| 00:02.20 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 00:03.40 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:03.53 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 00:03.53 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:04.09 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 00:05.30 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 00:08.23 | riddlebox | hrmm I was given 3 cisco 7940 phones today, but no power supplies |
| 00:08.59 | k-man | that sounds like a nice gift |
| 00:09.45 | k-man | riddlebox: are the POE capable? |
| 00:10.09 | riddlebox | k-man, yeah thats what the people were using them with |
| 00:10.35 | k-man | it might be cheaper to buy a POE switch than to buy the power supplies for them |
| 00:10.51 | riddlebox | I kinda looked on ebay, 20 bucks a piece |
| 00:10.59 | k-man | oh, thats not too bad |
| 00:11.04 | k-man | they look like pretty good phones |
| 00:11.19 | riddlebox | I am looking into turning them into SIP phones |
| 00:11.30 | k-man | are they not natively sip? |
| 00:11.39 | riddlebox | nope I believe its skinny |
| 00:11.52 | k-man | oh - don;t know what that means |
| 00:12.37 | riddlebox | thats their codec |
| 00:13.08 | k-man | oh, i see |
| 00:13.18 | k-man | so how would you convert them to SIP? |
| 00:13.39 | mcf3782 | I'm looking for some suggestions/guidance on how to troubleshoot an issue with my TDM400 card in my Asterisk box. |
| 00:14.31 | riddlebox | k-man, you have to load the firmware for it, which is what I am looking at now, I dont know if I am going to use them or not |
| 00:14.45 | riddlebox | mcf3782, it helps to know whats wrong |
| 00:14.57 | k-man | riddlebox: are they different to the 7940G phones on the cisco site? |
| 00:15.07 | mcf3782 | yep. :) |
| 00:15.21 | riddlebox | not sure |
| 00:15.46 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 00:16.25 | mcf3782 | When the FXO port goes off hook to make or receive a call from the POTS line; it loads down the line to the point that the DSL service that's delivered over that same POTS line loses sync with the DSLAM. |
| 00:17.01 | mcf3782 | This is a recent....bug... It's been working flawlessly for well over a year. Whatever this is just started recently. |
| 00:17.29 | mcf3782 | I've made no changes to config files on the box. |
| 00:17.59 | mcf3782 | I've swapped out the FXO daughter card on the TDM card to see if that changed anything. It does not. |
| 00:18.56 | mcf3782 | If I unplug the asterisk box from the line, and plug a phone directly into it; making a voice call has no affect on the DSL service. |
| 00:19.39 | mcf3782 | So it would seem that something is.....broken...with either the TDM card or the module (two bad ones in a row?) or.....? |
| 00:19.44 | mcf3782 | Any other suggestions? |
| 00:20.45 | eldorel | have you moved any of the dsl filters on the line recently? |
| 00:20.46 | lanning | are you using a DSL filter? |
| 00:21.41 | mcf3782 | The DSL signal is filtered at the telco demarc and goes to one single jack. The DSL modem is plugged directly into that jack. |
| 00:23.11 | mcf3782 | Well. I sure know how to kill a room don't I? :) |
| 00:23.15 | eldorel | there should be a filter on the line at the jack the fxo is plugged into. |
| 00:24.03 | lanning | not nessarily |
| 00:24.29 | mcf3782 | It's filtered before the signals ever leave the demarc. The voice signal goes to one place from the demarc and the DSL signal goes to another. They never travel together past the at&t side of the demarc. |
| 00:24.47 | eldorel | ok, |
| 00:26.31 | eldorel | so there is a filter, just on the opposite end of the cable. got it. Have you had any storms/power outages that could have damaged that filter? |
| 00:26.35 | mcf3782 | None that I know of. |
| 00:26.48 | mcf3782 | But in Atlanta in late winter/early spring... anything is possible. |
| 00:27.05 | Qwell | if the filter is doing its job, nothing on it *can* muck with the DSL signal. |
| 00:27.21 | eldorel | The reason i'm asking is the only things i've seen cause the problem you're having is when a device draws too much current off of the line, or when there was no filter/bad filter |
| 00:27.26 | Qwell | like, it's simply not possible. That's the entire point of the filter. |
| 00:27.29 | brunner | Why are calls to +44800 so expensive, if it's a free call in the UK? |
| 00:27.33 | mcf3782 | I'm going to open a trouble ticket with at&t in the morning; just trying to troubleshoot as much on my own before then as I can. |
| 00:27.51 | eldorel | do you have an o-scope? |
| 00:28.02 | Qwell | or, you know...another filter? :p |
| 00:28.13 | eldorel | or a management interface on the dsl modem that shows power levels? |
| 00:28.13 | Qwell | also, filters shouldn't be used in that manner. |
| 00:28.30 | Qwell | There should be a filter on each line, not many lines from a single one |
| 00:28.37 | *** join/#asterisk isamar (n=isamar@server1.dw7.telegate-americas.com) |
| 00:28.39 | Qwell | (unless things have changed significantly since I had DSL..) |
| 00:28.41 | mcf3782 | Qwell - wish I had one.. This is an at&t supplied one that's wired into the demarc. |
| 00:28.47 | eldorel | @qwell:depends on the filter, some have multiple outlets. |
| 00:28.49 | *** join/#asterisk ingenius (n=alektro@host2.190-31-177.telecom.net.ar) |
| 00:28.59 | k-man | if you purchase a license of the g729 codec, does that license last for ever or is it a subscription type license? |
| 00:29.04 | Qwell | if it's AT&T's, it's probably fine |
| 00:29.11 | Qwell | k-man: I do believe it is forever. |
| 00:29.14 | eldorel | mcf: they're about $8 us at most hardware/electronics stores.... |
| 00:29.19 | k-man | thanks Qwell |
| 00:29.25 | mcf3782 | Bellsouth (now the new at&t) wired this up in this config when the line was installed. |
| 00:29.34 | Qwell | k-man: of course, you'll want to verify with sales@digium.com |
| 00:29.42 | k-man | Qwell: yeah, thanks |
| 00:31.23 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 00:34.46 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 00:35.08 | mcf3782 | Doesn't sound like it rings any bells with anyone about other things to test... So I'll go have dinner and open a ticket with Bell tomorrow. |
| 00:35.32 | mcf3782 | Thanks for the extra set of eyes, folks. Much appreciated. |
| 00:35.37 | eldorel | any radioshack near you? |
| 00:35.47 | mcf3782 | Yep |
| 00:36.08 | Qwell | eldorel: If it's in a dmarc, it's going to be much higher quality than the crap at radio shack |
| 00:36.13 | eldorel | stop by and pick up a filter while you at dinner, $5 and it might fix it. |
| 00:36.16 | Qwell | (one would hope) |
| 00:36.30 | eldorel | yeah, but you can't replace that one tonight.... |
| 00:36.46 | *** join/#asterisk jsgoecke (n=jsgoecke@c-67-180-103-93.hsd1.ca.comcast.net) |
| 00:37.00 | mcf3782 | and just plug that filter in between the line from the demarc and the FXO port to see if the telco filter has gone bad...... |
| 00:37.02 | mcf3782 | good thought. |
| 00:37.12 | eldorel | good luck |
| 00:37.19 | Qwell | mcf3782: You will of course want to remove the old filter |
| 00:37.46 | mcf3782 | And I just found a reason to actually carry one of those cheapie DSL filters around with me now. :) |
| 00:38.05 | *** join/#asterisk docid (n=eris@69.196.68.142) |
| 00:38.14 | mcf3782 | Qwell - good point. |
| 00:38.43 | *** join/#asterisk RoPBX (n=nickserv@200.93.34.175) |
| 00:39.03 | mcf3782 | Now if only the local radio shack actually has one. Man that store has turned into a sad, sad, mostly useless shell of what that company once was. |
| 00:39.36 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-051d49cb24038fd0) |
| 00:39.53 | mcf3782 | Off to find dinner and a filter. Good suggestion guys. Thanks!! |
| 00:40.07 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 00:53.48 | *** join/#asterisk wcuestas (n=wcuestas@190.232.57.225) |
| 00:54.26 | wcuestas | hi, i got a question |
| 00:54.58 | wcuestas | is there any way to disable call waiting and have the target phone a missed call notification ? |
| 00:58.35 | lanning | that depends on what technology you are using. |
| 00:59.31 | lanning | the missed call notification is a feature of the phone, not the pbx |
| 00:59.54 | lanning | and requires call waiting/second line |
| 01:00.33 | wcuestas | we are using grandstream phones (GXP 280, 1200 and 2000) |
| 01:00.37 | *** part/#asterisk jchonig (n=jch@firewall.honig.net) |
| 01:00.54 | wcuestas | and have disabled call waiting, but, there is no missed call notification |
| 01:00.56 | jaytee | Radio Shack, back in the day the salespeople actually knew what a capacitor or zener diode was, "You've got questions? We've got answers!" has now turned into "Yeah, we got stuff!" |
| 01:01.24 | lanning | right, without call waiting the phone will never know that a call was attempted. |
| 01:01.43 | lanning | as I said, missed call notification is in the phone, not the pbx |
| 01:01.56 | wcuestas | yeap, you are very right |
| 01:02.15 | wcuestas | we have a manager asking this "feature" because he saw it in another company |
| 01:02.29 | wcuestas | but, we've tested it and it's like you said |
| 01:03.48 | wcuestas | thanks a lot! |
| 01:03.55 | lanning | one possibility is to have a phone register twice, using separate IDs. One can ring silently |
| 01:04.11 | lanning | but this all depends on the phone |
| 01:04.55 | lanning | you would implement a busy call forwarding to the second registration. |
| 01:05.56 | riddlebox | I wish I could find the firmware upgrade for these stupid phones |
| 01:06.36 | wcuestas | lanning: so, the phones should handle 2 accounts |
| 01:06.48 | lanning | ya |
| 01:06.57 | wcuestas | lanning: this is possible just in GXP 1200 |
| 01:07.28 | lanning | give the manager that one. :) |
| 01:07.39 | wcuestas | :) |
| 01:08.10 | rob0 | "You've got questions? We've got shrugs." |
| 01:08.13 | k-man | is there some way I could direct where voicemail messages get email based on the CID? |
| 01:08.52 | lanning | the voicemail app doesn't have that. |
| 01:09.30 | lanning | you could forward the email to a mail processor, that parses the email and routes based on content. |
| 01:10.19 | k-man | oh |
| 01:10.28 | k-man | ok - thats too much effort - doesn't matter |
| 01:10.54 | wcuestas | lanning: thanks so much, bye |
| 01:11.00 | lanning | bye |
| 01:11.07 | *** join/#asterisk CrazyTux (n=brandon@c-98-196-6-54.hsd1.tx.comcast.net) |
| 01:11.09 | *** part/#asterisk wcuestas (n=wcuestas@190.232.57.225) |
| 01:14.43 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 01:18.51 | k-man | how do i make a dialplan that will record a message until the user presses hash? i want it for recording prompts for IVR |
| 01:18.51 | jaytee | drmessano ping |
| 01:20.03 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 01:24.21 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 01:24.24 | joako | k-man: show application record |
| 01:25.15 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 01:25.56 | trnzmeta | guys: what does the flash operator panel actually do? |
| 01:26.36 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 01:27.10 | trnzmeta | ok got the explanation from web |
| 01:27.17 | trnzmeta | always confused cos it never worked |
| 01:27.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:28.08 | k-man | joako: thanks |
| 01:28.24 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 01:29.15 | k-man | if show file formats says "ogg_vorbis" is one of the formats available, would I be able to use that in voicemail recordings? |
| 01:31.35 | *** part/#asterisk `paul (n=kutimoy@121.97.99.151) |
| 01:39.53 | NovceGuru | wonders how datacenters get their "tier" status |
| 01:41.38 | NovceGuru | anybody have any issues with outbound vitelity today? |
| 01:41.51 | NovceGuru | er, voicepulse actually |
| 01:46.09 | KavanS | yeah |
| 01:46.11 | KavanS | I did.... |
| 01:46.19 | KavanS | I was thinking of dropping them actually |
| 01:46.27 | *** join/#asterisk BeeBuu (n=beebuu@59.38.96.249) |
| 01:46.29 | KavanS | NovceGuru: you have good luck with vitelity? |
| 01:47.16 | NovceGuru | vitelity or voicepulse? |
| 01:47.51 | KavanS | I had issues with voicepulse.... |
| 01:47.57 | KavanS | I was thinking of signing up with vitelity |
| 01:48.14 | KavanS | I'd like to find a "nice" voip carrier, but it seems to be a challenge |
| 01:48.33 | KavanS | I have one PSTN...but I'd like to augment with some reasonable voip provider |
| 01:49.43 | NovceGuru | voicepulse *usually* has been pretty good |
| 01:49.50 | NovceGuru | I just had a few calls today that just sat silent |
| 01:49.52 | KavanS | yeah I had found the same results.... |
| 01:49.59 | KavanS | it's been happening for a couple days for me |
| 01:50.04 | KavanS | I just narrowed it down today to voicepulse |
| 01:50.29 | NovceGuru | fuckers |
| 01:50.38 | KavanS | I called their tech support |
| 01:50.40 | KavanS | they acted like I was crazy |
| 01:50.41 | KavanS | assholes... |
| 01:50.59 | NovceGuru | their seem pretty arrogant when I talk to them |
| 01:51.03 | KavanS | lol yeah |
| 01:51.12 | NovceGuru | "o theres no WAY we can port your number if it doesnt say we can on the website" |
| 01:51.12 | KavanS | I'm going to port a number from them....then pull account |
| 01:51.23 | KavanS | oh yeah they pulled similar shit withme |
| 01:51.28 | NovceGuru | I just use them for outbound trunks |
| 01:51.29 | KavanS | I have toll-free for canada/upper 48 |
| 01:51.33 | KavanS | and they couldn't port for canada |
| 01:51.37 | KavanS | I'm like "WTF?" |
| 01:51.57 | KavanS | don't get me wrong, I understand "technical limitations" |
| 01:52.01 | KavanS | but when someone wants to spend money....most the time the answer involves "yes" |
| 01:52.19 | KavanS | I use them for the free incoming |
| 01:52.19 | KavanS | heh |
| 01:52.52 | KavanS | NovceGuru: do you use vitelity? |
| 01:53.07 | NovceGuru | yeah I use them for inbound, not many providers out here in BFE |
| 01:53.12 | NovceGuru | with local did |
| 01:53.28 | NovceGuru | voicepulse is nice because of their 4 channels |
| 01:53.52 | KavanS | yep |
| 01:54.01 | KavanS | I was thinking of using combination of voicepulse/vitelity/PSTN |
| 01:54.08 | KavanS | so I have *some* redundancy |
| 01:54.15 | NovceGuru | although vitelity has unlimited inbound channels if you pay per minute |
| 01:54.18 | NovceGuru | I wish their outbound was the same way |
| 01:54.32 | KavanS | yeah they prolly don't want call centers using em |
| 01:54.42 | KavanS | they seem to have good failover services too |
| 01:54.48 | NovceGuru | yeah that is nice |
| 01:54.53 | NovceGuru | voicepulse $20 extra for that |
| 01:54.54 | KavanS | the fact they don't charge for it, makes it nice |
| 01:54.59 | KavanS | yeah....20$ per *month* |
| 01:55.04 | NovceGuru | heh yeah |
| 01:55.06 | KavanS | lol |
| 01:55.08 | KavanS | I was like "wtf?" |
| 01:55.46 | NovceGuru | yeah I heard that and didnt belive it |
| 01:55.54 | NovceGuru | charge an exuberant amount per minute |
| 01:55.55 | NovceGuru | for it |
| 01:56.00 | KavanS | yeah lol |
| 01:56.06 | KavanS | so wait....I pay you if your system goes down? |
| 01:56.12 | KavanS | it's like bailing out banks... |
| 01:56.19 | NovceGuru | I think their system goes down you're fucked reguardless |
| 01:56.19 | KavanS | but back on topic :P |
| 01:56.22 | NovceGuru | it's more if yours goes down |
| 01:56.23 | KavanS | haha yeah |
| 01:56.27 | KavanS | they had some downtime |
| 01:56.31 | KavanS | and I asked about redundancy |
| 01:56.34 | KavanS | they are in like 2 data centers |
| 01:56.40 | KavanS | and they forward back to the primary apparently |
| 01:56.47 | KavanS | i.e. san jose is just a proxy for new jersey |
| 01:56.56 | KavanS | or new york/whatever east coast state they are in |
| 01:57.07 | NovceGuru | ahh yeah |
| 01:57.10 | KavanS | so if east coast goes down...they are fucked |
| 01:57.22 | KavanS | and/or I am... |
| 01:57.28 | NovceGuru | yeah |
| 01:57.51 | NovceGuru | I use inbound vitelity failing over to a distinctive ring pstn |
| 01:58.02 | KavanS | ahh nice |
| 01:58.03 | KavanS | that's cool |
| 01:58.05 | NovceGuru | (distinctive becuase it's usually a fax line) |
| 01:58.12 | KavanS | yeah I might make the switch then |
| 01:58.25 | NovceGuru | and I can route to a fxo card |
| 02:01.41 | riddlebox | is chan_sccp in asterisk 1.4.x by default? |
| 02:02.12 | carrar | moo |
| 02:02.32 | carrar | I think so yes |
| 02:04.12 | KavanS | heck yeah |
| 02:04.18 | KavanS | going to deal with vitelity tomorrow then... |
| 02:04.32 | riddlebox | carrar, why the moo? |
| 02:05.17 | riddlebox | I just got me a few free cisco 7940's and I ordered a power brick for one to test with first, it was only $20 on ebay |
| 02:06.06 | *** join/#asterisk seanmh (n=johndoe@70.90.202.94) |
| 02:08.18 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 02:09.00 | drmessano | jaytee: Lookin for me? |
| 02:09.22 | NovceGuru | KavanS: their cheap enough to give a try |
| 02:09.28 | NovceGuru | they dont have unlimited inbound in my area though :\ |
| 02:09.30 | NovceGuru | I blame drmessano |
| 02:09.39 | KavanS | anyone have any recommendations on voip provider? |
| 02:09.58 | KavanS | I had a list awhile back, but I've failed to find a "good enough" solution |
| 02:10.11 | jaytee | drmessano, yeah I was |
| 02:10.39 | jblack | If you don't care about your area code, you can get free unlimited from ipkall. |
| 02:10.39 | jaytee | drmessano, have you used the SPA8000 yet? |
| 02:11.16 | jblack | jaytee: I have one. |
| 02:11.33 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 02:12.11 | jaytee | jblack, do you have to use limitonpeers and call-limit=1 with them? I've seen two configs for Asterisk. One with and one without. |
| 02:12.47 | jblack | I don't have to do anything special with * with them. |
| 02:13.08 | drmessano | Its 8 different peers, IIRC, so no different than 8 phones or other devices, correct, jblack? |
| 02:13.11 | jblack | each port just registers as a separate sip device. I did limit them to two calls for call waiting purposes |
| 02:13.16 | drmessano | heh |
| 02:13.25 | Titanous | I've got a Aastra phone on a remote network (behind NAT), which registers to Asterisk server on my local network (also NAT'd). When I call the phone from my network it works great. When the remote phone tries to call Asterisk, the SIP works fine, but no RTP is ever sent or received by the Aastra phone. |
| 02:13.57 | jblack | I didn't even need insecure anything. Just forced them to ulaw. |
| 02:14.07 | jaytee | if I set call-limit=1 I can't conference if I do. I'm thinking of setting it to 2 for phones and 1 for the few fax machines I have on them. |
| 02:14.27 | Titanous | Any suggestions? (Already changed routers/firewalls, and all SIP providers can call me just fine) |
| 02:14.31 | jblack | You should be able to set the number of lines in the ATA itself, and not bother telling *. |
