.26 | toe | does anyone have a clue where to start troubleshooting? |
| 12:53.23 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:53.31 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 12:53.31 | WIMPy | toe: No. Does the registration ever succeed? |
| 12:54.28 | *** join/#asterisk Akiraa (~Akira@92.81.172.106) |
| 12:57.56 | toe | WIMPy: i don't know |
| 12:58.12 | toe | i will try with main (and working) sip account deactivated |
| 12:58.52 | WIMPy | Maybe yur registration attemps are ignored because of wrong credentials? |
| 12:59.42 | toe | wrong credentials? what does that mean? i am no native speaker ;) |
| 13:00.36 | toe | same problem with only those ekiga accounts active... |
| 13:01.00 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 13:01.46 | toe | credentials = username/password? |
| 13:02.14 | WIMPy | yes |
| 13:03.53 | toe | should be correct |
| 13:05.13 | toe | i will cross try it with a softphone |
| 13:06.50 | *** join/#asterisk Lord_Rahl (~Lord_Rahl@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 13:08.26 | Lord_Rahl | I have a weird problem. I have had paging working fine. Then I add a voip channel now everytime I try to age I get : [Dec 2 08:02:31] WARNING[4815]: app_meetme.c:803 build_conf: Unable to open pseudo device -- <Local/227@default-6192,1> Playing 'conf-invalid' (language 'en') [Dec 2 08:02:31] WARNING[4815]: file.c:1293 waitstream_core: Unexpected control subclass '-1' |
| 13:09.14 | WIMPy | Lord_Rahl: You don't have DAHDI running. |
| 13:09.32 | toe | credentials are correct |
| 13:10.09 | *** join/#asterisk cjk (~cjk@85.93.217.128) |
| 13:10.44 | cjk | hi, i have a pbx connected to my pri (asterisk) and when i send a fax connected to the pbx i receive the callerid in overlap dial (dtmf) but i get duplicates. any idea? |
| 13:10.59 | Lord_Rahl | WIMPy, Your right ,but how could you tell? I see no refernce to dahdi |
| 13:11.41 | russellb | "Unable to open pseudo device" - the pseudo device is a DAHDI thing |
| 13:11.48 | WIMPy | Lord_Rahl: The timing pseudo device is a dahdi device. |
| 13:12.04 | russellb | we should probably add "Is DAHDI loaded?" to that error message |
| 13:12.08 | joeyjones | is there a eay to tell what number forwarded a call to a DID? ie. gsm cell phone forwarding |
| 13:12.27 | Lord_Rahl | WIMPy, oohhh learn my first thing today :) Thanks |
| 13:12.29 | WIMPy | russellb: Sounds like a good idea. |
| 13:12.31 | tzafrir_laptop | simply: "unable to open DAHDI pseudo device"? |
| 13:12.44 | russellb | tzafrir_laptop: yeah, that works |
| 13:14.01 | WIMPy | russellb: I tried cahn_misdn yesterday, BTW: Didn't go too well. |
| 13:14.37 | russellb | WIMPy: I forwarded your feedback to our ISDN expert. He's taking a look. He actually sent me a response that I need to go through. |
| 13:16.17 | *** join/#asterisk jamko (~chatzilla@173-162-11-81-naples.hfc.comcastbusiness.net) |
| 13:17.05 | WIMPy | russellb: I also had a little chat with malcolmd. Otherwise I try to summarize my experiences to some sort of guide. |
| 13:17.25 | russellb | WIMPy: ah, good. That would explain some stuff that came from him yesterday :-) |
| 13:18.00 | WIMPy | Damn. Now you know it came from the same guy :-) |
| 13:18.13 | russellb | haha, yep. |
| 13:18.18 | russellb | you shouldn't have told me. |
| 13:18.43 | WIMPy | I guess it's too late now. |
| 13:18.49 | russellb | indeed. |
| 13:19.01 | jamko | Morning.... I have always accepted the default settings for MOH, but in my current version 1.6.12, the "robot dity" is a bit intense. I checked out the musiconhold.conf file, but I am not seeing how to change that song as the default. I tried moving the robot dity files out of the moh dir, but that just kicked up an error that it can't find robot dity. |
| 13:19.27 | russellb | jamko: reload after you remove the file |
| 13:19.50 | jamko | russellb: I did, but same result. Let me try again though. |
| 13:20.12 | russellb | module reload res_musiconhold.so |
| 13:20.18 | jamko | ok |
| 13:20.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:21.07 | russellb | "DAHDI" added to those app_meetme error messages btw, so hopefully that will help in the future |
| 13:21.27 | *** join/#asterisk bip (~bip@unaffiliated/bip) |
| 13:23.23 | jamko | russellb: reloading the module didn't do it, but restarting asterisk core did. Thanks! |
| 13:23.39 | russellb | hm, thanks ... i guess that could be a bug |
| 13:23.46 | russellb | i think a reload should handle that |
| 13:23.53 | russellb | feel free to report it on issues.asterisk.org if you feel up to it |
| 13:24.24 | WIMPy | Thinking of that, I think I once found that I had to unload, then load instead of just reload. |
| 13:24.48 | russellb | to be clear, reload is not unload+load |
| 13:25.06 | russellb | reload tells the module to reload its config |
| 13:25.44 | jamko | well I tried CLI>reload ... and CLI>module reload res_musiconhold.so |
| 13:26.07 | russellb | yep, that should have worked in an ideal world. |
| 13:26.11 | jamko | ok. |
| 13:26.24 | russellb | I blame leifmadsen. |
| 13:26.34 | leifmadsen | goddamnit no! |
| 13:26.36 | leifmadsen | not again! |
| 13:26.37 | jamko | Also, does asterisk choose the default song alphabetically? |
| 13:26.49 | leifmadsen | jamko: by default I think it does, but you can configure that to alpha or random |
| 13:26.57 | jamko | thanks. |
| 13:27.03 | leifmadsen | JamesHarrison: in musiconhold.conf |
| 13:27.06 | russellb | jamko: whatever order they are presented when the directory gets opened ... probably alphabetical, unless you set random=yes |
| 13:27.45 | leifmadsen | ;sort=random ; Sort the files in random order |
| 13:27.47 | jamko | I'm a big fan of the one that sounds like a rip off of Floyd's Dogs. |
| 13:27.54 | russellb | leifmadsen: yeah, that ... :-p |
| 13:27.56 | leifmadsen | :D |
| 13:27.58 | leifmadsen | 1.8 ftw |
| 13:28.03 | leifmadsen | russellb: you should write a book |
| 13:28.07 | russellb | i totally should |
| 13:28.09 | russellb | ~newbook |
| 13:28.09 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/ |
| 13:28.19 | leifmadsen | this is the note that I found most interesting: |
| 13:28.20 | leifmadsen | ;sort=alpha ; Sort the files in alphabetical order. If this option is |
| 13:28.20 | leifmadsen | ; ; not specified, the sort order is undefined. |
| 13:28.27 | leifmadsen | undefined! |
| 13:28.29 | russellb | \o/ |
| 13:28.38 | leifmadsen | coffee me! |
| 13:28.43 | russellb | it's whatever the open directory, iterate files code presents |
| 13:28.46 | leifmadsen | cD~ |
| 13:28.48 | russellb | it's not defined |
| 13:29.56 | cmnky | does that 3rd edition asterisk book cover 1.6.x ? |
| 13:30.07 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 13:30.17 | russellb | not directly, it's specifically on 1.8 |
| 13:30.24 | russellb | but 1.8 is 1.6 + some more stuff |
| 13:30.58 | cmnky | russellb, but 2nd edition only covers 1.4 ... right ? |
| 13:30.58 | leifmadsen | s/stuff/awesome stuff/ |
| 13:31.02 | russellb | correct |
| 13:31.06 | jamko | Has anyone accomplished or tried least cost routing with Func_ODBC? |
| 13:31.10 | russellb | but see, 1.4 is just 1.6 minus some stuff |
| 13:31.16 | leifmadsen | jamko: heck ya I have |
| 13:31.20 | cmnky | so id be better served to get 3rd edition and skip 2nd ? |
| 13:31.31 | russellb | shrugs |
| 13:31.36 | leifmadsen | 3rd edition is not done :) |
| 13:31.36 | *** join/#asterisk sbszulu (~dundubala@41.16.32.42) |
| 13:31.36 | cmnky | as is the usual case ;) |
| 13:31.40 | russellb | 2nd is available now, 3rd is still probably 3-4 months from print |
| 13:31.44 | russellb | get them both! |
| 13:31.45 | leifmadsen | ya that |
| 13:31.49 | leifmadsen | THAT! |
| 13:31.54 | cmnky | leifmadsen, when you finish it ... i'll buy a copy ... get to work ;) |
| 13:31.59 | russellb | 3rd edition is mostly a rewrite anyway |
| 13:32.02 | toe | do you have any recommendations for free sip providers? |
| 13:32.05 | russellb | leifmadsen: you just got burned |
| 13:32.05 | leifmadsen | you can pre-order! buy it now! |
| 13:32.21 | leifmadsen | I don't recommend free sip providers |
| 13:32.27 | leifmadsen | ...perhaps Google Voice :) |
| 13:32.31 | cmnky | leifmadsen, nice try ... i dont buy wal-mart cash cards either tho ;) |
| 13:32.33 | russellb | Google Voice, yes. :-) |
| 13:32.53 | russellb | but that's not SIP. |
| 13:33.00 | russellb | but it's free ... and probably not evil |
| 13:33.07 | leifmadsen | and probably works |
| 13:33.08 | jamko | leifmadsen: cool. I have someone trying to push me to Kamailio because of "advanced" features... But I am arguing that most of the same can be accomplished with asterisk. |
| 13:33.17 | leifmadsen | jamko: and easier |
| 13:33.20 | russellb | Kamailio and Asterisk are not the same thing at all |
| 13:33.22 | cmnky | google is totally evil ... if they werent, they would be advertising they arent ;) |
| 13:33.24 | leifmadsen | not even close |
| 13:33.25 | russellb | they are often used in combination |
| 13:33.35 | cmnky | er..wouldnt be ... |
| 13:33.39 | russellb | ~googlevoice |
| 13:33.39 | infobot | For information on setting up calls with google using Asterisk 1.8, see https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 13:33.50 | leifmadsen | russellb: look you owning the bot today |
| 13:33.56 | russellb | that's how i roll |
| 13:34.04 | leifmadsen | f7esh |
| 13:34.06 | russellb | leifmadsen: is that in the external services chapter, too? |
| 13:34.11 | leifmadsen | russellb: yes sir! |
| 13:34.17 | toe | hmm. any free iax provider recommendations then? ;) |
| 13:34.29 | leifmadsen | that is even less likely than a free sip provider me thinks |
| 13:34.50 | cmnky | i just got a free DID/incomming from ipcomms ... havent used it yet because im a noob |
| 13:34.51 | russellb | infobot: googlevoice is also see the External Services chapter: http://ofps.oreilly.com/titles/9780596517342/ch17.html |
| 13:34.51 | infobot | okay, russellb |
| 13:34.57 | leifmadsen | it is not free to interact with the PSTN |
| 13:34.57 | cmnky | you'll get a dixon, ca # |
| 13:35.29 | cmnky | its sip tho, not iax |
| 13:36.24 | WIMPy | Does DAHDI support NT-mode on any BRI cards other than Digiums b410p without third party addons or patches? |
| 13:37.00 | *** join/#asterisk nfi|ermes (~sdff@host54-26-static.93-94-b.business.telecomitalia.it) |
| 13:37.46 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:39.53 | toe | oh and forgot to mention: i don't need a landline phone number |
| 13:40.35 | jamko | russellb: right but I have a theory that Kamailio just lacks asterisk functions, being the main difference. |
| 13:40.40 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:40.56 | russellb | the main difference is that Kamailion is a SIP Proxy (an amazing one at that) |
| 13:41.02 | leifmadsen | toe: I still suggest Google Voice since you can get everything you're asking for. |
| 13:41.27 | leifmadsen | and that asterisk is a Back2Back User Agent (not a SIP proxy) |
| 13:41.38 | russellb | Asterisk is not a proxy ... it's a multiprotocol application server, PBX, B2BUA, whatever... |
| 13:41.43 | russellb | they're just for different things in a network |
| 13:41.45 | leifmadsen | ^^^ |
| 13:41.55 | russellb | they are both awesome. |
| 13:41.58 | russellb | <3 Kamailio and Asterisk |
| 13:42.24 | WIMPy | prefers to add a third: LCR |
| 13:42.25 | cmnky | i started a SIP server project ... but its *real* low on the priority list |
| 13:42.47 | russellb | I didn't mean to exclude other applications which may also be awesome :-) |
| 13:42.56 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 13:43.09 | toe | "Thanks for visiting Google Voice. We're not yet open for users outside the US, but are planning to expand our service to additional countries in the future." |
| 13:43.12 | toe | grmpf |
| 13:43.17 | russellb | burn |
| 13:43.59 | cmnky | toe, they haven't perfected the speech-to-text data mining algorithms for other languages yet ;) |
| 13:44.02 | toe | i am not even in nl, but my google account belongs there :D |
