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| 08:57.13 | joebee | how to implement text2wav replacement in ARI? |
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| 13:07.06 | LgK | hi |
| 13:09.34 | LgK | is someone can help me with MusicOnHold feature, have a little problem with it |
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| 13:23.11 | Demon_VoIP | Good day. Whether res_pjsip resolving the SRV records in asterisk 13.4? Where can I read about it? Found only test cases for asterisk 14 |
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| 13:23.46 | LgK | nobody for info about musiconhold feature ? |
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| 13:38.03 | MrFidget | hey. Anyone aronud ?? |
| 13:38.09 | [TK]D-Fender | nope |
| 13:38.25 | MrFidget | Been a while [TK]D-Fender |
| 13:38.47 | LgK | don't think |
| 13:38.56 | MrFidget | :-) |
| 13:39.06 | MrFidget | anyway, got a small challenge with 1.8 |
| 13:39.21 | file | Demon_VoIP, it does - it uses the built-in PJSIP resolver |
| 13:39.30 | file | Demon_VoIP, the support in 14 is more complete/advanced |
| 13:39.36 | file | and uses our own |
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| 13:42.51 | Demon_VoIP | file, I specify uri "sip:host" without port... and SRV records does not resolving.. Endpoint is Unavailable |
| 13:42.59 | MrFidget | <PROTECTED> |
| 13:43.09 | MrFidget | SIPPEER returns unknown |
| 13:43.33 | Demon_VoIP | file, How can I verify that the SRV lookup is happening? In the "core set debug 5" I found nothing interesting |
| 13:45.08 | file | Demon_VoIP, it should show up in PJSIP stuff, if you bump it up high enough - I use 8 |
| 13:48.12 | Demon_VoIP | file, thanks. I'll try |
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| 13:50.40 | Demon_VoIP | 13.4. the last few weeks, periodically (every 8 hours) the server hangs. netstat shows the queue for sending UDP. Very similar to DNS issues resolving, but in the logs there are no errors "getaddr". There are no errors :( |
| 13:51.11 | Demon_VoIP | Someone is already treated like this? |
| 13:53.23 | Demon_VoIP | Moreover, switching of all connected peers chan_sip or res_pjsip does not change the situation. |
| 13:55.23 | MrFidget | Dont worry. Found a default IP address in the handset sip.conf |
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| 14:03.47 | loko- | Does anyone have any pointers for dtmf with asterisk? I have tried inband, auto, rfc2833 and no matter what I try - I cannot connect into webex meetings. |
| 14:05.27 | mjordan | loko-: I highly doubt that the issue is Asterisk recognizing the DTMF. |
| 14:06.16 | loko- | I guess I can try a different VoIP provider to eliminate that. |
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| 14:08.12 | [TK]D-Fender | loko-: You need to look at each leg of your call. You have not specifed what you have on your inbound leg to * or what you are dialing out using |
| 14:09.19 | loko- | [TK]D-Fender, thanks. I think I figured it out changing it in the sip.conf for the provider |
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| 14:10.13 | cervajs2 | hello. is it possible simultaneously use periodic-announce in queue and ring agents? actual situation is sequential - ring agents, play announce (for 15 sec), ring agents , ... (i want connect agent with caller asap when agent is free) |
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| 15:44.16 | vader- | Are there any good providers that offer reseller programs for Hosted PBX? |
| 15:44.45 | [TK]D-Fender | Clarify that... |
| 15:45.22 | vader- | I have a few small businesses that are like 3-15 phones with an on premise Asterisk/FreePBX box. The phones are good but I want to get them into a hosted solution and resell the service |
| 15:45.32 | vader- | so i am looking for a good provider that I can resell |
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| 15:51.27 | [TK]D-Fender | There are hosting services |