| 02:14.48 | jblack | but yeah, I'd do that on the ATA side. |
| 02:15.12 | jblack | fax machines don't care much for call waiting. |
| 02:16.08 | jblack | btw, I suggest after doing much configuration over http and savings settings, you power it down afterwards. |
| 02:16.09 | jaytee | drmessano, were you here earlier when I was describing my "incident" that happened this afternoon? |
| 02:16.32 | drmessano | No, dolphin get flogged? |
| 02:16.39 | jaytee | hahaha |
| 02:17.46 | jaytee | no, one of my coworkers, MCSE, CCNA and makes more money sets up a new network printer. Accidentally sets the same static IP as the Asterisk server. For 15 minutes no calls and all the calls in progress dropped due to RTP timeouts. |
| 02:17.58 | drmessano | HAHAHHAHAHHAHAH |
| 02:18.02 | NovceGuru | hahaha WOOPS |
| 02:18.06 | drmessano | Must Call Someone Else? |
| 02:18.41 | drmessano | Nice |
| 02:19.39 | jaytee | yeah, boss was none too happy |
| 02:19.54 | KavanS | lol |
| 02:19.56 | KavanS | jesus.. |
| 02:20.04 | jaytee | because one of the dropped calls was the CEO |
| 02:20.05 | KavanS | you can never protect the network enough from an MCSE |
| 02:20.08 | KavanS | oh nice. |
| 02:20.12 | drmessano | LOL |
| 02:20.19 | KavanS | I always love when the execs get involved over the stupid coworker |
| 02:20.28 | drmessano | CEO, as in, doesnt even work onsite, but was calling in? |
| 02:20.41 | jblack | kavans: That's why I suggested he run a parallel network for *. :) |
| 02:20.53 | KavanS | jblack: lol |
| 02:20.59 | jaytee | no, as in on site and using his phone to make an outside call |
| 02:21.10 | KavanS | jaytee: you should suggest your boss give you a raise and fire the MCSE |
| 02:21.30 | KavanS | "think of it as a 'network bailout'" |
| 02:21.41 | jaytee | KavanS, I'd have better luck getting my hair to grow back. |
| 02:21.54 | jblack | Teach him the wonders of cmd.exe and "ping" |
| 02:22.13 | jblack | "Look! You can see if an IP is being used BEFORE you set it. Amazing!" |
| 02:22.39 | KavanS | yeah lol |
| 02:22.49 | KavanS | certs can't do everything |
| 02:23.08 | jaytee | no, in his defense he wasn't trying to set it to the same address, he was trying for one that ended in .147 and ended up with .47 |
| 02:23.10 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 02:23.33 | jblack | well, thar's your problem. An mcse trying to type something, instead of clickign with the mouse. |
| 02:23.41 | NovceGuru | obviously he didnt try to test it then afterwards |
| 02:23.57 | NovceGuru | jblack: "sorry is there a GUI for ping?" |
| 02:24.19 | jblack | I bet you can find one if you look hard enough. :) |
| 02:24.27 | NovceGuru | angry ip scanner |
| 02:24.39 | jblack | Here you go. http://www.gping.com/ |
| 02:24.50 | NovceGuru | fancy! |
| 02:25.57 | KavanS | that's where MS fucked up... |
| 02:26.02 | KavanS | is made it "easy" to admin their systems |
| 02:26.27 | KavanS | because you cannot differentiate (even with certificate) motherfuckers who *know* what is going on....and those who *act* like they know what is going on |
| 02:26.39 | KavanS | I assume this will stir all sorts of "well it's easy to verify..." |
| 02:26.49 | jblack | heh. They made it so "easy" that I won't touch a foreign windows system with a ten foot pole. Just say "sorry. The only thing I know about windows is that they're easily broken" |
| 02:26.54 | jaytee | yep, it's that race to the bottom. The Lowest Common Denominator Rules!!!!! |
| 02:27.08 | KavanS | lol yes windows admins are the plague of IT |
| 02:29.04 | jaytee | it took one of our outside Apple Macintosh consultants (my boss refuses to have IT support Macs) a full day to upgrade Adobe Creative Suite on four Macs. Can you say "milking"? Sure you can! |
| 02:29.35 | NovceGuru | yeah popping that disk in and hitting upgrade sure is tough! |
| 02:29.48 | pdmmm | were they pirated ? |
| 02:29.55 | *** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net) |
| 02:29.55 | Micc | Anyone know of a good voip phone for a receptionist that can have 50+ programmable keys? |
| 02:29.58 | jaytee | then the horror story. It's taken our primary admin (MCSE) almost 7 months to deploy Microsoft Operations Manager |
| 02:30.11 | pdmmm | Micc: Cisco + 7914 sidecar! |
| 02:30.38 | riddlebox | pdmm, can you get the sip firmware for cisco phones? |
| 02:31.14 | pdmmm | riddlebox: which one are u after |
| 02:31.14 | jaytee | can't ya just buy a Linksys phone? I think they're the same damn thing with a different logo |
| 02:31.14 | pdmmm | my CCO login is busted |
| 02:31.14 | riddlebox | pdmm, 7940 |
| 02:31.14 | imcdona | is it possible with a .tel to create a records? or is it just a fancy domain to point to a website with your contact info? |
| 02:31.15 | pdmmm | like any version? |
| 02:31.49 | riddlebox | pdmm, latest if best, I registered for the login, but it wont let me download it |
| 02:32.04 | pdmmm | u need service contract |
| 02:32.15 | pdmmm | i dont have the latest |
| 02:32.28 | pdmmm | i can get it tomorrow from the network guys @ work |
| 02:32.32 | riddlebox | any version would be great, otherwise I will try to use skinny |
| 02:32.37 | pdmmm | ok |
| 02:32.38 | pdmmm | hold |
| 02:33.22 | NovceGuru | Micc: I think polycoms can do 50 keys |
| 02:34.01 | pdmmm | hm |
| 02:34.07 | Micc | NovceGuru, which one? the 670 and 650 with 3 expantions won't get that many. |
| 02:34.10 | NovceGuru | lies |
| 02:34.18 | NovceGuru | I lied I mean |
| 02:34.22 | pdmmm | riddlebox: so i cant find the original tar |
| 02:34.30 | pdmmm | u want my tftp dir :) |
| 02:34.36 | riddlebox | yeah cool! |
| 02:34.39 | pdmmm | ok hold |
| 02:34.45 | NovceGuru | 42 with 3 cars and 6 on the phone, so close! |
| 02:35.02 | pdmmm | lemme clean out the other bs firmware shit in it |
| 02:35.09 | riddlebox | ok cool |
| 02:35.38 | NovceGuru | Micc: at that point i'd probably look at something like a pc based gui |
| 02:35.59 | NovceGuru | like the flash operator panel or someting |
| 02:36.16 | Micc | NovceGuru, or something like just parking call appearance. |
| 02:36.20 | NovceGuru | or astassistanc |
| 02:36.26 | NovceGuru | astassistant |
| 02:36.41 | Micc | yeah, maybe a nice web gui would be the best. |
| 02:37.01 | NovceGuru | drag/drop calls is nifty |
| 02:37.11 | NovceGuru | switchvox has a nifty interface but is pricey |
| 02:37.37 | Micc | NovceGuru, how many lines will it handle? switchvox that is. |
| 02:38.39 | NovceGuru | I think they have a system that'll do 100 concurerent calls |
| 02:38.45 | NovceGuru | I dont know if they actually limit the calls |
| 02:41.03 | [TK]D-Fender | Aastra wins on mass-presence consoles... |
| 02:41.14 | pdmmm | riddlebox: how u wanna download it |
| 02:42.03 | NovceGuru | [TK]D-Fender: yeah deff |
| 02:42.10 | pdmmm | riddlebox: i'm uploading to my colo |
| 02:42.18 | pdmmm | it'll be a min, its coming off my t1 |
| 02:42.23 | riddlebox | pdmm, I can download it anyway you want |
| 02:43.13 | pdmmm | 3 minutes |
| 02:43.31 | pdmmm | u should be able to use the whole tftp dir |
| 02:43.34 | riddlebox | cool thanks, I am still going to have to wait a week or so to get my first power brick |
| 02:43.35 | pdmmm | as is |
| 02:43.56 | pdmmm | its got the firmware for 7960/40, 7961, 7921 |
| 02:44.09 | NovceGuru | h8 cisco sccp2sip |
| 02:44.09 | pdmmm | the 7921 == SCCP |
| 02:44.12 | pdmmm | i fucking hate it |
| 02:44.24 | pdmmm | it uses embedded linux too |
| 02:44.36 | pdmmm | but cisco dont care |
| 02:45.08 | NovceGuru | Micc: http://www.asternic.org/demo.html |
| 02:45.10 | NovceGuru | wooh! |
| 02:45.36 | NovceGuru | clunky! |
| 02:46.24 | pdmmm | riddlebox: http://unf.net/riddlebox.tar.gz |
| 02:48.01 | NovceGuru | KavanS: heh, what was your sip server set to for vp 3 days ago? |
| 02:48.18 | riddlebox | pdmm, I just drop that 7940 into my tftp server then? |
| 02:48.22 | KavanS | NovceGuru: what do you mean? I use sjc-primary |
| 02:48.49 | NovceGuru | ok, because 3 days ago they disable connect01/02/03 outbound calling |
| 02:48.51 | pdmmm | riddlebox: the whole tarball |
| 02:48.54 | pdmmm | ya |
| 02:48.59 | pdmmm | its set to flash |
| 02:50.03 | pdmmm | to sip |
| 02:50.13 | pdmmm | find me when u do it |
| 02:50.18 | pdmmm | brb |
| 02:50.24 | riddlebox | cool, is it still better to use sip or is skinny better with asterisk |
| 02:50.36 | riddlebox | pdmm, can I get your email, it will be a week or so |
| 02:51.57 | Micc | NovceGuru, is this free? |
| 02:53.31 | *** join/#asterisk intralanman (n=intralan@99-196-39-200.cust.wildblue.net) |
| 02:53.58 | Micc | NovceGuru, I think I looked at this before, does it work for multiple customers on a single asterisk box? I think I remember it only works for one customer on one box. |
| 02:54.45 | NovceGuru | I think multiple, never used it |
| 02:56.30 | NovceGuru | Micc: if you want something supported, http://www.i9technologies.com/index.php?option=com_content&task=view&id=19&Itemid=40 |
| 02:58.37 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 03:01.48 | Micc | yeah I like isymphony. |
| 03:07.38 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 03:12.34 | *** join/#asterisk Nasra (n=maxshipp@201.229.245.231) |
| 03:14.08 | dan__t | Ok. |
| 03:14.12 | dan__t | I like FastAGI + xinetd |
| 03:14.17 | *** join/#asterisk BadHAL (n=nn@66.194.174.31) |
| 03:16.45 | *** join/#asterisk Assimilate (n=Assimila@72-24-150-105.cpe.cableone.net) |
| 03:28.20 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 03:34.41 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 03:34.42 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:34.49 | *** join/#asterisk tAnkOSX (i=tank@the.matrix.has-you.net) |
| 03:35.34 | *** join/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 03:37.31 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 03:45.43 | *** join/#asterisk jblack (n=jblack@71.181.164.209) |
| 03:46.01 | jblack | Oh man, the ipkall # for the #asterisk conference expired because _nobody_ called. :( |
| 03:51.17 | *** part/#asterisk eldorel (n=eldorel@ip68-108-238-144.br.br.cox.net) |
| 03:52.20 | drmessano | I would have, but nobody likes me. :( |
| 03:52.25 | drmessano | :`( |
| 03:52.31 | drmessano | :````( |
| 03:52.51 | drmessano | Im fresh out of emoticons |
| 03:54.58 | dan__t | haha |
| 03:55.05 | jblack | Since when does a closet troll care he's unloved?!?? |
| 03:55.15 | dan__t | Mar 9 20:32:47 mail named[573]: FORMERR resolving 'pt4.successfuldiscriminating.com/AAAA/IN': 124.207.41.201#53 |
| 03:55.52 | jaytee | awww, I like ya Danny! you always manage to make me laugh when I need to. |
| 03:56.17 | drmessano | lol |
| 03:56.21 | jblack | I love teh dr too. He keeps things interesting. |
| 03:56.35 | jaytee | and he's pretty sharp on the tech side too |
| 03:56.36 | drmessano | Im not a closet troll, just dont like people and I think they're complete and total morons |
| 03:56.48 | jaytee | we are! |
| 03:56.49 | drmessano | Wait.. |
| 03:56.54 | jblack | watches water proclaim that it's not wet |
| 03:56.58 | drmessano | I do have this bridge |
| 03:57.14 | jblack | it's ok. I'm a troll too. |
| 03:57.28 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 03:57.52 | drmessano | My goal in life is to make people think.. Biggest pet peeve is sheep, or helpless little vampires |
| 03:57.57 | drmessano | People can do better |
| 03:58.01 | jaytee | I'm short, stocky build and hairy all over except the top of my head. Does that count? |
| 03:58.20 | jblack | jaytee: Bah. You're not a troll. |
| 03:58.40 | jaytee | or do I have to do stuff like go into #gentoo and start singing the praises of OpenSUSE |
| 03:59.10 | drmessano | jaytee: It depends |
| 03:59.21 | jblack | See? If you were a troll, you'd go into #gentoo and ask how to emerge hourly builds of firefox. |
| 03:59.27 | drmessano | Find 10 ways OpenSuse is better than Gentoo |
| 03:59.32 | drmessano | and you're gold |
| 03:59.43 | drmessano | Go in there and yell Gentoo sucks, and youre a worthless troll |
| 03:59.44 | jblack | That could provide hours of enjoyment about "making sure your browser is truly secure" |
| 04:00.34 | drmessano | For example, I love pissing people off by having honest discussions about Firefox's shortcoming, since clearly, it has none |
| 04:01.15 | [TK]D-Fender | drmessano: FINALLY, someone else see it my way! |
| 04:01.17 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 04:01.26 | [TK]D-Fender | evangelizes |
| 04:01.32 | drmessano | I think it pisses people off to think |
| 04:01.33 | jaytee | I started playing YoVille, the Lame edition of Second (I have no fucking) Life on Facebook. |
| 04:01.46 | drmessano | or to face things they dont want to face |
| 04:01.52 | drmessano | Like Ubuntu isnt perfect |
| 04:02.01 | drmessano | Firefox 3 crashes like crazy |
| 04:02.03 | jaytee | it isn't? |
| 04:02.08 | [TK]D-Fender | drmessano: Denial.... it isn't just a river in Egyt you know... |
| 04:02.09 | jblack | [TK]D-Fender: I'd marry you if I thought you legs would look better in a skirt. |
| 04:02.15 | [TK]D-Fender | Egypt* |
| 04:02.25 | jaytee | oh, shit. Mark Shuttleworth will be devastated when he hears about this! |
| 04:02.35 | jblack | Screw him. |
| 04:02.40 | jaytee | not my type |
| 04:02.49 | drmessano | clubs drmessano with the Ubuntu gestapo and removes all traces of his existance |
| 04:02.50 | drmessano | Wait.. |
| 04:02.51 | jblack | Why? Got something against assholes? |
| 04:03.26 | jaytee | now Carl Weathers on the other hand....you know....the guy who played Apollo Creed? That's a whole nother story.....wait! better not go there! |
| 04:03.40 | [TK]D-Fender | YO ADDDDDDDDDRRRRRRRRRRRRIIIIIIIIIIIIIIIIIIIIIAAAAAAAAAAAAAAAAAAAAAANNNNNNNNNNNNNNNNNN!!!!!!!!!!!!!!!! |
| 04:03.47 | drmessano | Hell no |
| 04:03.54 | drmessano | Fuck Apollo, DUKE was THE MAN |
| 04:03.59 | drmessano | Duke was badass |
| 04:04.04 | jblack | Today, I made a list of the 30 things I need to get done, did 3 of 'em, and said "screw this" |
| 04:04.06 | jaytee | Duke? |
| 04:04.17 | jblack | Dude! |
| 04:04.19 | jaytee | the trainer? |
| 04:04.23 | drmessano | Apollos trainer.. He trained Rocky in Rocky IV |
| 04:04.30 | jaytee | oh, yeah! |
| 04:04.38 | drmessano | Rocky IV.. The best Rocky |
| 04:04.52 | drmessano | Drago should have won.. He was clearly better prepared |
| 04:05.18 | jblack | That was the russian one, right? |
| 04:05.27 | drmessano | He could punch with 4x the force of a normal man, yet, Rockys weak italian spine survived all those head jabs |
| 04:05.33 | drmessano | Yes |
| 04:05.33 | jblack | I must break vou. |
| 04:05.38 | drmessano | IF HE DIES |
| 04:05.40 | drmessano | HE DIES |
| 04:05.43 | drmessano | Love the dialog |
| 04:06.08 | drmessano | His trainer dude.. |
| 04:06.08 | jblack | You're just looking at it wrong. rocky is a love story. |
| 04:06.12 | drmessano | WHATEVER HE HITS... |
| 04:06.18 | drmessano | ... HE DESTROYS |
| 04:07.10 | jaytee | when you said Duke I thought at first you meant Bill Duke who played Cooke in Commando. "We'll take Cooke's car, he won't be needing it." |
| 04:07.39 | drmessano | That was back in the Georgio Moroder synthesizer soundtrack days.. when the whole computer-age thing was this wild, untamed, whimsical thing that no one understood, but it would change the world |
| 04:07.43 | drmessano | LOL |
| 04:08.23 | drmessano | In 1984, you could cast a computer in a movie and write a script around it |
| 04:08.27 | drmessano | Suck it, Steve Jobs |
| 04:08.45 | drmessano | "Make it look like a sunflower" |
| 04:09.02 | drmessano | They made it look like a trashcan on top of a typewriter and it won Oscars |
| 04:10.17 | drmessano | I think im still that stupid little kid with the commodore 64 who thought Wargames was the coolest movie ever |
| 04:10.40 | drmessano | Hmm... Kid, check.. Stupid, check... C64 in the closet, check.. |
| 04:10.42 | jaytee | I love some of the lines in Hackers |
| 04:10.44 | drmessano | yep... |
| 04:10.50 | drmessano | Hackers, lol |
| 04:10.55 | drmessano | Hackin the gibson |
| 04:11.05 | jaytee | it's so lame it's funny |
| 04:11.44 | drmessano | Im sure I will look back on 24 that way years from now |
| 04:12.01 | jaytee | haha |
| 04:12.06 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:13.25 | drmessano | But right now, 24 rocks balls |
| 04:13.35 | jaytee | what about the movie, Electric Dreams? |
| 04:13.44 | drmessano | Thats my fav movie of all time |
| 04:13.49 | drmessano | I WANT... TO MEET HER |
| 04:13.51 | drmessano | MOLES |
| 04:14.03 | jaytee | lemme guess, Virginia Madsen does it for you? |
| 04:14.13 | drmessano | Oh god yes.. shes never been sexier |
| 04:14.31 | drmessano | Check this out |
| 04:14.57 | jaytee | Princess Irulan in Dune, (the shitty version, not the SCI FI remake that was better) |
| 04:15.02 | drmessano | I have two DVDs of the movie, bootlegs, since its never been released on DVD, an offical WB VCD from Singapore, and the soundtrack, imported from Australia |
| 04:15.37 | drmessano | Somewhere I have a giant PNG of the movie poster, for a trip to Kinkos one day |
| 04:15.52 | drmessano | Also have a WMA of the trailer |
| 04:15.53 | drmessano | lol |
| 04:16.32 | jaytee | it took me forever to get Rain Man on DVD due to a rights ownership dispute, same for A Razor's Edge with Bill Murray and Farewell to the King with Nick Nolte. Good underrated flicks but lousy distribution. |
| 04:17.33 | drmessano | Theres a petition somewhere with like 70,000 signatures to get Electric Dreams on DVD |
| 04:17.38 | drmessano | Im sure no one reads those things |
| 04:17.42 | drmessano | But still |
| 04:18.15 | drmessano | Both of the DVDs I have, which are different packaging completely, have the same picture thats about 2 degrees tilted |
| 04:18.29 | drmessano | Same rip I am sure |
| 04:20.58 | jaytee | I wonder what it is about www.imdb.com that maxes out my cpu and turns Firefox into a boat anchor? |
| 04:29.16 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-ea9fa5fe887a41cd) |
| 04:29.43 | drmessano | Firefox |
| 04:29.57 | jaytee | probably |
| 04:30.39 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 04:30.51 | jaytee | that and this box has been up 11 days without a reboot |
| 04:31.44 | jaytee | brb |
| 04:32.11 | dan__t | haha, boat anchor |
| 04:39.31 | *** join/#asterisk luke-jr (n=luke-jr@2002:46bb:1a76:0:20e:a6ff:fec4:4e5d) |
| 04:40.05 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 04:40.52 | [TK]D-Fender | dan__t: http://www.youtube.com/watch?v=RCOlf_7BT-I @ 1:38 |
| 04:40.57 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 04:41.20 | [TK]D-Fender | jaytee: Enjoy the link as well |
| 04:42.29 | jaytee | has anyone seen the FedEx "alternate ending to Castaway" video clip? |
| 04:43.25 | *** join/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 04:43.48 | jaytee | hahahahaa |