| 13:44.05 | WIMPy | It actuially isn't. But I think it has the best potential in splitting different tasks to different applications that hopefully do their part better than a big one trying everything. |
| 13:44.37 | leifmadsen | toe: I just used a VPN to a server in the USA and then created my Google Voice account :) |
| 13:44.39 | toe | cmnky: my asterisk installation speaks only english for now ;) |
| 13:44.48 | leifmadsen | once it is setup, then you can use it from any server on the internet |
| 13:45.01 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 13:45.01 | toe | leifmadsen: that's kind of how i set up my netherlands account |
| 13:45.25 | cmnky | toe, uh ... yeah ... but google wants to evesdrop on everything anyone says ... get it ? |
| 13:45.31 | russellb | WIMPy: the architecture of Asterisk SCF may interest you, as well, then. |
| 13:45.31 | toe | sure |
| 13:45.54 | toe | that's why i am not that enthusiastic about using google voice and so on |
| 13:46.10 | russellb | well you're the one trying to get free service :-) |
| 13:46.14 | WIMPy | I've seen it, but not taken a real look so far. |
| 13:46.22 | toe | signing in just needed me to change my google cookies policy (from deny to allow :D) |
| 13:46.23 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:46.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:46.27 | cmnky | thats what i dont like about google ... their entire business model revolves around everyone telling them everything ... and they are worse than the NSA,CIA,KGB, etc in that regard ... because the users are all doing it voluntarily |
| 13:46.36 | jamko | russellb: Let me ask this then.. I have had scenarios where behind a NAT, if I did not specify asterisk as the "outbound proxy" in the phone's configuration, I would have problems. However after specifying asterisk as the outbound proxy int he phone, the issues would resolve. I know this may sound ridiculous but I always wondered why that was, considering asterisk is not proxy. |
| 13:46.58 | jamko | the phone being what was behind the nat. |
| 13:47.24 | russellb | I don't know, I guess I'd have to look closer ... presumably the phone was sending the calls to Asterisk, anyway |
| 13:47.28 | russellb | or it should have been ... |
| 13:47.42 | *** join/#asterisk Lipsum (~sengebret@77.40.154.242) |
| 13:48.17 | leifmadsen | and the phones word use of "proxy" does make asterisk a proxy :D |
| 13:48.32 | cmnky | "The SIP channel does not support encryption in this version of Asterisk. " |
| 13:48.40 | cmnky | anyway around that ? |
| 13:48.58 | leifmadsen | ya, use 1.8 |
| 13:49.10 | leifmadsen | beyond that, no |
| 13:49.15 | cmnky | 1.8 is your answer to everything leif ;) |
| 13:49.21 | leifmadsen | 1.8 contains SRTP |
| 13:49.24 | leifmadsen | 1.6.2 does not. |
| 13:49.44 | leifmadsen | although it's just code, so you could potentially backport chan_sip and all the functionality for SRTP to 1.6.2 :) |
| 13:49.59 | leifmadsen | cmnky: all your questions lead to an answer of 1.8 |
| 13:50.00 | WIMPy | But srtp in 1.8 crashes Asterisk if you use the wrong phone at the wrong time. |
| 13:50.10 | leifmadsen | WIMPy: issues filed? |
| 13:50.27 | WIMPy | Yes. I wan't the first to find out. |
| 13:50.38 | WIMPy | wasn't |
| 13:50.44 | leifmadsen | WIMPy: what are the issue numbers? Just curious since those would likely get a higher priority. |
| 13:51.01 | cmnky | um ... no ;) |
| 13:51.01 | bip | question: when people type the wron extension they receive a goodbye message in english, what should i do to send a differnt message ? |
| 13:51.21 | leifmadsen | bip: look at the dialplan that is being executed and change the prompt? |
| 13:51.44 | leifmadsen | bip: if you're using a GUI to make changes, I'd suggest you find the appropriate channel that supports your project |
| 13:52.19 | WIMPy | leifmadsen: 17976 |
| 13:52.30 | bip | I will leifmadsen , when they will eventually wake up ;-) |
| 13:52.45 | Lipsum | Is it OK to ask questions about Asterisk-Java API in here? |
| 13:54.11 | bip | it there a file where the dialplan is written ? |
| 13:54.24 | WIMPy | leifmadsen: And while you're at it, you can close 18402 and 18409. I was wrong there. |
| 13:54.26 | leifmadsen | WIMPy: you're using the same phone as reported in that issue right? It would be better if you could add a note clarifying exactly what hardware and firmware version you are using. |
| 13:55.22 | leifmadsen | Lipsum: sure, why not :) |
| 13:55.38 | leifmadsen | bip: yes ;) extensions.conf |
| 13:55.50 | leifmadsen | ok, I have a book to work on, I'm out |
| 13:56.08 | bip | yeah I had a look there |
| 13:56.13 | Lipsum | When using executeCliCommand(), shouldn't it be possible to execute shell commands (ie. ! commands)? |
| 13:56.32 | Lipsum | Everything works, except those - for me at least. |
| 14:04.58 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 14:08.29 | *** join/#asterisk l0st-soul (~lost@scylla.sysif.net) |
| 14:08.30 | l0st-soul | hello |
| 14:18.13 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
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| 14:23.30 | cjk | hi i have a pri in net mode and digits come in overlap. they are exactly double detected. any idea ? relaxdtmf did not change anything |
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| 14:37.14 | Katty | hi asl |
| 14:40.00 | *** join/#asterisk boch (~fseratti@190.220.65.19) |
| 14:40.04 | boch | hi all |
| 14:40.57 | boch | what can be the reason asterisk is not loading my custom application? how can i debug it ? |
| 14:44.32 | Katty | goooooooooooooooood morning |
| 14:44.44 | *** join/#asterisk Sipster_away (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
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| 14:52.45 | _Raptor_ | are there any known problems with iax from 1.6.2.14 to 1.8.0? |
| 14:53.24 | WIMPy | _Raptor_: I had problems, but I couldn't find out what was going on. |
| 14:54.02 | _Raptor_ | WIMPy: what kind of problems? the user on the 1.6.2.14 can't hear the 1.8.0 |
| 14:54.36 | WIMPy | very bad audio quality and excessive bandwidth usage. |
| 14:55.12 | WIMPy | It didn't happen between 1.8.0 and 1.4.21 however. |
| 14:55.38 | WIMPy | And using zoiper/iax to 1.8.0 was also working. |
| 14:56.53 | _Raptor_ | WIMPy: the 1.6.2.14 did use very much cpu |
| 14:57.16 | _Raptor_ | it still worked with 1.6.2.1 |
| 14:58.54 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
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| 15:23.35 | *** join/#asterisk Amnesia (~Amnesia@unaffiliated/amnesia) |
| 15:23.40 | Amnesia | hi |
| 15:23.50 | Amnesia | I've got a digium tdm410 card here |
| 15:24.11 | Katty | fancy |
| 15:24.13 | Amnesia | but I seriously do not know how to configure it to let incoming / outgoing call go through it |
| 15:24.13 | Katty | i got a dr pepper |
| 15:24.48 | Amnesia | any idea;/? |
| 15:25.08 | Katty | if it was me, i'd google for some samples |
| 15:25.26 | Amnesia | already did that, but can't find any suiting solutions |
| 15:27.20 | Katty | solutions is such a fun word. |
| 15:27.51 | Amnesia | :/ |
| 15:28.19 | Katty | perk up sunshine |
| 15:28.29 | Katty | if you're patient, someone might volunteer to help |
| 15:28.45 | goddva | Amnesia: asterisk version? |
| 15:29.01 | Amnesia | 1.6.2.7 |
| 15:29.43 | goddva | asterisk-learner: dahdi_hardware output? |
| 15:30.01 | goddva | Amnesia: dahdi_hardware ouytput? |
| 15:30.12 | Amnesia | Wildcard AEX410P |
| 15:30.31 | Amnesia | (guess you don't need the bus address:)) |
| 15:30.38 | goddva | + or - at the end? |
| 15:31.07 | Amnesia | ehm the kernelmodule = wctdm24xxp- |
| 15:31.14 | goddva | ahh.. so its not loaded |
| 15:31.32 | goddva | output from: lsmod | grep dahdi |
| 15:31.37 | Amnesia | I just saw:o |
| 15:31.41 | Amnesia | no it isnt loaded:$ |
| 15:31.44 | goddva | :) |
| 15:32.02 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:32.04 | Amnesia | hows the main module called? |
| 15:32.05 | goddva | try some modprobing... then restart asterisk to see if that helps.. |
| 15:32.27 | goddva | dahdi, dahdi_transcode and dahdi_echocan_mg2 |
| 15:32.59 | goddva | you have generated/created all the other files? |
| 15:33.04 | Amnesia | I'm getting "invalid argument" |
| 15:33.14 | Amnesia | what'd you mean with all the other files? |
| 15:33.15 | goddva | what are you trying |
| 15:33.23 | Amnesia | modprobing dahdi |
| 15:33.30 | Amnesia | sudo modprobe dahdi |
| 15:33.32 | asterisk-learner | goddva: ??? |
| 15:33.36 | goddva | like the /etc/dahdi/system.conf |
| 15:33.47 | goddva | asterisk-learner: sorry, was tabbing... |
| 15:33.56 | Amnesia | yep, did that |
| 15:34.02 | goddva | and the /etc/asterisk/dahdi-channels.conf |
| 15:34.05 | asterisk-learner | goddva: ok np :-) |
| 15:34.11 | goddva | ;) |
| 15:34.48 | goddva | try: modprobe wctdm24xxp |
| 15:35.01 | goddva | then: lsmod | grep dahdi |
| 15:35.27 | Amnesia | already tried that |
| 15:35.48 | Amnesia | getting " error inserting wctdm24xxp (/lib......0): invalid argument |
| 15:35.54 | *** join/#asterisk [netman] (~netman@131.Red-83-39-224.dynamicIP.rima-tde.net) |
| 15:36.04 | goddva | did you compile dahdi on this box? |
| 15:36.13 | Amnesia | nope |
| 15:36.20 | Amnesia | downloaded it through the ubuntu repositories |
| 15:36.35 | goddva | hmmm.. |
| 15:36.55 | Amnesia | the modules do exist |
| 15:37.53 | Amnesia | oh wait... |
| 15:38.00 | Amnesia | there has been a kernel update |
| 15:38.02 | Amnesia | perhaps that's it |
| 15:38.03 | goddva | ahhh |
| 15:38.07 | Amnesia | brb, gonna switch kernel |
| 15:38.09 | goddva | that was my net question |
| 15:38.15 | goddva | next |
| 15:38.23 | Amnesia | rebooting sec:) |
| 15:38.32 | goddva | hurry up - need to go soon :) |
| 15:38.48 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:38.54 | JerJer | who's your dahdi? |
| 15:39.02 | goddva | hehe |
| 15:39.54 | Amnesia | booting up |
| 15:39.57 | Amnesia | thanks for helping me out btw |
| 15:39.58 | Katty | i want a dahdi |
| 15:40.34 | Amnesia | crap udev is configured for the newer kernel got to recompile |
| 15:40.36 | bip | that reminds me a bruce sprinngsteen song actually |
| 15:40.49 | Katty | reminds me of something else ^_^ |
| 15:40.59 | bip | oh well |
| 15:41.22 | bip | evrybody is entitled to his/her own set of memories ;-) |
| 15:41.37 | Katty | mmmmmmmemmories. |
| 15:41.38 | bip | look ma how PC I am ;-) |
| 15:42.11 | goddva | Amnesia: good look then... I need to leave now |
| 15:42.22 | Katty | so i learned how to beatbox this morning, kinda |
| 15:42.31 | Amnesia | ah k |
| 15:42.32 | goddva | Amnesia: just make sure you can load the modules, then the rest should be easy |
| 15:42.32 | Katty | just say bootsncats repeatedly |
| 15:42.40 | Amnesia | well you brought me a little further I guess:) |
| 15:42.43 | Amnesia | okay thx:D |
| 15:46.42 | cmnky | <cmnky> is asterisk.conf the only config file that gets processed, unless inside it there are includes to others ? |
| 15:53.26 | tzafrir_laptop | Amnesia, did you install the dkms package? If so: do you have kernel headers (linux-headers-`uname -r`) installed? |
| 15:53.33 | tzafrir_laptop | Otherwise it will fail to build |
| 15:56.43 | Amnesia | yep |
| 15:56.47 | Amnesia | its loaded fine now |
| 16:03.54 | *** join/#asterisk brainiac (~brainiac@necrotox.in) |
| 16:04.54 | l0st-soul | any1 has an idea why my sip calls through a sip gateway would be cut after 85 minutes of communication ? |
| 16:04.57 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 16:05.02 | l0st-soul | i checked the timeout(absolute) and it is =0 |
| 16:05.23 | l0st-soul | i suspect its at the carrier's side, but he says its my side |
| 16:06.03 | yonahw | <PROTECTED> |
| 16:06.10 | *** join/#asterisk Besticles (~larry@209-58-227-178.static-ip.telepacific.net) |
| 16:07.06 | WIMPy | What kind of phones? |
| 16:07.11 | Besticles | Is there a command to list active AMI sessions? |
| 16:07.42 | WIMPy | manager show connected |
| 16:08.24 | Besticles | I think manager show connected shows aggregate sessions by user. was hoping to see how many connections a manager account has open. |