| 15:51.36 | [TK]D-Fender | I can't imagine anyone RESELLTING that |
| 16:05.29 | vader- | intermedia |
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| 17:01.44 | Resnik | DTMF qustion, on my SIP trunk if I call from mobile phone, tones to go through, but if I call from lets say other landline (I belive its inband configured), it does not register DTMF tones at all |
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| 17:38.20 | adeeln | as the func_odbc.conf file has the ability to list multiple DSN's for read/write queries, does that mean the preferred method of database failover is by having separate, distinct ODBC DSN's? |
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| 17:42.15 | adeeln | Resnik: from my experience, i've only ever really had success using RFC2833 for DTMF |
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| 17:56.26 | wanda_ | HI, I have a question aboit voip-info.org |
| 17:56.33 | wanda_ | how can we contribute? |
| 17:56.54 | [TK]D-Fender | Sign up |
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| 17:57.03 | wanda_ | just register? |
| 17:57.15 | wanda_ | inside I could see the good place? |
| 17:57.16 | [TK]D-Fender | And then realize that that wiki is decrepit crap |
| 17:57.22 | wanda_ | :) |
| 17:57.29 | wanda_ | decrepit huhhu |
| 17:58.41 | wanda_ | thank you <[TK]D-Fender> |
| 17:59.22 | qakhan | i am trying to get events through Asterisk Manager, here is my code http://pastebin.com/5pe9QrNc |
| 17:59.35 | qakhan | when i disconnect the manager there is warning message |
| 17:59.42 | qakhan | PHP Warning: fgets(): 8 is not a valid stream resource in /scripts/phpagi/phpagi-asmanager.php on line 158 |
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| 18:37.00 | eschmidbauer | anybody know where to find royalty free hold music? |
| 18:37.55 | robmal | opsound |
| 18:38.35 | eschmidbauer | cool thanks |
| 18:39.39 | robmal | Thank Digium ;-) |
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| 18:39.52 | eschmidbauer | digium setup opsound? |
| 18:40.05 | [TK]D-Fender | Not that I can see... |
| 18:40.10 | robmal | http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ |
| 18:40.47 | [TK]D-Fender | Of course googling "royalty free music" would have gotten you a ton of results with less typing than you spent asking the question.... |
| 18:40.58 | eschmidbauer | those results were bs |
| 18:41.01 | eschmidbauer | that's what i did |
| 18:41.10 | eschmidbauer | they were all sites asking for mney for royalty free music |
| 18:41.27 | robmal | Yes. And here his first result was the best possible. |
| 18:41.37 | robmal | better than google for once. |
| 18:42.31 | [TK]D-Fender | Just found one NOW |
| 18:42.38 | [TK]D-Fender | off the first page of results |
| 18:42.48 | [TK]D-Fender | And is REALLY free and I'm douwnloading MP3's NOW |
| 18:42.53 | [TK]D-Fender | Ther goes that idea... |
| 18:42.55 | eschmidbauer | you're better googler than me |
| 18:42.59 | eschmidbauer | good job |
| 18:43.03 | [TK]D-Fender | SAME search. |
| 18:43.05 | [TK]D-Fender | First page |
| 18:43.11 | [TK]D-Fender | I gave the EXACT words I used |
| 18:43.16 | eschmidbauer | cool |
| 18:43.19 | eschmidbauer | congratulations |
| 18:43.28 | eschmidbauer | want a trophy? |
| 18:43.31 | robmal | You know google results are personalised for at least a few years? |
| 18:43.35 | [TK]D-Fender | I'm sure I'll find more if I cared to look. |
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| 19:27.34 | Marquel | morning. i have a question about sip-providers. my sip-provider will host multiple, non-consecutive public phone numbers and forward all incoming calls through one sip-user/password combination. what will be an appropriate way to pick all these called numbers apart again in a recent asterisk? |
| 19:31.42 | eric_hill | Marquel, do they preserve the dialed number and hand you the call "to" that number? |
| 19:32.04 | Marquel | eric_hill: tbh. i don't know exactly. |