| 04:44.24 | jaytee | heheheheeheeeeeeeeeeee |
| 04:44.51 | [TK]D-Fender | jaytee: Oldie, but a goodie |
| 04:48.50 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 04:50.20 | jaytee | http://www.youtube.com/watch?v=x7PhJp3ciRQ&feature=related |
| 04:54.26 | [TK]D-Fender | jaytee: Yup, saved that one about a year ago :) |
| 04:54.44 | *** join/#asterisk Gopaul (n=chatzill@61.17.185.118) |
| 04:54.47 | jaytee | still funny |
| 04:55.04 | Gopaul | how to get the cdr event using java? |
| 04:56.24 | drmessano | HAHAHAH |
| 05:00.13 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 05:00.36 | carrar | better: http://www.youtube.com/watch?v=WlexVymRzk4&feature=related |
| 05:11.54 | drmessano | [TK]D-Fender |
| 05:12.11 | drmessano | You following this mess in the other channel? |
| 05:13.14 | [TK]D-Fender | drmessano: Oh God... he's still going at it |
| 05:13.40 | drmessano | Hes been pwn3d |
| 05:13.52 | drmessano | 50000 open ports cant be wrong |
| 05:17.10 | *** join/#asterisk Gopaul_ (n=chatzill@61.17.185.118) |
| 05:18.00 | [TK]D-Fender | drmessano: Just F-ing regurgitate his head, piss on it and move on.... |
| 05:18.28 | drmessano | lol |
| 05:25.45 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 05:27.38 | [TK]D-Fender | bed time. Later all |
| 05:37.51 | *** join/#asterisk Putzz (n=me@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 05:41.35 | *** join/#asterisk thelordmortis (n=lordmort@203.8.160.250) |
| 05:51.40 | *** join/#asterisk eject_ck (n=eject_ck@85.223.182.86) |
| 05:51.44 | eject_ck | Hi all |
| 05:52.49 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 05:56.03 | eject_ck | I have configured SPA-3102 with Asterisk (very simple config) where I have 2 sip accounts in sip.conf ( 100 for SPA and 101 for SIP client on my PC) with default extention. My SPA connected to PSTN and I can use it as VoIP->PSTN gateway, so when I make call to 100 and then dial number - I'm going to PSTN and voila. Now I'm playing with extentions.conf, and I need suggestion how to deal with this task. Should I use extensions.conf only to do it ? |
| 05:56.06 | eject_ck | Thank you! |
| 06:00.27 | *** join/#asterisk Gopaul (n=chatzill@61.17.185.118) |
| 06:01.16 | *** join/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 06:04.45 | eject_ck | or maybe I need do additional config on SPA ? |
| 06:15.55 | eject_ck | can anybody help to build dialplan? I need something like this exten => _80XXXXXXXXX,1,Dial(SIP/101/${EXTEN},12,tr) |
| 06:16.10 | eject_ck | but this not works for me :( |
| 06:16.57 | eject_ck | When I call to 101 from my 102 sip phone I need dial PSTN phone dumber which Is _80XXXXXXXXX |
| 06:19.39 | _BBV_ | exten => _80XXXXXXXXX,1,Dial(SIP/101,12,trd(${EXTEN})) |
| 06:19.42 | *** join/#asterisk Un1x_ (n=Un1x@CPE000024cb557c-CM00125572fe2a.cpe.net.cable.rogers.com) |
| 06:19.44 | _BBV_ | try this |
| 06:19.51 | eject_ck | thanks, trying |
| 06:20.09 | *** join/#asterisk Gopaul__ (n=chatzill@61.17.185.118) |
| 06:21.13 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 06:21.46 | eject_ck | _BBV_: same :( |
| 06:21.58 | _BBV_ | sorry big D |
| 06:22.00 | _BBV_ | exten => _80XXXXXXXXX,1,Dial(SIP/101,12,trD(${EXTEN})) |
| 06:22.43 | eject_ck | trying ... |
| 06:22.53 | eject_ck | great!!! |
| 06:22.58 | eject_ck | Works! Alot of thanks! |
| 06:23.20 | _BBV_ | :) |
| 06:23.31 | eject_ck | How can I allow use this exten only for 102 ? |
| 06:23.49 | eject_ck | e.g. only SIP account 102 can use this ext |
| 06:24.19 | _BBV_ | context |
| 06:24.48 | eject_ck | thanks! |
| 06:25.22 | _BBV_ | and set users 2 different context |
| 06:27.42 | eject_ck | I need call to only one direction - 103 -> 101, so I changed conext for 103 to custom and moved just added dialplan to custom context |
| 06:27.47 | eject_ck | Thanks a log again! |
| 06:31.10 | _BBV_ | ^) |
| 06:34.07 | eject_ck | is it possible to set call on hold from asterisk console ? |
| 06:34.10 | eject_ck | sip channel |
| 06:35.03 | eject_ck | I have established call and I need play moh for each end :) |
| 06:35.11 | eject_ck | when I need |
| 06:43.37 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 06:46.29 | Gopaul__ | how to get the cdr event with java? |
| 07:04.56 | *** join/#asterisk Gopaul (n=chatzill@61.17.185.118) |
| 07:04.57 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 07:06.06 | *** join/#asterisk eject_ck (n=eject_ck@85.223.182.86) |
| 07:09.55 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 07:13.32 | eject_ck | where I can read about asterisk server and firewall, I need port(s) to open ? |
| 07:14.14 | thelordmortis | what protocols are you using? |
| 07:14.31 | eject_ck | rtpstart=10000 rtpend=20000 ? |
| 07:14.32 | thelordmortis | (from asterisk to the people outside the firewall) |
| 07:14.48 | eject_ck | I'm using SIP only |
| 07:14.50 | thelordmortis | so you are using SIP ? |
| 07:14.56 | eject_ck | yes |
| 07:15.00 | thelordmortis | then yeah, the RTP ports and the SIP port itself |
| 07:15.13 | eject_ck | I need allow connections for external cusers |
| 07:15.17 | eject_ck | users |
| 07:15.18 | thelordmortis | i don't know what the SIP communications port is off the top of my head |
| 07:15.56 | eject_ck | so UDP 5060 and range which Is defined in rtp.conf |
| 07:18.43 | eject_ck | hm, where I should define rtp ports range ... I don`t see rtp.conf in config dir |
| 07:20.09 | *** part/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 07:20.39 | *** join/#asterisk jsgoecke (n=jsgoecke@c-67-180-103-93.hsd1.ca.comcast.net) |
| 07:21.55 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:28.07 | *** join/#asterisk xrmx__ (n=rm@host128-22-dynamic.15-87-r.retail.telecomitalia.it) |
| 07:35.48 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 07:43.27 | dude7064 | Why is there a need for a service provider ? Is it possible to do everything on my server ? |
| 07:44.07 | dude7064 | where can I learn more about this service provider issue ? and if possible , let my server handle the work they do ... |
| 07:45.45 | dude7064 | ~service provider |
| 07:46.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:56.14 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:58.14 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 08:00.26 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 08:05.44 | imcdona | I see that FreePBX adds "record_out=Always" in the sip.conf for a phone where recording is enabled. Why doesn't asterisk choke on this. "record_out=Never" is not a valid configuration option |
| 08:05.45 | dude7064 | I'm still stick on this service provider issue |
| 08:06.03 | dude7064 | could somebody please help ? |
| 08:06.14 | dude7064 | I am running AsteriskNow for the first time, |
| 08:06.16 | imcdona | whats going on dude? |
| 08:06.33 | dude7064 | Running the config of AsteriskNow for the first time, |
| 08:06.39 | dude7064 | I don't have hardware |
| 08:06.55 | dude7064 | need to make simple test calls to expirment with Asterisk |
| 08:07.10 | dude7064 | One of the first configuration pages is asking for a service proivder |
| 08:07.31 | imcdona | so you want to make some test calls between two extensions? |
| 08:07.33 | dude7064 | most of those i found who are service providers require payment of some sort |
| 08:07.50 | dude7064 | yes, on the same pc, or different PCs |
| 08:08.13 | imcdona | you can skip adding a provider and just add two extensions |
| 08:08.32 | dude7064 | I have my own server, why would I need a service provider ? this is something I still don't understand.. |
| 08:08.40 | imcdona | add two sip extensions then install xlite on both pcs |
| 08:08.57 | imcdona | you only need a sip provider if you want to place calls to the PSTN. |
| 08:09.02 | dude7064 | xlite doesn't come with AsteriskNow ? |
| 08:09.03 | mchou | you need a service provider only if you need to call pstn phone numbers |
| 08:09.06 | imcdona | like someones cell phone or home phone for example |
| 08:09.27 | dude7064 | I see |
| 08:09.35 | mchou | because pstn phones arent "ip" |
| 08:10.20 | dude7064 | I have 10 pcs in one country and 10 others in another country.. and I want them to commuincate with each other directly |
| 08:10.38 | dude7064 | i guess in this case I won't be needing a service provider, will I ? |
| 08:10.48 | mchou | lol |
| 08:11.03 | dude7064 | sorry, I guess I misinterpreted the situation |
| 08:11.27 | dude7064 | I own 10 pcs in one country and 10 others in another |
| 08:11.55 | mchou | ip doesnt care which countries they are in |
| 08:11.59 | dude7064 | goal is to initate calls from pcs in first country to some individuals phone lines/mobile phones in the other country |
| 08:12.14 | mchou | could be right next to each other for all they care |
| 08:12.42 | dude7064 | i have 10 pcs connected to the internet at the same time in the other country (where callees reside),, |
| 08:12.43 | mchou | dude7064: which part of PSTN "dont" you understand? |
| 08:13.27 | dude7064 | now the idea is , it is cheaper if I had my physical phone lines connected to the pcs in the other country to route the calls than using a service provider |
| 08:13.39 | mchou | if you are calling a PSTN phone number you will need ISTP (unless you want to become one yourself) |
| 08:15.00 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:15.01 | dude7064 | Is that hard to achieve ? |
| 08:15.08 | dude7064 | where can I learn more about this ? |
| 08:15.29 | mchou | ~book |
| 08:15.38 | jbot | [book] probably 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://tfot.leifmadsen.com or see ~buybook |
| 08:16.13 | mchou | ~itsp |
| 08:16.14 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 08:18.11 | dude7064 | mchnou> can I ask you a quetion in private ? |
| 08:18.20 | mchou | just ask here |
| 08:19.11 | dude7064 | I am working on establishing a call center in a country to serve companies in another country,, |
| 08:19.41 | dude7064 | Is there a simple application I can simply install and run, , and hire somebody to maintain ? |
| 08:19.43 | mchou | yeah right |
| 08:19.59 | mchou | you sound like a telemarketer |
| 08:20.13 | stintel | burn! |
| 08:20.14 | dude7064 | It'll be for accepting support calls |
| 08:20.18 | dude7064 | incoming calls |
| 08:20.27 | mchou | lol |
| 08:20.41 | mchou | outsourcing our jobs to india? |
| 08:21.00 | stintel | thats already happening too much |
| 08:21.15 | mchou | anyway, it aint an "application" |
| 08:21.20 | thelordmortis | ~itsplist-us |
| 08:21.20 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 08:21.35 | dude7064 | mchnou> I'm not talking about the US here |
| 08:21.40 | mchou | the industry buzzword is "solution" |
| 08:21.41 | *** join/#asterisk CrazyTux (n=brandon@c-98-196-6-54.hsd1.tx.comcast.net) |
| 08:22.33 | dude7064 | The countries I will be serving are not in Europe or USA |
| 08:22.34 | CrazyTux | Hey guys has anyone ever had an issue with delay when playing an audio file via Background() and receiving DTMF input for an appropriate extension? It seems to only occur of a specific audio file, I checked and compared file size etc against the other similar audio files, and they are all about the same size ~392Kb any ideas is appreciated. |
| 08:23.40 | mchou | CrazyTux: yeah, there's weird behavior |
| 08:24.05 | CrazyTux | mchou: Yea very weird... any thoughts? |
| 08:24.23 | mchou | CrazyTux: I dunno the root cause by preety much just gonna have to work around it |
| 08:24.27 | CrazyTux | It literally has about a 5-6 second delay before continue and "re-acting" to the DTMF |
| 08:24.39 | tzafrir_laptop | Anybody wants to build a fax SignatureFilter? http://article.gmane.org/gmane.linux.region.israel/36371 |
| 08:24.55 | CrazyTux | mchou: I sort of need to get this worked out... |
| 08:25.11 | mchou | CrazyTux: I cant help you with that |
| 08:25.46 | mchou | CrazyTux: just reencode that file and see if it happens again |
| 08:26.08 | *** join/#asterisk Great_Anta_Baka (i=c4219f53@gateway/web/ajax/mibbit.com/x-1317ce4c628fc424) |
| 08:27.40 | CrazyTux | mchou: already tried... |
| 08:28.03 | mchou | CrazyTux: well, I dunno what to tell ya |
| 08:28.16 | Great_Anta_Baka | i think its time to leave my job at my "cutting edge" "voip" company when they decide to take out the voip solution and put in a purely anolog pbx, "because digital doesnt work" |
| 08:28.19 | thelordmortis | what format is the audio file? |
| 08:31.01 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 08:31.03 | mchou | Great_Anta_Baka: that's interesting |
| 08:31.31 | CrazyTux | thelordmortis: .wav |
| 08:31.37 | mchou | Great_Anta_Baka: maybe they were using cisco proprietary junk :) |
| 08:32.02 | Great_Anta_Baka | no its not.. the project manager's only experience in IT is being a cable boy... its a shame.. they had sucha good model... this guy is retarted |
| 08:32.02 | Great_Anta_Baka | na |
| 08:32.02 | Great_Anta_Baka | it was asterisk solution |
| 08:32.12 | mchou | well, asterisk aint exactly scalable |
| 08:32.24 | mchou | cant be all things to all people |
| 08:32.27 | Great_Anta_Baka | wasnt about being scalable |
| 08:32.38 | Great_Anta_Baka | and its more scalable than the piece of crap they putting in |
| 08:32.57 | Great_Anta_Baka | http://www.hybrexsa.com/item.php?prodname=12 |
| 08:33.05 | Great_Anta_Baka | thats what they putting in.. i saw the thing |
| 08:33.17 | mchou | how could a cable boy be an advocate of analog PBX? |
| 08:33.26 | *** join/#asterisk TheMaestro (n=ymax@203.126.136.142) |
| 08:33.30 | Great_Anta_Baka | he is a damn good talker |
| 08:33.36 | Great_Anta_Baka | thats how he got the project manager job |
| 08:33.43 | CrazyTux | mchou: thelordmortis: http://rafb.net/p/94MqsV37.html sox stat, file that does not work (on top) and the one that does work (bottom) |
| 08:33.50 | CrazyTux | Great_Anta_Baka: :) lol |
| 08:33.51 | Great_Anta_Baka | and now supercedes all the more qualified ppl in the company |
| 08:33.57 | Great_Anta_Baka | :( |
| 08:34.00 | mchou | Great_Anta_Baka: oh lord |
| 08:34.09 | TheMaestro | hi..how do i get a cisco 7940 phone to work with asterisk? the phone has already been configured for SIP Calls |
| 08:34.25 | mchou | Great_Anta_Baka: someone needs to fire the dude that hired cable boy |
| 08:34.30 | thelordmortis | CrazyTux: that looks really strange! should both be the same |
| 08:34.36 | Great_Anta_Baka | hahaha.. it was the director :( :( |
| 08:34.50 | thelordmortis | CrazyTux: maybe file a bug if you figure out what exactly causes it? :P |
| 08:34.57 | thelordmortis | (as in, what situations) |
| 08:34.58 | CrazyTux | thelordmortis: will do |
| 08:35.28 | Great_Anta_Baka | and using an analogue pbx is kinda out of the question when you are managing over 20 different residential sites.. some of them hotels.. etc.. big big sites |
| 08:35.44 | mchou | Great_Anta_Baka: project manager's dont get to decide anything |
| 08:35.54 | Great_Anta_Baka | they do here |
| 08:35.58 | Great_Anta_Baka | i cant understand it |
| 08:36.01 | mchou | Great_Anta_Baka: they are implementors |
| 08:36.19 | Great_Anta_Baka | i really cant... its beyond me... i am not resigning my contracty |
| 08:36.37 | mchou | Great_Anta_Baka: you are a contractor? |
| 08:36.40 | mchou | lol |
| 08:36.44 | mchou | that's why |
| 08:36.54 | Great_Anta_Baka | well.. the rest of the people here arent |
| 08:37.05 | mchou | you are not in a position of influence |
| 08:37.12 | Great_Anta_Baka | i came here to take over their asterisk.. but after seeing this ... i am not going to do anything here |
| 08:37.20 | mchou | which is why cable boy gets to decide |
| 08:37.25 | Great_Anta_Baka | no |
| 08:37.45 | Great_Anta_Baka | you dont get it,.. there are more qualified people working here.. but somehow they are all doing DESKTOP support |
| 08:37.48 | Great_Anta_Baka | its a shame. |
| 08:38.17 | mchou | Great_Anta_Baka: then people at the top dont know what they are doing |
| 08:38.23 | Great_Anta_Baka | indeed.. |
| 08:38.34 | mchou | Great_Anta_Baka: which means you should leave regardless |
| 08:39.07 | mchou | Great_Anta_Baka: cause talent is too precious to waste on dopes |
| 08:39.57 | Great_Anta_Baka | the thing is.. these guys are part of the biggest telecommunications provider in this country.. if they do things like this here.. it makes me sad to think what else is out there |
| 08:40.10 | mchou | digital stuff doesnt work. that's rich |
| 08:40.13 | angryuser | mchou, the people on the top dont know what they do in detail yes, i think sometimes it is a communication problem |
| 08:40.31 | *** join/#asterisk Nobbie (n=chatzill@196.13.113.1) |
| 08:41.06 | mchou | angryuser: maybe. but if people on top cant utilize ppl to even 50% potential, they cant be good business ppl |
| 08:41.21 | Great_Anta_Baka | na.. its just this one guy.. his latest decision was to outsource the whole technical department and we will just do the faults... problem is the guy called me yesterday to ask me to help him set up a sip phone |
| 08:41.34 | Great_Anta_Baka | the oursourced guy |
| 08:41.36 | mchou | angryuser: cause by definition that's "bleeding money" |
| 08:41.38 | Great_Anta_Baka | out |
| 08:42.28 | mchou | well cisco sip phones are tough to provision :) |
| 08:42.46 | angryuser | mchou, all is for bleeding money anyway |
| 08:42.52 | Great_Anta_Baka | no.. i mean.. a crappy little normal sip phone with a web interface |
| 08:43.12 | Great_Anta_Baka | and they dont provision phones here |
| 08:43.27 | mchou | angryuser: I dont understand what you mean |
| 08:44.00 | mchou | angryuser: they objective is to bleed less than the other guy to remain competitive |
| 08:44.03 | angryuser | mchou, i mean at the end it is allways the question of $$ ;) |
| 08:44.09 | Great_Anta_Baka | i have seen one snom... and the rest are all no name brands lol... normally the box just says.. high performance voip phone |
| 08:44.29 | Great_Anta_Baka | well this guy is getting more money than the whole technical department was o_0 |
| 08:44.46 | mchou | Great_Anta_Baka: maybe that's why digital "doesnt work" :) |
| 08:44.55 | Great_Anta_Baka | hahaha |
| 08:45.16 | mchou | Great_Anta_Baka: I aint laughing |
| 08:45.19 | angryuser | i want to double my salary toooooo ;) |
| 08:45.49 | Great_Anta_Baka | haha |
| 08:47.01 | *** join/#asterisk sergey (n=sergey@sergey-home.iks.ru) |
| 08:52.29 | TheMaestro | guys..does anyone know how to configure a sip enabled cisco phone with asterisk? |
| 08:52.49 | mchou | TheMaestro: you dont |
| 08:53.03 | TheMaestro | mchou, eh? |
| 08:53.04 | mchou | TheMaestro: you use ftp :) |
| 08:53.23 | mchou | or tftp |
| 08:53.34 | TheMaestro | mchou, for configuration? |
| 08:53.48 | mchou | yup |
| 08:53.48 | TheMaestro | mchou, my requirement is to run sip calls over asterisk |
| 08:53.58 | mchou | so? |
| 08:54.39 | TheMaestro | okay i have no idea wht tftp is...can you help me out? |
| 08:54.53 | mchou | go read the cisco manuals |