| 16:11.23 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 16:12.13 | yonahw | so is anyone using 1.8? |
| 16:12.59 | WIMPy | yonahw: What kind of phones? |
| 16:13.02 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
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| 16:28.59 | jsjc | I been having some sip issues today where the connection dropped and the sip was not connected. Is there any way to have asterisk checking or "reconnecting" every onece in a while? |
| 16:30.04 | WIMPy | There is no sch thing as a SIP connection. |
| 16:30.14 | WIMPy | Please describe your issue. |
| 16:31.09 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 16:31.39 | jsjc | registerattempts=0 think will help it... |
| 16:33.32 | leifmadsen | has anyone used Atxfer with AMI? |
| 16:33.40 | leifmadsen | I can't seem to get it to work even though I see the DTMF coming in |
| 16:33.49 | leifmadsen | It does stuff, but no transfer happens |
| 16:34.38 | WIMPy | does not understand how AMI and DTMF come together. |
| 16:35.46 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
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| 16:39.24 | leifmadsen | WIMPy: it's the way the feature is implemented, it is an attended transfer, and Asterisk "types" the numbers to the extension being transferred for you..... like during an attended transfer |
| 16:40.18 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 16:40.26 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:40.43 | WIMPy | But how is that initiated? |
| 16:40.49 | jlewisat1 | I'm looking into an odd problem...a user (on a polycom 501) can do blind transfers, but when she tries an attended transfer, we get SIP/2.0 404 Not Found |
| 16:41.12 | WIMPy | And then you have an atxfer like vit inband features but send the DTMF via AMI? |
| 16:41.49 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.141.242.dsl.dyn.forthnet.gr) |
| 16:43.44 | leifmadsen | WIMPy: via Atxfer |
| 16:43.56 | yonahw | WIMPy: sorry the phones are Polycom 550's but this is on incoming calls from external |
| 16:43.56 | leifmadsen | WIMPy: nevermind, I have a better way to do it now anyways |
| 16:44.43 | WIMPy | leifmadsen: So the only point where AMI comes into play is to send (fake) DTMF? |
| 16:44.52 | leifmadsen | WIMPy: sure |
| 16:45.04 | leifmadsen | I've moved on from that command now |
| 16:45.19 | WIMPy | leifmadsen: Ah, ok. I thought there was some other magic feature I hadn't herd of. |
| 16:45.33 | WIMPy | yonahw: So how do the calls come in then? |
| 16:45.35 | leifmadsen | I don't know.... do: manager show command Atxfer |
| 16:47.00 | yonahw | WIMPy: I noticed that on DID's that were supposed to go directly to specific extensions if I answered the call and then called dial I had this issue. For that I removed the Answer and all works as expected. My problem now is I have calls coming in to an IVR, caller can dial any extension and get routed to that extension. Routing is happening via Dial(${EXTEN},25,wW). I have tried with 'r' as well and same issue, caller does not hear ringing. If I ... |
| 16:47.06 | yonahw | ... use 'm' then they hear music but I want ringing. |
| 16:48.02 | WIMPy | yonahw: So how do the calls come in then? - What technology? |
| 16:48.09 | yonahw | WIMPy: sorry, SIP |
| 16:48.31 | WIMPy | yonahw: Progress() might help. |
| 16:48.56 | yonahw | yeah leifmadsen mentioned Progress yesterday but it's not changing anything for me. |
| 16:49.00 | WIMPy | Hmm. in that case I'd ecpect the caller to generate it themselves. |
| 16:49.15 | leifmadsen | WIMPy: what I really wanted was Action: Redirect |
| 16:49.18 | brainiac | does anyone know what this means?: Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 5: Invalid argument (22) |
| 16:49.22 | brainiac | Did you forget that FXS interfaces are configured with FXO signalling |
| 16:49.24 | brainiac | and that FXO interfaces use FXS signalling? |
| 16:49.48 | WIMPy | leifmadsen: I know that one :-) |
| 16:50.11 | p3nguin | Dial(${EXTEN},25,wW) is not valid -- you didn't specify the channel tech. |
| 16:50.57 | WIMPy | yonahw: That might be one of thos 180/183 with or without sdp things. Try to get a sip debug for the incomming call. |
| 16:50.59 | SuPrSluG | brainiac: your signalling is incorrect in your config file |
| 16:51.08 | yonahw | WIMPy: why would the caller generate the ringing? As far as the caller is concerned they are already connected. Call was answered and is listening to IVR prompt. Asterisk is handling the call and dialing internally to another SIP connection. |
| 16:51.17 | brainiac | ok |
| 16:51.25 | yonahw | WIMPy: I will get a sip debug and post to pastebin |
| 16:51.27 | tzafrir_laptop | brainiac, grep -w 5 /etc/dahdi/system.conf; lsdahdi |
| 16:52.09 | WIMPy | yonahw: Ah, sorry that was after the ivr. Hmm. I'n not sure when I last tried to dial on a call that was already answered. |
| 16:52.33 | WIMPy | That also explains why progrss() doesn't help. |
| 16:52.36 | yonahw | WIMPy: what are my alternatives? How do people usually implement calling extensions from an IVR? |
| 16:52.42 | brainiac | fxsks=5 |
| 16:52.42 | brainiac | ### Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER) 1 FXO FXSKS (In use) RED |
| 16:52.53 | brainiac | <PROTECTED> |
| 16:52.53 | brainiac | <PROTECTED> |
| 16:52.53 | brainiac | <PROTECTED> |
| 16:52.53 | brainiac | ### Span 2: WCTDM/4 "Wildcard TDM400P REV I Board 5" |
| 16:52.53 | brainiac | <PROTECTED> |
| 16:52.53 | yonahw | Yeah Progress didn't seem to be right since call was answered already |
| 16:52.56 | brainiac | <PROTECTED> |
| 16:52.58 | brainiac | <PROTECTED> |
| 16:53.01 | brainiac | <PROTECTED> |
| 16:53.03 | brainiac | |
| 16:53.05 | russellb | ~pb |
| 16:53.05 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 16:53.30 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 16:54.04 | brainiac | I have an alarm here I think. Is that what RED means? |
| 16:54.08 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 16:54.45 | p3nguin | Red is the color of the blood that flows. |
| 16:56.07 | SuPrSluG | brainiac: your signalling is incorrect in your config file |
| 16:56.40 | brainiac | I'm looking at that now |
| 16:58.11 | SuPrSluG | brainiac: http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf |
| 16:58.34 | brainiac | SuPrSluG: thank you |
| 16:59.11 | *** join/#asterisk devyll (~paul@thpallady.net.hostway.ro) |
| 16:59.33 | SuPrSluG | brainiac: just replace zaptel stuff with dahdi |
| 17:00.19 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 17:00.58 | yonahw | WIMPy: what would you use instead of dial in my case? I can't be the first person to allow dialing extensions from an ivr with ringing instead of moh. |
| 17:01.31 | p3nguin | No, I am the first. |
| 17:01.47 | devyll | hello. My problem is that when the calling party hangs up (cause 16) asterisk doesn't warn me in any way. Usually you would expect a "busy tone" or something if the other person hangs up. Any way you can point me in the right direction for configuring a beep or anything else when the other person hangs up |
| 17:01.48 | yonahw | p3nguin: perfect, how did you get it to work |
| 17:01.55 | yonahw | :P |
| 17:02.22 | WIMPy | yonahw: Dial is surely the right way. But I have indeed never done it. |
| 17:02.44 | p3nguin | I used Dial(SIP/phonesname,30) |
| 17:02.51 | WIMPy | devyll: What kind of phone? |
| 17:03.07 | WIMPy | Hmmm. |
| 17:03.12 | WIMPy | Take that back. |
| 17:03.16 | yonahw | WIMPy: I hear, thanks for the assistance thus far |
| 17:03.27 | yonahw | p3nguin: and caller gets ringing? 1.8? |
| 17:03.44 | p3nguin | Of course the caller gets ringing, but it's not 1.8. |
| 17:04.17 | devyll | WIMPy, normal analog phone. (I'm using a sip gateway between the phones and the asterisk box; also the asterisk box has an E1 channel for outside calls) |
| 17:04.45 | WIMPy | devyll: Then you have to see what that sip gateway offers. |
| 17:04.51 | devyll | ok |
| 17:04.53 | devyll | thanks |
| 17:05.06 | yonahw | Where would I look to try and track down this bug or to at least file in bug tracker |
| 17:05.48 | WIMPy | devyll: Asterisk doesn't the concept of audio after disconnect. If a call is over, it's over. |
| 17:07.09 | devyll | WIMPy, so my best chance is a feature on the sip gateway or on the terminal phone |
| 17:08.05 | WIMPy | devyll: Yes, usually the SIP device will generate tones. |
| 17:11.35 | devyll | WIMPy, thanks a lot for your info |
| 17:13.17 | WIMPy | devyll: It one of the many things I am missing. But I'm not even sure if it's possible with sip. I geuss not. |
| 17:16.03 | *** join/#asterisk Mhaddog_ (~Mhaddog_@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 17:19.07 | cmnky | zapateller is part of the asterisk package, yes ? |
| 17:19.48 | WIMPy | yes |
| 17:19.57 | cmnky | ty |
| 17:20.16 | p3nguin | That would depend on what package you're talking about. |
| 17:20.51 | WIMPy | Ok, it's part of Asterisk. |
| 17:21.00 | cmnky | im just reading the www.asterisk.org/docs ... it told me all about zapateller and how cool it was ... just didn't say "what" it was |
| 17:21.16 | cmnky | and im a stoopid noob |
| 17:21.36 | cmnky | aspiring to be a real boy ;) |
| 17:22.56 | SuPrSluG | cmnky: for dealing with PITA calls |
| 17:23.27 | *** join/#asterisk huntbrandon (~huntbrand@66.232.70.40) |
| 17:23.55 | cmnky | yeah ... screw those pita people ... i dont wanna talk to pet loving vegetarians |
| 17:24.08 | cmnky | ;) |
| 17:24.15 | SuPrSluG | cmnky: er pain in the *** |
| 17:24.26 | cmnky | SuPrSluG, i know ;) |
| 17:24.56 | SuPrSluG | never know. lot of internations folks in here |
| 17:24.57 | *** join/#asterisk gregd (~gregd@188-220-38-34.zone11.bethere.co.uk) |
| 17:25.15 | SuPrSluG | s/internations/internationals/ |
| 17:25.20 | gregd | hello everyone. Can someone tell me why the transfer failed?: http://pastebin.com/cgCcNBrM I do not seem to find a reason |
| 17:27.00 | p3nguin | That's what you get for using FreePBX. |
| 17:29.35 | gregd | ;) |
| 17:29.39 | SuPrSluG | do you have this extension? SIP/pstn-0000008c to '95212' |
| 17:30.16 | gregd | yes, correct, that what is supposed to happen |
| 17:30.30 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 17:31.21 | SuPrSluG | well what is extension 95212 supposed to do? |
| 17:31.38 | cmnky | various loud monkey screechings <--- pure genius ... kudos to the person responsible |
| 17:31.55 | gregd | SuPrSluG: http://pastebin.com/tUBC42Cw |
| 17:32.30 | gregd | please note that the transfer works always when I try to test... just in real life example it seems to fail?! |
| 17:32.43 | *** join/#asterisk nevermind_quack (~nevermind@66.232.70.40) |
| 17:32.58 | gregd | SuPrSluG: here is a successful transfer log: http://pastebin.com/GDj2ZGQe |
| 17:34.02 | nevermind_quack | has anyone heard what the change was to google voice that broke chan_gtalk in the last couple days? |
| 17:39.10 | nevermind_quack | incoming calls still route just fine, but starting around tuesday afternoon outbound calls fail with a generic "Remote peer reported an error, trying to establish the call anyway" ... looks like a few people are posting to forums about it but 1.8 is new enough I don't think a lot of people are using it |
| 17:39.59 | SuPrSluG | gregd: can you describe how the successful call is made? looks to be from an internal extension. And the one that fails from the pstn? |
| 17:40.07 | citywok | nevermind_quack: are you using trunk, or 1.8.0? |
| 17:41.18 | gregd | SuPrSluG: the successful transfer: http://pastebin.com/GDj2ZGQe ? pstn called 102, and 102 transferred it to 95272 |
| 17:41.28 | nevermind_quack | citywok: 1.8.0 |
| 17:43.50 | *** join/#asterisk imox1234 (~imox1234@p4FC5C538.dip0.t-ipconnect.de) |
| 17:44.23 | citywok | ah, so google changed something themselves that broke it then? |
| 17:45.01 | nevermind_quack | sure looks like it ... i checked another person's box I set up with gv and it is broken too |
| 17:45.10 | trollasaurus | I'll try my GV now.. |
| 17:45.12 | nevermind_quack | some random forums are referencing it too -- http://www.dslreports.com/forum/r25146039-Asterisk-GV-outgoing-calls-with-Asterisk1.8 |