| 19:32.05 | eric_hill | TO field is different from SIP user/password. |
| 19:32.27 | Marquel | eric_hill: i can only try in about 24hrs. |
| 19:34.36 | [TK]D-Fender | Then you'll have to wait and see |
| 19:34.59 | [TK]D-Fender | Most will make the target # as part of the INVITE string itself and thus the extension being targeted |
| 19:35.46 | Marquel | [TK]D-Fender: in that case it would probably sufficient to have a context=inbound in [general] in sip.conf along with a single register => ? |
| 19:36.37 | [TK]D-Fender | Marquel: How to identfy the numebr has nothing to do with peer |
| 19:36.51 | [TK]D-Fender | Marquel: And calls always land in a context |
| 19:36.57 | [TK]D-Fender | targeting an extension |
| 19:37.03 | [TK]D-Fender | What that is, and where that is... depends on you |
| 19:38.49 | Marquel | [TK]D-Fender: according to sip.conf documentation, register => user:pass@sip-provider.com pushes all inbound calls from that provider to s-extension in the defined context. question is, whether that still holds or if the called number is in invite then i can have exten => number... and that extension will be used instead of s-exten. |
| 19:39.35 | [TK]D-Fender | Marquel: It doesn't. It tells the other side to send calls to "s" ebcause you didn't say otherwise. It does not FORCE them to. They can send to whatever they want |
| 19:39.56 | [TK]D-Fender | marso as I said.. you're just going to have to wait and SEE what calls from them look like |
| 19:42.02 | Marquel | [TK]D-Fender: which "other side"? i'm talking accepting calls and dispatch them to different internal extensions within asterisk. |
| 19:42.52 | [TK]D-Fender | your PROVIDER |
| 19:43.08 | [TK]D-Fender | You'll need to wait and see what the calls actually look like |
| 19:43.36 | Marquel | hrm. |
| 19:44.00 | eric_hill | Registration provides context, not a specific extension. The call can come from the provider in a myriad of different ways. Hopefully it will just be a simple invite to the original destination. |
| 19:45.22 | [TK]D-Fender | Registration does not provide "context" |
| 19:45.32 | *** join/#asterisk zapata (~zapata@2001:470:1f0b:11bc:4c7b:ada4:621:15e3) |
| 19:45.44 | [TK]D-Fender | the peer that the inbound attempt successfully matches against sets the context |
| 19:46.29 | Marquel | [TK]D-Fender: that would be me. and that's what all the questions are about. |
| 19:46.30 | eric_hill | You're splitting hairs. |
| 19:47.12 | [TK]D-Fender | If you're going to use words, don't misrepresent their meaning. |
| 19:47.34 | [TK]D-Fender | Because people will start assuming that it's correct and start forming other thoughts around it and then their entire understanding ends up a mess |
| 19:48.08 | Marquel | [TK]D-Fender: honestly at the moment you're confusing me more than helping. |
| 19:49.13 | [TK]D-Fender | Well if identifying what provides "context" confuses you.... well that's not a good sign for being propared to configure your system for those calls. |
| 19:51.02 | eric_hill | Apparently my understanding is messed up because I have a peer defined in sip.conf that has a context=whatever defined. My inbound calls (no matter what the number) comes in to the "whatever" context. How does the registration not provide context??? |
| 19:52.44 | [TK]D-Fender | Where do you see "context" on a REGISTER => line? |
| 19:52.54 | [TK]D-Fender | Calls get matched against peers and users |
| 19:53.04 | [TK]D-Fender | REGISTER => is not amatching system |
| 19:53.09 | Marquel | [TK]D-Fender: as i said before: there is a context defined in [general] as well as a register => line. |
| 19:53.11 | [TK]D-Fender | ~sipregister |
| 19:53.12 | infobot | [~sipregister] SIP registration is to tell your provider what IP address & EXTEN to send INCOMING calls to. Some ITSPs let you use a fixed address or host rather than registering. Registration is NOT normally needed to PLACE calls, as those are typically auth'ed independently. Others accept unauth'ed calls once you are registered (saves on negotiation BW). |