| 08:55.12 | mchou | tells you all about it |
| 08:55.53 | mchou | google works too |
| 08:58.19 | TheMaestro | mchou, hmm.ok.thanks |
| 08:59.31 | TheMaestro | one more thing...i need to enable call transfer for my sip channels. Now my phones have the feature enabled with a # code. But i get a 603 Policy decline from asterisk when i attempt the blind call transfer |
| 08:59.59 | TheMaestro | i added t option in the Dial function in extensions.conf |
| 09:00.09 | mchou | check features.conf |
| 09:00.20 | mchou | you have conflict there most likely |
| 09:00.46 | mchou | cisco phones have a dedicated transfer butoon. |
| 09:00.51 | mchou | button* |
| 09:01.06 | mchou | no need to use # whatever |
| 09:01.11 | TheMaestro | err..im actually using a different device for this feature |
| 09:01.23 | TheMaestro | analog phone connected to a WiMAX CPE |
| 09:04.23 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
| 09:10.39 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 09:18.32 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:18.36 | TheMaestro | okay..im getting a 603 Declined (policy) response for a REFER request for call transfer...it seems as if asterisk is disallowing call transfers |
| 09:30.52 | *** join/#asterisk ingenius (n=alektro@host2.190-31-177.telecom.net.ar) |
| 09:45.36 | TheMaestro | how do i enable sip call transfers with asterisk? |
| 09:48.45 | angryuser | TheMaestro, it is enabled by default , i dont even know how to forbid some sip peers to used the features defined in features.conf |
| 09:49.42 | TheMaestro | angryuser, it aint working in my case...im getting a 603 policy declined response for a REFER request |
| 09:50.35 | angryuser | TheMaestro, well have you got another phone where this feature works ? Compare the config |
| 09:50.57 | angryuser | TheMaestro, and compare the sip traces it is important |
| 09:51.24 | dude7064 | anybody here is operating a calling center ? |
| 09:53.57 | TheMaestro | angryuser, no its not working on any endpoint |
| 09:54.26 | angryuser | dude7064, yep |
| 09:54.39 | *** join/#asterisk Weazelon (n=deazel@mail2.tikalnetworks.com) |
| 09:55.28 | dude7064 | angryuser: Are you/or have you had experience operating a calling center ? |
| 09:55.59 | Weazelon | yes |
| 09:56.09 | norgs | TheMaestro: What context are these sip channels defined in? |
| 09:56.24 | dude7064 | would it be possible to give me a paid course into the fundamentals of everything ? |
| 09:56.30 | angryuser | dude7064, i am managing the strategic point who is destributing the calls to 5 call centers installed by us |
| 09:57.03 | Weazelon | sorry dude7064, i got no timed for courses tbh |
| 09:57.27 | TheMaestro | norgs, all in one custom context |
| 09:57.55 | dude7064 | Can you at least give me some guidance on where I should start ? |
| 09:58.12 | norgs | right, so if the calls aren't in the "from-internal" context then that will be your problem i'd say |
| 09:58.27 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 09:58.36 | norgs | dude7064: what's your current "level"? |
| 09:58.46 | TheMaestro | norgs, so should i just rename it to my custom name? |
| 09:59.24 | dude7064 | I do alot of programming in java/php/javascript and have the basics on networking and how things run,, |
| 09:59.30 | norgs | i'm by no means an expert TheMaestro |
| 09:59.53 | norgs | but the context idefines a lot of what's "doable" to the server |
| 09:59.54 | dude7064 | I did read alot from the book you suggested |
| 10:00.12 | norgs | for example you don't want incoming calls from trunks being able to transfer calls |
| 10:00.13 | dude7064 | but I still feel confused |
| 10:00.52 | TheMaestro | norgs, from-internal context is present in sip.conf |
| 10:00.53 | TheMaestro | ? |
| 10:01.08 | norgs | it sounds to me like your current context doesn't allow transferring |
| 10:01.09 | Weazelon | try asking a more specific question dude7064.. |
| 10:02.22 | dude7064 | ok.. here is a basic and a specific one: I need to emulate calls on my system just to get an understanding on how things run.. for a starter, the calls are originated from the same network |
| 10:02.37 | dude7064 | with no use of external service providers or any hardware |
| 10:02.47 | dude7064 | simply extension dialing another . |
| 10:03.00 | TheMaestro | norgs, that is evident ;) |
| 10:03.04 | dude7064 | I have setup AsteriskNow and configured everything |
| 10:03.11 | dude7064 | but how can i initiate the calls ? |
| 10:03.52 | dude7064 | In other words, how can I accept and originate calls without the use of hardware ? simply michrophone and speakers for now |
| 10:03.59 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 10:04.11 | Weazelon | you will have to use some sort of softphone |
| 10:04.13 | Great_Anta_Baka | do you know how to create extensions? |
| 10:04.38 | Great_Anta_Baka | http://www.zoiper.com/free.php |
| 10:04.40 | Weazelon | the Asterisk itself is the PBX, you still need a phone to dial into the pBX |
| 10:05.06 | Weazelon | Zoiper would be a nice freeware to start with |
| 10:05.26 | Weazelon | as the Great_Anta_Baka provided |
| 10:05.33 | Weazelon | 8) |
| 10:06.32 | rob0 | Digium does training courses. |
| 10:07.20 | rob0 | (followed by certification testing) |
| 10:10.14 | Gido-E | twinkle, ekiga or X-lite are softphones to. |
| 10:10.25 | Great_Anta_Baka | i dont dig those |
| 10:10.35 | Great_Anta_Baka | zoiper is simpler and smaller than most of those |
| 10:10.49 | Great_Anta_Baka | and/or |
| 10:13.06 | norgs | TheMaestro: Well, it's very hard to suggest anything that may help but, you may want to do a "sip show peer <peerid>" (assuming you are using SIP of course) and see if the Transfer mode is open |
| 10:14.00 | Gido-E | Great_Anta_Baka ? |
| 10:14.24 | *** part/#asterisk eject_ck (n=eject_ck@85.223.182.86) |
| 10:15.20 | Great_Anta_Baka | yo |
| 10:15.58 | Great_Anta_Baka | brb need to buy some food |
| 10:16.36 | angryuser | dude7064, to emulate a big number of call you can use sipsak, anyway what are you trying to achieve ? |
| 10:17.08 | TheMaestro | norgs, transfer mode is not open in any peer |
| 10:18.27 | norgs | that'll do it |
| 10:19.23 | norgs | you'll need a transfer = yes in the extension definitions for any channel whose call you want to be transferable, i would imagine |
| 10:19.50 | norgs | or if you want all, then tranfer = yes in the general section |
| 10:20.07 | *** join/#asterisk Mr_BOnD_007 (i=Mr_BOnD_@119.160.199.6) |
| 10:20.20 | norgs | then any you don't want set specifically transfer = no |
| 10:20.35 | kaldemar | there's no such parameter in sip.conf. |
| 10:21.17 | kaldemar | that would be allowtransfer, which is yes by default. |
| 10:25.40 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 10:25.51 | TheMaestro | any gurus out here? |
| 10:26.15 | TheMaestro | ive even enabled transfer in features.conf |
| 10:26.30 | TheMaestro | blindxfer => #30 |
| 10:27.31 | norgs | in the [general] section of peers.conf and users.conf, put in transfer=yes |
| 10:27.37 | Weazelon | did you reload the features ? |
| 10:27.56 | Weazelon | after a change in the features.conf you must reload the features module |
| 10:28.10 | norgs | anyhow, i have to go now, good luck |
| 10:28.23 | Weazelon | reload res_features.so |
| 10:32.16 | Weazelon | The Maestro, did you try to reload ? |
| 10:32.16 | *** join/#asterisk cousin_luigi (n=luigi@unaffiliated/cousinluigi/x-395723) |
| 10:32.16 | cousin_luigi | hello |
| 10:32.16 | Weazelon | hi there |
| 10:32.45 | cousin_luigi | is completely new to the matter and wondered whether it was possible to implement a low-traffic T.38 fax server without dedicated hardware |
| 10:37.29 | dude7064 | Here is the situation: calling center needs to be established in a country to serve the customers of compnaies in another country.. none of the two countries are in Europe or America, and most of the online service providers charge too much per min for calls. It is much cheaper to rent phone lines at one country which are charged a fixed amount per month and give unlimited local calls. The idea is to have the calls routed from a coun |
| 10:37.51 | *** join/#asterisk eaxxae (n=eaxxae@208.70.103.41) |
| 10:37.54 | dude7064 | Can somebody please explain to me what are the systems/components I will need ? |
| 10:38.25 | dude7064 | The expected number of employees will be around 200 at each given time |
| 10:38.35 | Weazelon | let me understand, you want 2 sites at different places to be able to place calls from the same PBX ? |
| 10:38.57 | dude7064 | not the same PBX,, no.. |
| 10:39.07 | dude7064 | one is going to be connected to the internet at all times |
| 10:39.25 | Weazelon | ok then, both sites are connected via Lan i persume ? |
| 10:39.44 | dude7064 | the other office can be connected to the internet if needed,, and will have as many physical phone lines as needed |
| 10:39.53 | dude7064 | yes, both are connected to LANs |
| 10:40.19 | Weazelon | well then, if I understand what you need correctly, then you should configure a trunk route between the sites |
| 10:40.55 | Weazelon | which will produce the calls over the LAN between sites using that trunk |
| 10:42.23 | dude7064 | There is no direct connection between the two offices, and the idea is to the use the internet .. what sort of hardware will I need ? |
| 10:42.52 | imcdona | why does asterisk restart the b channel on one of my spans every hour? |
| 10:43.38 | mchou | dude7064: got get a copy of "Asterisk for Dummies" |
| 10:43.51 | mchou | s/got/go |
| 10:44.14 | kaldemar | imcdona: do you have resetinterval configured in zapata.conf/chan_dahdi.conf? |
| 10:44.21 | Weazelon | if you are using the internet then replace LAN with WAN in what i wrote :D |
| 10:44.23 | Weazelon | same thing |
| 10:45.43 | mchou | wonders why people can't be bothered to read books instead of begging for handouts in this channel |
| 10:46.46 | angryuser | dude7064, you somehow confuse everything, it is hard to understand what do you want exactly , try to read the book and caome back when you know what to ask |
| 10:47.02 | imcdona | kaldemar: I see no option of resetinterval anywhere |
| 10:47.18 | imcdona | must be somewhere though cuz its exactly every hour |
| 10:47.53 | imcdona | why would anyone want to reset their b channels regularyly anyhow? |
| 10:47.55 | kaldemar | imcdona: maybe it's the other end that engages the restart? that's not uncommon in any way with PBXs. |
| 10:48.14 | imcdona | ahh...this is connected to another pbx via crossover. thanks kaldemar |
| 10:48.16 | dude7064 | I will get Asterisk for Dummies and come back with questions if I have any,, |
| 10:49.05 | imcdona | Why do they restart the b channels though? Is it like rebooting an NT server every month? Just for good measure? |
| 10:50.09 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) |
| 10:50.12 | ThoMe | hello |
| 10:50.21 | ThoMe | have: exten => s,1,NoOp(SkillTime-Fax Anruf: ${CALLERID(num)} Angerufene Nummer: ${CALLERID(dnid)}) |
| 10:50.25 | ThoMe | result: "SkillTime-Fax Anruf: 08989223760 Angerufene Nummer: 2244892e0") |
| 10:50.45 | ThoMe | but i would like the number, not my sipid von sipgate (2244892e0) |
| 10:50.46 | ThoMe | is it posible? |
| 10:50.49 | ThoMe | the calling number? |
| 10:51.11 | angryuser | ThoMe, called number ? |
| 10:51.15 | ThoMe | angryuser: emm |
| 10:51.26 | angryuser | ThoMe, its incoming ? |
| 10:51.30 | imcdona | Asterisk will reset unused channels once an hour by default. This is |
| 10:51.30 | imcdona | intended to keep channel state mismatches between the two endpoints from |
| 10:51.30 | imcdona | persisting indefinitly. The is harmless most of the time. |
| 10:51.34 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 10:51.54 | ThoMe | when I (08989223760) dial the number 089123456 then is this my account at sipgate(.de) and would get the 089123456 not my sipid 2244892e0 |
| 10:52.22 | angryuser | ThoMe, ${EXTEN} |
| 10:52.47 | ThoMe | angryuser: -- Executing [s@fax-skilltime:1] NoOp("SIP/2244892e2-09cea670", "SkillTime-Fax Anruf: 08989223760 Angerufene Nummer: s") in new stack |
| 10:52.50 | ThoMe | <PROTECTED> |
| 10:53.15 | angryuser | ThoMe, i suppose you use macro ? |
| 10:53.45 | ThoMe | angryuser: jep. my question is: the number which i have dial is not in the sip header hm? |
| 10:54.34 | kaldemar | ThoMe: try ${MACRO_EXTEN} |
| 10:54.51 | angryuser | ThoMe, try to pass ${EXTEN} var to that macro and do a SET ${VAR1} = ${EXTEN} |
| 10:55.04 | angryuser | or as kaldemar said |
| 10:55.20 | ThoMe | is no makro: |
| 10:55.21 | ThoMe | kaldemar: exten => 2244892e0,1,Goto(fax-skilltime,s,1) |
| 10:55.21 | kaldemar | if that doesn't work, show some more information (=cli output of a call) so we don't have to guess. |
| 10:57.05 | ThoMe | kaldemar: and when i try the noop before goto then i have in EXTEN "SkillTime-Fax Anruf: 08989223760 Angerufene Nummer: 2244892e0") |
| 11:00.20 | kaldemar | based on what you give, maybe your provider doesn't send the number. but, since you know the number anyway, what is the problem? |
| 11:01.05 | angryuser | ThoMe, to do the following exten => 2244892e0,1,Goto(fax-skilltime,${EXTEN},1) and change fax-skilltime, s to mach ${EXTEN} |
| 11:01.42 | ThoMe | angryuser: yes. EXTEN is 2244892e0. i think sipgate(.de) set this. |
| 11:02.37 | angryuser | ThoMe, So i dont see the problem then ;) |
| 11:03.03 | ThoMe | [Mar 10 12:05:29] NOTICE[21007]: chan_iax2.c:6196 update_registry: Restricting registration for peer 'fax-IAX6' to 60 seconds (requested 300) |
| 11:03.04 | angryuser | ThoMe, you receive fax by sip ? |
| 11:03.15 | ThoMe | is this posible set the 60 seconds higher? |
| 11:03.19 | ThoMe | angryuser: jep. |
| 11:03.27 | Weazelon | is the peer set on IAX Thome ?? |
| 11:03.34 | angryuser | ThoMe, do you use t38 is some way ? |
| 11:03.40 | Weazelon | cause IAX and SIP wont mix |
| 11:03.42 | angryuser | in* |
| 11:04.06 | cousin_luigi | is still looking for any informations on t.38 btw:) |
| 11:04.14 | ThoMe | Weazelon: i have only a connection to sipgate.de with SIP |
| 11:04.17 | ThoMe | Weazelon: and IAX local |
| 11:05.03 | angryuser | ThoMe, forget it , do you have ping with you provider in a range of 10 ms ? if no forget about faxin over ip |
| 11:05.21 | cousin_luigi | oh |
| 11:05.31 | *** join/#asterisk zz_Buglouse (n=Buglouse@CPE-69-76-77-243.wi.res.rr.com) |
| 11:06.01 | ThoMe | angryuser: i recieve more as 200 faxe / day |
| 11:06.03 | ThoMe | over 1 xear |
| 11:06.05 | ThoMe | year |
| 11:06.08 | ThoMe | works perfectly! |
| 11:06.34 | angryuser | ThoMe, i must say you are the lucky bastard if so |
| 11:06.39 | angryuser | ;) |
| 11:06.42 | ThoMe | :P |
| 11:06.55 | angryuser | ThoMe, and how many failed ? |
| 11:07.14 | ThoMe | angryuser: all fax recieved |
| 11:07.45 | angryuser | ThoMe, even the 100% digital lines does not have 100% received, maybe you dont know ? ;) |
| 11:07.59 | ThoMe | angryuser: is posible :-)) |
| 11:09.56 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 11:16.10 | Weazelon | anyone has a fair idea on the Music on Hold in Asterisk|Freepbx ? - for some reason i'm uploading new wav files, and getting silence instead of the music. i've tried to upload Mp3 and Wav(PCM 16 8,000 mono) |
| 11:16.33 | Weazelon | but to no avail :( |
| 11:17.36 | Weazelon | i have also created a new group in the MoH Freepbx |
| 11:17.40 | norgs | Weazelon: how are you uploading the files? |
| 11:17.52 | Weazelon | using the Freepbx upload mechanism |
| 11:18.13 | norgs | and the files are ending up in the folder on the server? |
| 11:18.54 | Weazelon | well yea.. i've just upgrade to asterisk 1.4 from 1.2 which i didnt have so many problems with the MoH |
| 11:19.31 | Weazelon | is there a new folder for the MOH in 1.4 ? |
| 11:20.14 | *** part/#asterisk cousin_luigi (n=luigi@unaffiliated/cousinluigi/x-395723) |
| 11:20.33 | Weazelon | i've checked just now to verify its at - /var/lib/asterisk/mohmp3/ |
| 11:20.41 | Weazelon | same folder as 1.2 |
| 11:23.30 | norgs | when you do a "moh show files |
| 11:23.52 | norgs | in the asterisk cli, you can see your new class and the uploaded files? |
| 11:25.29 | Weazelon | yip its there |
| 11:25.51 | Weazelon | hmm here's something interesting. I now see that there is a folder at /var/lib/asterisk/moh aswell as the previous stated |
| 11:26.27 | Weazelon | Class: Weazel |
| 11:26.27 | Weazelon | File: /var/lib/asterisk/mohmp3/Weazel//orig_01MrFrustrationMan |
| 11:29.41 | *** part/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 11:33.00 | TheMaestro | has anyone got call transfer working for sip calls in asterix? |
| 11:33.08 | Weazelon | of course |
| 11:33.11 | norgs | so does your default moh class have files? |
| 11:33.17 | Weazelon | yea |
| 11:33.23 | Weazelon | currently 1 file and it works |
| 11:33.25 | norgs | TheMaestro: pretty much everyone has working transfer |
| 11:33.33 | Weazelon | its the default asterisk file |
| 11:33.43 | TheMaestro | guess im the odd one out then.. ;( |
| 11:33.55 | TheMaestro | ive broken my head on this for close to 4 hrs now |
| 11:34.14 | TheMaestro | tried everything from dialplan changes to sip.conf |
| 11:35.20 | TheMaestro | i have a code #30 followed by the number to transfer the call |
| 11:35.47 | TheMaestro | asterisk is taking 30 as part of the number and giving me an "extension not found" error |
| 11:35.58 | Weazelon | TheMaestro, : have you tried to reload after applying the change ? |
| 11:36.01 | TheMaestro | yes |
| 11:36.13 | TheMaestro | restarted the server also |
| 11:36.29 | TheMaestro | ive enabled blindxfer in features.conf |
| 11:37.00 | TheMaestro | tried tT option in Dial function of the relevant extensions |
| 11:38.47 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 11:38.51 | Weazelon | TheMaestro, are u using FreePBX or ? |
| 11:40.12 | norgs | Weazelon: so your default MOH works, but when you SetMusicOnHold(Weazel) then the called party puts the caller on hold you hear nothing? |
| 11:40.19 | TheMaestro | Weazelon, asterisk |
| 11:40.24 | TheMaestro | 1.4 |
| 11:40.52 | TheMaestro | oops 1.6.1 |
| 11:43.14 | Weazelon | norgs, : i'm choosing the Music on hold in a ringgroup settings not via console |
| 11:47.43 | Weazelon | never worked with 1.6.1 before |
| 11:48.10 | Weazelon | TheMaestro, just to double check "reload res_features.so" is the command u used in the Asterisk CLI ? |
| 11:48.31 | TheMaestro | Weazelon, i just did reload |
| 11:48.40 | TheMaestro | and then restart now |
| 11:50.03 | TheMaestro | there isnt any file by that name in /etc/asterisk |
| 11:55.10 | Weazelon | reload res_features.so is a command in the asterisk CLI |
| 11:57.17 | dude7064 | I have added a new user to AsteriskNOW using the front-end gui,, but what type of user was added ? SIP or IAX ? it doesn't say on the front end what type of users have been added,, |
