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| 00:50.24 | ctpdump | Hello. I have a strange issue. Everything is working correctly (can dial, I can hear properly) but in the logs I always get: Spawn extension [..] exited non-zero on [..]" after hangup |
| 00:50.54 | ctpdump | I searched for the error message but all people have issues with dialling or call not connecting when they get the non-zero exit code |
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| 01:03.37 | file | it's not an error message |
| 01:05.37 | ctpdump | the thing is I'm trying to create a .call file and due to this error, the file gets retried incorrectly (as it detects non-zero exit) |
| 01:05.52 | ctpdump | [Mar 10 14:04:33] NOTICE[7893]: pbx_spool.c:388 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) |
| 01:06.06 | ctpdump | although the .call file was executed correctly |
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| 10:15.52 | gavimobile | my telco wants to know if im calling in e164 format. I read up about e164 online and from what I understand asterisk only works using e164 format. is this correct? |
| 10:15.59 | dermont | Hi, is it possible to set an extension (fixed number or name) you are calling as the callerid instead using the usual caller number |
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| 10:17.03 | gavimobile | dermont: are you refering to spoofing your caller id? |
| 10:17.14 | WIMPy | gavimobile: Asterisk can use anything. It doesn't care. |
| 10:17.44 | gavimobile | WIMPy: I need to read more on what's e164 then cause im not understanding it |
| 10:18.19 | WIMPy | dermont: Extensions are destinations you can call. And you can set the Caller ID to whatever you want. If whoever you're talking to will accept it is another question. |
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| 11:36.11 | y_ateya | Hi, how does SIP/Asterisk perform on bad channel before having bad voice? Assume g729 codec. |
| 11:36.31 | danrogl | I've got an old ugly asterisk server and in the process of moving away to a newer internal server, the old server today is filling log files with: WARNING[9855] tcptls.c: Accept failed: Socket operation on non-socket |
| 11:40.22 | y_ateya | Do you use tcp or tls on sip? |
| 11:41.15 | danrogl | tcp |
| 11:41.34 | y_ateya | If you don't use tcp/tls (sip over secured tcp channel), add these to sip.conf: |
| 11:41.34 | y_ateya | tcpenable=no |
| 11:41.34 | y_ateya | tlsenable=no |
| 11:41.57 | danrogl | that was meant to be a no :-) not tcp |
| 11:43.16 | *** join/#asterisk [[thufir]] (~thufir@96.48.128.162) |
| 11:43.19 | y_ateya | ah, you use tcp :). Try tlsenable=no |
| 11:43.36 | y_ateya | which version do you use? |
| 11:45.38 | danrogl | I'm having to deal with an old trixbox, with Asterisk 1.6.0.26 |
| 11:48.54 | y_ateya | 1.6 is pretty old. If calls are going fine ignore this warning. If it is breaking something calls will not work. |
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| 11:51.48 | danrogl | mainly filling the log files |
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| 11:54.16 | danrogl | I've only recently been given an ok to move away from it, so hopefully I can make it last long enough |
| 11:57.26 | y_ateya | Well, disable warnings :D . It is not a good advice, but stops the annoying messages. |
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| 12:01.18 | mjt | "requested format = gsm, requested prefs = (), actual format = g726, host prefs = (), priority = mine" -- what does this mean and how to fix it? |
| 12:02.22 | y_ateya | It means someone is sending you a call with not-supported codec. |
| 12:02.24 | *** join/#asterisk freemanls (~if@93.152.164.60) |
| 12:02.32 | mjt | and is it possible for the caller to see or hear the error ? Right now it jsust a short "beep" and hangup |
| 12:02.42 | mjt | that someone would be me ;) |
| 12:03.13 | y_ateya | no, call will not be established. codec negotiation is done before starting call. |
| 12:03.54 | y_ateya | you use sip or pjsip? |
| 12:04.18 | mjt | yes, I understand the call wont be established, but at least it should be possible to send back some understandable text or displayable code? |
| 12:04.46 | mjt | it is linphone -> sip -> asterisk1 -> iax2 -> asterisk2 -> sip -> hw phone |
| 12:05.02 | mjt | it worked yesterday before asterisk2 server reboot :) |
| 12:06.12 | danrogl | annoying messages stopped, must keep server going till everyones igrated away |