| 19:53.16 | [TK]D-Fender | ^ |
| 19:53.31 | [TK]D-Fender | There is no "context" in a REGISTER => line |
| 19:54.14 | Marquel | *sigh* |
| 19:54.22 | [TK]D-Fender | http://svnview.digium.com/svn/asterisk/branches/13/configs/samples/sip.conf.sample?revision=434654&view=markup |
| 19:54.24 | [TK]D-Fender | Line #744 |
| 19:54.47 | Marquel | again: i have a context defined in sip.conf's [general] section. in that same section i do a register. |
| 19:54.51 | [TK]D-Fender | calls always get matched against peers/users |
| 19:54.59 | Marquel | is that good with you, [TK]D-Fender? |
| 19:55.05 | [TK]D-Fender | and you can have a register line an NOT peer defined whatsoever |
| 19:55.15 | [TK]D-Fender | because they are in fact 2 separate things |
| 19:55.24 | superscrat | eric_hill, registration and context are two different concepts. registration tells this asterisk instance to register itself with another sip provider (host). |
| 19:55.29 | [TK]D-Fender | you can also have just a bunch of peers and never register to anything at all as well. |
| 19:55.31 | file | ew, SVN |
| 19:55.38 | superscrat | wassup file? |
| 19:55.51 | file | not tacos |
| 19:55.57 | superscrat | oh, yeah, git r done... |
| 19:56.06 | superscrat | i ate nachos. |
| 19:56.39 | [TK]D-Fender | Marquelagain: i have a context defined in sip.conf's [general] section. in that same section i do a register. <_ that would not be "by the register". If the call comes and hits that context it means it didn't match any defined peer and your system is accepting completely unidentified calls |
| 19:56.40 | eric_hill | I give my carrier a set of credentials that they connect to a sip peer with and they hand calls through that peer. I don't use the register => mechanism. Hence my confusion. |
| 19:56.48 | [TK]D-Fender | Marquel: Which is consider "not great" |
| 19:57.25 | [TK]D-Fender | eric_hill: indeed registration is optional depending who you are dealing with. |
| 19:57.36 | [TK]D-Fender | eric_hill: wchi is an aspect that botlet tells you |
| 19:58.06 | Resnik | DTMF qustion again, on my SIP trunk if I call from mobile phone, tones do go through, but if I call from lets say other landline (I belive its inband configured on that end), it does not register DTMF tones at all, I use dtmfmode=rfc2833 in configuration... |
| 20:01.52 | Marquel | [TK]D-Fender: if i cannot give my provider credentials to use, then what options do i have? |
| 20:03.30 | [TK]D-Fender | Marquel: Normally you don't give the provider credentials. They give them to you. This includes the very common possiblity that they send you un-authed calls, and you should be matching by host alone without any user/pass check |
| 20:04.25 | Marquel | [TK]D-Fender: is there any page describing how to handle this case? possibly comparing to register =>? |
| 20:04.51 | [TK]D-Fender | Marquel: Remember, there is NO comparison. How they send you calls is how they send you calls. |
| 20:05.12 | Marquel | [TK]D-Fender: i mean configuration-wise. |
| 20:05.26 | [TK]D-Fender | Marquel: In a peer you setup for them to send calls via that provider you use "insecure=port,invite" to accept un-authed call from their IP. |
| 20:05.31 | Marquel | [TK]D-Fender: obviously i can use register => or define a peer. |
| 20:06.05 | Marquel | so i want to know what the difference between the two ways are the pro's and con's. |
| 20:10.25 | [TK]D-Fender | Marquel[TK]D-Fender: obviously i can use register => or define a peer. <- NO. This is the misunderstanding |
| 20:10.33 | [TK]D-Fender | matching the incoming call is ALWAYS the peer |
| 20:10.39 | [TK]D-Fender | Register does not factor in at all |
| 20:10.46 | Marquel | incoming. |
| 20:10.49 | Marquel | how? |
| 20:11.23 | [TK]D-Fender | I just explained that |
| 20:11.27 | Marquel | no |
| 20:11.38 | [TK]D-Fender | the call comes in and is ALWAYS matched against your peer definitions. |
