| 00:00.17 | kfife | kunwon1: : Tried that too. The automixmon filename appears to be a composite of 2 channel variables, plus a unique ID |
| 00:00.36 | kfife | The unique ID appears to be a timestamp, |
| 00:00.54 | kfife | But the timestamp is not the same timestamp that is the ${UNIQUEID} of the call |
| 00:01.12 | kfife | I believe it's the timestamp that the AutoMixMon was invoked |
| 00:01.34 | kfife | However, I can't effing find that effing number, even though it's right there on the console. |
| 00:01.48 | kfife | Correction: effing console. |
| 00:01.52 | kunwon1 | the only other thing i could suggest is try to add a hangup handler and run DumpChan() there - i think some things aren't set until later in call processing, especially when it comes to recording, but this is just a guess |
| 00:02.03 | kunwon1 | otherwise it may not be available in dialplan |
| 00:02.50 | kunwon1 | i found this which has some variables concerning monitor filenames, but don't think it's all too helpful https://wiki.asterisk.org/wiki/display/AST/Various+application+variables |
| 00:03.04 | kfife | kunwon1: Good idea, but I tried that too. Piissing me off somethign fierce. There are more than one of unanswered posts about this. |
| 00:03.49 | kunwon1 | okay, turn on verbose logging, write a shell script to pull it from the log, and call that with System() :) |
| 00:04.04 | kunwon1 | please don't actually do that |
| 00:04.12 | kfife | I'm trying to not need to write my own dynamic feature if it's not necessary. Otherwise I have to try to guess the filename or some ugly ${SHIT} |
| 00:04.24 | kunwon1 | yeah i think ugly things may be the only options left |
| 00:04.26 | kfife | Like your clever suggestion. Still lauging. |
| 00:04.28 | kunwon1 | or delving into the source code |
| 00:05.22 | file | what version? |
| 00:06.07 | kfife | 12. I looked in the source, but couldn't C where it was mapped to a channel variable. |
| 00:06.26 | file | TOUCH_MIXMONITOR_OUTPUT apparently |
| 00:08.35 | kfife | Righ?? Problem is, the little bastard is always null. |
| 00:08.49 | file | it's likely on the other channel |
| 00:09.19 | file | and you mentioned a different variable before |
| 00:09.32 | kfife | If I use chandump in the dialplan |
| 00:10.00 | kfife | which channel is it dumping? |
| 00:10.06 | kfife | There's no arg for that. |
| 00:10.09 | file | the channel on which it is executed. |
| 00:11.28 | kfife | file: I see your ponit. Let me try to grab the other channel |
| 00:13.48 | kfife | file; calling party is allowed to record in Dial(,,,X) , and the calling party dumps the channel on hangup. |
| 00:14.41 | kfife | Are you suggesting that automixmon is populating TOUCH_MIXMONITOR_OUTPUT on the channel the call is bridged to? |
| 00:14.49 | kfife | file ^^ |
| 00:14.57 | file | I don't know that code, but yes |
| 00:15.15 | kfife | Hmmm... Trying |
| 00:15.46 | file | I believe that MixMonitor is placed on the bridged channel, and the dialplan variable set there |
| 00:16.40 | kfife | I'd expect it to be placed on the bridged channel if x were passed, originating channel if X were passed. |
| 00:17.11 | file | that just controls who can initiate it |
| 00:17.30 | kfife | roger |
| 00:18.09 | kfife | -- AutoMixMonitor used to record call. Filename: auto-1516148212-3122977003-13124455902.wav |
| 00:18.58 | kfife | the 1516... prefix not in DumpChan() |
| 00:20.59 | kfife | and TOUCH_MIXMONITOR_OUTPUT still null. |
| 00:21.30 | kfife | Sh1t |
| 00:28.12 | kfife | ...although the prefix is the uniqueID, rounded up to the nearest second. |
| 00:28.24 | kfife | on this call at least. |