| 12:06.31 | y_ateya | "requested format = gsm, requested prefs = (), actual format = g726, host prefs = (), priority = mine" is pretty understandable for me :D. just kidding. SIP will reject you call with "Not Acceptable Here" |
| 12:06.59 | mjt | it is the asterisk2 who rejects the message |
| 12:07.04 | mjt | call* |
| 12:07.32 | y_ateya | it depends which asterisk gave you this error |
| 12:08.00 | mjt | I call from linphone - see above the call chain |
| 12:08.22 | y_ateya | this message appears in linphone? |
| 12:08.34 | mjt | nope, in debugging output of asterisk2 |
| 12:09.12 | y_ateya | ah, this might be misconfiguration between asterisk2 and hw-phone. |
| 12:09.13 | mjt | linphone produces just a short beep and is ready for the next call, ie, nothing visible |
| 12:09.51 | mjt | I just called this way: linphone -> sip -> asterisk2 -> sip -> hw phone -- it works |
| 12:09.54 | y_ateya | do you have Answer() santanza in extensions.conf of asterisk1 or asterisk 2 |
| 12:10.04 | mjt | no |
| 12:10.28 | mjt | should i? |
| 12:10.39 | y_ateya | no, you shouldn't. |
| 12:11.01 | y_ateya | Do you have allow=all in sip.conf and iax.conf of asterisk1 and asterisk2? |
| 12:11.28 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:11.33 | mjt | exten => _2[03]X,hint,(SIP/${EXTEN}); same => 1,Ringing(); same => n,Dial(SIP/${EXTEN}) -- that's all the relevant dialplan |
| 12:11.37 | mjt | on asterisk2 |
| 12:11.50 | mjt | with s/;/\n/g ofcourse |
| 12:12.20 | mjt | I don't have allow= lines in there |
| 12:12.31 | y_ateya | try to remove the Ringing() , it gives a misleading info to linphone that call is in progress. |
| 12:12.40 | y_ateya | allow=all in in sip.conf and iax.con, not extensions |
| 12:12.49 | mjt | yup re allow all |
| 12:12.54 | [TK]D-Fender | mjt: that won'tt work as shown |
| 12:13.02 | mjt | but it works? :) |
| 12:13.17 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-aetuvjjahfhwbyac) |
| 12:13.20 | [TK]D-Fender | The hint shouldn't |
| 12:13.33 | mjt | I think it doesn't work yes |
| 12:14.06 | [TK]D-Fender | mjton asterisk2 <--- Not a actual version or branch |
| 12:14.12 | [TK]D-Fender | an |
| 12:14.21 | *** join/#asterisk mjordan (mjordan@nat/digium/x-hkrxnsucqxwjfsap) |
| 12:14.21 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:14.22 | *** join/#asterisk yun1989 (5f5c784c@gateway/web/freenode/ip.95.92.120.76) |
| 12:14.26 | yun1989 | hello alll |
| 12:14.27 | mjt | y_ateya: without ringing there's no indication whatsover of a call progress.. |
| 12:14.46 | yun1989 | it's possible when one user leave the conference close the conference ? |
| 12:14.53 | mjt | [TK]D-Fender: version is 1.13.1 on both |
| 12:15.06 | mjt | [TK]D-Fender: 11.13.1 :) |
| 12:15.54 | mjt | hmm. I think I know what's going on, but don't really know how to fix it right |
| 12:16.03 | y_ateya | mjt: no, putting Ringing gives the indication of ringing even if the call is not ringing. I prefer to wait for the remote party to start ringing to indicate REAL ringing. |
| 12:16.14 | mjt | ah |
| 12:16.43 | mjt | one of the last changes I did was commenting out "bandwidth = medium" in iax.conf on asterisk2 |
| 12:16.59 | [TK]D-Fender | yun1989: which kind of conference |
| 12:17.35 | [TK]D-Fender | mjt: What is the issue you're working on? |
| 12:18.26 | mjt | [TK]D-Fender: "requested format = gsm, requested prefs = (), actual format = g726, host prefs = (), priority = mine" shown in asterisk2 debug when doing call linphone -> sip -> asterisk1 -> iax2 -> asterisk2 -> sip -> hw phone |
| 12:18.52 | [TK]D-Fender | Show us the peers and actual call debug |
| 12:19.29 | yun1989 | @[TK]D-Fender one normal conference with two tipes of users |
| 12:19.50 | [TK]D-Fender | yun1989: What KIND of conference? MEETMEE? CONFBRIDGE? |
| 12:20.15 | yun1989 | I use confbridge |
| 12:20.27 | yun1989 | and two tipes of users |
| 12:20.28 | yun1989 | http://pastebin.com/jZ5HkZC8 |
| 12:22.07 | mjt | [TK]D-Fender: http://paste.debian.net/160543/ -- this is on asterisk2. I removed other sip peeers from there and added mnemonic names reflecting my example |
| 12:23.14 | [TK]D-Fender | mjt: I don't see anything that feels like an error in there...What's wrong with it? |
| 12:23.29 | mjt | the call is being rejected |
| 12:23.55 | [TK]D-Fender | mjt: I don['t see that anywhere... you have no SIP or IAX2 debug enabled there |
| 12:24.58 | [TK]D-Fender | mjt: You're also dialing SIP/201 which isn't in the list we see up top at all... |