| 20:11.43 | Marquel | ah. |
| 20:11.48 | [TK]D-Fender | If it MATCHES.. then that is the one whose context is used |
| 20:12.04 | Marquel | so, if i do not have a peer? |
| 20:12.18 | [TK]D-Fender | If it doesn't match ANY then it is up to your setup to allow un-authed calls in the first place and [general] defines where those go |
| 20:12.26 | [TK]D-Fender | Which is a SHITTY way of doing things |
| 20:12.36 | [TK]D-Fender | You should always be able to identify the source of your calls |
| 20:12.43 | Marquel | ah, _now_ i start to get a picture. |
| 20:12.59 | [TK]D-Fender | What happens when some spammer starts flooding your system with calls trying to get out? |
| 20:13.15 | [TK]D-Fender | And then because you included some stuff you shouldn't have they get to dial back OUT your provider |
| 20:13.30 | [TK]D-Fender | then you're on the hook for 300$ worth of calls before you know it |
| 20:13.37 | [TK]D-Fender | (or $3000) |
| 20:13.55 | *** join/#asterisk pchero (~pchero@0x555140b5.adsl.cybercity.dk) |
| 20:14.57 | Marquel | [TK]D-Fender: not gonna happen. |
| 20:15.27 | [TK]D-Fender | Lack of understanding and 1 little "oops".... and it can |
| 20:15.32 | Marquel | contexts are either inbound and can only call local clients or the other way round. an external->internal context can never dial back out again. |
| 20:15.41 | [TK]D-Fender | So be sure of what you're doing and what calls are supposed to look like |
| 20:16.13 | [TK]D-Fender | And that you don't make a silly copy-paste error and then espeailly leave that in a context taht stuff like sip.conf's [general] points to. |
| 20:16.21 | Marquel | [TK]D-Fender: so i need to define a peer, which does not have credentials for inbound and i can _only_ match with their hostname(!) |
| 20:16.42 | [TK]D-Fender | generally you can use the same as you use for outbound. |
| 20:16.52 | Marquel | okay. |
| 20:16.53 | [TK]D-Fender | insecure= tells it not to auth calls from that host |
| 20:17.00 | [TK]D-Fender | based on what I gave you |
| 20:17.04 | Marquel | understood. |
| 20:18.18 | [TK]D-Fender | And still lets you auth out |
| 20:18.45 | Marquel | one peer definition which has a context for inbound calls and is used for outbound calls. and then i just have to hope that they tell me the called number so the context can decide which extension to use? |
| 20:19.32 | [TK]D-Fender | No, they ALWAYS send you SOMETHING as the extension |
| 20:19.45 | [TK]D-Fender | it' might be the "s" your register suggested (if you register) |
| 20:19.57 | [TK]D-Fender | They might IGNORE that and just the DID you pay them for as the extension. |
| 20:20.23 | [TK]D-Fender | They might send somethng else entirely and have the target # in another header field. In that case you'll have to strip it out the hard way in the dialplan itself. |
| 20:20.35 | [TK]D-Fender | So like I said... you're just going to have to wait and see |
| 20:20.58 | Marquel | i figure i want isdn back. that was easy. |
| 20:24.35 | [TK]D-Fender | This COULD be easy. Just wait and see. |
| 20:25.21 | Marquel | oh well, i wonder what happens if i try to transfer a t38 fax over that line ;) |
| 20:27.13 | eric_hill | T.38 works if the carrier can prove it actually works with your number on your server. In all other cases, the sales people lie. |
| 20:27.14 | [TK]D-Fender | Does your provider support T.38? |
| 20:27.53 | Marquel | honestly - don't know and don't really care. |
| 20:28.54 | [TK]D-Fender | Not caring is a GREAT thing |
| 20:28.59 | [TK]D-Fender | Means les disappointment |
| 20:29.03 | [TK]D-Fender | less |
| 20:29.38 | Marquel | [TK]D-Fender: if it works, that's cool, keeps my faxing machine running and as i've learned in the not so distant past, having one is sometime a good thing. |
| 20:29.39 | *** join/#asterisk mjordan (mjordan@nat/digium/x-bbohdozyphtvqckx) |
| 20:29.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:29.58 | Marquel | but then i've used a two to five times max in the 4 years since i bought it. |