| 00:29.08 | kfife | But not on the next one. Close though. |
| 00:34.59 | kfife | It's clearly just a timestamp of the time the button is pressed. |
| 00:38.23 | file | https://github.com/asterisk/asterisk/blob/12/bridges/bridge_builtin_features.c#L321 |
| 00:38.32 | file | that's the code that triggers it and generates the filename |
| 00:40.04 | kfife | You're right. I see it now. |
| 00:40.22 | *** join/#asterisk TriJetScud (~TriJetScu@69.172.162.252) |
| 00:41.05 | kfife | You're also right about asking the wrong channel. |
| 00:41.14 | kfife | I can get the variable from the CLI |
| 00:41.23 | kfife | But not the dialplan |
| 00:41.53 | kfife | Have to check to be sure it's doing what I think it is. |
| 00:42.29 | kfife | I see it now. This call has 3 channels |
| 00:42.40 | kfife | Thanks for your help |
| 00:43.00 | file | moar channels |
| 00:43.18 | kfife | mo channels, mo problems |
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| 01:21.49 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:21.49 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 02:48.30 | kfife | Got it: So if my dialplan logic is executing on one channel (say, SIP), and it creates a call invoking say, a dahdi channel Dial(DAHDI/G1/...,X), the variable MIXMONITOR_FILENAME is part of the Dahdi channel, not the SIP channel where my dialplan logic is running. file was exactly right here. Question is, can I query a property of the peer channel from within the dialplan that's in the SIP channel? I think I can do this by c |
| 02:48.31 | kfife | alling using the local channel, but wondering is the former also possible? |
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| 06:55.37 | Dirk23 | [TK]D-Fender: Hi |
| 06:56.15 | Dirk23 | [TK]D-Fender: https://pastebin.com/MHkiu3fF Log from beginning of the call to Hangup, but no oneway audio |
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| 07:32.05 | Dirk23 | [TK]D-Fender Samot : https://pastebin.com/iB618XkR this one is with a one way audio, comming from external trunk |
| 07:34.32 | [TK]D-Fender | You don't have any debug for the actual inbound call. |
| 07:34.44 | [TK]D-Fender | You've missing half of the entire picture |
| 07:34.55 | [TK]D-Fender | and ... it's bed-time... |
| 07:34.57 | [TK]D-Fender | BBL |
| 07:34.57 | Dirk23 | yes, missed the incoming call |
| 07:35.08 | Dirk23 | but i recorded the ony way |
| 07:35.11 | [TK]D-Fender | You're also comparing 2 differnt kinds of calls |
| 07:35.22 | [TK]D-Fender | your first was private-private |
| 07:35.26 | [TK]D-Fender | This other one is different |
| 07:35.33 | Dirk23 | same errors |
| 07:35.43 | [TK]D-Fender | You shouldn't be comparing differnt things like that |
| 07:35.43 | *** join/#asterisk mubbashar84 (~mubbashar@115.186.165.218) |
| 07:35.48 | Dirk23 | ok |
| 07:38.48 | mubbashar84 | Hi All, I need some help regarding reading packets in asterisk. I need to read the answer packet to check for headers. Is there any possibility to do that? |
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| 08:43.55 | mubbashar84 | Hi All, I need some help regarding reading packets in asterisk. I need to read the answer packet to check for headers. Is there any possibility to do that? |
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| 11:31.39 | jamesaxl | HEllo |
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| 11:31.50 | jamesaxl | Where can we donate for asterisk? |
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| 11:54.42 | matt_ | does anybody know how to set a channel variable from an agi script? |