| 12:24.59 | mjt | gosh. That's a LOT of debugging |
| 12:25.14 | mjt | [TK]D-Fender: it's the same as 202 - i removed the wrong line :) |
| 12:25.17 | [TK]D-Fender | mjt: Since I see it dialing SIP/201 I'd start with SIP |
| 12:25.26 | [TK]D-Fender | mjt: Since it made it that far to start |
| 12:27.01 | freemanls | mjt: if you have other calls going on and you want verbosity of debug messages just for 1 ip, you can do that |
| 12:27.53 | mjt | here's an unmodified debug (sip & iax2). Here, asterisk1 is "tls", asterisk2 is "panda". http://paste.debian.net/160546/ -- that's an attempt to deal 201 from 146 which is behind tls (asterisk2). |
| 12:29.30 | mjt | dial* :) |
| 12:29.57 | [TK]D-Fender | mjt: This looks very wrong |
| 12:30.27 | mjt | asterisk1 = 192.168.177.2 = tls, asterisk2 = panda = 192.168.19.1, call is 146 linphone -> tls asterisk1 -> panda asterisk2 -> 201 hw phone |
| 12:30.37 | [TK]D-Fender | mjt: I see a hangup with NO reequest from the other side and thee call was proceeding in what I see theere |
| 12:30.51 | [TK]D-Fender | mjt: Is taht debug from both sides spliced together? |
| 12:31.02 | mjt | nope, it is all on panda asterisk2 |
| 12:31.33 | mjt | two calls -- might be some other call has been injected at the same time. but not 3 |
| 12:32.13 | mjt | I think I'm missing something basic |
| 12:33.07 | [TK]D-Fender | 69 = dial, 110= confirmation of having sent invite, 112 = IAX2 HANGUP?!?! Your dialplan did not "run out", nor did the dial abort. |
| 12:33.10 | [TK]D-Fender | This is not sane at all. |
| 12:33.23 | [TK]D-Fender | And I'm having a lot of trouble trusting what I'm seeing |
| 12:33.40 | mjt | *oh, asterisk2 = tlspz, not panda, it is named tlspz in the config) |
| 12:33.56 | mjt | um, trusting? I just cut-n-pasted it from the terminal, unmodified... |
| 12:35.24 | [TK]D-Fender | * only just sent out the invite. It didn't time out, retry or even wait for the response beefore we see a hangup. |
| 12:35.37 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-zjkvbohlkhhkocni) |
| 12:36.58 | [TK]D-Fender | 112 = a claim of IAX2 hangup, 123 = SIP/102 reports back Trying before this * can even send a CANCEL (137) concerning the hangup |
| 12:37.27 | mjt | um, 123 is another peer |
| 12:37.41 | [TK]D-Fender | line #123 |
| 12:37.46 | mjt | ah |
| 12:37.58 | [TK]D-Fender | The timee-line I'm seeing doesn't make sense |
| 12:38.21 | [TK]D-Fender | I see an IAX2 hangup nowhere near debug to explain the reason. |
| 12:38.35 | mjt | maybe it will be clearer if I say that 192.168.19.8 is a dual-sip hw phone, both 201 and 202 extensions are registered to it |
| 12:38.49 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 12:38.54 | [TK]D-Fender | Doesn't mattere |
| 12:39.01 | [TK]D-Fender | * dialed an IP for it |
| 12:39.16 | [TK]D-Fender | And stopped the call before the Trying even came back... and it DID... |
| 12:39.21 | [TK]D-Fender | This doesn't make sense |
| 12:40.25 | marceloamorim | hello guys, could I use gotoif with condition n+1 like same => n,GotoIf($["${CALLERID(num)}" = "553799841120"]?n+1:callback) if condition false go to the next line to the dialplan |
| 12:41.12 | marceloamorim | I used to the ael, so I'm getting little difficult to transfer all my ael dialplan for normal dialplan |
| 12:42.00 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.125.24) |
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| 12:43.10 | *** join/#asterisk [[thufir]] (~thufir@96.48.128.162) |
| 12:44.06 | freemanls | marceloamorim: i had some issues using GotoIf with asterisk13 ended up using ExecIf(condition?Goto(true):Goto(false)) |
| 12:44.12 | *** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com) |
| 12:46.10 | [TK]D-Fender | marceloamorim: put NOTHING there |
| 12:46.31 | marceloamorim | could be ?:callback) |
| 12:46.52 | [TK]D-Fender | freemanls: I've never heard of any bugs reported with that option ever. |
| 12:46.57 | [TK]D-Fender | marceloamorim: Yes |
| 12:48.24 | marceloamorim | thx [TK]D-Fender, and thx freemanls for this tip about gotoif, if I have any problem with gotoif I'll report for you guys and change for execif while we try to figured out the problem |
| 12:48.31 | marceloamorim | thx a lot one more time |
| 12:48.50 | [TK]D-Fender | marceloamorim: GotoIf works. Every time. |
| 12:49.09 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 12:49.11 | [TK]D-Fender | marceloamorim: If you run into a failure you can't figure out, just show us |