| 11:58.55 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 11:58.58 | kaldemar | dude7064: #asterisknow or #asterisk-gui is a better place to ask |
| 12:00.13 | *** join/#asterisk vncsnvs (n=vncsnvs_@189.27.17.113.dynamic.adsl.gvt.net.br) |
| 12:00.24 | vncsnvs | hello, someone works with VICIDIAL/ |
| 12:00.26 | vncsnvs | hello, someone works with VICIDIAL? |
| 12:03.09 | *** join/#asterisk seanmh (n=johndoe@70.90.202.94) |
| 12:03.17 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 12:03.20 | dude7064 | no response from asterisk-gui or asteriskNOW :( |
| 12:03.35 | *** join/#asterisk maxagaz (n=maxagaz@soho.i-xanadu.com) |
| 12:07.29 | vncsnvs | hello, someone works with VICIDIAL? |
| 12:09.27 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:11.31 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 12:14.17 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 12:14.38 | *** join/#asterisk Buglouse (n=Buglouse@CPE-69-76-77-243.wi.res.rr.com) |
| 12:16.40 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 12:21.48 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 12:24.47 | awk_r | dude7064, i responded...there is a note in the asterisk-gui title saying that no one will probably be in til 10am EST. please read that before you go around giving us a bad name :-P |
| 12:24.52 | *** join/#asterisk maxagaz (n=maxagaz@soho.i-xanadu.com) |
| 12:25.22 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 12:26.45 | awk_r | dude7064, not to mention the other note saying, "is no one answering? say 'awk_r' in your sentence'"...just saying... |
| 12:27.47 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:28.14 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:29.33 | *** join/#asterisk Great_Anta_Baka (i=c4219f53@gateway/web/ajax/mibbit.com/x-eb09ebf194043443) |
| 12:29.48 | *** join/#asterisk lbruno (n=me+irc@pa1-84-91-3-125.netvisao.pt) |
| 12:29.54 | lbruno | waves |
| 12:30.54 | russellb | awk_r: no sleep for you! |
| 12:31.04 | lbruno | howdy; what's the best current practice to use an HFC-4S with asterisk? chan_capi from junghanns? |
| 12:34.18 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 12:36.18 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:37.50 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 12:38.06 | *** join/#asterisk Holister (n=ryan@70.89.17.121) |
| 12:38.23 | awk_r | russellb, heh yea i saw the commits chat. i was watching a horrible movie and decided i'd make myself feel like i'm not wasting time by coding as i watch. |
| 12:38.33 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 12:39.06 | phix | hey |
| 12:39.24 | Holister | My asterisk is sending back 407 proxy authentication required messages to my sip provider for incoming calls seemingly randomly (i.e. most of the time it sends 200 ok, but every once in a while it rejects it)... any idea how to fix this? |
| 12:39.29 | *** join/#asterisk _BBV_ (n=boris@213.138.71.254) |
| 12:39.30 | phix | ZAP hang up detect is influenced by busydetect right? |
| 12:42.13 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 12:43.23 | phix | hi |
| 12:44.57 | Holister | also, all of the asterisk scripts seem to use Net::SMTP to send to localhost, is there any way to get them to use ssmtp instead so I don't have to run a daemon??? |
| 12:45.04 | russellb | awk_r: it's all good. It's okay to be addicted. :-) |
| 12:45.16 | *** join/#asterisk propellerhead (n=yogurt2u@host215.190-138-92.telecom.net.ar) |
| 12:46.05 | awk_r | russellb, totally. |
| 12:46.09 | tzafrir_laptop | Holister, ssmtp exposes /usr/sbin/sendmail , right? |
| 12:46.25 | Holister | yes |
| 12:46.36 | tzafrir_laptop | Holister, which Asterisk script uses Net::SMTP? |
| 12:46.53 | Holister | all ones that mail :) |
| 12:47.02 | Holister | like the voicemail one for instance |
| 12:47.03 | *** join/#asterisk questprojects (n=mg@82-71-8-45.dsl.in-addr.zen.co.uk) |
| 12:47.37 | Holister | audio-email.pl and fax-process.pl |
| 12:48.08 | qp | angryuser? fixed that timer/stutter/conf room issue, had to reboot the vmware host in the end, then everything was ok again. think we shall just get dedicated 1u's instead |
| 12:48.46 | Holister | it may be freepbx and not stock asterisk |
| 12:49.08 | qp | also, zttest runs a lot faster now, 1 hit every few seconds rather than 1 a second |
| 12:50.10 | angryuser | qp, virtualising and dummy module do not produce good results in general |
| 12:51.42 | angryuser | qp, you can try with a special kernel optimized for this, i head amazon cloud had such an install, search google if you are interested |
| 12:51.48 | angryuser | heard* |
| 12:54.43 | [TK]D-Fender | Holister: Indeed not part of * |
| 12:59.50 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.19 | brunner | What's the most efficient language to use to run a FastAGI server? |
| 13:07.35 | [TK]D-Fender | brunner: SOLDER |
| 13:08.13 | [TK]D-Fender | brunner: Here, have some flux, on me... |
| 13:08.32 | brunner | I'm not even sure I should look that up... is this going to be assembly or something? |
| 13:08.58 | brunner | oh, you meant like a SOLDERing gun... |
| 13:09.13 | brunner | I thought it must be an acronym |
| 13:09.25 | brunner | anyway, seriously, what's the best way to go? |
| 13:09.36 | brunner | C or C++? |
| 13:09.57 | [TK]D-Fender | brunner: Want some more kerosene to light the "what language is the best" flamewar? |
| 13:10.14 | brunner | I'm clearly not asking what language is "best" |
| 13:10.19 | [TK]D-Fender | brunner: * is the weak link, not the language you choose |
| 13:10.21 | brunner | I'm asking which is the *most efficient* |
| 13:10.27 | brunner | for this specific purpose |
| 13:10.43 | brunner | well that's why I'd run * on its own server |
| 13:10.43 | jasonwoot | this is why I can't monitor freenode all day |
| 13:10.57 | [TK]D-Fender | brunner: its a friggen socket to process DIALPLAN. My analog watch can do this. |
| 13:11.09 | Holister | My asterisk is sending back 407 proxy authentication required messages to my sip provider for incoming calls seemingly randomly (i.e. most of the time it sends 200 ok, but every once in a while it rejects it)... any idea how to fix this? |
| 13:11.15 | brunner | I just don't want 500 separate PHP processes on my FastAGI server |
| 13:11.32 | [TK]D-Fender | Holister: Post actual debug for people to look at. |
| 13:11.56 | Holister | hard to replicate...I have sip packet capture though |
| 13:12.08 | [TK]D-Fender | brunner: Then feel free to pick "whatever". Don't know too many people using PHP as a listening daemon, thats for sure |
| 13:12.33 | tzafrir_laptop | brunner, you forgot assembly |
| 13:12.36 | brunner | [TK]D-Fender: well that's why I'm asking what would be more appropriate |
| 13:12.43 | *** join/#asterisk [gnubie] (n=[gnubie]@cm92.omega113.maxonline.com.sg) |
| 13:12.46 | brunner | tzafrir_laptop: actually, I mentioned that earlier |
| 13:12.47 | [TK]D-Fender | brunner: ANYTHING ELSE |
| 13:12.52 | [gnubie] | waves |
| 13:13.00 | brunner | uhg, I give up, just disregard my question |
| 13:13.13 | [TK]D-Fender | brunner: PERL would be a much more popular choice I suspect |
| 13:13.36 | *** join/#asterisk dpryo (n=hn@donatello.ondskap.net) |
| 13:13.42 | brunner | I want something that can use threading instead of me creating a million different processes for each session |
| 13:14.02 | brunner | or, xinetd creating, rather |
| 13:14.19 | tzafrir_laptop | brunner, I sense premature optimization |
| 13:14.49 | tzafrir_laptop | But you can use threading in just about any language nowadays. Well, maybe not in PHP |
| 13:14.50 | jasonwoot | do they make pills for that? |
| 13:14.54 | dpryo | Is there any modules or profiling software that let's me monitor memory usage within asterisk? |
| 13:14.55 | brunner | tzafrir_laptop: well I'm not going to implement it right now, but I will have to handle more than 200 calls at a time |
| 13:15.09 | [TK]D-Fender | brunner: Again, we're talking about DIALPLAN. overkill much? |
| 13:15.42 | brunner | [TK]D-Fender: this is going to way too complicated to put it all in extensions.conf, and it's much easier to interface with a database in PHP |
| 13:16.16 | brunner | okay, maybe not way too complicated to put in extensions.conf, but *much* *much* easier to code in a real language |
| 13:16.22 | [TK]D-Fender | brunner: then go right ahead with that. |
| 13:16.40 | brunner | [TK]D-Fender: I certainly will |
| 13:16.55 | [TK]D-Fender | brunner: And as i said FastAGI is just dialplan. Nothing more. |
| 13:17.13 | tzafrir_laptop | brunner, PHP and thread safety: http://blog.madism.org/index.php/2009/02/28/181-how-to-not-make-a-function-safe |
| 13:17.19 | [TK]D-Fender | brunner: the language you choose is likely going to have no relevence to the outcome. |
| 13:17.43 | brunner | [TK]D-Fender: you don't see a problem with having 500 or more PHP processes running on one machine? |
| 13:18.06 | *** join/#asterisk msetim (n=setim@200.195.162.210) |
| 13:18.17 | brunner | tzafrir_laptop: thanks |
| 13:18.21 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:18.28 | tzafrir_laptop | brunner, read it. It's fun |
| 13:18.35 | [TK]D-Fender | brunne there are ways to deal with common overhead ETC IIRC |
| 13:18.40 | msetim | hi guys.. what mean "ast_log(LOG_WARNING, "Did not read data.\n");" in res_features.c line 930 at branches? |
| 13:19.08 | msetim | Cannot it read DTMF? |
| 13:19.32 | Katty | yawns |
| 13:20.03 | Weazelon | YAWNS even more |
| 13:20.42 | Katty | has teh sleepies :< |
| 13:20.57 | msetim | someelse know? |
| 13:21.22 | WeazelON | msetim, where did you get that error ? |
| 13:22.04 | *** join/#asterisk Khratos (n=khratos@190.166.103.111) |
| 13:22.54 | [gnubie] | tzafrir_laptop sorry to pm you. do you know how i can enable exim4 to send voicemail attachments to my e-mail address at gmail on lenny? |
| 13:23.00 | Holister | [TK]D-Fender: ok, interesting thing, I'm not able to duplicate the problem now with sip debug on, BUT, I set sip debug peer for just that peer, and incoming calls, are not being debugged...just register's... so it's not associating them with that peer for some reason? does that narrow down the problem? |
| 13:24.03 | [TK]D-Fender | Holister: Clearly things aren't matching right <- |
| 13:24.22 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 13:24.23 | [TK]D-Fender | Holister: So enable global SIP debug and go see what is really happening. |
| 13:24.46 | [TK]D-Fender | [gnubie]: That wasn't a PM |
| 13:25.07 | [gnubie] | [TK]D-Fender: yeah, i mistyped sorry.. |
| 13:25.18 | [TK]D-Fender | [gnubie]: And the sample voicemail.conf shows you the obvious hooks for using whatever maill app/script/whatever you want |
| 13:25.50 | *** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu) |
| 13:25.59 | jasonwoot | wonders when gnubie will type a password into a channel |
| 13:26.26 | Holister | [TK]D-Fender: how do I turn it off? |
| 13:26.31 | [gnubie] | [TK]D-Fender: i have configured already my voicemail.conf but i don't have a mail server |
| 13:26.51 | [TK]D-Fender | Holister: "sip set debug off" |
| 13:26.53 | [gnubie] | has a password of qwerty |
| 13:26.58 | brunner | is it a bad idea to put the asterisk realtime DB on a dedicated database server? |
| 13:27.41 | [TK]D-Fender | brunner: As in not putting other data than reatime in the DB? |
| 13:27.41 | *** join/#asterisk nima0102 (n=nima@91.98.223.104) |
| 13:28.07 | *** join/#asterisk Tili (n=tili@WimaxUser379-61.wateen.net) |
| 13:28.24 | Tili | how can I specify a number range in extensions.conf like 1200-1370 |
| 13:28.27 | brunner | yeah |
| 13:28.29 | tzafrir_laptop | suspects [gnubie]'s keyboard is dvorak |
| 13:29.59 | [TK]D-Fender | Tili: takes about 3 different patterns to match |
| 13:30.58 | Tili | ah. that is not good. |
| 13:31.01 | Holister | [TK]D-Fender: http://rafb.net/p/ZMRmhP94.html |
| 13:32.29 | [TK]D-Fender | Holister: Congratulations it can't tell who its coming from |
| 13:32.42 | Holister | [TK]D-Fender: provider is callcentric...they use round-robin dns for load balance... will insecure=very fix it? |
| 13:34.07 | [TK]D-Fender | Holister: Only if it can match the rest |
| 13:34.20 | [TK]D-Fender | Holister: and it shouldn't for this last sample |
| 13:34.24 | Holister | [TK]D-Fender: That is the only thing besides the raw SIP packets that I already had that debug info gave...I don't know what you were expecting to see |
| 13:34.38 | [TK]D-Fender | Holister: that wasn't a 407 failure, that was a "not even going to ask... WTF are you?" |
| 13:34.50 | Holister | [TK: then allowfrom for their subnet? |
| 13:34.53 | *** join/#asterisk davevg (n=davevg__@74.94.3.214) |
| 13:35.03 | Holister | [TK]D-Fender: it didn't reject the call...that call went through 200 ok just fine |
| 13:35.20 | [TK]D-Fender | Holister: Go fix your peer and the norm is "insecure=port,invite", not "very". That is long since deprecated |
| 13:35.43 | Holister | [TK]D-Fender: what to fix? |
| 13:36.06 | [TK]D-Fender | Holister: Allow ranges/hosts, "insecure", etc |
| 13:36.14 | *** join/#asterisk j_kroon (n=jkroon@dsl-242-5-74.telkomadsl.co.za) |
| 13:36.26 | *** part/#asterisk fdz (n=francois@c2cpc3.camptocamp.com) |
| 13:39.22 | msetim | WeazelON, atxtransfer |
| 13:40.29 | qp | does anyone know how I could go about joining two sip calls together from our own softphone? at the moment I am moving them into a conference room, but its a bit clunky. would like to use the CLI or something else to make the "join" at the * level |
| 13:40.48 | qp | voip only |
| 13:44.02 | [TK]D-Fender | qp: How would you know who to choose to "join", and what would them be doing prior to this? |
| 13:44.44 | j_kroon | hi all, i've got a strange issue. call comes in on DAHDI, rings at some SIP extension, gets answered, put on hold, extension dials second extension, then one of two things happens: 1. press transfer - one-way voice. 2. return to other call (flash button), press transfer, then the second called person stays on hold and the caller gets cut off. |
| 13:45.03 | qp | its our softphone, so we have lines 1 and 2 with active calls. at the moment we move each into a conference room |
| 13:45.27 | qp | managed transfer kind of thing |
| 13:45.40 | [TK]D-Fender | qp: then your soft-phone sucks. Polycom phones can put them on 3-way, and the drop the call leaving them bridged |
| 13:46.09 | qp | its our softphone, but the sip library we use doesn't seem to have this |
| 13:47.16 | [TK]D-Fender | qp: Get coding! |
| 13:49.02 | qp | ok will do! so the main point is, this can't be done in the CLI ? |
| 13:49.12 | qp | as I can execute CLI stuff easily enough |
| 13:49.13 | [TK]D-Fender | qp: in 1.6 look at "bridge" |
| 13:49.22 | qp | ok thanks Fender |
| 13:51.42 | [TK]D-Fender | j_kroon: With no info on your phone's environment anything we offer would be a complete guess |
| 13:52.03 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 13:52.04 | [TK]D-Fender | j_kroon: I might immediately suspect NAT / other networking issues |
| 13:52.36 | j_kroon | [TK]D-Fender, no NAT on the LAN... disallow=all, allow=ulaw. all phones (GS GXP280) configured for only PCMU. |
| 13:53.12 | j_kroon | canreinvite=no for all phones. |
| 13:54.51 | j_kroon | [TK]D-Fender, what kind of info would be useful in tracing it? SIP debug trace for two of the phones in question? |
| 13:54.58 | [TK]D-Fender | j_kroon: that thing uses the hook to transfer? Doubtful, and even if... GAH |
| 13:55.12 | *** join/#asterisk jcoffi (n=jcoffi@75.147.154.145) |
| 13:55.16 | [TK]D-Fender | j_kroon: And yes, it would help for you to be actually looking at the problem |
| 13:55.21 | j_kroon | no, it's got a transfer button on the phone. |
| 13:55.26 | j_kroon | quickly gets some traces. |
| 13:55.48 | [TK]D-Fender | j_kroon: its got a lot of buttons.. I doubt you should need the hook for this |
| 13:56.35 | *** join/#asterisk AndyML (n=quassel@pool-98-111-140-229.phlapa.fios.verizon.net) |
| 13:56.36 | j_kroon | well, we use the "flash" button to put on hold + get a second line ... maybe that's the problem. |
| 13:58.04 | *** join/#asterisk AndyML (n=quassel@pool-98-111-140-229.phlapa.fios.verizon.net) |
| 14:00.28 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 14:02.11 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.33.119) |
| 14:13.47 | j_kroon | [TK]D-Fender, i'm only just getting reading on this one as well, it's with SIP debug on two of the peers involved, no dahdi debugging. |
| 14:13.50 | j_kroon | http://pastebin.co.za/10894 |
| 14:14.08 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:16.02 | Katty | has anyone seen a polycom say online where the date should be? |
| 14:17.17 | rob0 | probably just set the time zone / daylight savings ... oops, I need to do that with my sipuras too. |
| 14:17.39 | rob0 | Then the callerID sets the time. |
| 14:17.42 | Katty | i don't think it's that... none of the other phones in the building are doing it. |
| 14:17.51 | Katty | plus the phone can't call out |
| 14:17.56 | rob0 | hmm |
| 14:18.16 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 14:19.58 | qp | had a look, but is there a nice page with all the 1.4 vs 1.6 differences listed? |
| 14:20.57 | qp | actually, found it on russellbryant.net |
| 14:22.27 | [TK]D-Fender | j_kroon: SIP/2.0 401 Unauthorized <- you are getting a LOT of these. Not looking like you're phones are set up right at all |
| 14:22.38 | [TK]D-Fender | qp: in the TARBALL |
| 14:22.39 | j_kroon | goes looking. |
| 14:22.51 | j_kroon | ta [TK]D-Fender ... |
| 14:23.17 | *** join/#asterisk jcoffi1 (n=jcoffi@208.87.0.146) |
| 14:26.19 | j_kroon | [TK]D-Fender, could it be a sip.conf issue? |
| 14:26.24 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:27.46 | *** join/#asterisk jcoffi (n=jcoffi@75.147.157.49) |
| 14:28.13 | *** join/#asterisk iratik (i=6337e516@gateway/web/ajax/mibbit.com/x-0e9b8d760527c32e) |
| 14:31.14 | *** join/#asterisk KavanS (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:31.35 | j_kroon | [TK]D-Fender, i'm only getting it when DAHDI is involved, pure internal SIP calls are working fine... how can I figure out why it's getting SIP/401 though? |
| 14:31.54 | *** join/#asterisk ayeso (n=chatzill@216.65.195.52) |
| 14:33.28 | ayeso | I have configured SMDI.conf and added the appropriate info in voicemail.conf. But when I leave a voicemail I dont see any info about asterisk trying to send mwi on the terminal.. does this mean its not trying or is there some way to see what asterisk is doing with mwi? |
| 14:36.11 | [TK]D-Fender | ayeso: Because MWI is determined by the DEVICE |
| 14:36.24 | [TK]D-Fender | ayeso: go look at your DEVICE setup |
| 14:36.54 | ayeso | [TK]D-Fender: Im trying to send MWI via SMDI, not SIP |
| 14:37.45 | j_kroon | [TK]D-Fender, if I set the "Force INVITE" option to Yes, which causes the phone to use INVITE instead of UPDATE which now makes the 401's go away, end result is the same. |