| 17:47.11 | trollasaurus | Yep, failed |
| 17:47.52 | SuPrSluG | gregd: might not be making it into the context you want it to. |
| 17:48.01 | trollasaurus | JABBER: asterisk INCOMING: <iq type="error" to="michelle5601@gmail.com/asteriskEE1B802D" from="+nnnnnnnnnnn@voice.google.com/srvres" id="aabfw"><session type="initiate" initiator="xxxx@gmail.com/asteriskEE1B802D" id="65fd229c30f87506" xmlns="http://www.google.com/session"><description xml:lang="en" xmlns="http://www.google.com/session/phone"><payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000"/><payload-type id="100" name="EG711U" clockrate="8000" |
| 17:50.32 | nevermind_quack | hrm ... did the dev team have any inside info on the google api? any asterisk jabber developers in here? |
| 17:51.19 | SuPrSluG | garymc: try http://fonality.com/trixbox/forums/trixbox-forums/sip-and-iax-trunks-and-providers/gradwell-sip-trunk-settings |
| 17:51.50 | citywok | it looks like dvossel and pabelanger did most of the work on chan_gtalk |
| 17:52.10 | citywok | recently anyways |
| 17:54.59 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 17:55.29 | nevermind_quack | any thoughts on where to go from here? |
| 17:55.48 | MrTelephone | Can someone fill me in on how asterisk authenticates when a caller uses *67 to black blank our their contact information? The problem I'm having is when there is 2 SIP accounts behind the same IP address. It confuses Asterisk as it uses the IP address to verify the client making the call. |
| 17:56.38 | MrTelephone | What is the proper method of authentication if the client provides "Anonymous" anonymous@invalid.in for the From: header and Contact header? |
| 17:58.47 | trollasaurus | nevermind_quack, http://www.google.com/session/phone -- why 404 ? :-P |
| 17:59.30 | MrTelephone | *67 behind an endpoint with 2 sip accounts configured. If anyone has problems with this please shout out. |
| 18:00.52 | MrTelephone | Here is an example of the logged error message |
| 18:00.53 | MrTelephone | [Dec 1 19:01:08] WARNING[25071] chan_sip.c: username mismatch, have <508>, digest has <500> |
| 18:01.44 | MrTelephone | 508 turns on privacy and asterisk is trying to authenticate 508 with 500 credentials. I'm guessing because it matches the sip account based on IP Address |
| 18:03.18 | *** join/#asterisk beefpastry (~beefpastr@74-129-198-56.dhcp.insightbb.com) |
| 18:04.11 | jlewisat1 | I'm back with my call transfer question :)...I think I've half figured it out. It appears to be NAT related. sip show peer on this user's phone is showing the public IP under full contact...and so her call IDs have the public IP in them...but when she tries an attended transfer, the Refer-To Replaces= uses the call ID@private-IP, and so * gets confused and says it can't find that call-id |
| 18:05.08 | jlewisat1 | so...the question appears to be, what would cause her sip registration to not expose her phone's private IP? |
| 18:06.08 | nevermind_quack | trollasaurus: no idea and I can't find any record of google changing anything |
| 18:08.17 | trollasaurus | nevermind_quack, Were you searching google ? :-P |
| 18:08.36 | nevermind_quack | haha :) |
| 18:08.55 | nevermind_quack | dear google, what have you done? |
| 18:10.19 | cmnky | nevermind_quack, evil ... to be sure ... evil ;) |
| 18:11.54 | *** join/#asterisk arielb27 (~chatzilla@63.214.236.169) |
| 18:12.24 | Holos | When I have an incoming call, that hasn't been answered in 10 seconds, I'd like to call our Paging unit to play a tone and a quick message so the people in the shop before ringing their phone. Can I somehow fork from the dialplan and call their paging extension (which answers right away) and play them a message and hangup without having the call Joined to the paging horn's extension? |
| 18:13.56 | Holos | I know I can call the LOCAL channel and use the WAIT command, but I need to call the phone and playback a message without joining the call. |
| 18:14.02 | *** join/#asterisk hfb (~hfb@pool-98-112-232-180.lsanca.dsl-w.verizon.net) |
| 18:23.39 | Holos | [TK]D-Fender: Thanks for the tip on the Originate() cmd, this should do what I need. |
| 18:26.16 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 18:27.28 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 18:34.50 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 18:35.03 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 18:36.54 | *** join/#asterisk nix8n82 (~nate@63.162.28.112) |
| 18:37.30 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 18:38.14 | Katty | peeks |
| 18:38.15 | Katty | in |
| 18:38.23 | thehar | herros? |
| 18:38.56 | timholum | does anyone know does * work in config files, for example could I do include => "clients/*.conf" to include all the conf files under a folder clients? |
| 18:39.25 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:41.12 | leifmadsen | timholum: yes, you want #include subdir/myfile.conf |
| 18:41.23 | *** join/#asterisk Godfather_ (~godfather@108.Red-88-6-252.staticIP.rima-tde.net) |
| 18:41.24 | leifmadsen | you can't do *.conf though (i don't think) |
| 18:41.34 | Godfather_ | hi |
| 18:41.40 | leifmadsen | #include (for files) vs. include => (for contexts) |
| 18:41.48 | pabelanger | nevermind_quack: Yes, looks like google may have changed something on their side |
| 18:42.19 | Godfather_ | how should i change the default sip port (5060)? in [general] section using port= or updbindaddr=0.0.0.0:port ? |
| 18:42.26 | leifmadsen | pabelanger: did chan_gtalk stop working? |
| 18:42.40 | leifmadsen | Godfather_: for outbound, inbound? |
| 18:42.52 | leifmadsen | Godfather_: udpbindaddr is probably the better way to change the listening port |
| 18:42.59 | Godfather_ | leifmadsen, listening port |
| 18:43.00 | timholum | thanks leifmadsen |
| 18:43.01 | pabelanger | leifmadsen: looks like it, about to create an issue on mantis. |
| 18:43.16 | pabelanger | leifmadsen: Trying to hunt down jabber error codes too |
| 18:43.16 | *** join/#asterisk Lord_Rahl (~Lord_Rahl@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 18:43.22 | nevermind_quack | google changed something in their call processing ... inbound calls work, outbound calls fail |
| 18:43.40 | leifmadsen | damnit :( |
| 18:43.44 | nevermind_quack | started tuesday afternoon methinks |
| 18:43.51 | leifmadsen | gotcha, thanks for noticing |
| 18:44.07 | Godfather_ | leifmadsen, ok, then i'll try with udpbindaddr=0.0.0.0:port |
| 18:44.12 | Godfather_ | ty |
| 18:44.54 | *** join/#asterisk brainiac (~brainiac@necrotox.in) |
| 18:46.11 | Godfather_ | leifmadsen, anyway, i dont understand the difference between both :-? |
| 18:46.52 | *** part/#asterisk d-fens (~jscalia@su-nat.int.smq.datapipe.net) |
| 18:47.15 | leifmadsen | Godfather_: port is the old way of setting it |
| 18:47.38 | leifmadsen | Godfather_: pretty much deprecated now |
| 18:47.56 | Godfather_ | leifmadsen, ahh ok |
| 18:50.48 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.136.171) |
| 18:55.01 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 18:55.12 | Godfather_ | leifmadsen, i can't connect, whats the point of that 5060 port? http://pastebin.com/EHLdHd1s |
| 18:55.47 | leifmadsen | Godfather_: sip show settings |
| 18:56.02 | leifmadsen | Godfather_: the port you're seeing is what the peer is listening on, not what asterisk is listening on |
| 18:56.18 | leifmadsen | is your peer trying to connect to the port you're listening on? is your firewall open? |
| 18:56.22 | *** join/#asterisk dinesh___ (~dinesh@46-126-192-144.dclient.hispeed.ch) |
| 18:56.41 | dinesh___ | hi all, I'm getting the following error message while trying to place an outgoing SIP call, "SIP/voipalot-088eab28 is circuit-busy", any idea why ? |
| 18:56.41 | leifmadsen | Godfather_: you should also look at the sip trace via 'sip set debug ip xxx.xxx.xxx.xxx |
| 18:56.41 | Godfather_ | leifmadsen, http://pastebin.com/80VFH084 |
| 18:56.58 | Godfather_ | leifmadsen, boths pcs are on the same lan, no firewall |
| 18:57.07 | leifmadsen | ok so asterisk is ilstening on 9999 -- is your phone trying to connect to 9999 ? |
| 18:57.25 | leifmadsen | dinesh___: the circuit is busy? need more information such as a sip trace |
| 18:57.55 | Godfather_ | well, i'm using ekiga, i enter the port in the Register field like this -> 192.168.1.2:9999 |
| 18:58.16 | Godfather_ | i dont know if its allowed, cause i have no "port" field in ekiga |
| 18:59.22 | Godfather_ | let me download x-lite and try it... |
| 19:03.05 | *** join/#asterisk sam_affable (sam@202.53.10.123) |
| 19:03.09 | sam_affable | hello |
| 19:03.37 | sam_affable | hey can anyone point me how to setup google voice and ipkall DID in asterisk |
| 19:03.50 | dinesh___ | leifmadsen, http://pastebin.com/WtTGEK5p , I did set the debugging on |
| 19:03.56 | Katty | heellllllllllllllllllllllloooooooooooooooooooo nurse |
| 19:04.51 | citywok | is she sexy? :P |
| 19:07.08 | dinesh___ | Dec 2 18:24:20] WARNING[12170]: chan_sip.c:11860 handle_response_invite: Received response: "Forbidden" from '" |
| 19:07.15 | dinesh___ | i'm also getting this before the congested message |
| 19:08.00 | dinesh___ | (i'm trying a new sip provider) |
| 19:09.14 | citywok | dinesh___: are you registering with the provider? |
| 19:10.31 | dinesh___ | no, i'm only placing outgoing calls |
| 19:11.05 | Katty | leifmadsen is sexy. |
| 19:11.10 | Katty | but not a she. |
| 19:11.32 | Lord_Rahl | I am donig a NoOp on the callerid : exten=s,1,NoOp(Open) exten=s,n,Answer() exten=s,n,NoOp(${CALLERID(all)}) but all I get is the number. I am using 1.4 I have tried callerid(name) & all still just the number |
| 19:11.47 | Katty | leifmadsen: i'd hug ya. |
| 19:11.52 | dinesh___ | registring works |
| 19:12.14 | citywok | so it works if your register? |
| 19:13.41 | carrar | leifmadsen is one sexy bitch? |
| 19:13.47 | carrar | heh |
| 19:14.22 | Katty | idk i don't think he could pull off the whole bitch persona |
| 19:15.12 | Godfather_ | leifmadsen, it was an ekiga problem using a different sip port, with linphone there is no error |
| 19:15.21 | carrar | With the help of Firefox he can just download a new perona |
| 19:15.27 | carrar | perrsona |
| 19:15.29 | carrar | persona |
| 19:15.59 | carrar | Yakuza Paint! |
| 19:16.36 | carrar | http://www.getpersonas.com/en-US/persona/322488 |
| 19:19.06 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.wlan.noare-1.holmedal.net) |
| 19:20.07 | nevermind_quack | pabelanger: do you have an id in mantis yet? need any assistance collecting info? |
| 19:20.36 | pabelanger | nevermind_quack: no, but feel free to create an issue. I'll triage it from their |
| 19:24.27 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.138.173) |
| 19:25.17 | dinesh___ | no it didn't help to register, but the registration itself worked citywok |
| 19:25.24 | dinesh___ | so the user/pass are correct |
| 19:26.57 | dinesh___ | i just purchased some credits but that didn't help, i expected to get a voice saying that i don't have enough credits, but i got this forbidden and congested error message instead |
| 19:30.38 | *** join/#asterisk mandragor (~ergudicsu@70.158.116.58) |
| 19:30.51 | *** part/#asterisk mandragor (~ergudicsu@70.158.116.58) |
| 19:38.53 | *** join/#asterisk BenC[UK] (~bencummin@cpc1-lock3-2-0-cust367.6-1.cable.virginmedia.com) |
| 19:39.22 | BenC[UK] | Evening guys, I am getting a error when calling saying "All Circuits are busy" |
| 19:39.30 | BenC[UK] | is that a sound file in asterisk? |
| 19:39.50 | Qwell | BenC[UK]: Are you using FreePBX? |
| 19:39.56 | Qwell | try asking in #freepbx. |
| 19:40.07 | BenC[UK] | I am, tanks |
| 19:40.10 | BenC[UK] | thanks* |
| 19:40.38 | *** join/#asterisk jamko (~chatzilla@173-162-11-81-naples.hfc.comcastbusiness.net) |
| 19:44.06 | jamko | when does asterisk create the mysql config files, ie - cdr_mysql.conf , res_mysql.conf .. During the main asterisk install, or during the addons install? |
| 19:44.13 | Qwell | neither |
| 19:44.31 | Qwell | They can be installed with `make samples` in asterisk-addons. |
| 19:44.47 | leifmadsen | which is just copying and renaming the files from the configs/ subdir of your asterisk source |