| 12:50.03 | mjt | [TK]D-Fender: I added "disallow = all; allow = gsm" to asterisk2 (panda, tlspz, 192.168.19.1) iax.conf and it now works fine. |
| 12:50.27 | [TK]D-Fender | mjt: ok...... |
| 12:51.15 | [TK]D-Fender | yun1989: show us a failed test for your marked users. |
| 12:51.19 | mjt | [TK]D-Fender: but I understand this is not a good solution, not a right onw |
| 12:51.21 | mjt | one |
| 12:51.40 | [TK]D-Fender | mjt: Setting codes is alwqaya appropriate... and I never saw what you actually set |
| 12:51.52 | [TK]D-Fender | And * didn't throw any kind of error during that from what we saw |
| 12:52.11 | mjt | I never told it anything about codecs |
| 12:52.13 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 12:52.29 | [TK]D-Fender | mjt: That is already a ..... |
| 12:52.31 | [TK]D-Fender | ~soso |
| 12:52.31 | infobot | [~soso] Shoot-On-Sight Offense |
| 12:52.33 | [TK]D-Fender | :p |
| 12:53.06 | mjt | I just didn't know which codecs to use, and discovered it works without that (until some time anyway) |
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| 12:58.01 | freemanls | [TK]D-Fender: i know i am the only one v13.2.0, very strangee |
| 12:58.24 | [TK]D-Fender | freemanls: Show me |
| 12:58.43 | freemanls | okay, let me try again |
| 13:00.52 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:02.41 | freemanls | [TK]D-Fender: just tested again, i dont have problems with it anymore, sry for the offtopic |
| 13:06.46 | freemanls | Guys I am experiecing sound dropping (Background messages disappear) while I am testing by going crazy inside an IVR menu clicking on options |
| 13:07.31 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 13:07.49 | freemanls | for example if the message is "hello this is an ivr menu", sometimes I hear just "IVR menu" the end of the message |
| 13:07.59 | mjt | [TK]D-Fender: why did you say my hint wont work? -- exten => _2[03]X,hint,(SIP/${EXTEN}) |
| 13:08.03 | freemanls | I do experience this only Cisco Phone 7960 |
| 13:09.12 | dermont | hi, is it possible to do a call till someome answers or if noanswer to extend caller 1 and call 2 different people |
| 13:09.12 | [TK]D-Fender | mjt: Does it? |
| 13:09.41 | dermont | right now I am having trouble with cellphone and SIP as caller B and caller C |
| 13:09.44 | mjt | [TK]D-Fender: it doesn't, and I confirmed that already, but I want to understand the reason, ie, what I did wrong... |
| 13:09.53 | [TK]D-Fender | dermont: As in dial #1. Then after a while ADD #2 to the dialing while continuing to ring #1? |
| 13:09.59 | dermont | yes |
| 13:10.01 | [TK]D-Fender | mjt: () <------------- |
| 13:10.36 | dermont | it does call 1 then hangs up calls #2(Cell) and #3 |
| 13:10.39 | [TK]D-Fender | dermont: Dial at least 1 local channel as one of thee devices. in the dialplan for that local channel do a Wain in front. |
| 13:10.44 | [TK]D-Fender | Wait* |
| 13:10.45 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 13:11.04 | dermont | if #2 rings it hangs up #3 |
| 13:11.17 | mjt | [TK]D-Fender: aww, is it the extra comma before the ()s? :) |
| 13:11.18 | dermont | probably because of cellphone using |
| 13:11.51 | [TK]D-Fender | dermont: Your description needs a little fixing there |
| 13:12.06 | [TK]D-Fender | dermont: Please start over.... |
| 13:12.11 | dermont | sec |
| 13:12.16 | [TK]D-Fender | dermont: From the beginning of what you want |
| 13:13.41 | dermont | Call extension 99 |
| 13:13.41 | dermont | Call #1 on SIP Phone |
| 13:13.41 | dermont | Wait 8sec |
| 13:13.41 | dermont | Call #2 (cellphone) and #3 SIP Phone while #1 is still ringing |
| 13:14.09 | dermont | I am having trouble with step 4 |
| 13:14.23 | dermont | if #2 is ringing it kills #3 |
| 13:14.44 | dermont | though I just do DIAL(SIP/#2&SIP/#3) |
| 13:15.05 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:17.09 | [TK]D-Fender | Show us the call |
| 13:17.32 | dermont | you mean the log? |
| 13:17.42 | [TK]D-Fender | yes |
| 13:17.51 | [TK]D-Fender | We need to see what #2 is doing |
| 13:20.31 | dermont | in here or on query |
| 13:20.47 | freemanls | pastie |
| 13:21.06 | freemanls | sip set debug 3 && core set debug on |
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| 13:24.14 | dermont | http://pastie.org/private/8oyxp2jxjgqvefxlrmhhq |
| 13:28.25 | freemanls | oops, please do these: core set debug 3 && sip set debug on |
| 13:29.03 | [TK]D-Fender | dermont: -- SIP/gateway01-000059a7 answered SIP/99-000059a5 |