| 11:55.57 | matt_ | oh, set variable, didn't work :( |
| 12:00.22 | matt_ | ok it does, the script was failing |
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| 14:24.48 | bananapie | If I call WAIT FOR DIGIT repeatedly, but a user hits a DTMF between two "WAIT FOR DIGIT", I lose the digit. So if the user enters, say a PIN of 1234. But while I am parsing the 1 in my AGI, he enters 2 before I call WAIT FOR DIGIT, WAIT FOR DIGIT returns 3. This is a problem on my asterisk box because of it's slow processor where processing time can exceed time it takes to enter the next dtmf. |
| 14:26.58 | [TK]D-Fender | Don't use weak junk for your server. Or pick another method of entry |
| 14:27.10 | [TK]D-Fender | s/or/AND/ |
| 14:27.23 | [TK]D-Fender | darn... |
| 14:27.30 | [TK]D-Fender | there goes the punch-line |
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| 14:53.51 | *** mode/#asterisk [+o kharwell] by ChanServ |
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| 14:55.53 | *** mode/#asterisk [+o bford] by ChanServ |
| 14:56.02 | *** join/#asterisk guest1062 (~bounceman@185.32.9.250) |
| 14:56.41 | guest1062 | Hi, are srv records working/implemented in Asterisk 11? I can see an option that says DNS SRV Records: Yes in the sip settings so that is my qualified guess. |
| 14:57.12 | guest1062 | The issue still is when I try to call out it gives me acl.c:833 resolve_first: Unable to lookup 'mysrv.foobar.io' |
| 14:57.41 | guest1062 | I mean I can dig it from the server |
| 14:57.43 | guest1062 | works great |
| 14:58.04 | [TK]D-Fender | 11 = dead |
| 14:58.11 | Samot | There are no SRV records in Asterisk. |
| 14:58.17 | [TK]D-Fender | should not be used anymore |
| 14:58.32 | Samot | SRV lookup means it tags the _sip._ucp. part. |
| 14:58.40 | Samot | On to the rest of the domain. |
| 14:58.48 | Samot | It formats the FQDN for a SRV lookup |
| 14:58.52 | guest1062 | Damn, that is a shame. |
| 14:59.10 | guest1062 | Is there a workaround for this in Asterisk? |
| 14:59.16 | Samot | What do you mean? |
| 14:59.20 | Samot | What is the actual issue? |
| 14:59.36 | guest1062 | Maybe I understood you, sorry. So asterisk prepares the domain and the system performs the lookup? |
| 14:59.58 | Samot | If you have host=sip.domain.com and SRV Lookups enabled.. |
| 15:00.12 | Samot | Asterisk will do a DNS lookup with _sip._udp.sip.domain.com |
| 15:00.32 | Samot | It prepends the SRV prefix. |
| 15:01.03 | Samot | Do a DIG on sip.domain.com is not an SRV lookup |
| 15:01.11 | Samot | You have to do the lookup with the SRV prefix |
| 15:01.18 | Samot | And look for the SRV record. |
| 15:05.46 | guest1062 | Yeah my SRV record looks like this: _sip._udp.example.com |
| 15:07.26 | guest1062 | so I added example.com as my trunk, it did not like that and it did not seem to put the call through. |
| 15:07.49 | guest1062 | Then as srv is enabled it should add _sip._udp. to that domain |
| 15:09.01 | guest1062 | subscriber absent |
| 15:09.14 | guest1062 | dig srv _sip._udp.example.com gives me back my two records |
| 15:09.46 | file | chan_sip has shortcomings in SRV support where it will resolve things in a non-SRV fashion when it shouldn't |
| 15:12.12 | *** join/#asterisk jkroon (~jkroon@vc-nat-gp-n-41-13-66-80.umts.vodacom.co.za) |
| 15:21.11 | znoteer_ | I've an asterisk machine behind NAT with an iax2 connection to my provider. When I first set it up, some command I ran reported what IP address my provider saw me as, ie, my pubic IP. I thought it was "iax2 show registry". Does anyone know what I'm talking about, or am I completely lost. I'd like to see how my provider sees me |