| 19:44.57 | Qwell | asterisk-addons* source |
| 19:45.13 | leifmadsen | cp ~/src/asterisk/configs/cdr_mysql.conf.sample /etc/asterisk/cdr_mysql.conf |
| 19:45.18 | leifmadsen | for example |
| 19:45.32 | leifmadsen | or ~/src/asterisk/addons/ (if using 1.8 or later) |
| 19:45.40 | leifmadsen | I believe that's the path for addons |
| 19:45.45 | nevermind_quack | created bug #0018412 for chan_gtalk -- I'm not sure what additional info to add, nothing seems useful :| |
| 19:46.24 | jamko | ok great thanks. |
| 19:47.14 | *** join/#asterisk user1324 (44e4c5b5@gateway/web/freenode/ip.68.228.197.181) |
| 19:47.27 | Lord_Rahl | in a incoming call it is displaying the callers number I need it to display the calls name how would I make do that? Asterisk 1.4 |
| 19:48.40 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 19:49.24 | user1324 | any fixes for jabber/gtalk/google voice trunk call initiation? |
| 19:52.58 | nevermind_quack | user1324: i just submitted a bug, and nobody seems to know what to do about it at this point. all signs point to something changed on google's side |
| 19:55.26 | user1324 | that's what i suspected. i guess workarounds aren't a big deal but you lose the elegance. |
| 19:56.42 | nevermind_quack | any thoughts on helpful debugging for the developer who picks up the gtalk issue? |
| 19:56.50 | user1324 | nevermind_quack: you have a link to bug report? |
| 19:57.08 | nevermind_quack | user1324: https://issues.asterisk.org/view.php?id=18412 |
| 19:57.27 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 19:57.50 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.142.101) |
| 19:57.52 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 19:58.15 | nevermind_quack | none of the debugging i've done so far has shown anything beyond the generic error |
| 19:58.26 | user1324 | nevermind_quack: hard to say. other than comparing sniffed traffic from the gmail plugin i don't have any good suggestons. |
| 19:59.29 | user1324 | they might have implemented some kind of two factor auth in conjunction with the web login or could be as simple as a protocol syntax |
| 20:01.47 | citywok | somebody posted that it had to do with being connected 24/7 and they were blocking those users. is that not the case? |
| 20:03.51 | nevermind_quack | citywok: no I think that was an issue from a few years ago, my dev box uses a gv account that has only been online for about an hour in the last week |
| 20:05.18 | user1324 | anybody seen any traffic logs from the gmail call plugin? |
| 20:11.58 | *** join/#asterisk engrxyz (~trettr433@host81-150-217-170.in-addr.btopenworld.com) |
| 20:13.30 | *** join/#asterisk ccesario (~ccesario@187.10.57.94) |
| 20:20.59 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 20:21.45 | Docfxit | Hi, |
| 20:22.56 | Docfxit | What is a good way to check for open UDP ports in Ubuntu for sip phones |
| 20:23.07 | *** join/#asterisk gregd (~gregd@cpc3-sgyl29-2-0-cust457.sgyl.cable.virginmedia.com) |
| 20:26.46 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 20:28.43 | pabelanger | Docfxit: netstat ? |
| 20:29.01 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:30.37 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.4.226) |
| 20:30.46 | drmessano | So here's one for ya |
| 20:31.06 | drmessano | My MoH doesn't work in 1.8 with res_timing_dahdi loaded |
| 20:31.33 | atan | Anyone here mess with the nortel 1535 phones? I can't get mine to register. I bet I missed a setting in it somewhere =D |
| 20:31.35 | drmessano | But if I don't load it, my system goes unresponsive after a while, lol |
| 20:32.30 | *** join/#asterisk gregd (~gregd@cpc3-sgyl29-2-0-cust457.sgyl.cable.virginmedia.com) |
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| 20:35.52 | *** part/#asterisk DelphiWorld (~VoIpGuy@41.200.4.226) |
| 20:36.10 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 20:37.42 | drift- | i've got interesting question , i have SMC router / cable modem running to > managed switch and under that a 24 port switch..... question of the hour is i have polycom 501 phones and if you know these phones you know there is a PC connection on back of them to connect the computer, i've got office here with like 8 rooms how can do QoS on phones and internet, right now my entire inet is slow as shit and phones work perfect :( |
| 20:38.23 | jlewisat1 | as I suspected...customer's router had a buggy SIP ALG that was hosing things |
| 20:38.35 | jlewisat1 | one little check box, and the call transfer issue is fixed |
| 20:38.37 | drift- | any suggestions on how to rate limit? is it possible? on swr2008 linksys managed switch to do this? i need 10 computers working with internet and 10 phones... |
| 20:40.05 | p3nguin | I think you'll have to create VLANs for separating computers and phones. Voice should get priority, but computers should get bandwidth if the phone load is low. |
| 20:40.10 | MrTelephone | drift-, if your congested your phones should work crappy? |
| 20:40.27 | drift- | when you say vlans does that mean i have to have seperate connection for phone and seperate connection for computer? |
| 20:40.36 | drift- | cause i have jacks in every office for 1 |
| 20:40.53 | drift- | phones work fine actually better then ever |
| 20:41.06 | MrTelephone | where does your phone traffic go? |
| 20:41.21 | drift- | i dont know thats the problem |
| 20:41.25 | p3nguin | You can create a VLAN for phones and another VLAN for computers and use both VLANs on the same port. The difference would be in the subnet. |
| 20:41.38 | drift- | interesting p3nguin |
| 20:41.55 | MrTelephone | I seperate the phones/data via vlans and have a cable modem for each vlan |
| 20:42.01 | p3nguin | Unless you are overloading a port, it'll work. |
| 20:42.09 | drift- | well here is thing |
| 20:42.11 | nevermind_quack | if your phones work fine you don't have an issue that requires QOS ... voice quality _always_ the first thing to go south if you have congestion |
| 20:42.17 | drift- | right now i have 3 ports in use on this manged switch 8 port switch |
| 20:42.30 | drift- | 1 for cable modem > 2 for asterisk server > 3 for 24 port switch with computers and phones |
| 20:42.37 | drift- | 5 other ones are free |
| 20:42.37 | p3nguin | I doubt you are overloading each switch port; it sounds like you are just overloading your internet connection (since you said it is slow). |
| 20:42.37 | MrTelephone | I can't see him conjesting 100mbit ports |
| 20:42.51 | drift- | yeah we have 50mbit comcast business line |
| 20:43.00 | drift- | 10 browsing machines and 10 phones |
| 20:43.02 | p3nguin | Create your VLANs. |
| 20:43.07 | MrTelephone | remember that each ip from the cable modem doesn't give you a different service flow |
| 20:43.14 | MrTelephone | oh |
| 20:43.33 | MrTelephone | 50mbit down. 256kbps up |
| 20:43.44 | p3nguin | heh |
| 20:43.54 | p3nguin | I hope that's not right. |
| 20:43.58 | MrTelephone | lol |
| 20:44.07 | drift- | no |
| 20:44.09 | drift- | its like 3 - 5 up |
| 20:44.19 | p3nguin | I have 60 down and 5 up on RESIDENTIAL cable... I would hope their business class is at least that good. |
| 20:44.33 | drift- | yeah that sounds about right |
| 20:44.42 | MrTelephone | do you even get those speeds, that's crazy |
| 20:44.52 | p3nguin | I do. |
| 20:45.06 | MrTelephone | it takes an entire 6mhz band at 256 qam to do 30mbits |
| 20:45.27 | MrTelephone | nuts |
| 20:45.35 | p3nguin | We use bonded channels. |
| 20:45.57 | MrTelephone | so your home alone can use 12 mhz of bandwidth |
| 20:45.57 | *** join/#asterisk ningia (~gain@host143-85-dynamic.11-79-r.retail.telecomitalia.it) |
| 20:45.59 | MrTelephone | nice |
| 20:46.07 | p3nguin | 4 channels |
| 20:47.04 | Docfxit | pabelanger Thank you. Shouldn't I see 5060 UDP listening for sip phones? |
| 20:47.06 | p3nguin | Downstream Modulation QAM256 QAM256 QAM256 QAM256 |
| 20:47.38 | MrTelephone | are you in some kind of high rise or something? |
| 20:47.51 | MrTelephone | each street gets it's own 4 channel bonding I guess |
| 20:47.55 | p3nguin | Me? No. This is regular high speed cable. |
| 20:48.20 | p3nguin | Mabye each ones gets its own... |
| 20:48.22 | drift- | http://www.compusa.com/applications/SearchTools/item-details.asp?EdpNo=2855850&CatId=5227 < would this do what i need it to do? |
| 20:48.23 | p3nguin | no idea. |
| 20:48.35 | nevermind_quack | drift-: so if your voice quality is fine what is your problem exactly? |
| 20:48.44 | drift- | computers are getting no speed |
| 20:48.48 | drift- | like 1mbit if they are lucky |
| 20:48.51 | drift- | ping is like 600 to google |
| 20:48.58 | drift- | i need to figure out way to speed up internet |
| 20:49.05 | nevermind_quack | call your ISP |
| 20:49.08 | Lord_Rahl | how would I display the callers name and not the name in caller id? is it in a config some were? I have try to set({$CALLERID(name)}) but that did not seem to work |
| 20:49.35 | drift- | there is no form of traffic shaping |
| 20:49.44 | p3nguin | http://www.speedtest.net/result/1054719421.png |
| 20:49.46 | drift- | or anything its just bunch of computers n phones plguged into 2 switches |
| 20:49.49 | nevermind_quack | QOS and vlans will accomplish nothing ... traffic shaping is only designed to slow things down |
| 20:50.28 | MrTelephone | do you have any switch lights going crazy? |
| 20:50.35 | nevermind_quack | try the speed from a single computer plugged directly into the modem with everything else disconnected |
| 20:51.39 | atan | Anyone here know how to setup a Nortel IP1535 to connect to Asterisk? As far as I can see I've input the settings correctly but Asterisk doesn't even get the registration request =( |
| 20:53.31 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 20:53.37 | MrTelephone | Your node is conjested, your upstream is noisey, or one of your computers is using up all your upstream |
| 20:53.44 | Lord_Rahl | drift-, you may want to out a patch capture software on the network and see what being sent you may have a rouge program or virus flooding your network. I can not think of the free one right now google it. |
| 20:55.26 | MrTelephone | new patch cord from modem to switch? set the switch port to the proper duplex/speed for your cable modem. Maybe the cable modem is made by cisco and it can't auto sense properly. hah |
| 20:57.34 | MrTelephone | drift-: check the port statistics on the web interface |
| 20:58.41 | drift- | now phones arent working |
| 20:58.46 | drift- | ever outgoing call is going busy |
| 20:59.26 | MrTelephone | time to pull out your network A+ manual |
| 20:59.35 | MrTelephone | Asterisk A+ |
| 21:00.27 | p3nguin | Maybe you mean Asterisk+ |
| 21:01.08 | drift- | well its bandwidth related |
| 21:01.12 | Lord_Rahl | drift-, I have had a Polycom 301 phone go bad and grab and ip that was outside DHPC scope and crash or server. so it possible that you have a bad phone somewhere doing weird stuff |
| 21:01.14 | drift- | nothing else then that |
| 21:01.32 | drift- | well heres thing i dont have any rules setup |
| 21:01.34 | drift- | its open network right now |
| 21:01.41 | drift- | inet > managed switch > computers n phones |
| 21:01.58 | drift- | so everyone is grabbing all bandwidth they want |
| 21:02.04 | drmessano | How many phones? |
| 21:02.09 | drmessano | How many PCs? |
| 21:02.11 | drift- | 10 phones 10 computers |
| 21:03.00 | drift- | <PROTECTED> |
| 21:03.08 | drmessano | Ok, I have like 3 times that on a 3 meg link and barely hit 85% during the day.. Fire someone or look for basic Network 101 crap like a cable looped through the switch twice or something |
| 21:03.09 | Lord_Rahl | drift-, wireshark - www.wireshark.org |
| 21:04.03 | drmessano | or you misconifured your managed switch |
| 21:04.17 | drmessano | misconfigured |
| 21:05.17 | drift- | well |
| 21:05.22 | drift- | there isnothing configured on the manged switch |
| 21:05.29 | drift- | i just have it online thats it |
| 21:05.36 | drift- | its not doing shit at moment cause id ont know what to configure |
| 21:05.40 | drift- | i'm little lost |