| 13:29.11 | [TK]D-Fender | dermont: it ANSWERED. |
| 13:29.20 | dermont | so it says |
| 13:29.25 | dermont | but actually it is just ringing |
| 13:29.37 | dermont | SIP/gateway01-000059a7 = #2 |
| 13:29.41 | [TK]D-Fender | dermont: Yes well whatever you are dialing sent back an answeer and is sending ringing in-bad |
| 13:29.49 | [TK]D-Fender | dermont: There is nothing this system can do about that |
| 13:29.55 | [TK]D-Fender | dermont: What is this gateeway? |
| 13:31.15 | dermont | that is our firewall and dhcp server |
| 13:32.14 | [TK]D-Fender | dermont: SIP/gateway01 <------ |
| 13:32.34 | dermont | yep |
| 13:32.39 | [TK]D-Fender | that is a SIP SEERVICE |
| 13:32.48 | [TK]D-Fender | not a "DHCP service", or "firewall" |
| 13:33.02 | [TK]D-Fender | what device/provider/etc is that? |
| 13:33.08 | dermont | so, is there a workaround on it? |
| 13:33.16 | [TK]D-Fender | Not as long as it acts like that |
| 13:35.05 | dermont | ok |
| 13:35.06 | dermont | thx |
| 13:35.38 | mjt | [TK]D-Fender: re my codec mismatch issue: shouldn't something like `allow=all' fix that for good? |
| 13:35.47 | mjt | (and I thought it is the default?) |
| 13:36.13 | [TK]D-Fender | Depends on [general] as well. Also ... don't evre do that |
| 13:36.27 | mjt | to defaults or to "all" ? |
| 13:36.33 | freemanls | to all |
| 13:36.35 | [TK]D-Fender | Define your codes. Never think that you can get away with not actually making these kinds fo choices explicitly |
| 13:36.42 | [TK]D-Fender | codecs* |
| 13:37.57 | mjt | that was one of my very first questions: which codecs to use. It is a question of compatibility, network efficiency and sound quality |
| 13:38.23 | mjt | (I'd enable opus ;) |
| 13:39.25 | *** join/#asterisk sruffell (sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:39.25 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:39.45 | [TK]D-Fender | As soon as you say quality & compatibility then the answer is G.711 |
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| 13:40.43 | mjt | heh. And right after that there are ulaw and alaw :) |
| 13:40.54 | [TK]D-Fender | those ARE G.711 |
| 13:41.10 | [TK]D-Fender | ulaw = G.711u, alaw=G.711a |
| 13:41.48 | [TK]D-Fender | chosen based on where you are and what is used by any service provider you use |
| 13:42.06 | [TK]D-Fender | to avoid even nominal loss of conversion between them |
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| 13:49.27 | mdhas_laptop | Good morning, does anyone know why this might happen and what I can do to resolve it? So yesterday our did provider seems to have made some changes and as a result I had to make some changes to our trunk. I edited each trunk and changed the hostname and issued the new remote host and the new password and saved it. When I do sip show registry from the CLI I see the same number twice, once with the old hostname a |
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| 13:52.13 | [TK]D-Fender | mdhas_laptop: Might ahve to restart * entirely to flush the old one from memory |
| 13:57.07 | mdhas_laptop | D-Fender: You mean like a service restart? |
| 13:57.17 | mdhas_laptop | or more of a server restart? |
| 14:03.37 | [TK]D-Fender | service |
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| 14:20.37 | bulletxt | [TK]D-Fender: nat= comedia ? |
| 14:20.45 | [TK]D-Fender | yes |
| 14:20.52 | bulletxt | I tied but nothing changes |
| 14:21.00 | bulletxt | I put nat=comedia for user 300 in sip.conf |
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| 14:28.33 | bulletxt | [TK]D-Fender: any ideas ? |
| 14:28.36 | bulletxt | :( |
| 14:29.01 | [TK]D-Fender | PB the peer & the call |
| 14:30.01 | bulletxt | ok, ill be able to do that later ill let you know |
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| 14:47.12 | yun1989 | @[TK]D-Fender hello |
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| 14:48.28 | yun1989 | i need to catch the event when the user leave the conversation |
| 14:48.40 | yun1989 | but i don't know how |
| 14:49.01 | mjt | like a hangup, or a conference call? |
| 14:49.10 | [TK]D-Fender | yun1989: I asked you to show your attempt |
| 14:50.47 | yun1989 | when I receive a call i put this call in conference |
| 14:51.11 | yun1989 | and in this call exist three users |
| 14:51.31 | yun1989 | for example (10007/1002 and 1002REC |
| 14:52.03 | yun1989 | the 1002REC is one extension that record the call between 10007 and 1002 |
| 14:52.48 | yun1989 | but when the 10007 (sofphone) hangup the call, the 1002 and 1002REC continuos in conference |
