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| 00:40.02 | cleb | I have 2 Asterisk PBX's, with an IAX trunk between. PBX A has a Sip trunk to the PSTN, and extensions 76XX. PBX B has extensions 2XXX. I'm trying to set the caller ID of calls bound for the PSTN SIP Trunk based on the extensions that come from PBX B, like this exten => _1NXXNXXXXXX/2XXX,1,Set(CALLERID(num)=2035002020). Its not setting the call ID correctly. |
| 00:41.18 | cleb | When I call 76XX from 2XXX, the 76XX phone shows the correct 2XXX CID extension, so thats coming over the IAX trunk, its just not matching it when I make outbound calls over the SIP trunk from PBX B. |
| 00:42.36 | Gosmac | Hey i have an interesting topic to discuss here. |
| 00:42.36 | Gosmac | The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . |
| 00:42.36 | Gosmac | the problems that i faced with this is the following and i hope i could get an advise here. |
| 00:42.38 | Gosmac | asterisk 13 vanilla version has some issues marking the video packets this complain web browser specially VP8 codecs so a friend of mine help me to patch res_rtp_asterisk and now asterisk is marking video streams :) it just mark video packets not touch anything else and web browser show video on web page now Iâm using online demo http://tryit.jssip.net/ is stable and get more updates than sipml5. so i try echo() dialplan test and everything |
| 00:42.41 | Gosmac | <PROTECTED> |
| 00:42.42 | Gosmac | i have two questions and i hope you could give me some advise. |
| 00:42.45 | Gosmac | 1) after marking video packet Iâm able to make Dial() between two webrtc peers but i get one way audio and video on callee party, âafter 3 minutes on callâ i get two way audio and video on all parties seems to be not just a problem on a missing keyframe. |
| 00:42.46 | Gosmac | <PROTECTED> |
| 00:42.48 | Gosmac | <PROTECTED> |
| 00:42.50 | Gosmac | 2) after marking video packets i realize that when you make a call with video and you involve on dialplan an application like playback or music on hold any application that played audio files (audio and video never work). |
| 00:42.52 | Gosmac | <PROTECTED> |
| 00:42.54 | Gosmac | 2.1) asterisk is muggling the audio and video streams ? |
| 00:42.56 | Gosmac | This is good information for all guys out there that wants to support video on webrtc in asterisk 13 |
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| 02:41.35 | LemensTS | Someone wants me to send a cdr data via IP to hyperterminal on another server. Would I want to user Asterisk manager API for this? |
| 02:42.32 | WIMPy | I think you should re-phrase that. |
| 02:42.52 | oli-work | ^ |
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| 09:42.59 | bkruse | bkruse |
| 09:43.02 | bkruse | I am |
| 09:43.29 | bkruse | mjordan: Thank you! I will check that |
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| 10:12.28 | mjt | so, do I understand it right that in order to hear/see something sensible on the callee's phone when Dial() somehow failed, I should add something like Goto(xx-${DIALSTATUS}) after the Dial() and define xx-NOANSWER, xx-BUSY, xx-CHANUNAVAIL etc extens too? |
| 10:13.02 | mjt | on the calleR's phone ofcourse :) |
| 10:13.14 | mjt | the one who makes the call that is |
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| 10:21.49 | Marquel | mjt: probably yes. |
| 10:22.54 | mjt | probably? ;) |
| 10:24.39 | WIMPy | Generally there is DIALSTATUS and HANGUPCAUSE to find out what happened. |
| 10:25.00 | Marquel | mjt: depends on what you want to achieve. |
| 10:25.43 | mjt | right now I have Dial() at the end of the dialplan. And when it fails, the caller don't see anything, his phone just hangs up |
| 10:26.09 | mjt | so he repeats dialing and finally asks the administrator why his phone doesn't work |
| 10:26.27 | WIMPy | Classic |
| 10:27.11 | mjt | not even Busy signal (the short beeps) is heard |
| 10:27.39 | WIMPy | That's not the way SIP works. You have to read your phones display. |
| 10:27.51 | WIMPy | And usually you have to be fast at that :-( |