| 21:05.45 | *** join/#asterisk diegomad (~mad@190.147.221.138) |
| 21:06.17 | drmessano | You should be able to hit the UI of the switch and look at port utilization |
| 21:06.24 | drmessano | That would be a good start |
| 21:08.35 | drift- | i dont see any stats really... on this switch |
| 21:09.00 | atan | ' chan_sip.c:3707 sip_write: Asked to transmit frame type 4, while native formats is 0x1 ' =\ |
| 21:10.02 | drift- | drmessano: you have this in your house? these machines and phones? |
| 21:10.15 | drift- | or is this office? |
| 21:10.19 | drmessano | No, I don't have 30 users at home |
| 21:10.26 | drift- | heh |
| 21:10.40 | drift- | but you probably have better router and switch |
| 21:10.55 | drift- | you have polycom running from wall jack running to phone to pc? |
| 21:10.58 | drmessano | Has nothing to do with it |
| 21:11.17 | drmessano | If you have congestion, you need to get to the source of it |
| 21:11.34 | Lord_Rahl | atan, has it worked before? |
| 21:14.11 | Lord_Rahl | atan, http://www.voipuser.org/forum_topic_13350.html |
| 21:14.16 | drmessano | drift-: Again, irrelevant.. You're talking about 20 devices.. that's a drop in the bucket |
| 21:14.43 | drift- | yeah |
| 21:15.00 | BenC[UK] | what does "Prodding channel failed" mean? |
| 21:15.01 | drmessano | drift-, start looking at DATA and stop trying to get a 1000ft view or look for someone to sympathize.. You need to troubleshoot and not just panic |
| 21:16.17 | Lord_Rahl | drift-, you run any fat apps like citrix or heavy sql base apps |
| 21:18.45 | cmnky | is it fair to say ... a dialplan ... is a program ... that begins execution .... when the phone is picked up ... and stops executing when hung up ? |
| 21:19.34 | drift- | nope |
| 21:20.01 | drift- | this is very basic network , asterisk and windows machines |
| 21:20.08 | p3nguin | cmnky: I think it depends on your interpretation of the terminology. |
| 21:20.15 | Lord_Rahl | cmnky, yes but after number are sent to asterisk |
| 21:20.51 | atan | Lord_Rahl, =S |
| 21:21.04 | cmnky | well... like my local extension ... i pick it up ... the local ext dialplan starts executing ? |
| 21:21.23 | cmnky | (its not necessarily a seperate process/task/thread) ... just a context |
| 21:22.12 | tzanger | hmm |
| 21:22.36 | p3nguin | cmnky: No, Asterisk is not a key system. |
| 21:22.51 | tzanger | odd question about Asterisk: I know that there are GUIs that edit and reload various parts of asterisk with varying degrees of success, but do any of them provide some kind of API to programmatically make the changes? |
| 21:23.06 | drmessano | FreePBX does |
| 21:23.07 | cmnky | im a programmer ... has nothing to do with key systems |
| 21:23.17 | tzanger | The question just came up and in all the years I've used Asterisk I don't know of one that actually exposes some kind of API |
| 21:23.25 | p3nguin | cmnky: Phones are devices. Phones are not extensions. Extensions are the rules for processing calls when they are sent into Asterisk. |
| 21:23.45 | tzanger | drmessano: really? I did not know that |
| 21:23.46 | cmnky | what happens when i pick up my ATA box connected phone ... a SIP session begins for it, does it not ? |
| 21:23.51 | p3nguin | cmnky: no |
| 21:24.06 | drmessano | cmnky, no, it goes BRRRRRRRRRRRRRRRRRRRRRRRRRRRRR |
| 21:24.11 | tzanger | I was thinking maybe AsteriskNOW had something since its got a JSON-y kind of interface |
| 21:24.13 | cmnky | p3nguin, then why do packets go from the ATA to asterisk ? |
| 21:24.20 | cmnky | when i pick up the phone ? |
| 21:24.24 | p3nguin | cmnky: Nothing happens between the ATA and Asterisk until you dial a number and the ATA sends the call to Asterisk. |
| 21:24.25 | malcolmd | tzanger: switchvox exposes a large set of api calls via xml and json that manipulate functionality, but it may not be what you're looking for (free) |
| 21:24.33 | drmessano | tzanger, AsteriskNOW is not a GUI.. You're either referring to FreePBX or Asterisk-GUI |
| 21:24.49 | cmnky | p3nguin, thats not true ... because it sends packets .. and ive not pushed a button yet |
| 21:24.59 | tzanger | drmessano: well AsteriskNOW is a distro with the GUI, yes I suppose you're right, I have my terms cross-wired |
| 21:25.07 | p3nguin | cmnky: It's not sending the call until a call is placed. |
| 21:25.09 | tzanger | malcolmd: aha. |
| 21:25.15 | atan | Lord_Rahl, you were on to something though. We're off to the races now!! Thanks =D |
| 21:25.17 | tzanger | malcolmd: is there any documentation on this? |
| 21:25.20 | cmnky | p3nguin, if i turn asterisk off ... and reset the ATA ... i wont get a dialtone again until asterisk is running |
| 21:25.23 | tzanger | (public, before-you-buy kind of stuff) |
| 21:25.32 | *** join/#asterisk laidback_01 (~jax@209.206.245.109) |
| 21:25.35 | cmnky | p3nguin, i didnt say it was sending a call |
| 21:25.37 | malcolmd | tzanger: http://developers.digium.com/switchvox/wiki/index.php/WebService_methods |
| 21:25.41 | Lord_Rahl | atan, np :) glad I can help |
| 21:25.47 | cmnky | p3nguin, i didnt say anything about calls ... ive been talking about "dialplans" |
| 21:25.57 | atan | Lord_Rahl, the issue was I didn't tell sip.conf what codec the phone could use |
| 21:25.58 | atan | =D |
| 21:26.02 | p3nguin | cmnky: There could be some misc packets, but dialplan is not touched until a call begins. |
| 21:26.04 | tzanger | malcolmd: very nice documentation site! |
| 21:26.25 | p3nguin | cmnky: I'll repeat: Dialplan is not touched until a call begins. |
| 21:26.33 | cmnky | p3nguin, so i cant control what happens if i push 3 on my phone ? |
| 21:26.41 | Lord_Rahl | cmnky, you could say that but picking up the handset does nothing but open a channel to asterisk. The phone simulates the dial tune. the dial plan is exicuted when you send numbers that are match to dial plan than it start |
| 21:26.58 | p3nguin | cmnky: If the ATA interprets the dialing of 3 as the entire number and starts a call, you can. |
| 21:27.10 | malcolmd | tzanger: i'll send compliments to the team :) |
| 21:27.13 | drmessano | cmnky, an ATA does nothing when you simple off-hook the attached device... not until you enter a pattern match for the dialplan in the ATA, which then initiates a SIP session |
| 21:27.15 | p3nguin | cmnky: It won't be instant, though. |
| 21:27.25 | Lord_Rahl | atan, that will do it :) |
| 21:27.30 | cmnky | Lord_Rahl, programs on unix begin at "main()" ... not the first line inside 'main()" |
| 21:27.40 | tzanger | malcolmd: please do. :-) |
| 21:27.45 | cmnky | and dialplans begin when i pick up the phone ... not after i do something with it |
| 21:27.48 | p3nguin | cmnky: The ATA notices you've dialed the entire number, and then starts the call. |
| 21:28.10 | p3nguin | Dialplan does not start when you pick up the phone. |
| 21:28.14 | laidback_01 | say, I've never worked with asterisk, but... I'm at work. We have an all Nortel network, with Nortel IP phones -model 1140e. We've got some features missing, no redial, no last caller, but we have caller id, etc. All it says for the advanced features is "Web server not available" Any case, I'd like to have a server - pbx of some kind that I can play with. This seems like an opportunity to learn to me |
| 21:28.27 | *** join/#asterisk greezmunkey (~greezmunk@67.208.107.210) |
| 21:28.42 | tzanger | malcolmd: nothing like that for the Asterisk GUI with AsteriskNow? Could I fake out the JSON calls it uses? |
| 21:29.01 | cmnky | p3nguin, you claimed no packets were sent when i picked up the phone ... but that was wrong ... are you one of the asterisk programmers ? |
| 21:29.19 | malcolmd | tzanger: negative. :( asterisk gui isn't that advanced |
| 21:29.24 | drmessano | p3nguin, technically, that's not true.. Your digit timeout starts from the time you off-hook the device.. but we're still on the ATA side of the call, and no SIP sessions have been harmed or molested until something matches in said ATA's dialplan |
| 21:29.28 | p3nguin | No CALL is sent until AFTER you dial the fucking number. Why is this hard to understand? |
| 21:29.33 | Lord_Rahl | cmnky, well in that context you could say the dial plan starts then asterisk reloads. |
| 21:30.17 | drmessano | cmnky, your ATA is always sending keep-alives, etc.. nothing is sent because you off-hook the attached device. |
| 21:30.19 | p3nguin | The ATA must accept the number dialed before the call starts. |
| 21:30.39 | cmnky | isnt a dialplan context unique ( as far as execution logic, not configed code ) for each session starting when i pick up the phone, and then hang it up ? |
| 21:31.01 | drmessano | Um, huh? |
| 21:31.13 | cmnky | drmessano, so asterisk doesnt know ive picked up the phone ? |
| 21:31.14 | p3nguin | Contexts unique per session? No. |
| 21:31.38 | drmessano | cmnky, no, not unless you've loaded res_fluxcapacitor |
| 21:31.44 | cmnky | p3nguin, dude ... you couldnt have multiple "sessions" of anything if the contexts for each "session" werent unique |
| 21:31.50 | cmnky | p3nguin, if you arent a programmer ... stop talking |
| 21:31.55 | p3nguin | hahahaha |
| 21:32.10 | p3nguin | Right. Like you telling me to stop talking is going to stop me from talking. |
| 21:32.19 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 21:32.48 | cmnky | drmessano, so at what point does a "dialplan" begin executing ? |
| 21:33.01 | cmnky | drmessano, for an outgoing call ... forget about incomming |
| 21:33.09 | p3nguin | Asterisk knows when I pick up _my_ phone because chan_sccp.so tells it. If I used an ATA, Asterisk wouldn't know. |
| 21:33.13 | laidback_01 | so.. what I'm wondering is: Can someone point me to the FAQ, or instructions for how to build a smallish server to take stuff over a NIC, and send it on to my phone - whilst captureing the intformation of callers, etc. Is this even possible? |
| 21:33.30 | laidback_01 | stuff over a NIC == IP phone calls |
| 21:33.33 | p3nguin | All calls are "incoming" calls in relation to Asterisk. |
| 21:34.32 | cmnky | p3nguin, and im asking about asterisk in relation to what im doing ... not what asterisk is doing ... i cant speak for asterisk ... im making an outgoing call ... it isnt .. its processing logic and data |
| 21:34.36 | drmessano | cmnky, the dialplan IN THE ATA starts a digit timeout when you pick the handset up.. which is part of the DIALPLAN OF THE ATA... NOTHING is passed to ASTERISK until you do something that matches the DIALPLAN in the *ATA*, which could be "Timeout and then dial BLAH" or "Match these digits".. otherwise, in about 10 seconds you're going to get a busy signal and fail |
| 21:35.21 | cmnky | drmessano, ok ... so when does the asterisk dialplan start ? |
| 21:35.31 | leifmadsen | cmnky: when the INVITE is authenticated |
| 21:35.37 | drmessano | After the ATA matches SOMETHING and passes the call to Asterisk |
| 21:35.40 | leifmadsen | cmnky: the entire number is passed as a single string |
| 21:35.41 | *** join/#asterisk superm1 (~superm1@ubuntu/member/superm1) |
| 21:35.47 | voxter | Can anyone suggest a paid-for (or free if it actually exists) Asterisk CDR reporting engine that can tell me stats on things like # of calls per inbound DID per month, etc? |
| 21:35.50 | drmessano | If the ATA doesn't get a match for SOMETHING.. NOTHING happens |
| 21:36.13 | cmnky | in order to avoid a big circle and an argument over semantics here ... would reading the SIP RFC answer this question ? |
| 21:36.17 | atan | I want 1000 to dial this extension... where did this go wrong? exten =>_1000,1,DIAL(SIP/Akiga}, 60, WT) |
| 21:36.25 | p3nguin | drmessano: But packets are flowing from the ATA to Asterisk. I saw it on my network screening device! |
| 21:36.29 | drmessano | cmnky, both myself and a developer both answered your question |
| 21:36.29 | leifmadsen | voxter: you could try these guys? http://www.dthvoipbilling.com/call-rating-service.aspx |
| 21:36.39 | drmessano | cmnky, it doesn't get better than that |
| 21:36.42 | leifmadsen | atan: no spaces |
| 21:36.47 | cmnky | but just now you failed to answer my question |
| 21:36.53 | leifmadsen | atan: (in the Dial() ) |