| 14:52.53 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:53.30 | yun1989 | i need when one user 10007 or 1002 hangup the call, terminate the confence but i don't know why |
| 14:53.37 | [TK]D-Fender | yun1989: PASTEBIN and actual call of one fo each of the memebers joining and show what is actually happening. |
| 14:55.37 | yun1989 | http://pastebin.com/nExRF0nz |
| 14:56.55 | yun1989 | if you need i can to show my configurations |
| 14:57.11 | [TK]D-Fender | yes, those as well |
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| 14:59.13 | yun1989 | http://pastebin.com/yRLj3dF0 |
| 14:59.19 | yun1989 | my extensions.conf |
| 15:00.09 | [TK]D-Fender | your confibridgee configs <- |
| 15:00.15 | [TK]D-Fender | confbridge* |
| 15:00.48 | yun1989 | http://pastebin.com/yv5ZEHh6 |
| 15:01.04 | yun1989 | my confbridge is it |
| 15:01.18 | yun1989 | i have two type of users |
| 15:01.35 | [TK]D-Fender | I don't see the END of this conference. |
| 15:01.51 | [TK]D-Fender | You didn't show where the actual failuree was happening |
| 15:03.31 | yun1989 | yes the problem is it i don't close the conference but i dont know when i make this |
| 15:03.39 | yun1989 | is in extensions.conf ? |
| 15:03.44 | yun1989 | in confbridge ? |
| 15:04.49 | [TK]D-Fender | Show me a complete call including the failure |
| 15:04.58 | yun1989 | for example when hangup the call in 1002 ( my application webrtc) I hangup the 1002 and 1002REC the conference and the conference is closed |
| 15:05.15 | yun1989 | the call not fail |
| 15:05.52 | yun1989 | the call works fine, the problem is close conference, why i don't receive the event hangup the participants in CLI |
| 15:07.03 | [TK]D-Fender | ..... |
| 15:07.33 | *** join/#asterisk freemanls (~if@93.152.164.60) |
| 15:07.42 | [TK]D-Fender | THERE IS NO HANGUP IN YOUR CALL DEBUG |
| 15:08.08 | yun1989 | yes but why ? |
| 15:08.26 | [TK]D-Fender | You did NOT hangup in that debug |
| 15:08.34 | [TK]D-Fender | You are not showing a hangup happeningh |
| 15:09.07 | yun1989 | yes why I don't receive the hangup |
| 15:09.21 | freemanls | [TK]D-Fender: would you help me with an interesting issue concerning asterisk 13.2.0 and Cisco 7690 sip firmware(8.12.00) |
| 15:09.24 | yun1989 | i have configured the hangup in extensions.conf |
| 15:09.27 | [TK]D-Fender | You did not show a COMPLETE call. |
| 15:10.18 | yun1989 | i show the complet call |
| 15:10.49 | [TK]D-Fender | yun1989: Then those channels are still in progress |
| 15:10.56 | freemanls | [TK]D-Fender: i've got a tcpdump of an example call that is dropping sound, as far as i am reading all day it is about an RTP timestamp issue, could I change asterisks code so I can move on from that or how could I get the SNTP server working on the Cisco7690 it shows 00:00 as an current hour no matter what |
| 15:11.58 | [TK]D-Fender | yun1989: You are not showing up to the point where one of those SIP devices involved HANGS UP. |
| 15:12.23 | [TK]D-Fender | Yun You showed all 3 making it into the conference. You did NOT show a single one of them LEAVING |
| 15:12.44 | [TK]D-Fender | freemanls: no idea |
| 15:13.17 | freemanls | [TK]D-Fender: I am really stuck with this and would need some second pair of eyes to look at what i have found on the net |
| 15:13.26 | freemanls | appears like there were patches for sip.c |
| 15:13.27 | [TK]D-Fender | freemanls: I don't touch Cisco <- |
| 15:13.38 | freemanls | lucky you |
| 15:13.48 | [TK]D-Fender | I prefer the term "smart" :) |
| 15:13.57 | yun1989 | why i receive this event |
| 15:13.58 | yun1989 | -- Auto fallthrough, channel 'SIP/1000REC-00000011' status is 'UNKNOWN' |
| 15:13.59 | yun1989 | ? |
| 15:14.08 | freemanls | i consider myself as smart too, should i quit job now :D |
| 15:14.22 | [TK]D-Fender | yun1989: I don't know. It isn't in what you've shown me in that pastebin.... and I don't see anything ELSE around it. |
| 15:14.42 | [TK]D-Fender | yun1989: just showing that line alone doesn't say why things got where the did where you saw it. |
| 15:15.01 | [TK]D-Fender | freemanls: \o/ |
| 15:15.03 | freemanls | [TK]D-Fender: could you spot the sound dropping in RTP packets, if I post the tcpdump ? |
| 15:15.09 | yun1989 | the complete call |
| 15:15.10 | yun1989 | http://pastebin.com/QXXhV98s |
| 15:15.32 | yun1989 | in this call between 1002 and 10007 |
| 15:15.34 | [TK]D-Fender | freemanls: Are you getting NO audio at all? |