| 15:22.10 | [TK]D-Fender | They'll see you as your WAN IP based on whatever your router uses for your server |
| 15:22.16 | [TK]D-Fender | Which you should already know |
| 15:26.40 | znoteer_ | yes, I already do know it, but it's a dynamic IP that changed recently, and I want to make sure that my provider has the new IP |
| 15:27.09 | znoteer_ | I'm looking for the command that reports how we are seen by the remote end |
| 15:28.06 | znoteer_ | I seem to recall that some command returned a column labelled, "Sees us as" or something like that |
| 15:30.40 | jkroon | has anybody seen cases where the AMI (over http at least) becomes non-responsive? |
| 15:31.40 | jkroon | ok, looks like it's rather the http module that's problematic rather than ami. |
| 15:32.21 | [TK]D-Fender | "sees us as" means nothing |
| 15:32.39 | [TK]D-Fender | if you REGISTER to them they see based on the IP you last sent that registration via |
| 15:33.11 | znoteer_ | well then, how do I resend the registration? |
| 15:33.29 | [TK]D-Fender | If your registration interval is short enough it will eventually expire and reregister and they'll see you coming from a possibly new IP |
| 15:33.32 | jkroon | znoteer_, iax2 shows registry shows it for iax2, not aware of anything similar for SIP because the information isn't available from the protocol. |
| 15:35.17 | znoteer_ | [TK]D-Fender: and the registration interval is the "Refresh" column when you run "iax2 show registry"? |
| 15:35.37 | znoteer_ | jkroon: I'm using iax2 |
| 15:35.52 | [TK]D-Fender | What does it say? |
| 15:36.13 | jkroon | as far as I know it's smart enough to due to the ping-poing nature of the qualify statement know when to re-register. |
| 15:36.18 | znoteer_ | 60 |
| 15:36.44 | znoteer_ | I guess that's seconds |
| 15:36.49 | jkroon | yes |
| 15:39.32 | znoteer_ | well here's the problem I'm trying to get to the bottom of. Incoming calls no longer make it to my asterisk server. I can't think of anything that I've changed recently. If you call my DID, it just rings and rings. The console on my end shows no activity, no incoming call. My firewall is set to forward iax traffic to the asterisk box, but nothing appears to reach me |
| 15:39.59 | jkroon | iax2 set debug on, and correlate with tcpdump. |
| 15:40.07 | jkroon | if that doesn't get to you, then something is wrong on your firewall. |
| 15:40.30 | jkroon | then, check and make sure your connection tracking on your firewall isn't buggered (old linux kernel bug). |
| 15:41.07 | jkroon | https://forum.mikrotik.com/viewtopic.php?t=43557 for one, and http://jkroon.blogs.uls.co.za/it/voip/connection-tracking-problems |
| 15:41.43 | znoteer_ | jkroon: I thought of setting iax2 debug on, but there is nothing to debug on the asterisk console. Do you think it is still worthwhile in that case? |
| 15:42.12 | jkroon | yes, because the debug will tell you what iax2 packets asterisk sends and receives, and just as importantly: if and why it discards them. |
| 15:42.35 | znoteer_ | ok, I see. I'll try it |
| 15:43.03 | jkroon | to confirm the connection tracking bug, run tcpdump just outside your firewall and see if the source IP address of the packet is correct. |
| 15:43.40 | jkroon | if you can't do that, assume it's buggy. workaround: shut your asterisk server for ~10 minutes, and start it back up. |
| 15:43.59 | znoteer_ | isn't connection tracking only useful for outgoing traffic to make sure that any related reply comes through? |
| 15:44.01 | jkroon | if that works, point your firewall admins at the above two links. |