| 10:27.55 | RPerre | Hi. How do I disable transcoding? |
| 10:28.22 | WIMPy | RPerre: By using the same CODEC on both call legs. |
| 10:29.08 | RPerre | WIMPy, thanks |
| 10:37.57 | Marquel | honestly forgot what he does in his dialplans. |
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| 11:01.07 | Rad | Hi , how can i send a sip response with a specific message , our voip operator sends same response code but with different messages (Ex : SIP/2.0 404 unknown user account / SIP/2.0 404 Not Found) , same code (404) but different messages |
| 11:01.10 | Rad | how can i do that ? |
| 11:01.45 | Rad | I know i can pass an argument to HangUp cmd, which is the cause code , but what about the message ? |
| 11:04.24 | WIMPy | rad: You can't |
| 11:07.37 | Rad | Its about limiting outgoing calls to a peer , so i need to inform the client that they have reached outbound calls limit. what hangup cause code is the most clear to "too many outgoing calls" ? (I am not using call-limit but GROUP_COUNT functions) |
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| 11:09.41 | WIMPy | 34 |
| 11:10.06 | Rad | Thank you |
| 11:10.48 | WIMPy | 37 would also work. Don't know what text Asterisk adds. |
| 11:11.02 | WIMPy | 47, sorry. |
| 11:11.58 | Rad | any link on which i can see all differents codes and texts ? |
| 11:12.12 | WIMPy | Q.850 |
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| 11:12.45 | Rad | thx |
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| 11:15.48 | Marquel | fun fact: you can tell isdn exchanges fake codes and they are honored. HangUp(1) on an ISDN causes the network to tell the caller "the number you have dialed is not assigned", though exactly that number has just told the isdn to do that... |
| 11:16.29 | WIMPy | Which does make sense if you have DDI with unused numbers. |
| 11:17.02 | WIMPy | But not all cause values will be accepted. |
| 11:18.23 | Rad | our customer needs to detect when they have reached the limit of outgoing calls by scanning sip frames |
| 11:18.30 | Rad | so they need a specific message |
| 11:19.12 | WIMPy | And then there's the location which would reveal to the caller that the message didn't come from the network. Locations are filtered. |
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| 11:21.35 | r00f | can't they just count number of calls on respective sip peer? why do they need to scan sip frames? |
| 11:22.17 | Marquel | WIMPy: nope, PTMP connection, so either an MSN is assigned or it is not (and in the latter it wouldn't reach my pbx). |
| 11:23.22 | Marquel | WIMPy: let's say most callers don't have the equipment to figure this out and are happy to believe what the kind network voice tells them. |
| 11:23.55 | WIMPy | Yes, I like to use to to reject telemarketeers. |
| 11:23.56 | Rad | r00f our client is a diy pbx , they usually have problems on their system , so they would like to know what cause precisely blocks theirs calls |
| 11:25.09 | Rad | r00f i also like to be crystal clear when sending responses... i need the customers to know the reason , instead of calling our tech every now and then |
| 11:25.12 | WIMPy | Marquel: But you can even fool your telco, wich is fun :-) |
| 11:25.44 | Marquel | WIMPy: that's one, annoying relatives is another ;) |
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| 11:28.47 | r00f | Rad: so make HangUp(34) and let them know that there is congestion, and they should interpret it as "you are out of capacity" (which 34 code actually means, as i understand it) |
| 11:29.45 | r00f | if their diy pbx cannot parse hangup code, still they can see it inside of sip frames, as you wanted |
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| 11:31.22 | cervajs2 | hi, i'm migrating to asterisk 13. i was using exten h to add technical info to the CDR (like packet loss, jitter, ...). this isnt possible now. i must change "endbeforehexten" to no which creates second CDR. is there some way to update the first cdr? |
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| 11:41.47 | newtonr | cervajs2, I don't understand your question. Are you trying to update CDR for Party A or B? |