| 20:30.07 | [TK]D-Fender | yup, an unfortunate reality |
| 20:32.04 | Dovid | Anyone here use libss7? |
| 20:35.06 | [TK]D-Fender | And... it's checkout time, BBIAB |
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| 22:07.22 | TazzNZ | Dovid, what part of it ? |
| 22:08.24 | Dovid | TazzNZ: i have it set up on a sangoma card. it has been working for years. now i need add a link and I cant get the second link set up. no matter how i configure the second linkset it wont come up |
| 22:18.39 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:25.45 | TazzNZ | Dovid, can you pastebin your configs ? |
| 22:26.35 | Dovid | TazzNZ: http://pastebin.com/Vq0aqjsN |
| 22:26.45 | Dovid | i changed it a few times trying to figure it out |
| 22:28.42 | *** join/#asterisk rockydoggy (~rockydogg@c75.152.0-147.clta.globetrotter.net) |
| 22:32.00 | rockydoggy | Hi, is there anyway i can change my Contact URI port? My PBX reply to ACK request on 5060 but I want him to use 6000 instead |
| 22:34.06 | TazzNZ | Dovid, what does "dahdi_hardware" show ? |
| 22:34.27 | TazzNZ | rockydoggy, you want a port per contact ? |
| 22:34.37 | Dovid | TazzNZ: pci:0000:0a:01.0 wanpipe- 1923:0040 Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 22:34.42 | TazzNZ | or do you simply want to move SIP to another port ? |
| 22:34.52 | Dovid | TazzNZ: Port 1 is fine, i cant seem to get the second link up |
| 22:35.20 | TazzNZ | Dovid, yip - but where is your SS7 cards |
| 22:35.21 | TazzNZ | ? |
| 22:35.31 | Dovid | on the box. using a sangoma a101 |
| 22:35.51 | TazzNZ | you dahdi_hardware only showed 1 card thougt ? |
| 22:37.33 | *** join/#asterisk krapper (krapper@gibson.krapper.net) |
| 22:38.18 | krapper | Good evening.. how would one call many channel variables via one dialplan line entry? |
| 22:38.43 | Dovid | TazzNZ: It has two port son it |
| 22:38.43 | rockydoggy | TazzNZ: my inbound NAT is dooing 6000 to 5060 for incomming traffic, this is perfect. // My equipment doesn't support NAT for outbound traffic, is there anyway I can ask PBX to reply on 6000 instead of 6000? or set iptables to do it? |
| 22:38.44 | [TK]D-Fender | krapper, Define "call them" |
| 22:38.45 | krapper | aside from an include with many Set commands is there anything to look in? |
| 22:38.58 | rockydoggy | instead of 5060* |
| 22:39.18 | TazzNZ | Dovid, but that is an Analog card that is shown |
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| 22:39.32 | TazzNZ | ss7 (from what I know) only runs on digital :) |
| 22:39.38 | Dovid | TazzNZ: I know. seems thats how the BIOS see's it but It's an 102 |
| 22:40.21 | krapper | Hey [TK]D-Fender, looking to define many variables and load them into the channel with one dialplan entry |
| 22:40.21 | Dovid | TaaNZ: http://pastebin.com/WkcnPKca |
| 22:40.41 | TazzNZ | rockydoggy, I havn't heard of a way to tell asterisk to change the port for the external IP |
| 22:41.00 | TazzNZ | I would suggest that you try setting the external IP in sip.conf with the port |
| 22:41.06 | TazzNZ | as example, 1.2.3.4:6000 |
| 22:41.35 | TazzNZ | Dovid, ah - that is better :) |
| 22:41.50 | [TK]D-Fender | krapper, One at a time is how you'll have to do this. |
| 22:44.32 | TazzNZ | Dovid, I suspect that your issue is because the card is reported wrong back to dahdi |
| 22:44.33 | rockydoggy | TazzNZ this part is working well, the register is beeing sent on 6000 and I can make calls perfectly. After 30 sec, the calls are disconnecting, because my ACK reply still show 5060. |
| 22:44.51 | Dovid | TazzNZ: Then how do you explain th other port working OK? |
| 22:44.54 | TazzNZ | rockydoggy, read my messages |
| 22:45.06 | TazzNZ | Dovid, since you are using 1 port, 1 port is mapped ok |
| 22:45.18 | Dovid | TazzNZ: mapped == ? |
| 22:45.30 | TazzNZ | or presented |
| 22:46.28 | TazzNZ | this is my machine, with 2 analog cards: |