| 15:44.21 | jkroon | znoteer_, yes, and you did mention that you're behind a firewall. |
| 15:44.32 | jkroon | and you implied that it performs NAT. |
| 15:44.57 | jkroon | NAT => connection tracking. |
| 15:44.58 | znoteer_ | yes, but outgoing is working just fine. I can make calls, I just can't receive them |
| 15:45.06 | znoteer_ | Ah, ok |
| 15:46.00 | jkroon | ok, that is odd. what does "iax2 show registry" say? |
| 15:46.20 | jkroon | ok, so regarding my issue, it seems there is an issue in main/http.c |
| 15:46.25 | jkroon | just not sure what ... :( |
| 15:47.11 | *** join/#asterisk yoavz (~yoavz@185.187.161.165) |
| 15:47.17 | jkroon | netstat shows exactly 11 open connections, with send data, but the other side is long dead. |
| 15:49.18 | *** join/#asterisk jeev (~j@unaffiliated/jeev) |
| 15:54.28 | znoteer_ | jkroon: here's the output of "iax2 show registry" https://paste.debian.net/hidden/fc9ce32b |
| 15:55.44 | [TK]D-Fender | You're registered. Go prove your firewalls aren't screwed |
| 15:55.48 | [TK]D-Fender | Go place a call |
| 15:55.52 | [TK]D-Fender | go get a dump from it |
| 15:56.01 | [TK]D-Fender | go ask your provider where they are sending to |
| 15:56.03 | *** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com) |
| 15:56.17 | jkroon | no, that can't be right. is both the Host and Perceived values the same? |
| 15:56.28 | jkroon | or did your mangling just hide stuff from us? |
| 15:56.34 | [TK]D-Fender | Go prove the IP's are right |
| 15:56.36 | znoteer_ | jkroon: yes, they are the same |
| 15:56.43 | jkroon | can't be. |
| 15:57.19 | znoteer_ | that is my recollection too, that they weren't the same when I first set this up, but they are the same now. I don't know why |
| 15:57.22 | jkroon | perceived is where your provider will send to (including port number), so what you're saying is that your provider is seeing you as them - which makes absolutely no sense. |
| 15:58.02 | jkroon | x.y.4.8:4569 Y gracepoint a.b.137.14:4569 60 Registered <-- working example. |
| 15:58.40 | jkroon | ask them for the output of "iax2 show peer 182775". |
| 15:58.42 | znoteer_ | [TK]D-Fender: OK, I'll try all the dump stuff. I've already asked my provider twice what they saw me as, and they just avoided the question both times |
| 15:59.21 | znoteer_ | ok, I'll ask them again |
| 16:00.29 | znoteer_ | how is it possible for the perceived address to be theirs. Would it be an error on my end, or on theirs? |
| 16:01.26 | jkroon | probably theirs. |
| 16:04.43 | znoteer_ | [TK]D-Fender: how do I prove the IP's are right. I've already done a reverse lookup on the IP address and it comes back as belonging to my provider |
| 16:05.20 | [TK]D-Fender | huh? |
| 16:05.32 | [TK]D-Fender | reverse of what? |
| 16:10.23 | znoteer_ | [TK]D-Fender: sorry, I got a little confused with all the things I've tried. What I did was to ping using the host name of my provider's server. The address returned by that ping is the same as what appears on the paste. So at least that IP is correct, as far as I understand it |
| 16:10.42 | *** part/#asterisk retentiveboy (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 16:10.44 | [TK]D-Fender | ***YOUR IP*** |
| 16:10.48 | [TK]D-Fender | verify YOUR IP |
| 16:10.55 | znoteer_ | I have |
| 16:11.09 | *** join/#asterisk retentiveboy (~retentive@c-73-82-30-193.hsd1.ga.comcast.net) |
| 16:11.24 | znoteer_ | xxx.yyy.150.230 |
| 16:11.29 | [TK]D-Fender | Go enable IAX2 debug and watch for a registration |
| 16:12.45 | znoteer_ | if I do "core set debug 10" will that affect "iax2 set debug on"? in other words, can I change the level of verbosity of "iax2 set debug on"? |