| 11:42.24 | cervajs2 | there is only one CDR for simple partyA->partyB call |
| 11:44.36 | Demon_VoIP | i have try using AGI and db write... No success :( CDR is not in table |
| 11:44.52 | cervajs2 | the change from pre-asterisk12 is "When a CDR is finalized, no further modifications can be made to the CDR by the user or Asterisk." this was possible on pre-asterisk12 |
| 11:45.06 | cervajs2 | https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification#Asterisk12CDRSpecification-CDRLifetime |
| 11:46.25 | Demon_VoIP | now i also loose this usefull info :( Because subscribe to the issue |
| 11:46.25 | cervajs2 | i'm asking if there are ways to modify this CDR. maybe i can listen to AMI/ARI and update it from another application like node.js |
| 11:47.23 | Demon_VoIP | cervajs2, cdr writes in separate thread (i guess).. and you cannot be sure it exists from AGI |
| 11:47.44 | cervajs2 | i'm not talking about AGI |
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| 11:48.48 | Demon_VoIP | AMI has some events about channel stats. I caught them noticed, but did not have time to study |
| 11:51.17 | newtonr | cervajs2, I'm not sure. You'll want to ask mjordan when he is around. |
| 11:51.26 | Demon_VoIP | This technical information is very important. It would be wonderful if all the field is declared in the CDR table it was written automatically |
| 11:53.14 | newtonr | cervajs2, in the meantime you might try from a hangup handler to see if you can do it from there. https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers |
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| 11:56.13 | r00f | cat /etc/iproute2/rt_tables |
| 11:56.36 | r00f | ohh, my bad. |
| 11:57.14 | cervajs2 | i tried hangup handlers. it's the same as h exten. CDRs are finalized immediatelly after hangup |
| 11:57.43 | Rad | r00f thank you |
| 12:00.54 | Demon_VoIP | cervajs2, newtonr, where is no AMI events: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+AMI+Events Event RTCPReceived is called very often and provides current information rather than "final" |
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| 12:12.26 | rox | Hello, does anybody have an idea, how to get Asterisk to log the caller ID that was changed in the dialplan? |
| 12:13.02 | WIMPy | Verbose() |
| 12:13.14 | WIMPy | Or Log() |
| 12:13.27 | rox | e.g. a call comes in, in the dialplan a Set(CALLERID(num)=12345) is called and a Dial() afterwards |
| 12:13.57 | rox | asterisk still logs the call with the caller Id that came in, rather than the caller Id that was assigned in the Set() call in dialplan |
| 12:14.17 | rox | WIMPy: i meant the CDR log |
| 12:14.28 | WIMPy | You should have said so. |
| 12:14.51 | rox | WIMPy: in messages, the correct caller ID is logged, in CDR, however, it is nearly impossible to get Asterisk to log what you want it to |
| 12:15.00 | WIMPy | I don't think that CDR field is writable. But You can check the CDR function. |
| 12:15.19 | rox | WIMPy: I apologize for lack of clarity |
| 12:16.04 | rox | WIMPy: My issue is this: there are two calls, the one that comes in, and the one that goes out, the one thast comes in is naturally logged with the caller ID that I assigned |
| 12:16.26 | rox | WIMPy: the call, that goes out, however, goes out with the assigned caller ID, but is logged with the caller ID, that came in |
| 12:16.43 | rox | WIMPy: it is weird |
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| 12:17.26 | rox | the callee's device displays the caller ID, that was assigned in the dialplan, but Asterisk still logs the previous caller ID |
| 12:17.40 | newtonr | cervajs2, I asked mjordan and it looks like the documentation is accurate. You can't add anything to the finalized CDR after hangup. You can only add to a new CDR for that channel. You would need to do post-processing if you wanted to merge those two CDR records. |
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| 12:23.25 | cervajs2 | newtonr, ok thank you. its clear now |
| 12:23.52 | rox | WIMPy: you were right, CDR(src) is not writable |