| 15:15.40 | mdhas_laptop | D-Fender: Unfortunately restarting the service hasn't helped those DIDs' are still being listed twice |
| 15:15.48 | freemanls | it is like cutting first three words of the message |
| 15:15.53 | freemanls | or seven words depends |
| 15:15.59 | freemanls | and total silence |
| 15:16.04 | yun1989 | after the call established i hangup 10007 and the conference is not closed |
| 15:16.09 | freemanls | in asterisk's log everything seems perfect |
| 15:16.26 | freemanls | it is behaving on it's side just correctly |
| 15:16.40 | yun1989 | in 1002 the call is running normally but the 10007 don't are in call |
| 15:16.45 | freemanls | the thing is that the phone is dropping some frames or messing around with the timestamps |
| 15:17.33 | yun1989 | after this I hangup the call in 1002 |
| 15:17.42 | yun1989 | and receive the -- Auto fallthrough, channel 'SIP/1000REC-00000011' status is 'UNKNOWN' |
| 15:17.42 | [TK]D-Fender | yun1989: We do NOT see 10007 hanging up in there. |
| 15:18.04 | yun1989 | yes and i don't understand why |
| 15:18.11 | [TK]D-Fender | 10007 DID NOT HANGUP IN THERE. |
| 15:19.05 | yun1989 | the extensions.conf are not configurated correctly for hangup ? |
| 15:19.33 | [TK]D-Fender | yun1989: "sip set debug on" <- Go look at what's actually happening with your SIP comms to it. If you supposedly hung up... * isn't seeing it or your aren't actually showing mee COMPLETE debug. |
| 15:19.45 | [TK]D-Fender | yun1989: You dialplan is IRRELEVANT# |
| 15:19.58 | [TK]D-Fender | yun1989: You SIP DEVICE did NOT end the call |
| 15:20.00 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 15:20.15 | [TK]D-Fender | NO HANGUP |
| 15:20.19 | [TK]D-Fender | Are we clear on this? |
| 15:21.25 | yun1989 | http://pastebin.com/Ld5jpHvy |
| 15:21.31 | [TK]D-Fender | yun1989: "core show application confbridge" <- |
| 15:21.48 | yun1989 | i think the problem is here but i don't understand why occours this |
| 15:22.11 | [TK]D-Fender | freemanls: is it only the first few words and the rest of the call is fine? |
| 15:22.24 | freemanls | yes while going thru the ivr menu |
| 15:22.40 | freemanls | for example first 2 menus are ok and on the 3nd level you get cutted of some part of the background message |
| 15:23.07 | [TK]D-Fender | freemanls: Clarify that.... |
| 15:23.31 | [TK]D-Fender | freemanls: You are losing audio in thee MIDDLE of the call>? |
| 15:23.37 | freemanls | yep |
| 15:23.41 | freemanls | exactly |
| 15:23.43 | [TK]D-Fender | That;s odd... |
| 15:23.47 | freemanls | it is very odd |
| 15:24.10 | yun1989 | @[TK]D-Fender http://pastebin.com/Ld5jpHvy you see it ? |
| 15:31.04 | freemanls | <PROTECTED> |
| 15:31.37 | freemanls | oops, sry got disconnected from my tmux session :D |
| 15:35.08 | freemanls | [TK]D-Fender: the audio loss is happening only on this series of phones cisco 7690 and not on anything other |
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| 15:41.09 | ghostlines | howdy, anyone know what message I should look for to see if Asterisk restarted? |
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| 15:48.51 | newtonr | ghostlines, sometime like: [Mar 10 10:47:15] Asterisk SVN-branch-13-r432404 built by root @ debian1 on a i686 running Linux on 2015-01-23 22:44:23 UTC |
| 15:49.07 | newtonr | obviously with your Asterisk installation's specifics |
| 15:49.30 | ghostlines | newtonr ahh, thanks will check for something similar |
| 15:52.34 | newtonr | also, if you mean an actual administrator initiated restart, then there is a more obvious message of "[Mar 10 10:47:15] VERBOSE[2728] asterisk.c: Asterisk is now restarting..." |
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| 16:20.43 | mdhas_laptop | D-Fender: So just to see if it would help I did a server restart. Still have those DID's listed with the original server |
| 16:20.53 | mdhas_laptop | D-Fender: Any other ideas? |
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| 16:55.08 | SamDaMan | I'd like to thank WIMPy for his sugestion yesterday, you put me on the right path to solving my problem. When an extension is called my dialplan copies a .call file into /var/spool/asterisk/outgoing/ then rings the called extension the .call file places a call to the paging system that plays a ring sound a couple of time and hangs up. Much appreciated. |
| 16:55.40 | [TK]D-Fender | copy = fail |
| 16:55.45 | [TK]D-Fender | You need to MOVE it |
| 16:56.11 | file | I like to move it move it |