| 16:13.11 | [TK]D-Fender | no |
| 16:13.17 | [TK]D-Fender | there is no such thing as verbosity |
| 16:21.03 | jeev | is it possible to have blind transfers have different ringtones? |
| 16:21.54 | jeev | i'm setting alert info on a DID and it's working fine, when they call eachother's extensions, the original ringtone is there but if they blind transfer the call that came in, it passes the alert info ringtone |
| 16:22.51 | [TK]D-Fender | check if the call was blind transfered in your dialplan |
| 16:23.22 | jeev | yes, it is blind transferred |
| 16:24.23 | [TK]D-Fender | check if the call was blind transfered in your dialplan |
| 16:29.01 | jeev | "Blind Transfer: , Attended Transfer: , User: , Alert Info: <Triplet>") in new stack |
| 16:29.36 | jeev | looking still |
| 16:30.12 | *** join/#asterisk DanB (~DanB@clt-195.192.207.77.ip-anschluss.net) |
| 16:30.23 | znoteer_ | [TK]D-Fender: here's the debug output of an iax2 registration: D-Fender: here's the debug output of an iax2 registration: |
| 16:30.35 | znoteer_ | https://paste.debian.net/hidden/53728ce0/ |
| 16:31.31 | [TK]D-Fender | and the status now? |
| 16:31.59 | [TK]D-Fender | APPARENT ADDRES : IPV4 xxx.yyy.150.230:4569 |
| 16:32.51 | znoteer_ | iax2 show registry still shows my provider's IP as the perceived address. It's the same as before |
| 16:33.32 | [TK]D-Fender | Go show us the forwarding on your router side |
| 16:33.33 | znoteer_ | if that's what you meant by status |
| 16:33.36 | [TK]D-Fender | and place a call |
| 16:34.08 | jeev | "Blind Transfer: SIP/117-00000054, Attended Transfer: , User: , Alert Info: <Triplet>") in new stack |
| 16:35.07 | znoteer_ | to show the forwarding means using tcpdump? |
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| 16:37.14 | [TK]D-Fender | <znoteer_> to show the forwarding means using tcpdump? <- show us your ROUTER CONFIG FOR IT |
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| 16:51.43 | znoteer_ | [TK]D-Fender: sorry for being so clueless, but showing the router config mean showing the iptable rules? |
| 16:52.52 | [TK]D-Fender | .... |
| 16:53.05 | [TK]D-Fender | the ROUTER your server connects through |
| 16:54.14 | znoteer_ | yes, that part I understand. What I don't understand is exactly what configuration you are looking for. Are you looking for my firewall rules? |
| 16:54.46 | [TK]D-Fender | [TK]D-Fender> Go show us the forwarding on your router side <-------- |
| 16:55.29 | *** join/#asterisk samoied (~samoied@177.96.33.129) |
| 16:55.36 | samoied | Hello Guys! |
| 16:55.52 | *** part/#asterisk kharwell (~kharwell@user-24-214-15-130.knology.net) |
| 16:56.10 | znoteer_ | simply repeating what you've already said doesn't help. I need more detail. I haven't been doing this as long as you. I may not understand the jargon well. Please be more specific |
| 16:56.29 | samoied | I'm trying to use COnfBridge for a large system, where most of conferences will be just 2 channels. |
| 16:56.55 | znoteer_ | do you want to see forwarding via a tcpdump or via my firewall configuration or something else? |
| 16:57.02 | samoied | Tried to use bridge_native_rtp, but always I create a ConfBridge, a CBAnn channel appears |
| 16:57.12 | [TK]D-Fender | SHOW THE RULE FORWARDING IAX2 TO YOUR SERVER |
| 16:57.27 | znoteer_ | thank you |
| 16:57.37 | samoied | I disable all announcements in confbridge.conf, but it keeps appearing when I start a new ConfBridge. |
| 16:57.45 | samoied | Any toughts in how to disable this? |
| 16:58.49 | file | ConfBridge will never use bridge_native_rtp, it will always use mixing |