| 12:25.12 | r00f | i also rely on that information in cdr. didn't know it was cancelled in 13. will have to stick to 11 until its end of life, hopefully there will be some how-tos by that time |
| 12:27.26 | mjordan | It isn't going to change. |
| 12:27.28 | mjordan | sorry. |
| 12:27.50 | mjordan | Unless someone wants to write a patch for it, and I can say with certainty that you are going to have a difficult time getting that to work without breaking 8 billion other things. |
| 12:28.18 | mjordan | You can add post-processsing information, but you have to do it on a new CDR for that channel. If you do that, then you have to post process your records. |
| 12:29.19 | r00f | yeah, that is clear |
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| 13:27.59 | bkruse | Hey guys - the /sounds/soundID thing that mjordan was talking about earlier - does that function actually check to see if the file exists at that time? |
| 13:28.43 | bkruse | If I have a message that is comprised of 10 audio files, checking for 10 audio files before each call may be pretty invasive (me thinks) |
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| 15:42.35 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 15:48.57 | *** join/#asterisk ipalmer (~IceChat77@host81-133-140-79.in-addr.btopenworld.com) |
| 15:50.26 | ipalmer | Hi all, may be a silly question but is it possible to make a single call ring in 2 different places at once, ie call comes in enters queue after 30 seconds, leaves queue and at the same time enters an overflow queue and dials across to a different PBX? |
| 15:54.19 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 15:54.50 | WIMPy | You can use local channel to dial back in to the dialplan. |
| 15:55.48 | ipalmer | WIMPy: thanks, so in effect I'll have the same call ringing in 2 different places? |
| 15:56.16 | WIMPy | As many as you want. |
| 15:56.47 | ipalmer | ok, I have a lab setup here I'll give that a try. Thanks |
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| 16:17.53 | Kunsi | oh, just found out i've got an issue regarding snom phones and call pickup ⦠i've set notifycid=yes in sip.conf, but when calling *876 to pick up calls from 76, i don't get the number of the caller, just *876 in display. how to debug that? |
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| 16:30.24 | [TK]D-Fender | that option is not related |
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| 16:30.33 | Kunsi | hm? |
| 16:30.55 | Kunsi | i've read if you enable notifycid, you'll get the callers number in your snom display |
| 16:33.57 | [TK]D-Fender | while watching a call arrive via presense |
| 16:34.25 | [TK]D-Fender | This has nothing to do with updating your CID when actually placing a call |
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| 16:45.04 | *** join/#asterisk kchehab (~kchehab@77.42.223.180) |
| 16:45.07 | kchehab | hi ppl |
| 16:45.57 | Kunsi | so the snom phone should receive caller id at same time when it's getting notified about ringing state? (when subscribed to an extension) |
| 16:46.16 | rrittgarn | rpidupdate would deliver what you're looking for Kunsi |
| 16:47.31 | kchehab | i am trying to bind 3 bluetooth hci devices to three mobile phones using chan_mobile ,only the first mobile can be scanned as phone .at the first place i pair the three devices to asterisk server, by setting the interfaceces discoverable and pair each phone to the device |
| 16:47.46 | [TK]D-Fender | Kunsi: Yes |
| 16:48.01 | kchehab | i set the asteris discoverrable using this command |
| 16:48.03 | kchehab | dbus-send --system --type=method_call --print-reply --dest=org.bluez /org/bluez/hci1 org.bluez.Adapter.SetMode string:discoverable |
| 16:48.11 | Kunsi | [TK]D-Fender: how to verify that? sip debug? |
| 16:48.21 | [TK]D-Fender | yes |
| 16:48.32 | kchehab | and i repeat the same command with changing /org/bluez/hci2 & /org/bluez/hci0 |
| 16:48.48 | kchehab | please can you advise ? |