| 14:37.47 | ayeso | [TK]D-Fender: Im trying to find a way to see if * is even trying to send the SMDI out the serial port. |
| 14:38.59 | [TK]D-Fender | ayeso: plug another PC into it with a terminal active |
| 14:39.18 | ayeso | [TK]D-Fender: thats not a bad idea |
| 14:39.42 | [TK]D-Fender | j_kroon: Trying to force something doesn't sound like "configuring it properly" so much as "beating it with a big stick and hoping it'll give in" |
| 14:40.31 | j_kroon | ok, so where can i find out what i need to have set how? cause right now the big stick is sounding very, very inviting. |
| 14:40.46 | j_kroon | i know the stick is wrong, but i'm pretty desparate atm. |
| 14:42.50 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:45.57 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:49.14 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 14:49.30 | *** join/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 14:52.02 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:52.02 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:52.50 | mcf3782 | Hey Qwell - good call last night by you and the others (who's screen names I can't remember now :( ) about replacing the demarc DSL filter with a cheapie from Radio Shack to see if the demarc one had failed. Sure enough.. that cleared the problem right up! Someone from at&t will be out before noon to replace the defective one. |
| 14:53.03 | ayeso | Ok so my * box is not trying to send SMDI for some reason, is there something I have to do other than set up the smdi.conf file and voicemail.conf file? Do I have to have some external smdi app? |
| 14:53.28 | ayeso | documentation on the web is poor/old |
| 14:54.14 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 14:55.41 | iratik | I have a client who has 2 DSL connections at 400 kbps upstream... Using asterisk with teliax... but the box can only have 1 default gateway ... is there any way to use the second DSL connection for additional bandwidth? |
| 14:57.03 | [TK]D-Fender | iratik: Setup ML-PPPoE on it |
| 15:01.03 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
| 15:01.27 | [TK]D-Fender | ayeso: Everying in the sample configs imply that *'s use for SMDI is for receiving, not sending |
| 15:01.35 | iratik | Thank you |
| 15:04.15 | *** join/#asterisk GringoChapin (n=GringoCh@65-103-48-47.cdrr.qwest.net) |
| 15:04.52 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:05.12 | jplank | ML-PPPoE is only going to work if a) both circuits are from the same carrier, b) the carrier supports it, which most dont |
| 15:05.14 | *** join/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) |
| 15:05.44 | GringoChapin | Hi all. Is Asterisk 1.4 still recommended for production environments? |
| 15:06.19 | GringoChapin | or is 1.6 stable enough? |
| 15:06.20 | anonymouz666 | it is THE version recommended for production environments. |
| 15:06.26 | jplank | I have many production environments running 1.4 |
| 15:06.35 | jplank | thats all we use |
| 15:06.46 | stintel | sure, if it ain't broken, don't fix it |
| 15:06.50 | GringoChapin | k. Thanks. |
| 15:07.07 | jplank | I thought 1.6.0.1 or whatever it is is supposed to be stable "enough" |
| 15:07.17 | GringoChapin | Yeah, going with 1.4 was my first inclination. |
| 15:07.32 | jplank | we'd still be using 1.2 if it wasn't for that nasty DTMF bug |
| 15:07.41 | [TK]D-Fender | jplank: Yeah... thats why we're at 1.6.0.6 now... |
| 15:08.20 | Katty | jaytee: i saw a posting about your zoo this morning |
| 15:08.29 | Katty | jaytee: http://icanhascheezburger.files.wordpress.com/2009/02/funny-pictures-elephants-are-washing-your-car.jpg |
| 15:08.35 | jplank | isnt there a RC1 out for 1.6? |
| 15:09.05 | GringoChapin | Was going to try putting it on Amazon EC2, but all the prebuilt AMIs are 1.6. |
| 15:09.37 | ayeso | [TK]D-Fender: oh wow.... thats not good, do you know of a way to have asterisk send SMDI? |
| 15:09.44 | jaytee | Katty, hehehe. We don't have Asian elephants at our zoo, just African elephants. The size of the ears is a dead giveaway :-) |
| 15:10.03 | [TK]D-Fender | ayeso: You've got the source just like the rest of us... |
| 15:10.18 | GringoChapin | That's ok though. I'd rather roll my own anyway. Easier to fix when something breaks that way. |
| 15:10.43 | ayeso | [TK]D-Fender: true, didn't know if there was anything out there already that you knew of. |
| 15:10.53 | Katty | jaytee: i thought you'd appreciate it ;) |
| 15:10.53 | [TK]D-Fender | ayeso: Noting on my side... |
| 15:16.34 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 15:16.47 | Chesther | So, is anyone working with SIP/TLS? I seem to be out on the bleeding edge here. |
| 15:17.43 | [TK]D-Fender | Chesther: Gauze & Polysporin are in the cabinet to your left |
| 15:18.43 | Chesther | I think at the moment I mostly need an icepack for my forehead where it keeps colliding with my desk. :-) |
| 15:18.57 | *** join/#asterisk arpu (n=arpu@91-113-87-127.adsl.highway.telekom.at) |
| 15:19.09 | angryuser | iratik, try to use pfsense load balancing as fails back is supported |
| 15:19.31 | jaytee | after yesterday afternoon I'm beginning to think that VOIP is a bad idea all around for any serious business use, nice toy though. |
| 15:19.33 | angryuser | iratik, pfsense or zeroshell or ipcop or m0n0wall |
| 15:20.35 | iratik | thanks |
| 15:21.14 | norgs | jaytee: voip is almost 100% necessary for businesses these days |
| 15:21.32 | mort_gib | <angryuser> Use OpenBSD instead.... |
| 15:21.54 | norgs | especially IT based or global businesses |
| 15:22.29 | angryuser | mort_gib, why should i ? ;) |
| 15:22.42 | norgs | the main question is if you have the it infrastructure to support it |
| 15:22.51 | mort_gib | <angryuser> That's where th load balancer in PFsense/monowall comes from |
| 15:22.54 | jaytee | norgs, until you have an MCSE setting up a network printer cripple your asterisk server by assigning the printer the same static IP address. |
| 15:23.01 | mort_gib | s/monowall/m0n0wall |
| 15:23.24 | angryuser | mort_gib, yes, i know, you dont like pfsense ? |
| 15:23.25 | norgs | that's simply a matter of sgregation |
| 15:23.29 | jaytee | the only thing that's ever taken my old Nortel pbx completely out of the picture was a lightening strike and that was only for 15 minutes. |
| 15:23.46 | mort_gib | <angryuser> :-) Got me there! |
| 15:23.52 | jjshoe | what is for incoming calls on sip/iax. peer or user? |
| 15:24.20 | norgs | did you run your nortel pbx on the same wires as your computers? or did you use a seperate set of circuits? |
| 15:24.22 | jaytee | norgs, I agree but i'm just a little asterisk/telephony guy that "does what he's told". I literally had to threaten to quit just to get VLAN and QoS. |
| 15:24.41 | jaytee | half the network runs to workstations weren't even on CAT5 wiring. |
| 15:24.58 | angryuser | mort_gib, well it took me 30 min to install pfsense and setup a failback with load balancing, i am not sure that i could do it with netbsd at all (without reading 500 pages of manual) |
| 15:25.02 | jaytee | place is a shoddy joke and run like a third world dictatorship |
| 15:25.56 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 15:25.56 | mort_gib | angryuser, but when you HAVE done that once you understand how it works, and if anyone asks you what the limits for that technology is you don't go -Uhm, I dunno |
| 15:26.34 | norgs | see, there's a difference between sloppy installation and lack of forethought or funding to do it right and doing everything right, buying proper hardware, installing stuff properly and with enough redundancy to support any problems |
| 15:27.01 | norgs | not that i'm saying it needs to be expensive |
| 15:27.19 | angryuser | mort_gib, yes, but in this case i dont care how shaper works, i dont deal with *bsd everyday, so it would be an overkill to learn all by myself for only one install |
| 15:29.32 | *** part/#asterisk AwayML (n=quassel@pool-98-111-140-229.phlapa.fios.verizon.net) |
| 15:30.45 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 15:31.15 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 15:41.00 | *** join/#asterisk RMod (n=nicolasj@unaffiliated/rmod) |
| 15:41.27 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 15:44.40 | *** join/#asterisk ronr (n=ron@82-204-104-166.fttx.bbeyond.nl) |
| 15:45.53 | ronr | hi, I setup a custom message to the voicemail in asterisk, but after my message I still hear the default voice for a while, I just want the beep to sound immediately after my message. How can I achieve that? |
| 15:47.02 | *** join/#asterisk abatista (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net) |
| 15:50.12 | ronr | found it: jgfi |
| 15:51.13 | *** part/#asterisk GringoChapin (n=GringoCh@65-103-48-47.cdrr.qwest.net) |
| 15:51.15 | [TK]D-Fender | ronr: Google? No, * CLI <- |
| 15:51.37 | ronr | [TK]D-Fender: that's what google told me :D |
| 15:53.47 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 15:54.53 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:58.26 | *** join/#asterisk nima0102 (n=nima@91.98.205.6) |
| 16:16.18 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 16:18.01 | Qwell | ronr: Google told you to google it? |
| 16:18.06 | Qwell | It's become sentient... |
| 16:18.31 | Qwell | You know, that would be an epic april fools joke. Every search result comes back "JFGI" |
| 16:20.06 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 16:20.29 | *** join/#asterisk trippssss (n=tripps@c-98-201-149-5.hsd1.tx.comcast.net) |
| 16:24.19 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:27.53 | angryuser | hello, is there any way to do a multi vide conference with asterisk ? (3 more persons) ? |
| 16:27.53 | angryuser | video* |
| 16:29.03 | jplank | polycom is setup to use ftp by default, is there any way to change that through DHCP, or will each phone have to be configure separately for it? |
| 16:29.10 | *** join/#asterisk r0d3nt (i=astrutt@pinky.ratman.org) |
| 16:29.25 | Qwell | angryuser: not currently, no |
| 16:30.05 | ronr | Qwell: lol, actually google told me to cli it |
| 16:30.21 | angryuser | Qwell, is there any way to do it ? let's say i can buy hardware and integrate to * |
| 16:31.42 | Qwell | angryuser: I'm not familiar with any video conferencing stuff |
| 16:48.21 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
| 16:53.25 | mcf3782 | Qwell: Did you see my message earlier about the DSL filter issue from last night? |
| 16:53.34 | Qwell | no |
| 16:53.53 | mcf3782 | <PROTECTED> |
| 16:54.06 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 16:54.07 | mcf3782 | easier to copy and paste than type it again. |
| 16:54.08 | mcf3782 | :) |
| 16:54.24 | Qwell | to be fair, I was somewhat against the idea initially :p |
| 16:54.35 | mcf3782 | They replaced it with a new one. All better now. |
| 16:54.41 | Qwell | though getting one for testing is good, as you've discovered |
| 16:54.46 | Qwell | so, cool |
| 16:55.19 | mcf3782 | Yep. Not something I'd thought of before. But now I know. :) |
| 16:55.49 | KavanS | [TK]D-Fender: could you tell me what I'm doing wrong? I have a problem with slashes in my ast_monitor filename, but I'm not sure how to grep/sed/replace the bad text: ex. http://www.pastebin.ca/1357467 |
| 16:56.48 | [TK]D-Fender | kava":" is not legal |
| 16:57.20 | KavanS | hrm, it only errors on the "/" |
| 16:57.27 | KavanS | I already have filenames with ":" in them |
| 16:57.46 | KavanS | is there any way to remove "/" from ever being used in monitor? |
| 16:58.18 | KavanS | can I assign a variable to the output of a sed/grep command |
| 17:01.08 | *** join/#asterisk mvanbaak (i=mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 17:03.43 | [TK]D-Fender | KavanS: Where do you see it erroring on the slash? |
| 17:05.58 | KavanS | [TK]D-Fender: updated.... http://www.pastebin.ca/1357467 |
| 17:06.04 | KavanS | I also included filenames....it's not the ":" |
| 17:07.00 | [TK]D-Fender | KavanS: NOT updated |
| 17:07.05 | [TK]D-Fender | KavanS: NEW link please |
| 17:07.06 | KavanS | pastebin.ca being slow....try now |
| 17:07.12 | KavanS | sorry...I clicked it was taking awhile |
| 17:07.16 | [TK]D-Fender | KavanS: NEW link please <---- |
| 17:07.18 | KavanS | ok |
| 17:07.48 | KavanS | [TK]D-Fender, http://pastebin.com/m3a20553a |
| 17:08.18 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 17:08.38 | KavanS | I'm pretty sure it's the slash....I need a way of removing/changing the slash ;\ |
| 17:08.59 | [TK]D-Fender | KavanS: why is there a LEADING slash? |
| 17:09.19 | [TK]D-Fender | #1 : |
| 17:09.20 | [TK]D-Fender | #1 : [Feb 5 04:59:42] WARNING[15631] res_monitor.c: Cannot change monitor filename of channel IAX2/LINUX-2 to OUTGOING_2009-02-05-04:59_/LI_5083660000, monitoring not started |
| 17:09.32 | KavanS | ok... |
| 17:09.38 | KavanS | I'm not sure why there is a leading slash |
| 17:09.39 | [TK]D-Fender | #2 : [Mar 10 07:46:04] WARNING[22327] res_monitor.c: Could not create file /OUTGOING_2009-03-10-07:46_/LI_8665572828-in |
| 17:09.48 | [TK]D-Fender | KavanS: I think you need to pay some attention here |
| 17:10.16 | KavanS | why is it recording for all other contexts? |
| 17:10.22 | KavanS | that's what I'm confused about |
| 17:10.52 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:15.30 | [TK]D-Fender | KavanS: I'm not looking at THSOE, am I? |
| 17:15.33 | [TK]D-Fender | THOSE* |
| 17:15.46 | KavanS | right... |
| 17:15.56 | KavanS | but I'm saying...how can I remove the slashes? |
| 17:16.12 | KavanS | I understand your point |
| 17:18.44 | [TK]D-Fender | KavanS: You are failing to show where you ASSIGN the recording file in the first palce |
| 17:18.53 | KavanS | ok, let me paste that section |
| 17:20.02 | KavanS | [TK]D-Fender, http://pastebin.com/m705424cb |
| 17:21.29 | [TK]D-Fender | KavanS: COMPLETE CALL please |
| 17:22.55 | Qwell | ${CHANNEL:4:3} ? |
| 17:23.08 | Qwell | There's your problem. |
| 17:23.37 | KavanS | Qwell: how can I get it so it only shows the outgoing number? that was the solution I found for zaptel/sip calls to parse the correct number |
| 17:23.51 | Qwell | Because zap and sip have 3 letters. |
| 17:24.32 | KavanS | doh... |
| 17:24.50 | Qwell | and the 3 will only get you...3 characters. Linux has...5 |
| 17:25.14 | KavanS | so, the best method is to change the context? |
| 17:25.55 | KavanS | hrm, IAX2 has 4 chars, vs 3 |
| 17:26.16 | Qwell | and LINUX has 5 |
| 17:26.38 | *** join/#asterisk brunner (n=chris@66.35.172.123) |
| 17:27.07 | KavanS | ok, so you are suggesting to change LINUX to LIN? |
| 17:27.12 | Qwell | use ${CUT()} |
| 17:27.24 | Qwell | don't depend on lengths of anything |
| 17:29.46 | KavanS | nice, okay I will try that |
| 17:33.44 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-188-106.telkomadsl.co.za) |
| 17:34.25 | jasonwoot | http://pastebin.com/m81d580c are the arguments being passed to this macro invalid? |
| 17:34.50 | KavanS | Qwell, [TK]D-Fender: thank you both for your assistance, trying out this CUT function |
| 17:37.39 | *** join/#asterisk twisted (n=twisted@router.asteriasgi.com) |
| 17:37.39 | *** mode/#asterisk [+o twisted] by ChanServ |
| 17:38.09 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:41.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:41.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:42.01 | *** topic/#asterisk by lmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.6 (2009/02/23), 1.4.23.2 (2009/03/10), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 17:42.08 | *** join/#asterisk mellow-yellow (n=mellow-y@mycomp.norris-stevens.com) |
| 17:42.14 | lmadsen | Asterisk 1.4.23.2 (security update) has been released |
| 17:42.18 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 17:42.18 | *** mode/#asterisk [+o Mog] by ChanServ |
| 17:42.37 | lmadsen | In addtion, 1.4.24-rc1, 1.6.0.7-rc1, and 1.6.1.0-rc2 are also available! |
| 17:42.43 | Katty | lmadsen: your sanity has also been released. |
| 17:42.51 | lmadsen | Please test those release candidates, and report issues to http://bugs.digium.com/ |
| 17:42.53 | jasonwoot | burn |
| 17:42.54 | lmadsen | Katty: barely |
| 17:44.34 | Katty | lmadsen: i found it at 976-Babe street |
| 17:45.22 | Katty | jaytee: and as for YOU |
| 17:45.55 | jaytee | huh? |
| 17:45.56 | Katty | jaytee: thank you (= |
| 17:46.05 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 17:46.10 | Katty | jaytee: if i were not several hundred miles away, i would insist on taking you to dinner. |
| 17:46.31 | jaytee | s'ok. I don't have much of an appetite lately |
| 17:47.04 | Katty | i hope it comes back. |
| 17:47.39 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 17:48.50 | jaytee | ever since I switched over one of my PRI's that used to connect to my Nortel pbx to route incoming calls to asterisk and outbound calls through asterisk to the telco to be a pri_cpe for my primary inbound span I've been getting dropped calls every so often but nothing shows up in the logs or on the console and it's a mix of inbound and outbound calling so it's either span. I'm stumped. |
| 17:49.18 | Katty | how often does it happen a day? |
| 17:49.28 | jaytee | Katty, it's sporadic |
| 17:49.32 | jaytee | and it's not a load issue |
| 17:49.33 | Katty | hrm. |
| 17:49.42 | Katty | can't reproduce it then, eh? |
| 17:49.50 | jaytee | not forcibly, no |
| 17:49.50 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 17:50.06 | Katty | darn )= there goes the idea of getting the telco's help |
| 17:50.11 | Katty | hi mike |
| 17:50.52 | MikeJ | heya Katty |
| 17:51.16 | Katty | MikeJ: what's shakin |
| 17:52.47 | MikeJ | cool stuff |
| 17:53.25 | coppice | MikeJ: don't worry. your weather should soon warm up |
| 17:53.50 | MikeJ | heh.... I don't buy that coppice |
| 17:54.09 | *** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net) |
| 17:54.12 | coppice | its nice and warm here. I'm naked right now |
| 17:54.23 | Katty | sounds like napping weather. |
| 17:54.25 | [intra]lanman | eeewwwwwww |
| 17:54.27 | Katty | i'd go for a nap. |
| 17:54.49 | [intra]lanman | just got the most horrible visual |
| 17:55.11 | *** join/#asterisk [Rated-R] (n=b0red@bookit-dev.com) |
| 17:55.13 | Katty | [intra]lanman: you should go to naked camp |
| 17:55.34 | MikeJ | coppice: thats why england isn't warm :D |
| 17:55.45 | [intra]lanman | Katty: i used to be a member of a nudist colony... |
| 17:55.50 | [8none1] | jaytee: you may be having problems with locked channels. I was having a similar issue and kept seeing this error in my logs : Ring requested on channel 0/4 already in use or previously requested on span 1. Attempting to renegotiating channel. |
| 17:56.04 | Katty | [intra]lanman: did everyone stay generally healthy? |
| 17:56.07 | [intra]lanman | Katty: but some people you just don't wanna see naked |
| 17:56.23 | coppice | MikeJ: you mean nature is trying to mitigate how the women there look? |