| 17:00.03 | samoied | file: understood... But I still not need CBAnn. Any way to disable this channel? |
| 17:00.30 | file | don't think so, no |
| 17:04.29 | znoteer_ | [TK]D-Fender: here's the iptables output relative to iax2 forwarding : https://paste.debian.net/hidden/015648c1/ |
| 17:07.45 | [TK]D-Fender | And that is from? |
| 17:08.25 | znoteer_ | that is from the router that incoming traffic passes through on its way to my asterisk server |
| 17:12.28 | znoteer_ | "iptables -L -vn" on the firewalling router |
| 17:12.51 | [TK]D-Fender | You're running a linux firewall? |
| 17:12.57 | [TK]D-Fender | firewall/router |
| 17:13.00 | znoteer_ | yes |
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| 18:11.09 | znoteer_ | [TK]D-Fender: here's the output of tcpdump during an attempt to call my DID. The paste has 3 parts: tcpdump on the * box, tcpdump on the public interface of the router, tcpdump on the private interface of the router: https://paste.debian.net/hidden/f8abfe30/ |
| 18:25.25 | samoied | another silly question. When a user using PJSIP channel and G.722 codec enters a COnfBridge room, channel do a Write and Read transcode to slin@8000. If I set internal_sample_rate it works, but I want to keep auto. Any toughts? |
| 18:26.30 | samoied | As channel uses G.722, the bridge needs to use slin@16000 to keep quality. |
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| 20:38.06 | *** join/#asterisk paule32 (~paule@2a00:c1a0:8502:2800:224:81ff:fe12:72e5) |
| 20:38.13 | paule32 | hello |
| 20:38.49 | paule32 | i have a fritz.box router/modem from avm, i registered on it with the apbx internally |
| 20:39.06 | paule32 | ich configure a echo test under phone 100 |
| 20:39.15 | paule32 | the echo answer my call |
| 20:39.29 | paule32 | i can follow the call in the console |
| 20:39.51 | paule32 | but, after the message, i speak into phone, i get poor sound back |
| 20:40.23 | paule32 | allow=gsm/ulaw are on |
| 20:41.03 | paule32 | is it possible to make the echo test clearler |
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| 20:46.36 | GMLudo | Hi everybody, we have a very strange problem, but before to open an issue, I want to validate it isn't a bad configuration issue: We are migrating from Asterisk 15+chan_sip for WebRTC to 15+chan_pjsip. AT SIP level, everything is OK. But when the call is picked, we have no sound. In the Asterisk console we have this error: |
| 20:46.41 | GMLudo | [Jan 17 14:41:53] ERROR[13613]: pjproject:0 <?>: icess0x7f4abc08efe8 ..Error sending STUN request: Invalid argument |
| 20:47.24 | GMLudo | After some dig, when we use WebRTC+chan_sip, we see the WebBrowser sent a STUN packet to Asterisk and Asterisk answers |
| 20:48.20 | GMLudo | But with WebRTC+chan_pjsip, when the WebBrowser sends the STUN packet, no answer from Asterisk, and we see "Error sending STUN request: Invalid argument" in Asterisk console |
| 20:49.11 | GMLudo | we have taken a capture on the Asterisk server, no STUN packet is emitted |
| 20:54.00 | GMLudo | Debug logs before the error message: https://gist.github.com/GMLudo/10d672b8bcf565467a28e3e15dd143b2 |
| 20:54.44 | GMLudo | Somebody uses WebRTC+pjsip on production ? |
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| 22:04.56 | paule32 | hello |
| 22:05.18 | paule32 | why blocks asterisk linux sound pulseaudio? |
| 22:05.44 | paule32 | every time, i get asterisk running, sound on the system is off |
| 22:06.06 | paule32 | then, when i kill asterisk process, sound on system is ok |
| 22:06.29 | paule32 | have asterisk a pulseaudio sound port? |
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