| 21:37.05 | drmessano | [16:35] <drmessano> After the ATA matches SOMETHING and passes the call to Asterisk |
| 21:37.06 | drmessano | [16:35] <leifmadsen> cmnky: the entire number is passed as a single string |
| 21:37.07 | leifmadsen | atan: you also don't need _ in the extension because you're not doing any pattern matching |
| 21:37.08 | drmessano | ^^^^ |
| 21:37.15 | cmnky | i'll go read the rfc ... thanks everyone |
| 21:37.20 | drmessano | No, thank you |
| 21:37.44 | p3nguin | It shouldn't be this difficult. |
| 21:37.48 | drmessano | It's not |
| 21:37.53 | leifmadsen | +1 |
| 21:38.03 | atan | It says [Dec 2 21:37:45] WARNING[23377]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 21:38.04 | atan | <PROTECTED> |
| 21:38.07 | voxter | leifmadsen: thanks, ill give em a peek. |
| 21:38.16 | leifmadsen | atan: means the other end is not available |
| 21:38.20 | leifmadsen | atan: sip show peers |
| 21:38.40 | atan | Akiga shows registered in that list |
| 21:38.42 | atan | and the ip & so on |
| 21:38.58 | *** join/#asterisk tzervo (~tzervo@77.49.80.174.dsl.dyn.forthnet.gr) |
| 21:39.08 | *** part/#asterisk tzervo (~tzervo@77.49.80.174.dsl.dyn.forthnet.gr) |
| 21:39.13 | leifmadsen | atan: and it's named [Akiga] in sip.conf? is there a firewall? what does the sip trace say? |
| 21:39.47 | atan | sip.conf has the user starting with [Akiga] ; Desktop |
| 21:40.14 | leifmadsen | atan: look at the sip trace then and see why it is getting rejected |
| 21:40.21 | Lord_Rahl | voxter, http://samreports.com/ - http://www.cdr-stats.org/ |
| 21:41.03 | atan | leifmadsen, how do I trace it? =) |
| 21:41.23 | atan | I suppose I can change it to another device to see if it works |
| 21:41.25 | atan | one sec =D |
| 21:41.30 | leifmadsen | atan: sip set debug peer Akiga |
| 21:42.09 | nevermind_quack | atan: unless you typed it wrong here you have an extra right curly brace } after the device name |
| 21:42.24 | leifmadsen | nevermind_quack: oh snap nice catch |
| 21:44.02 | atan | nevermind_quack, NICE FIND! thanks =D |
| 21:45.42 | superm1 | nevermind_quack, i didn't see bt from earlier, re #18412 did you make any other progress in identifying what happened? |
| 21:48.32 | nevermind_quack | superm1: nothing, I'm lost -- packet dumps aren't helpful because of TLS, debug messages are generic |
| 21:48.54 | atan | What could I start all internal extensions with to prevent them from ever getting confused or mixed up with real world numbers? 000 ? |
| 21:49.11 | superm1 | nevermind_quack, dang :( |
| 21:49.37 | tzanger | malcolmd: how hard is it to incorporate a proprietary DAHDI hardware driver into switchvox? I'm writing the driver, I can write for a specific version of DAHDI, but how difficult is it to provide what's essentially a "driver disk" for switchvox? |
| 21:50.11 | superm1 | nevermind_quack, as a quick hack, i have to wonder if maybe google started looking for anyone with /asterisk in their identifier |
| 21:50.32 | superm1 | and just returning a 400 code if they tried to initiate a call |
| 21:51.00 | superm1 | because calls can still be initiated from the standard client in gmail just fine behind the same IP (they're not blacklisting IPs) |
| 21:51.58 | nevermind_quack | how are you getting the xml details? i haven't figured that piece out. |
| 21:52.54 | superm1 | that's just a speculative theory, i haven't been able to get xml details either. putting jabber in debug just spits out a few extra details like the error code |
| 21:52.56 | Holos | So.. Echo on SIP trunks.. anyone have a thought on what causes it? Is it the upsteam provider I need to go after? |
| 21:53.47 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 21:54.05 | cmnky | has an occasional awful echo on his FIOS landline |
| 21:54.11 | p3nguin | What the crap... my outgoing calls no longer have ringing sound. I thought I noticed that a couple weeks ago, but now I've confirmed it. |
| 21:54.25 | p3nguin | I guess the ITSP probably changed something. |
| 21:54.40 | _Corey_ | p3nguin: you may need progressinband or something |
| 21:55.19 | p3nguin | _corey_: Where do you propose I configure that? |
| 21:55.20 | _Corey_ | p3nguin: i had this happen spontaneously with one carrier |
| 21:55.21 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:55.38 | _Corey_ | p3nguin: the sip peer |
| 21:55.45 | p3nguin | I'm not using any SIP. |
| 21:55.55 | _Corey_ | ah, when you said ITSP I assumed... |
| 21:56.00 | p3nguin | SCCP from the phone to Asterisk, IAX2 from Asterisk to the ITSP. |
| 21:56.10 | _Corey_ | ah |
| 21:59.29 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 22:04.21 | *** join/#asterisk tyman (~tyler@173-14-203-39-fresno.hfc.comcastbusiness.net) |
| 22:04.31 | *** part/#asterisk tyman (~tyler@173-14-203-39-fresno.hfc.comcastbusiness.net) |
| 22:08.13 | p3nguin | _corey_: I switched it to SIP and still got no ringing sound. I added progressinband=yes in the peer, sip reload ... and I still get no ringing sound. |
| 22:09.07 | _Corey_ | hm, that did it for us with Global Crossing |
| 22:09.20 | _Corey_ | Maybe your carrier is having a problem upstream |
| 22:10.47 | p3nguin | Adding the 'r' option in the Dial() gives back the ringing, but I hate doing that. |
| 22:13.44 | _Corey_ | You could debug the SIP messaging a little bit if you want to see if the carrier is attempting to provide in-band ringback |
| 22:15.53 | nevermind_quack | superm1: found a good resource -- http://code.google.com/apis/talk/call_signaling.html -- reading through it now |
| 22:18.10 | atan | Is there something special within the dialplan that passes video? |
| 22:18.13 | p3nguin | We go from 100 Trying to 183 Session Progress. |
| 22:18.26 | atan | I have two video phones but when they call eachother the caller shows their video on the called, but not the other way around |
| 22:18.55 | *** join/#asterisk blebleble (~blebleble@s7.n194.n222.n216.static.myhostcenter.com) |
| 22:19.02 | superm1 | nevermind_quack, interesting, so the error code 400 is actually coming from the 'candidates' part where they are trying to negotiate RTP ports it seems |
| 22:19.34 | blebleble | I have a number of did's dumping into the same queue. Is there a way to give a certain did priority over the others when entering the queue? Like put them to the front of the line? |
| 22:19.47 | atan | No, sorry, the video call goes through as an audio call according to the dialed phone |
| 22:20.03 | nevermind_quack | superm1: call flow starts with initiate then proceeds to candidates ... if you look back you should see the 400 message in reply to each request |
| 22:20.26 | superm1 | oh you're right there are 2 400 message replies |
| 22:21.01 | *** join/#asterisk prudhvi (~nobody@nullpointer.in) |
| 22:21.31 | prudhvi | Hi, when using Dial() with 'W' option in Dial plan, is it possible to define the name of the output file? |
| 22:22.15 | nevermind_quack | superm1: on a forum i read that calls to buddies still work, i can't verify that but it suggests a syntax problem (?) |
| 22:22.27 | nevermind_quack | er ... with just pstn calls that is |
| 22:22.48 | superm1 | nevermind_quack, a syntax problem? but things worked no changes a day or two ago |
| 22:23.14 | superm1 | unless google stopped supporting something |
| 22:25.01 | superm1 | could you double check the exact PSTN's they're dialing? are they appending a + or something? |
| 22:25.30 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.156.49) |
| 22:26.08 | *** join/#asterisk slidesinger-lt (~jtatum@12.146.80.48) |
| 22:27.03 | nevermind_quack | superm1: <bad-request/> -- the sender has sent XML that is malformed or that cannot be processed (e.g., an IQ stanza that includes an unrecognized value of the 'type' attribute); the associated error type SHOULD be "modify". |
| 22:27.32 | nevermind_quack | http://tools.ietf.org/html/rfc3920#section-9.3.3 |
| 22:27.42 | superm1 | i was just gonna ask where you got that from |
| 22:28.20 | superm1 | interesting |
| 22:30.02 | *** join/#asterisk dan__t (~dant@vpn.withparity.net) |
| 22:30.05 | dan__t | Hello. |
| 22:30.52 | dan__t | Kind of just a question on good practice but.... what are the more common ways of storing dialplan information when the dialplan is just plain gigantic? Sure you could use macros and reuse a lot of contexts and stuff, but then what? |
| 22:31.03 | dan__t | Is it common to store this data in SQL maybe? Something like that? |
| 22:35.38 | user1324 | nevermind_quack: there is no error text though which seems to suggest they don't want you to know what's wrong |
| 22:35.47 | WIMPy | dan__t: That's called realtime |
| 22:35.53 | atan | Do you need something special in your extensions.conf to pass video when you Dial() ? |
| 22:36.16 | WIMPy | atan: No |
| 22:36.34 | atan | My video calls are connecting but no video is being sent |
| 22:36.40 | atan | Can I force it to go through the server and not peer to peer? |
| 22:37.03 | WIMPy | atan: directmedia=no |
| 22:37.15 | dan__t | Whoa, that's pretty cool. |
| 22:38.25 | atan | WIMPy, sip reload would do it eh? The phones shouldn't need to re-register? |
| 22:38.30 | dan__t | Way cool, actually. |
| 22:39.06 | WIMPy | atan: Correct |
| 22:39.28 | atan | Okay still no video happening for me. Hmm. Could 'Proxy type' mess this up on the phone? |
| 22:39.55 | atan | There are 5 options inside it. It's set to CS2K right now. Audio calls are running fine =) |
| 22:40.20 | WIMPy | atan: No generic option? |
| 22:40.47 | atan | It offers MCS5100, MCS5100 4.0, AS5200, and CS2K |
| 22:40.54 | atan | No generic stuff is listed in there. |
| 22:42.20 | superm1 | nevermind_quack, i just did some experiment and at least match the behavior from forums, I can call someone's @gmail.com address from asterisk and it appears to at least ring and not spit back any errors, i actually see the 'result' response and it ringing |
| 22:42.31 | superm1 | so it is only happening for pstn |
| 22:43.06 | *** join/#asterisk vimpulse (63fdf6a8@gateway/web/freenode/ip.99.253.246.168) |
| 22:43.58 | nevermind_quack | superm1: k i will add a note to the bug tracker |
| 22:45.53 | vimpulse | hi all. A client of mine has four employees and are looking to become five. They own a Norstar 6x16 PBX, but they don't like that moves, adds, and changes are difficult. |
| 22:45.53 | vimpulse | They pay for three trunks from Bell Canada (total is about US$115/mo), plus US$30-40/mo or so for long distance, and have cable-modem Internet service. Should they buy another Norstar phone plus some spares, or look into replacing their phone system? |
| 22:46.26 | vimpulse | (i.e. perhaps with Asterisk or some other VoIP-based PBX) |
| 22:46.45 | _Corey_ | Yeah, get Asterisk |
| 22:46.51 | _Corey_ | This is an Asterisk forum, so... |
| 22:47.41 | vimpulse | Are you sure it'd be cost-effective? What kind of broadband Internet access would they need to upgrade to, for one thing, and how much might that cost per month? |
| 22:48.55 | nevermind_quack | superm1: what is your "to=" value for the buddy call? what are you using for a dial pattern? i've been using the recommended +1NPANXXXXXX@voice.google.com |
| 22:49.10 | nevermind_quack | maybe that's what changed (?) |
| 22:50.37 | vimpulse | _Corey_: it seems to me that for VoIP they'd need to move to a T1 at US$400 or more per month. Correct? |
| 22:51.39 | Qwell | vimpulse: for 3 analog channels? very little bandwidth is required for that. |
| 22:51.51 | p3nguin | vimpulse: How many simultaneous incoming calls will you have? |
| 22:51.58 | p3nguin | have or need |
| 22:52.03 | vimpulse | p3nguin: three or four |
| 22:52.05 | _Corey_ | vimpulse: Not necessarily, it depends on location and broadband availabiliy |
| 22:52.22 | vimpulse | _Corey_: Toronto, Canada, a city of a few million people. |
| 22:52.25 | p3nguin | vimpulse: If you get a T1, you'll get 19 or 20 more than you require. |
| 22:52.35 | p3nguin | vimpulse: That's a lot of extra. |
| 22:52.51 | vimpulse | so what would my client buy instead of a T1? I'm new to this. |
| 22:53.36 | _Corey_ | DSL, Cable modem, FIOS, ... |