| 17:56.26 | Katty | everyone looks the same in irc. |
| 17:56.32 | Katty | i don't care what coppice does ;) |
| 17:56.38 | jaytee | [8none1], thanks but I'm not getting squat on either the CLI or the messages log in /var/log/asterisk |
| 17:56.39 | [intra]lanman | unless you know what they look like |
| 17:58.05 | Katty | how about we wear sombraos, have mexican, and listen to mariachi remix songs.. and pretend we're at a party on friday night instead. |
| 17:58.16 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 17:58.49 | Katty | bring your own margaritas. |
| 17:59.07 | Katty | coppice: and your pants, please. |
| 17:59.29 | mosty | jaytee, what about pri debug? |
| 17:59.46 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 18:00.18 | MikeJ | [8none1]: are they really locked or are they really in use? |
| 18:00.36 | *** join/#asterisk arpu (n=arpu@chello080109017107.12.14.vie.surfer.at) |
| 18:00.39 | MikeJ | that error could be a bug but just as well could be channel setup glare issue |
| 18:01.14 | coppice | it could be a glaring bug |
| 18:02.30 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:05.44 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 18:06.28 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 18:07.24 | jaytee | what's the usage for pri set debug file? if i type pri set debug /tmp/spans31009.txt I get No such command 'pri set' (type 'help' for help) |
| 18:07.52 | jaytee | i'd rather dump pri debug to a file than to the CLI if I can |
| 18:09.19 | jaytee | nevermind |
| 18:10.16 | jaytee | pri debug span 1 |
| 18:10.21 | jaytee | oops |
| 18:10.25 | jasonwoot | anyone else experience lots of parity errors with 410p on Dell equip? |
| 18:10.25 | jaytee | disregard :-) |
| 18:14.29 | mosty | jaytee, try pri intense debug if you can't see anything useful (and you can handle that amount of logging) |
| 18:15.21 | jaytee | mosty, I've got pri debug logging on my inbound span at the moment. I'm not seeing anything unusual coming across the CLI |
| 18:16.19 | jaytee | would activating a second NIC on a separate VLAN from the server's primary NIC cause the primary to reset? I get no indication in the system logs to that effect. |
| 18:16.45 | jaytee | because I did that right around the time of the last reported dropped calls |
| 18:16.47 | Katty | pakistan put sharia law into effect :< |
| 18:17.02 | mosty | jaytee, i don't see how NIC settings would affect PRI |
| 18:17.20 | jaytee | saudi arabia is going to give a 75 year old woman lashings and a prison term for "mingling" |
| 18:17.33 | Katty | [TK]D-Fender: how many women can you fit into your basement? |
| 18:17.36 | jaytee | mosty, neither do I but it doesn't hurt to ask |
| 18:17.52 | Katty | jaytee: she won't live through that |
| 18:18.16 | [TK]D-Fender | Katty: Brothel's full, sorry... |
| 18:18.16 | *** join/#asterisk {Sean} (n=sean@99.49.211.86) |
| 18:18.31 | jaytee | religion of peace my ass! I say nuke the Saudis and we'll just drill through the radioactive glass to get to the oil afterwards |
| 18:18.45 | Katty | i definately think we need to cut all funding. |
| 18:18.58 | jaytee | funding to who? they fund us. |
| 18:18.58 | Katty | and hide as many women as can fit into fender's basement |
| 18:19.09 | Katty | mine can probably hold at least 30 |
| 18:19.33 | jaytee | hell, they practically have us all by the "balls" economically. The US govt kisses the Saudi King's ass on a daily basis. |
| 18:19.47 | Katty | psh |
| 18:19.54 | Katty | the bush administration gave pakistan 2 billion dollars |
| 18:20.01 | *** part/#asterisk {Sean} (n=sean@99.49.211.86) |
| 18:20.11 | [TK]D-Fender | jaytee: Screw it. use this as another reason to go completely green, dump gasoline, vreate a new world economy and let them realize that without gas they have nothing else to offer and they'll die off on their own. |
| 18:20.26 | [TK]D-Fender | jaytee: Death by IRRELEVANCE |
| 18:20.32 | jaytee | anyways, I've got to get back to figuring out this crap with dropped calls instead of ranting against "cave people in turbans" |
| 18:21.00 | Katty | k |
| 18:21.05 | Katty | we can rant later |
| 18:21.09 | [TK]D-Fender | "If it is weak, kill it or ignore it. Anything else honours it" |
| 18:21.37 | Katty | i want to go on a manhunt over there. |
| 18:21.40 | Katty | kill any male that i see. |
| 18:22.08 | Katty | i'll make a flash game out of it |
| 18:23.06 | [TK]D-Fender | Katty: MAN-Slaughter, a 1st-degree game! |
| 18:23.35 | Katty | [TK]D-Fender: excellent suggestion. |
| 18:25.00 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 18:28.08 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:29.01 | KavanS | nice, CUT worked |
| 18:29.05 | KavanS | thanks qwell! |
| 18:31.50 | Qwell | That'll be $299.99 |
| 18:34.18 | *** join/#asterisk msetim (n=setim@200.195.162.210) |
| 18:34.19 | jjshoe | what? why are you giving him a discount? you made me pay $599.99!!!! |
| 18:34.19 | *** join/#asterisk ob123 (n=ob1@mini.forked.net) |
| 18:34.29 | msetim | hi guys.. |
| 18:34.57 | msetim | Log full is getting "Internal timing is disabled". What does it mean? |
| 18:35.19 | ob123 | well now that i passed the test of registering a nickname to get into this channel, anyone willing to help me figure out if asterisk will work for my hypothetical setup? |
| 18:35.22 | Qwell | jjshoe: You also go "extra" services. |
| 18:35.23 | Qwell | got* |
| 18:35.50 | jjshoe | ob123 hypotheticaly someone here will help |
| 18:35.53 | ob123 | :) |
| 18:35.56 | jaytee | this afternoon I enabled a second NIC on my asterisk server that is in the same VLAN and ip subnet as all my phones and ATA's. I changed my phone to register to asterisk at that NIC's address instead of the one we've been using which was in our primary VLAN. Now calls from my phone show up on another's with my name and then sip:xxxx@xxx.xxx.xxx.xxx . I've masked the extension and ip address but you get the idea. Prior to this change callerid on the |
| 18:35.56 | jaytee | <PROTECTED> |
| 18:36.17 | Qwell | ob123: As long as you hypothetically ask a question. |
| 18:36.28 | *** join/#asterisk Joeppie (i=sgiebels@61-246.surfsnel.dsl.internl.net) |
| 18:36.46 | mosty | jaytee, are both ip's on the same subnet? |
| 18:37.16 | ob123 | here goes: i'd like to use my voip client enabled iPhone (via wifi) to connect to asterisk and have asterisk (running on an xserve) and have the xserve dial out on a standard telephone line |
| 18:37.39 | Qwell | ob123: sure |
| 18:37.40 | jaytee | no, one is 128.68.100.47 and the other is 128.68.116.2 |
| 18:37.51 | ob123 | i'll be travelling out of country and it would be nice to call my family without paying tons |
| 18:38.15 | ob123 | Qwell: what extra hardware might I need? I imagine something to connect said standard phone line to the xserve |
| 18:38.48 | Qwell | an ATA, unless the xserve is x86 and running Linux |
| 18:38.56 | jaytee | both are separate VLANs but the trunking on the Cisco Catalyst is set to let traffic traverse to the server at 128.68.100.47 from any node on 116 |
| 18:39.02 | ob123 | no, it's running OS X |
| 18:39.03 | Qwell | (an ATA with an FXO - there are only a few of those) |
| 18:39.03 | jjshoe | jaytee did you adjust your localnet? |
| 18:39.06 | mosty | ob123, unless you will make a lot of calls, it's probably cheaper to get a sip account with a voip service provider |
| 18:39.07 | ob123 | (os x server) |
| 18:39.37 | *** join/#asterisk Dimens (n=Dimens@93-81-5-11.broadband.corbina.ru) |
| 18:39.40 | ob123 | ah.. so it's kinda like spending lots to save a bit eh |
| 18:39.48 | jjshoe | skype? |
| 18:39.57 | jaytee | jjshoe, I enabled the second NIC and that's all I've done. |
| 18:40.00 | mosty | jaytee, ok just checking (asterisk doesn't work with multiple ip's on the same subnet) |
| 18:40.04 | jjshoe | jaytee adjust your localnet. |
| 18:40.15 | jaytee | jjshoe, sorry, not sure how to do that |
| 18:40.29 | jaytee | jjshoe, lemme go google that |
| 18:40.51 | ob123 | Qwell just for kicks, any idea of a brand that would be good for this? and what would the difference be if it were running linux? |
| 18:41.07 | Qwell | You could use a PCI card in the box. |
| 18:41.08 | mosty | ob123, if you look at the cost of an ATA or pci FXO card, it will probably me more than you are hoping to save |
| 18:41.09 | jjshoe | jaytee in one of your configs you probably have a localnet=<ip range here> |
| 18:41.30 | jaytee | jjshoe, asterisk configs? |
| 18:41.37 | jaytee | or system configs |
| 18:41.40 | jjshoe | jaytee * |
| 18:41.47 | jaytee | ok, I'll check |
| 18:41.53 | ob123 | ok, that's the thing; i have no idea what the costs are.. attempting to google :) |
| 18:44.41 | *** join/#asterisk Imo (n=Imo@brln-4db839be.pool.einsundeins.de) |
| 18:44.53 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 18:45.00 | Dimens | Hi, could somebody take an answer for follow: I've installed asterisk 1.4, tune it for my needs, and installed Xlite on WinXP. When I call to some number, and speak to microphone, at the end of line hear my voice quiet, although the mic switcher is on maximum in Xlite. What can you advice me? |
| 18:45.09 | Imo | hello i have problems with asterisk 1.6.0.6 current |
| 18:45.36 | Imo | when i get incomming calls its all ok but when i call out i can only call for one minute |
| 18:45.43 | Imo | what can i do ? |
| 18:46.38 | harry_v | wait untill you give is the config into on pastebin.ca |
| 18:46.47 | mosty | ob123, digium tdm410 with one FXO module, or sangoma a200 with one FXO module. and you'll regret getting either of those without the hardware echo cancellation option |
| 18:46.49 | harry_v | info |
| 18:47.23 | Qwell | mosty: he doesn't have a box to put a PCI card in |
| 18:47.44 | ob123 | Qwell boxes are easy to come by, xserve just happens to be a server that i have running now |
| 18:48.07 | ob123 | i could just throw linux on a box i've got lying around |
| 18:48.14 | ob123 | whould this be a potential option? http://tr.im/hdou |
| 18:48.16 | mosty | Qwell, i have never looking inside an xserve- do they not even have pci-express? |
| 18:48.18 | Dimens | Somebody could help me? |
| 18:48.25 | Qwell | mosty: It's neither x86 nor running Linux |
| 18:48.30 | Qwell | ob123: NO |
| 18:48.39 | *** join/#asterisk cart (n=cart@200.252.20.30) |
| 18:48.40 | jaytee | jjshoe, I don't have localnet set in sip.conf at all |
| 18:48.46 | ob123 | mosty: let me googhle that |
| 18:49.19 | ob123 | (assuming Linux on x86, not xserve) |
| 18:49.20 | mosty | ob123, apparently those won't work on your xserve, so add that to the cost of an x86 linux box |
| 18:49.29 | [TK]D-Fender | Dimens: Get a better mic or soundcard |
| 18:49.52 | harry_v | how long has xserv been around |
| 18:49.53 | jjshoe | jaytee hm.. interesting |
| 18:50.03 | ob123 | wow $600 bucks heeh |
| 18:50.11 | ob123 | not really a hobby purchase, eh |
| 18:50.53 | ob123 | ok, welp. thanks for the help everyone |
| 18:50.55 | harry_v | ob123, are you talking about the xserve costing 600 dollars? |
| 18:51.19 | ob123 | harry_v, i quick googled digium tdm410 and one came up at 600 or so.. i later found a few at 300 |
| 18:51.27 | ob123 | either way, more than the $50 i was expecting |
| 18:51.31 | mosty | ob123, there are cheaper, ancient single port fxo pci cards, but they're so crappy you wouldn't want to waste your time |
| 18:51.31 | Dimens | [TK]D-Fender, maybe exists some settings in asterisk that volume up the voice? |
| 18:51.39 | harry_v | okay, didnt think a server like that would be so cheap :) |
| 18:52.08 | mosty | ob123, the tdm410 comes with different options, the at $300 it's probably just the base card, you'd have to buy an FXO module for it |
| 18:52.09 | [TK]D-Fender | ob123: http://www.telephonydepot.com/Catalog/Linksys-Analog-Adapters/Linksys-SPA3102 |
| 18:52.16 | ob123 | mosty, well it's mostly for fun.. not for production |
| 18:52.25 | [TK]D-Fender | ob123: All other solutions = massive overkill for you |
| 18:52.34 | ob123 | yeah that's what i was thinking |
| 18:52.34 | [TK]D-Fender | ob123: Use this and you'll be fine |
| 18:52.42 | ob123 | [TK]D-Fender * checking |
| 18:53.23 | ob123 | oh neat, so basically plug phone cable in, ethernet comes out to box? |
| 18:53.31 | Dimens | [TK]D-Fender, maybe exists some settings in asterisk that volume up the voice? |
| 18:54.00 | [TK]D-Fender | Dimens: Hardly. Your souce has issues. Look in your mixer for a gain boost option. Most soundcards have this |
| 18:54.05 | [TK]D-Fender | ob123: Yup |
| 18:54.39 | Dimens | [TK]D-Fender, thank you, I'll look |
| 18:56.01 | eppigy | hello |
| 18:56.03 | eppigy | i am dave |
| 18:57.25 | Katty | hello, dave. |
| 18:57.34 | jplank | hi dave |
| 18:57.47 | Imo | Imo: hello i have problems with asterisk 1.6.0.6 current |
| 18:57.47 | Imo | [7:45pm] Imo: when i get incomming calls its all ok but when i call out i can only call for one minute |
| 18:58.19 | jplank | sounds like a invite issue |
| 19:07.26 | msetim | Hi guys, I'm getting: [2009-03-10 15:03:05] DEBUG[28023] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=-1) |
| 19:07.30 | msetim | it's ok? |
| 19:08.36 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:09.07 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 19:09.23 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 19:10.43 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 19:14.59 | kannan | hello, i have a agi script for call logs, but now want to use the default cdr-csv. The Master.csv doesnt use the server time , but is having some other time in the records. Where to start trouble shooting? |
| 19:15.06 | jaytee | I have my asterisk server as 128.68.100.47, all my phones are in the 116 vlan which includes the 128.68.116 and 128.68.117 subnets. When ever I enable my second NIC on my * server with a static IP on the 116 subnet it causes callerid on the phones to appear as "name of person sip:xxxx@xxx.xxx.xxx.xxx" instead of "name of person xxxx" |
| 19:16.24 | jaytee | with just eth0 enabled on my * server and the traffic traverses from VLAN 116 to VLAN 1 callerid looks normal. I have no localnet= setting in my sip.conf |
| 19:16.44 | [TK]D-Fender | jaytee: This is a polycom issue when the server is not the same as the phone's subnet. |
| 19:17.02 | [TK]D-Fender | jaytee: Even if it routes it will look like a foreign subnet and pin the IP on it |
| 19:17.19 | [TK]D-Fender | jaytee: Point the phone to the server's IP on that subnet |
| 19:18.37 | jaytee | [TK]D-Fender, think you misunderstood my previous message. The callerid gets messed up when I enable the NIC ON THE SAME SUBNET. when it traverses to the 100 subnet to the other NIC callerid is fine. |
| 19:19.36 | [TK]D-Fender | jaytee: Look at the SIP debug for that call... its whenever the reported subnet is different between the phone & * |
| 19:19.43 | [TK]D-Fender | jaytee: I used to get this a lot here. |
| 19:22.10 | Khratos | Have anyone here faced the situation where, having everything working on different Asterisk boxes linked by sip blocks, things stop working (Authentification, on INVITES) after changing g729 ? |
| 19:22.14 | [TK]D-Fender | kannan: UTC |
| 19:22.25 | Imo | can anybody help me please ?? |
| 19:22.52 | [TK]D-Fender | Khratos: that question should ahve come with a failed call's SIP DEBUG in a pastebin attached... |
| 19:23.11 | kannan | [TK]D-Fender . thanks |
| 19:24.43 | Khratos | I know [TK]D-Fender, let me see (note: everything was working ok, perfectly, after changing everything to g729 it got into troubles, SIP debug told me 'Proxy Authentification required', but it is set, and it was working before.. ) |
| 19:25.12 | [TK]D-Fender | Khratos: Sorry, i don't do stry-time :) |
| 19:25.14 | [TK]D-Fender | Story* |
| 19:25.36 | Khratos | ? |
| 19:25.44 | [TK]D-Fender | Khratos: PAStebIN <- |
| 19:25.54 | *** join/#asterisk kerx (n=kerx@adsl-69-104-17-222.dsl.irvnca.pacbell.net) |
| 19:26.02 | Khratos | Im on it right now |
| 19:27.36 | brutuz | asterisk contains gazillion config files.. |
| 19:28.19 | brutuz | and i thought qmail is weird.. |
| 19:28.20 | Katty | bored bored bored |
| 19:28.50 | brutuz | katty: break your asterisk.. and build it again.. |
| 19:29.02 | Katty | that's a horrible suggestion |
| 19:29.29 | bkw_ | so anyone here coming to cluecon ? |
| 19:29.40 | Katty | no |
| 19:30.04 | brutuz | it would keep you busy.. |
| 19:32.05 | *** join/#asterisk docid (n=eris@whthyt253-26.northwestel.net) |
| 19:33.46 | *** join/#asterisk Woody2143 (n=Brandon_@machine76.Level3.com) |
| 19:33.49 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 19:38.04 | *** join/#asterisk aenaus_ (n=hdgfghf@91.140.106.239) |
| 19:47.30 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 19:48.46 | Khratos | [TK]D-Fender, http://pastebin.com/m3ef1276e |
| 19:51.29 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 19:52.03 | [TK]D-Fender | Khratos: i SEE NO DIALPLAN EXECUTION, AND IF THERE WAS A CODEC ISSUE IT'D BE 484 |
| 19:54.25 | Khratos | Sorry about extensions (working on it right now), I forgot. About codec Issue, I know what you say; but suddenly it stopped getting authenticated |
| 19:57.45 | Khratos | http://pastebin.com/m52a6e275 extensions.conf added |
| 20:00.53 | *** join/#asterisk b14ck (n=comradeb@72.37.252.50) |
| 20:01.00 | b14ck | hey all--quick question about dial plans |
| 20:01.32 | b14ck | i want to have a dial plan that basically allows users to dial any number, then it puts the call through to that number, but after the other end picsk up, i want it to play a soundfile, then hangup |
| 20:01.42 | b14ck | i know how to do the play soudfile, and hangup part |
| 20:02.01 | b14ck | any ideas? :D |
| 20:02.03 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:02.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:03.14 | kaldemar | b14ck: core show application dial |
| 20:03.38 | kaldemar | as in the asterisk command. you'll find a suitful option for the dial command. |
| 20:04.20 | Khratos | b14ck, maybe : A(x) - Play an announcement to the called party, using 'x' as the file. (application: Dial) |
| 20:04.34 | b14ck | ah, thanks |
| 20:06.14 | b14ck | another quicky: lets say i want the user to be able to directly pass the number they dial through to my voip provider as the number to call, and lets say i assume that the number is of the form: 1 818 xxx xxxx, my dial plan looks like: _.,1,Dial(SIP/whatgoeshere) <--- what is the environment variable set by that command which contains the number they entered? |
| 20:06.53 | Khratos | ${EXTEN} |
| 20:07.03 | b14ck | oh yey! that solves my problem! |
| 20:07.05 | b14ck | thanks again <3 |
| 20:07.18 | Khratos | np |
| 20:09.31 | kaldemar | b14ck: if you have _.,1 in your dialplan, replace _. with something else. that matches to special extensions such as t (timeout) and h (hangup) and will cause you unexpected behaviour sooner or later. _X. will match to any number. |