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| 16:53.42 | stefan27 | does increasing verbose=0 in asterisk.conf to verbose=3 have any effect on load at all if you never open a remote console? |
| 17:03.47 | Kunsi | stefan27: yes. there |
| 17:03.57 | Kunsi | there's verbose logging, for example |
| 17:05.05 | mjordan | stefan27: you're changing the logging in the core of Asterisk. Logging incurs overhead and work. |
| 17:05.18 | *** join/#asterisk ASUchander1 (~asuchande@cpe-071-070-227-251.nc.res.rr.com) |
| 17:06.18 | ASUchander1 | Hi, I have an asterisk system that has been working great for a long time. However in the last few days I've found that when I make a phone call (even from one extension to another) I found that outbound audio does not work for the extension that receives the call. any ideas as to what might cause this? |
| 17:06.30 | pjensen00 | NAT |
| 17:06.37 | pjensen00 | one way audio has usually been NAT issues for me |
| 17:07.22 | pjensen00 | Change anything with your network recently? |
| 17:07.44 | ASUchander1 | That's the wierd thing - the network config and firewall have been more or less static for years (literally, years) |
| 17:07.50 | pjensen00 | Hrm. |
| 17:08.05 | [TK]D-Fender | Look anyway |
| 17:08.06 | pjensen00 | Well, NAT is definitely not the only thing that causes one way audio. Just the most common in my experience |
| 17:08.12 | ASUchander1 | I rebooted the firewall and the asterisk box...no dice there |
| 17:08.44 | ASUchander1 | [TK]D-Fender: pjensen00 - I spent a bunch of time looking at firewall logs, and onfigs and AFAICT all look good/normal. |
| 17:08.46 | pjensen00 | Another reason I've stumbled across was that our router happened to put VOIP traffic on low priority one day |
| 17:09.47 | [TK]D-Fender | ASUchander1: That is one set of eyes and nothing else to consider. |
| 17:11.35 | ASUchander1 | pjensen00: Thanks - that was it. For some reason the hsfc queue had a low priroity |
| 17:11.53 | pjensen00 | glad to help! |
| 17:11.53 | ASUchander1 | decreasing it seemed to fix things. |
| 17:12.19 | pjensen00 | Being bitten in the butt by these things makes me immediately start finger pointing these things. :P |
| 17:13.37 | ASUchander1 | Very odd that that would be the case - theres not really enough traffic on the network to need to drop those packets, etc. |
| 17:15.51 | ASUchander1 | Actually, perhaps I need to take that back. I have one DID where inbound audio works. The other DID's it does not. WFT? |
| 17:17.03 | pjensen00 | still? |
| 17:17.15 | pjensen00 | Hrm. |
| 17:17.30 | ASUchander1 | Yeah, the wierd thing is that they're all on the same SIP registration w/flowroute |
| 17:17.50 | ASUchander1 | So aside from the routes, they ought to all work the same.. |
| 17:18.01 | pjensen00 | Is flowroute having any issues currently? I've had a lot of issues with them in the past. |
| 17:18.06 | pjensen00 | *grumbles slightly |
| 17:22.10 | pjensen00 | I can't speak for everyone, but even with low traffic on our router having VOIP set to a low priority messed up a lot of stuff for us. Are you getting absolute silence on your one way audio or is it crackly |
| 17:22.59 | ASUchander1 | pjensen00: absolute silence |
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| 17:24.06 | pjensen00 | All using the same SIP profile correct? Are the physical endpoints all on the same network including the one that IS getting two way audio? |
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| 17:28.16 | ASUchander1 | Yes. Same network, they both route to the same phone, same asterisk server |
| 17:28.27 | ASUchander1 | Literally, the same SIP registration with flowroute |
| 17:28.35 | ASUchander1 | It's bizarre |
| 17:29.34 | pjensen00 | .... that one has me stumped. |
| 17:36.01 | hirogen | anyone seen silverlight ? |
| 17:36.05 | hirogen | not seen him |
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| 17:40.05 | pjensen00 | oh, thought you meant the technology at first. I was about say "...... yes." |
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| 17:51.49 | kchehab | please anyone can elighten me :P for multi bluetooth hci with asterisk chan_mobile.so |