| 22:53.39 | p3nguin | vimpulse: Cable modem or DSL service would do the job as long as you get fast enough service. A T1 will give you 23 (or 24) channels. You need one channel per simultaneous call. |
| 22:54.17 | vimpulse | there's no fiber optic service here, only cable or DSL. Are cable or DSL reliable enough to run a business's phone system with? |
| 22:54.29 | p3nguin | vimpulse: Sometimes. |
| 22:54.48 | vimpulse | what do you do in the cases where they're not reliable enough? |
| 22:55.06 | p3nguin | vimpulse: In many cases, DSL and cable are fine. In a few rare instances, there are major problems. |
| 22:55.06 | ChannelZ | tin can + string |
| 22:55.23 | vimpulse | ChannelZ: :) |
| 22:55.52 | _Corey_ | Basically, Cable+DSL is much less expensive than a T1 and much more resilient |
| 22:56.38 | vimpulse | what do you mean "resilient"? |
| 22:56.40 | ChannelZ | I dunno about resilient |
| 22:56.53 | p3nguin | You should be able to get a suitable DSL circuit for $30 or less per month. |
| 22:56.58 | ChannelZ | T1's cost a lot more and generally you get some sort of SLA at least |
| 22:57.32 | superm1 | nevermind_quack, i've tried both +1ZZZXXXXXXX@voice.google.com and 1ZZZXXXXXXX@voice.google.com |
| 22:57.35 | ChannelZ | Cable/DSL, it really depends. Business cable they pay attention to you a bit more when you call.. |
| 22:57.37 | superm1 | both fall over just the same |
| 22:57.46 | _Corey_ | By having two types of circuit, (providing you have a router that leverages both) you will be able to survive more outage scenarios than any single circuit |
| 22:57.49 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 22:58.00 | p3nguin | Then you'll pay an ITSP for minutes of use. |
| 22:58.02 | ChannelZ | Oh. You meant having both. |
| 22:58.16 | _Corey_ | i.e. if your local CO has a problem, you may have a problem with T1 and DSL |
| 22:58.20 | _Corey_ | but not cable... |
| 22:59.47 | vimpulse | how much does it cost for a router that lets you connect your business to both a DSL modem and a cable modem at the same time? |
| 23:02.12 | p3nguin | Depends on your throughput. |
| 23:02.54 | *** join/#asterisk rs100 (~first.las@66.202.175.254) |
| 23:02.57 | p3nguin | I have an appliance sitting here with dual wan (with failover) that will do about 14 Mbps WAN-to-LAN that I'd sell you for about $25 plus shipping. |
| 23:03.26 | p3nguin | I can't use it because I have 60 Mbps service. |
| 23:03.27 | _Corey_ | We normally do Cisco 1800 series routers, i'm not sure how much they cost |
| 23:03.51 | _Corey_ | There are a wide range of products that would fit though |
| 23:04.15 | rs100 | Hi all, I have a problem with DAHDI and hardware echo cancel, should you have to specify an echo canceler in system.conf for DAHDI if you have the hardware addon for an AEX2400, or just enable echo canceling in chan_dahdi.conf? |
| 23:06.20 | vimpulse | p3nguin: good price. Thanks for the offer, but I'd need to do much more research on my own first, so not now. My email address is unforgettableid@gmail.com. May I ask yours, in case I decide to buy your appliance? |
| 23:08.05 | p3nguin | vimpulse: You can reach me here if you decide you want it. It is a Netgear FVS124G. It works well as long as you're aware of the throughput limitation. |
| 23:10.10 | superm1 | nevermind_quack, it looks like reading through some packet captures from the official client, it does use +1ZZZXXXXXX@voice.google.com internally |
| 23:11.23 | rs100 | I ask because I'm seeing this error: [Dec 2 17:54:36] WARNING[4913]: chan_dahdi.c:2685 dahdi_enable_ec: Unable to enable echo cancellation on channel 8 (No such device) which I've never seen before |
| 23:11.31 | p3nguin | vimpulse: I should add that I do not know how compatible it is with VoIP, so it would be a good idea to specifically look for that while you are doing your research. |
| 23:12.10 | p3nguin | I've never used it on a network with VoIP traffic. |
| 23:12.37 | nevermind_quack | superm1: how are you capturing from the official client? anything jump out as being different with the initiate request? |
| 23:13.39 | atan | Anyone see an error in my sip.conf? http://pastie.org/private/fzbyxn6zzfxyuvoxx4atq |
| 23:14.32 | superm1 | nevermind_quack, wireshark, i was just able to catch some plain text in the stream |
| 23:14.48 | superm1 | it's not all nice though, it's pretty raw, so just a few things stand out |
| 23:14.58 | nevermind_quack | hmm ... gotcha |
| 23:18.00 | vimpulse | So, let's assume that cable and DSL are each US$30 per month here in Toronto. It sounds like moving to VoIP for incoming and outgoing calls, and using a router that supports two WANs (one cable, one DSL), and failover, might be cheaper than the $115/mo my client pays Bell. It'd be only $60/mo plus the charges my client pays their ITSP. But is it worth moving for such small savings? |
| 23:18.15 | vimpulse | ($60/mo for cable plus DSL) |
| 23:20.19 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.158.131) |
| 23:22.08 | p3nguin | vimpulse: What's the current situation that you're considering switching to DSL and cable? |
| 23:23.24 | vimpulse | p3nguin: the receptionist said it took her an hour to log onto Google Apps this morning. I don't know whether it was because of unreliable cable or not. I find the cable here in Toronto to be completely reliable, based on personal experience. |
| 23:25.26 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 23:27.45 | p3nguin | vimpulse: Does that mean you were wanting to add a second cable modem dedicated to the phones? I doubt the phone usage over the current cable modem was dragging down the computers' traffic. |
| 23:28.25 | vimpulse | p3nguin: the phone service now goes through analog Bell Canada trunks. |
| 23:28.52 | *** join/#asterisk Godfather_ (~Godfather@108.Red-88-6-252.staticIP.rima-tde.net) |
| 23:29.16 | p3nguin | I refer you back to my question: What's the current situation that you're considering switching to DSL and cable? |
| 23:29.16 | Godfather_ | hi |
| 23:29.44 | p3nguin | If you were satisfied with the way things are now, you wouldn't be looking for a change. |
| 23:30.16 | vimpulse | p3nguin: maybe there is no such situation. It's nice to know, though, that it'll be possible to switch to a dual-WAN router with failover, later, if my client needs it. |
| 23:30.56 | Godfather_ | in sip.conf in general section i have: disallow=all allow=alaw, and in a client (x-lite) i only enable alaw codec, but when i try to play moh or something i get NOTICE[23617]: chan_sip.c:8497 process_sdp: No compatible codecs, not accepting this offer!, any ideas? |
| 23:31.04 | p3nguin | vimpulse: I was under the impression that you needed a new internet connection for phones. I guess I got confused somewhere. |
| 23:32.06 | vimpulse | p3nguin: I initially thought we needed a T1. I was wrong. :) |
| 23:34.18 | Godfather_ | is there any way to see what codec is offering a client? (x-lite in this case) |
| 23:35.08 | p3nguin | godfather_: If you are using SIP, it is shown in the sip debug. |
| 23:35.45 | cmnky | vimpulse, i went looking for a dual wan router ... the reviews on newegg pretty much says they all suck |
| 23:35.47 | p3nguin | for example: Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) |
| 23:35.51 | *** join/#asterisk Docfxit (ExUser@netblock-208-127-208-174.dslextreme.com) |
| 23:36.20 | cmnky | vimpulse, suck, as in, buggy firmware that doesnt work right and is impossible to fix |
| 23:36.21 | superm1 | nevermind_quack, you appear to have gotten your identification string to list as a real Talk client would according to those snippets (/Talk30658F9F), so i guess that rules out google intentionally filtering asterisk clients |
| 23:37.11 | beek | cmnky: check out vyatta |
| 23:37.40 | cmnky | beek, is that a linux distro ? |
| 23:37.55 | beek | debian-based commercial/open source router distro |
| 23:37.58 | beek | www.vyatta.com |
| 23:38.10 | beek | I use it everywhere. Great stuff. |
| 23:38.22 | Godfather_ | p3nguin, Capabilities: us - 0x0 (nothing), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) |
| 23:38.28 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 23:38.33 | cmnky | beek, yeah ... i was talking about the "dual wan router" appliances from newegg ... not sure if they had vyatta or not |
| 23:38.51 | beek | cmnky: You can assemble one yourself easily. I use the Alix boards for that purpose in a small environment. |
| 23:39.05 | cmnky | they all had one problem or another ... like working for 2 weeks then crapping out / hogging ram ... not switching / balaneing well, etc. |
| 23:39.14 | Docfxit | Hi, |
| 23:39.30 | vimpulse | beek: interesting idea. |
| 23:39.40 | Godfather_ | p3nguin, what is wrong? here is the full debug sip, http://pastebin.com/XxjsbiHF |
| 23:39.41 | cmnky | beek, yes .. i have my own router/firewall box ... just cautioning vimpulse on the appliances that claim to solve the problem for X dollars ... but suck |
| 23:40.41 | beek | vimpulse: Old computer + NIC + managed switch with VLAN support = solution to your problem. |
| 23:41.21 | DeHackEd | router on a stick? |
| 23:41.28 | p3nguin | godfather_: You don't have any available codecs. |
| 23:41.44 | p3nguin | us - 0x0 (nothing) |
| 23:41.52 | nevermind_quack | superm1: yeah I don't see anything to tip them off that they're talking to asterisk |
| 23:41.56 | Godfather_ | p3nguin, 'us' what means? |
| 23:42.12 | p3nguin | godfather_: us. Us. As opposed to them. |
| 23:42.23 | p3nguin | godfather_: Asterisk is considered to be "us." |
| 23:42.40 | Docfxit | I don't have port 5060 or sip listening on Ubuntu/Asterisk. In sip.conf I have bindport = 5060 Iptables shows nothing is blocked . What else would stop asterisk from listining on port 5060? |
| 23:42.48 | p3nguin | godfather_: The peer is the device or "other side" which you are trying to communicate with. |
| 23:42.56 | Godfather_ | p3nguin then the problem is on the asterisk side? |
| 23:43.21 | p3nguin | godfather_: In your peer entry in sip.conf, try adding disallow=all and allow=alaw,ulaw in the peer entry. |
| 23:44.15 | p3nguin | godfather_: Save the file, then run "sip reload" on the asterisk console. |
| 23:45.08 | Godfather_ | p3nguin, http://pastebin.com/g43Udiu5 |
| 23:45.38 | p3nguin | What a jacked up sip.conf. |
| 23:45.44 | Godfather_ | if i add allow=ulaw then it works, but i want to use alaw, and in the side of x-lite the only enabled codec is alaw |
| 23:46.15 | Godfather_ | p3nguin im testing it, i dont know why it doesnt work |
| 23:46.44 | p3nguin | On the debug you showed me, the peer was offering both ulaw and alaw. |
| 23:47.04 | p3nguin | <PROTECTED> |
| 23:47.05 | Godfather_ | p3nguin can you put me this line? where the peer |
| 23:47.10 | Godfather_ | aha |
| 23:48.20 | p3nguin | The line in the debug shows first "us" (this is asterisk). Then it shows "peer" (this is the device). Then it shows "combined" (this is what both asterisk and the device support together). |
| 23:49.10 | p3nguin | If Asterisk supports only ulaw and the peer supports ulaw/alaw, combined will be ulaw. |
| 23:49.44 | p3nguin | If Asterisk supports only g729 and the peer supports ulaw/alaw, combined will be nothing. |
| 23:49.49 | p3nguin | Et cetera |
| 23:49.58 | Godfather_ | p3nguin, i think i solved it, i dont know the problem, but seems working |
| 23:50.02 | Godfather_ | 192.168.1.4 16 NjI5OGQ0YzkwYmQ 0x8 (alaw) No Rx: ACK |
| 23:51.01 | Godfather_ | thanks for the explanation |
| 23:51.56 | Godfather_ | p3nguin, what do you refer with "Jacked up"? |
| 23:52.08 | p3nguin | screwed |
| 23:53.20 | Godfather_ | why? |
| 23:53.30 | p3nguin | It's incomplete. |
| 23:54.55 | Godfather_ | p3nguin what are you missing in that file? |
| 23:57.02 | Godfather_ | p3nguin, i mean, i just change the bind sip port, and defined a few peers... i dont understand your point of "screwed" |
| 23:57.19 | p3nguin | Changed it from what? The default? |
| 23:57.37 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 23:58.06 | Godfather_ | p3nguin what is imcomplete? |
| 23:58.27 | carrar | ~book |
| 23:58.27 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 23:59.13 | Godfather_ | p3nguin, i'm ussing the template "sets"... |