| 20:13.51 | Qwell | kaldemar: any number followed by anything. |
| 20:15.19 | kaldemar | yes. i was not complete. |
| 20:17.25 | *** join/#asterisk bmoraca (n=chatzill@209.60.253.58) |
| 20:26.31 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 20:26.58 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 20:27.00 | Chainsaw | Can someone translate from autoconf-speak to english for me please: |
| 20:27.01 | Chainsaw | configure.ac:27: warning: AC_COMPILE_IFELSE was called before AC_USE_SYSTEM_EXTENSIONS |
| 20:27.01 | Chainsaw | ../../lib/autoconf/specific.m4:386: AC_USE_SYSTEM_EXTENSIONS is expanded from... |
| 20:27.01 | Chainsaw | configure.ac:27: the top level |
| 20:27.19 | Chainsaw | I am completely unable to rebuild the configure script shipped with Asterisk 1.6.0.6 |
| 20:28.35 | MikeJ | waht version of autoconf are you using? |
| 20:28.35 | MikeJ | and do you ahve multiple versions installed |
| 20:28.37 | *** join/#asterisk SparFux (n=raoul@e182020098.adsl.alicedsl.de) |
| 20:28.48 | MikeJ | looks kind of like one version of autoconf using macros from a different version |
| 20:29.17 | Chainsaw | MikeJ: Using 2.63, I have both 2.63 and 2.13 installed. |
| 20:29.57 | MikeJ | try manually supplying the path to the 2.63 |
| 20:30.19 | MikeJ | I doubt 2.13 works but its been a long time sense I've been in ast autodrool |
| 20:32.05 | Chainsaw | amalthea / # /usr/bin/autoconf --version |
| 20:32.05 | Chainsaw | autoconf (GNU Autoconf) 2.63 |
| 20:32.12 | Chainsaw | Okay, will override. |
| 20:32.36 | *** join/#asterisk cgardner (n=cgardner@64.239.154.240) |
| 20:32.48 | Qwell | Chainsaw: Why are you rebuilding the configure script? |
| 20:33.07 | Chainsaw | Qwell: I'm trying to fix a bug in the net-snmp configure test. |
| 20:33.17 | Qwell | using bootstrap.sh? |
| 20:33.25 | Chainsaw | Qwell: Tried that as well, yes. |
| 20:33.33 | Qwell | and what did that say? |
| 20:33.36 | Chainsaw | Qwell: Right now through eautoreconf. |
| 20:33.54 | Chainsaw | It also gave a flood of AC_COMPILE_IFELSE was called before AC_USE_SYSTEM_EXTENSIONS. |
| 20:34.17 | Chainsaw | Only that one is not explicit about failure and will not abort in a visible way. |
| 20:36.02 | *** join/#asterisk BlargMaN00 (n=blargman@12.234.16.130) |
| 20:36.22 | Chainsaw | I've explicitly turned off all patching. This is a vanilla 1.6.0.6 codebase now. |
| 20:36.33 | Qwell | What distro are you on? |
| 20:36.37 | Chainsaw | Gentoo. |
| 20:37.18 | MikeJ | try removing autoconf 2.13 |
| 20:37.20 | Qwell | try an older autoconf - 2.61 works for me |
| 20:37.29 | Qwell | or Gentoo is being stupid and choosing 2.13 |
| 20:37.30 | MikeJ | this stinks of using 2 diff versions at once |
| 20:37.40 | MikeJ | heya Qwell |
| 20:38.25 | Chainsaw | Downgrading 2.63 -> 2.61 |
| 20:39.37 | Chainsaw | That runs fine, yes. |
| 20:39.46 | Chainsaw | Well, up to the netsnmp test that explodes. |
| 20:39.55 | Chainsaw | Upgrading 2.61 -> 2.62 to see where the problem starts. |
| 20:40.58 | Chainsaw | 2.62 does also explode, indeed. |
| 20:41.17 | Chainsaw | So whatever is going wrong happened 2.61 -> 2.62 |
| 20:41.33 | *** join/#asterisk LND (n=LND@89.193.143.243) |
| 20:42.07 | Chainsaw | Qwell, MikeJ: One of you able to temporarily upgrade autoconf and see whether that introduces the bug for you? |
| 20:42.58 | MikeJ | Chainsaw: no.. but... |
| 20:43.04 | MikeJ | perhaps the error message is right? |
| 20:43.25 | Chainsaw | Let me get you the full error message, at any rate. |
| 20:45.12 | Chainsaw | MikeJ: http://www.vroon.org/autoconf.error.txt |
| 20:45.29 | Qwell | What is line 27? |
| 20:45.34 | *** join/#asterisk tAnkOSX (i=tank@the.matrix.has-you.net) |
| 20:46.02 | *** join/#asterisk bob_slacker (n=root@189.27.17.113.dynamic.adsl.gvt.net.br) |
| 20:46.03 | Qwell | AC_GNU_SOURCE? |
| 20:46.07 | Chainsaw | AC_USE_SYSTEM_EXTENSIONS # note- does not work on FreeBSD |
| 20:46.21 | bob_slacker | ne1 uses VICIDIAL ? |
| 20:46.30 | Chainsaw | The line before is AC_GNU_SOURCE though. |
| 20:46.48 | Qwell | Chainsaw: remove the AC_GNU_SOURCE line. What happens? |
| 20:47.34 | bob_slacker | ne1 uses VICIDIAL ?? |
| 20:47.46 | Chainsaw | Qwell: configure:6259: error: possibly undefined macro: _AC_PATH_PROG_FEATURE_CHECK |
| 20:47.46 | Chainsaw | <PROTECTED> |
| 20:47.46 | Chainsaw | <PROTECTED> |
| 20:48.06 | Chainsaw | Which is a few lines into: |
| 20:48.07 | Chainsaw | # This macro is just copied into our local acinclude.m4 from libtool.m4 so that |
| 20:48.07 | Chainsaw | # the developers regenerating the configure script don't have to install libtool. |
| 20:48.49 | ipchains | ne1 has installed VICIDIAL successfully? |
| 20:49.04 | Chainsaw | ipchains: Asking every two minutes isn't really helpful. |
| 20:49.08 | Qwell | ipchains: ...we heard you the first three times |
| 20:49.28 | Qwell | Chainsaw: This reeks of an autoconf regression. I don't know what changed though |
| 20:49.32 | ipchains | sorry, lots of people, no answer |
| 20:49.40 | ipchains | i go mad |
| 20:49.45 | *** join/#asterisk horvath (n=horvath@74-51-45-109.telnetcommunications.com) |
| 20:50.17 | Chainsaw | Qwell: *nod* Bit unfortunate. Perhaps I can short-circuit the existing configure script for now then. |
| 20:50.49 | Chainsaw | Qwell: It's hackish and ugly, but that sure beats saying "no Asterisk 1.6.0.6 ebuild in the tree because it won't work on autoconf-2.63". |
| 20:52.19 | Qwell | There are a lot of packages that are going to get bit by this |
| 20:52.20 | Chainsaw | Qwell: You're right about the regression, others have found it: |
| 20:52.21 | Chainsaw | http://www.mail-archive.com/bug-autoconf@gnu.org/msg01971.html |
| 20:54.51 | Qwell | Chainsaw: When you removed AC_GNU_SOURCE, what version of autoconf did you use? |
| 20:55.00 | Chainsaw | Qwell: 2.63 |
| 20:55.04 | Chainsaw | Qwell: It did then continue. |
| 20:55.13 | ipchains | Qwell, may i ask again? |
| 20:55.15 | Chainsaw | Qwell: But it found new things to hate on. That macro. |
| 20:55.19 | Qwell | ipchains: You may not. |
| 20:55.54 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:56.26 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:56.38 | horvath | Qwell: wow he actually left |
| 20:56.59 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:03.18 | KavanS | Qwell: your advice helped me fix my problem earlier, cheers man |
| 21:04.15 | *** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net) |
| 21:04.18 | *** join/#asterisk bougyman (i=bougyman@pdpc/supporter/monthlygold/bougyman) |
| 21:07.48 | *** join/#asterisk Kage` (n=Kage@c-71-60-67-135.hsd1.wv.comcast.net) |
| 21:09.27 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 21:10.57 | *** join/#asterisk BadHAL (n=nn@66.194.174.31) |
| 21:13.52 | *** join/#asterisk mesfet (n=iw3grx@host165-3-static.25-87-b.business.telecomitalia.it) |
| 21:15.21 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:22.14 | *** join/#asterisk DarkRift (n=dark@65.92.249.12) |
| 21:22.36 | Kobaz | mesfet: no bri, just t1/pri |
| 21:25.08 | Carlos_PHX | Video phones...anyone use one they like? We are trying the Grandstream GXV3000 and have some issues with the handset (video works great). Looking for other good options. |
| 21:29.08 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 21:36.29 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 21:37.32 | jblack | I'm having occasional trouble with a sangoma card. I think the company providing the PRI is botching things up somehow so that on rare occasion, zap locks up on * (zap anything on asterisk -r gets stuck, no new channels are sohwn, etc). Restarting asterisk seems to restore intended behaviour. Is there some sort of reset option for zapata.conf I can add? |
| 21:39.09 | jaytee | jblack, you can put resetinterval=number of seconds for each group of channels or set it to never |
| 21:39.22 | *** part/#asterisk Khratos (n=khratos@190.166.103.111) |
| 21:41.03 | jblack | And that'll reset the card if necessary? |
| 21:41.14 | Qwell | Chainsaw: test a patch for me? |
| 21:41.22 | Chainsaw | Qwell: Sure. |
| 21:41.26 | Arsenick- | Hi all, I'm having some problem getting my polycom phone boot using the boot-server option given by my dhcp server... I just want to know if I need special setting to allow the phone to get and read the option with the user/password in it.. |
| 21:41.43 | Arsenick- | kind of option boot-server "ftp://user:XXXX@10.10.2.34" |
| 21:42.00 | Qwell | Chainsaw: remove the AC_GNU_SOURCE from configure.ac, and apply http://pastebin.com/m5f0e81c9 |
| 21:42.22 | bougyman | Carlos_PHX: i haven't found any handsets that work as good as the soft video phones. |
| 21:42.26 | jblack | cause it seems like the * <->zapata interface is locking up. typing "zap ?" in asterisk -r (which show show a list of zap options) locks up |
| 21:42.41 | *** join/#asterisk zenfox (n=steven@c-98-213-240-12.hsd1.il.comcast.net) |
| 21:43.42 | Chainsaw | Qwell: I'm getting a weird "patch unexpectedly ends in middle of line". |
| 21:43.48 | Chainsaw | Qwell: Could you e-mail your diff to tony@linx.net please? |
| 21:44.26 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 21:44.26 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 21:45.26 | Qwell | Chainsaw: sent |
| 21:45.50 | Chainsaw | Received, thank you. |
| 21:46.28 | Chainsaw | Still fails to apply though. |
| 21:46.37 | Chainsaw | Will do it manually. |
| 21:47.10 | Qwell | don't try to follow the syntax...you'll be much happier |
| 21:47.11 | *** join/#asterisk sjobeck (n=sjobeck@97-115-86-44.ptld.qwest.net) |
| 21:48.25 | Chainsaw | Qwell: Confirmed, that builds. |
| 21:48.33 | Qwell | after a bootstrap? |
| 21:48.36 | Chainsaw | Yes. |
| 21:48.58 | Chainsaw | And all I'm getting warning-wise is: |
| 21:48.58 | Chainsaw | configure.ac:143: warning: AC_CACHE_VAL(GNU_MAKE, ...): suspicious cache-id, must contain _cv_ to be cached |
| 21:58.08 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 21:59.12 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:02.51 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 22:05.56 | *** part/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 22:06.05 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 22:06.08 | Qwell | Chainsaw: svn branches/1.6.0/ revision 180946 |
| 22:06.59 | Chainsaw | Qwell: Cool, thanks. |
| 22:07.27 | Qwell | should fix all the problems you're seeing |
| 22:07.44 | Chainsaw | Qwell: Excellent. I filed this for dahdi-tools btw: http://bugs.digium.com/view.php?id=14638 |
| 22:08.29 | Qwell | Don't do that then? |
| 22:08.49 | Chainsaw | Pretty much every Gentoo system has that LDFLAG. |
| 22:08.54 | Chainsaw | It has a patch. |
| 22:09.31 | mesfet | Kobaz: about zaptel and 2.6.28, renaming expires field in _expires (line 202 of ztdummy, which produce the error) lead to a compilation success. I don't know if it really work. |
| 22:09.52 | Kobaz | maybe |
| 22:10.06 | mesfet | Kobaz: I've seen that in the linux header hrtimer.h the field of the structure hrtimer "expires" was renamed into "_expires". |
| 22:13.06 | *** join/#asterisk chigambamukoko (n=junk@fl-76-4-144-4.dhcp.embarqhsd.net) |
| 22:13.24 | chigambamukoko | Greetings to all in the name of the creator |
| 22:13.32 | chigambamukoko | I have set up freeswitch and asterisk with FS as the proxy server and asterisk as the client, I can't seem to call PSTN from asterisk although I can call asterisk fine from PSTN through FS |
| 22:13.48 | chigambamukoko | any pointers? |
| 22:14.00 | chigambamukoko | I yelp help |
| 22:22.37 | *** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 22:27.32 | *** join/#asterisk t_corr (n=richard@d199-74-67-104.nap.wideopenwest.com) |
| 22:33.49 | chigambamukoko | Any professional asterisk support people in here wanting to make a few dollars helping me? |
| 22:34.06 | new_to_perl_boy | just ask |
| 22:34.10 | new_to_perl_boy | you might get it for free |
| 22:34.13 | seanbright | not if it's a freeswitch question |
| 22:34.25 | seanbright | i heard freeswitch kills puppies |
| 22:34.29 | seanbright | that's second hand though... |
| 22:34.34 | Mog | chigambamukoko, /me is in money need situation |
| 22:34.57 | chigambamukoko | Ok, Mog PM me if you can |
| 22:36.23 | new_to_perl_boy | wonders why people don't just ask? |
| 22:36.27 | *** join/#asterisk jmmills (n=jason@74.85.239.62) |
| 22:36.59 | seanbright | he already asked, actually. |
| 22:37.15 | jmmills | Has anyone done any asterisk+memcache stuff yet? like dial plan storage? announcements? recording? etc |
| 22:37.18 | seanbright | "I have set up freeswitch and asterisk with FS as the proxy server and asterisk as the client, I can't seem to call PSTN from asterisk although I can call asterisk fine from PSTN through FS" |
| 22:37.55 | new_to_perl_boy | lol wut? |
| 22:38.07 | new_to_perl_boy | why use freeswitch? |
| 22:39.11 | chigambamukoko | new_to_perl_boy: well its a long story but what i need is very simple by your standards i'm sure of it |
| 22:39.19 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 22:39.33 | new_to_perl_boy | lol, think again. i've been battling the same 3 problems for like 9 months now. |
| 22:40.06 | new_to_perl_boy | speaking of which, i think i have one of them solved - the sucky provider one - i switched to broadvoice for our sip "trunking" and they rock. |
| 22:40.36 | mercutioviz | chigambamukoko: did you ask your question over in #freeswitch? |
| 22:41.24 | chigambamukoko | no sir, |
| 22:42.02 | mercutioviz | it might be a matter of configuring both ast and fs to do what you want |
| 22:42.05 | mercutioviz | sec, |
| 22:42.45 | mercutioviz | chigambamukoko: how about this? http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk |
| 22:43.22 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 22:43.35 | *** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 22:43.46 | chigambamukoko | i tried that mercutioviz, but can't seem to make any progress |
| 22:43.53 | mercutioviz | hmm |
| 22:44.22 | chigambamukoko | since i'm a novice, i think i missing something simple i think |
| 22:44.26 | mercutioviz | you might want to ask in #freeswitch and then use their pastebin to post your configs, errors, etc. |
| 22:44.28 | mercutioviz | possibly |
| 22:44.34 | mercutioviz | usually is something simple |
| 22:44.35 | drmessano | Ok, coolest person on twitter: http://twitter.com/themime |
| 22:44.44 | chigambamukoko | true, true |
| 22:46.54 | chigambamukoko | mercutioviz: that link you just pasted seem to focus on calling from asterisk to freeswitch, do you have any suggestions on making it go through FS rather than to FS? |
| 22:47.45 | mercutioviz | chigambamukoko: yeah, you're now talking about the dialplan; I'd recommend joining #freeswitch because this is definitely not a #asterisk subject |
| 22:47.50 | De_Mon | I seem to be missing something... I've updated my sip.conf context but when I dial a number, it's using the old context still |
| 22:48.01 | tehsuck | sip reload? |
| 22:48.02 | De_Mon | I did a sip reload and rebooted the phone, so what gives? |
| 22:48.20 | tehsuck | pastebin sip.conf and extensions.conf |
| 22:49.16 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:49.41 | pdmmm | sip show peer <peername> |
| 22:49.44 | pdmmm | what context does it say |
| 22:49.56 | De_Mon | I just looked at that and it has the correct context |
| 22:50.10 | De_Mon | so it's something with the contexts I've screwed up then, I didn't miss any steps as it were |
| 22:51.02 | tehsuck | a common problem i used to run into was either not having ,1, or having multiple ,1,'s |
| 22:51.04 | pdmmm | extensions reload |
| 22:51.22 | drmessano | ~sipnat |
| 22:51.23 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:51.53 | pdmmm | sip behind nat is terrible |
| 22:52.03 | tehsuck | i'm faring alright |
| 22:52.28 | tehsuck | got a sip provider and my asterisk server is behind nat - i just forward everything and it seems to behave |
| 22:53.03 | pdmmm | not me |
| 22:53.15 | drmessano | It works fine if youre not a dumbass and set it up in the 20 seconds it takes |
| 22:53.30 | tehsuck | also i set externip and localnet |
| 22:53.49 | tehsuck | and nat=yes in the peers. but really that was the extent of the nat-specific setup. |
| 22:54.03 | drmessano | externip/externhost, and localnet.. takes 20 seconds |
| 22:54.06 | drmessano | or 3 months |
| 22:54.15 | drmessano | Depending on how dense one is |
| 22:54.16 | tehsuck | heh. |
| 22:54.31 | drmessano | You either fix it the first time |
| 22:54.38 | pdmmm | or just put an interface on a public ip |
| 22:54.42 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:54.42 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 22:54.48 | drmessano | or come back every 2 weeks bitching about one-way audio and how asterisk sucks |
| 22:54.56 | drmessano | and SIP sucks |
| 22:54.58 | drmessano | and NAT sucks |
| 22:55.02 | drmessano | and life sucks |
| 22:55.05 | drmessano | etc |
| 22:55.05 | tehsuck | i had one way audio issues for a while but it turned out to be my provider |
| 22:55.10 | pdmmm | wow |
| 22:55.12 | tehsuck | i thought it was me, but no. |
| 22:55.14 | pdmmm | salty, eh? |
| 22:55.25 | drmessano | Nope |
| 22:55.29 | De_Mon | hrmm http://pastebin.ca/1357757 |
| 22:56.14 | De_Mon | its still executing @local, this is so weird. |
| 22:57.29 | De_Mon | the sip user is using a default template that has context=local with a local override to context=local-secure, and * is setting the context correctly, but it sure isn't dialing within that context. |
| 22:57.37 | De_Mon | I'm gonna try removing the template |
| 22:59.36 | De_Mon | it's like, asterisk kept the authentication credentials etc from before I rebooted the phone. |
| 23:12.50 | tehsuck | try /etc/init.d/asterisk stop |
| 23:15.30 | De_Mon | yeah, just tried that... its STILL going to local. |
| 23:15.56 | De_Mon | I have to be doing something stupid |
| 23:28.39 | jmmills | Anyone know of any web based astdb editors out there? |
| 23:31.02 | De_Mon | so, I just renamed my sip peer and it definately broke the user. |
| 23:32.47 | De_Mon | The ONLY place I have local is on this one template, and even when I change that one location and reload it still goes to local. could it be getting the context from something other than sip.conf? |
| 23:35.17 | Chainsaw | Qwell: 1.6.0.6 in the portage tree now. Thanks again :) |
| 23:38.09 | De_Mon | I'm going to chalk this up 1.4.21.1, and hope that upgrading fixes it. |
| 23:45.21 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 23:47.57 | *** join/#asterisk ManxPower (n=manxpowe@user-24-236-95-236.knology.net) |
| 23:52.58 | *** join/#asterisk youngproguru (n=root@cpe-76-180-148-178.buffalo.res.rr.com) |
| 23:53.41 | youngproguru | Hmmmmm |
| 23:55.34 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 23:56.06 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 23:59.50 | *** part/#asterisk youngproguru (n=root@cpe-76-180-148-178.buffalo.res.rr.com) |