| 18:05.24 | kchehab | its working by using one bluetooth devices |
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| 18:28.00 | ASUchander1 | FYI, from flowroute: We have seen some changes in the way that our peering network is currently handling audio deliverance. Your system is currently accepting multiple codecs in itâs answer signal, creating an ambiguous codec selection on the session. This is causing trouble with the audio stream. You can resolve this by updating your Flowroute trunk peer details in your phone system configuration to accept onl |
| 18:28.01 | ASUchander1 | y a single codec in the answer signal. |
| 18:28.23 | ASUchander1 | Humph. I'll try to change my config to use a single codec and see what happens. pjensen00 - thanks for the help |
| 18:29.15 | pjensen00 | ASUchander1: Boy that stinks. Flowroute's customer service actually being helpful is nice to hear though |
| 18:29.25 | pjensen00 | good luck |
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| 18:30.40 | ASUchander1 | BTW, switching to ulaw only fixed the issue for me |
| 18:30.49 | ASUchander1 | I had it on ulaw and g729 before.. |
| 18:31.02 | pjensen00 | Hrm, I think I have... let me check. |
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| 18:32.01 | pjensen00 | http://pastebin.com/XH0psb4E |
| 18:32.38 | pjensen00 | We're experimenting with webRTC so that's where the vp8 is coming from. |
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| 18:57.05 | kchehab | how to solve my problem if my bluetooth devices is paiing and not connectin |
| 18:57.10 | kchehab | connectiong |
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| 19:41.08 | pjensen00 | When connecting to the ARI web socket, is there a way to gracefully shut down the socket via a command to asterisk? Currently I just tear down the connection client side and asterisk says "Looks like this socket is no longer active" |
| 19:41.22 | pjensen00 | No performance issues I've noticed, but wondering if that's the "correct" way to do it. |
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| 19:48.28 | DivideBy0 | pjensen00: there's a close method in the spec, but I just tear down the connection too - what language are you using? |
| 19:48.39 | pjensen00 | Perl. |
| 19:49.15 | pjensen00 | where is the close method? Maybe I'm just daft, I didn't see it when I searched |
| 19:49.46 | DivideBy0 | https://tools.ietf.org/html/rfc6455#section-5.5.1 |
| 19:50.10 | DivideBy0 | which perl module are you using? Protocol-WebSocket? |
| 19:50.29 | pjensen00 | AnyEvent::WebSocket::Client |
| 19:50.50 | mutilator | http://www.dxsoul.com/product/gl-inet-wi-fi-router-openwrt-ssh-usb-64mbyte-ram-wireless-repeater-pcb-board-green-901346531#.VQHt7PnF9EY anyone know where i could get this.... except that provides POE? |
| 19:51.04 | pjensen00 | I see, it's not an asterisk specific command. |
| 19:51.17 | DivideBy0 | it's disconnect in protocol-websocket, I can look in anyevent |
| 19:51.22 | pjensen00 | Ok, that's what I was wondering. thanks! |
| 19:51.28 | pjensen00 | no need, I can take it from here. |
| 19:51.50 | DivideBy0 | welcome! good luck |
| 19:52.18 | pjensen00 | danke mein freunde |
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| 21:25.28 | y_ateya | Hi, I got these errors multiple times "[Mar 12 22:59:25] WARNING[31657][C-00086736] res_rtp_asterisk.c: RTP Read too short (-243, expecting 72 |
| 21:25.28 | y_ateya | [Mar 12 22:59:25] WARNING[31657][C-00086736] res_rtp_asterisk.c: RTP Read too short (-243, expecting 48" |
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| 22:09.24 | AL13N | i don't understand... i have zero tls anywhere in asterisk, it's all configured, but no ports are opened... (in netstat -lntup ) ... not in http.conf ; not in sip, nothing... how is this failing? |
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| 22:21.42 | newtonr | AL13N, for anyone to help, you'll probably need to pastebin your configs and an Asterisk log showing the SIP trace and verbose output |
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