| 22:46.33 | TazzNZ | # dahdi_hardware |
| 22:46.33 | TazzNZ | pci:0000:03:01.0 wanpipe- 1923:0040 Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 22:46.33 | TazzNZ | pci:0000:03:02.0 wanpipe- 1923:0040 Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 22:47.08 | [TK]D-Fender | A200 = analog |
| 22:47.14 | TazzNZ | I would expect to see at least the right model in there on your system |
| 22:47.42 | TazzNZ | Dovid, what if you run "dahdi_span_types list" |
| 22:48.39 | Dovid | TazzNZ: Command not found. This is my config: http://pastebin.com/Jvgs7TQe |
| 22:49.10 | TazzNZ | Dovid, you don't have dahdi_span_types ? |
| 22:49.34 | Dovid | TazzNZ: No very old version of dahdi. i cant mess with it since it's [rpduction |
| 22:49.40 | *** part/#asterisk kharwell (kharwell@nat/digium/x-kfkuveztiojmudwz) |
| 22:50.08 | TazzNZ | how old is very old ? |
| 22:51.26 | [TK]D-Fender | Dovid, What file is that? |
| 22:51.38 | TazzNZ | Dovid, also - your clocking source is "wrong" on the second span |
| 22:51.49 | Dovid | TazzNZ: what line is that on? |
| 22:51.59 | TazzNZ | [TK]D-Fender, looks like chan_dahdi.conf |
| 22:52.14 | [TK]D-Fender | TazzNZ, I want proof |
| 22:52.19 | TazzNZ | Dovid, span=2,2,0,ccs,hdb3 should read span=2,1,0,ccs,hdb3 - imho |
| 22:52.25 | [TK]D-Fender | I will NOT accept an assumption about it |
| 22:52.40 | Dovid | TazzNZ: Let me try that. one sec |
| 22:52.42 | TazzNZ | fair enough |
| 22:54.00 | Dovid | http://pastebin.com/Jvgs7TQe is system.conf |
| 22:56.28 | Dovid | TazzNZ: Changing the timing didnt work |
| 22:57.24 | TazzNZ | Dovid, you still didn't say how old the dahdi is ? |
| 22:58.37 | [TK]D-Fender | krapper, Not going to happen |
| 22:58.47 | Dovid | dahdo 2.4.0 |
| 22:58.48 | Dovid | brb |
| 22:58.57 | [TK]D-Fender | needs to want out for accidental scroll-backs... |
| 22:59.36 | TazzNZ | that doesn't make sense ? |
| 23:00.43 | TazzNZ | geepers Dovid - that is almost 3 years old |
| 23:00.49 | Dovid | yup |
| 23:00.55 | Dovid | now asterisk wont start |
| 23:00.58 | Dovid | i get [2015-06-16 00:59:47] ERROR[25986] chan_dahdi.c: Unable to register channel '32-47' |
| 23:01.08 | Dovid | so i guess asterisk doesent see the second port |
| 23:01.11 | TazzNZ | yeah - that is what I suspected at the start |
| 23:01.18 | TazzNZ | the "mapped" thing |
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| 23:02.33 | TazzNZ | Dovid, did you get that version from within Asterisk ? |
| 23:02.42 | Dovid | i looked at what i had in /usr/src |
| 23:03.15 | Dovid | TazzNZ: Nope that was an issue in my config |
| 23:03.52 | Dovid | but even before asterisk comes up wanrouter is showing one port up and the other down so it's either a wanrouter issue or a dahdi issue |
| 23:04.34 | TazzNZ | well - normally (and this happened on another system I help maintain) wanpipe and dahdi is downloaded and updated at install time |
| 23:04.37 | TazzNZ | and never again |
| 23:04.56 | TazzNZ | I am surprised that it shows both ports |
| 23:05.30 | Dovid | One other interesting note: http://pastebin.com/xKcdAmBg |
| 23:05.35 | TazzNZ | but it wont allow you to config them |
| 23:05.38 | Dovid | but agai the first port works OK |
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| 23:20.41 | TazzNZ | Dovid, imho, I would arrange an upgrade of wanpipe and dahdi |
| 23:21.06 | TazzNZ | other than that, you could log a bug with Sangoma re the card and see if they have another option |
| 23:22.57 | Dovid | TaaNZ: Thanks. I have a ticket opne with them |
| 23:29.20 | *** join/#asterisk iulhk (~iulhk@182.191.181.240) |
| 23:31.26 | iulhk | can i add custom sip header in manager.conf , or i have to add in dialplan? i just want if client A stop video, then client B should get the intimation that client A has stopped video ? |
| 23:40.07 | *** join/#asterisk italorossi (~Adium@179.211.186.236) |
| 23:58.23 | TazzNZ | iulhk, that sounds like a manager event |
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