| 16:56.22 | SamDaMan | you are correct it is being copied to a temp then moved |
| 16:56.45 | Elio19 | file are you quoting that Reel 2 Reel song from the 90s? |
| 16:57.07 | Elio19 | BTW, im trying to hire an Asterisk expert for my project, plz PM if interested |
| 16:57.39 | file | maybe |
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| 17:16.02 | Elio19 | ...did i mention i've got Bitcoins? and my project is relatively simple |
| 17:16.58 | PaybackTony | I was asking about this yesterday but didn't end up getting it working the way I had hoped. http://pastebin.com/0iEssKGT That is the base config for a peer, my goal was to enable allowing multiple peers with the same name but different domains. |
| 17:17.26 | PaybackTony | With that conf I can't register as test@1.sip.demo.com, but I can as 1_test@1.sip.demo.com |
| 17:17.38 | PaybackTony | Is that not how it works or am I doing something wrong in this config? |
| 17:19.39 | [TK]D-Fender | PaybackTony: defaultuser=test@1.sip.demo.com <- don't put the domain in there |
| 17:25.03 | PaybackTony | [TK]D-Fender: I'm still unable to register with just test@1.sip.demo.com with the domain removed from the defaultuser parameter, same as above, can register with 1_test but not "test" |
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| 17:34.38 | krapper | Asterisk 13 Fax to email.. compile spandsp? Any direction appreciated. Not finding much here in my research. |
| 17:35.53 | [TK]D-Fender | krapper: If you have the required libs then app_fax will compile with support for it. |
| 17:36.07 | [TK]D-Fender | krapper: Cheeck your menuselect and it'll tell you what you're missing |
| 17:38.55 | krapper | Hey [TK]D-Fender: I just skimmed my menuselect and not seeing app_fax.. also core show applications don't list much regarding fax |
| 17:39.13 | krapper | so clearly missing something :-) |
| 17:39.20 | [TK]D-Fender | Do better than "skim" |
| 17:39.27 | [TK]D-Fender | That sounds like "slacking" .... |
| 17:39.44 | krapper | i'll be honest... it was more than a skim |
| 17:40.03 | krapper | ok... XXX |
| 17:40.51 | [TK]D-Fender | AMAZEBALLZ |
| 17:41.27 | krapper | tots McGoats |
| 17:45.32 | mjordan | I don't always fax, but when I do, I prefer res_fax. |
| 17:45.42 | mjordan | stay thirsty. |
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| 17:49.39 | krapper | and some spandsp's for that ash |
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| 17:55.32 | Elio19 | mjordan, i dont know what you are talking about. If the tap water is bad, do not "stay thirsty," go and get a bottle of water and prepare to catch water when it rains. |
| 17:56.57 | Elio19 | ( BTC Reward ) Still looking for someone to lend me a hand with my Asterisk project. ( Schmexperts need no apply ) |
| 17:57.06 | Elio19 | Bascially looking for a voicemail solution to delive mp3 messages via email that can extend to support live calls down the line. |
| 17:58.29 | Elio19 | s/bad/less than desirable/ |
| 17:58.51 | [TK]D-Fender | From what you describe "Voicemail" isn't technically required. |
| 17:59.54 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-quwpdnsxayqjlmja) |
| 18:00.23 | *** join/#asterisk volga629 (~bendersky@CPE4c5e0cb3c2a0-CM0c473dd46e50.cpe.net.cable.rogers.com) |
| 18:01.10 | Elio19 | ok , but what is [TK] ? are you in a "gaming clan" or something? |
| 18:02.14 | volga629 | Hello Everyone, is possible send presence states to presence server ? |
| 18:02.33 | [TK]D-Fender | I haven't been a gamer for almost a decade now. |
| 18:02.47 | Elio19 | What did you play? |
| 18:03.09 | [TK]D-Fender | volga629: if some other presence server subscribes to * yes, othrewise no. |
| 18:05.05 | volga629 | I have edge proxy which can maintain presence, I wonder if I can bridge asterisk and remote client through edge proxy for presence state |
| 18:05.55 | [TK]D-Fender | volga629: * doesn't TAKE presence from devices. |
| 18:15.07 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 18:15.07 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 18:23.05 | *** join/#asterisk goddva (~goddva@2001:840:4259:a53:2ab2:bdff:feef:adf8) |
| 18:30.23 | krapper | I now have option for app_fax, res_fax and res_fax_spandsp of for which I'm selecting. After a make install, I'm still not seeing any fax relation applications on a core show applications. |
| 18:31.06 | [TK]D-Fender | check your modules.conf to make sure you're autoloading them |
| 18:40.48 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
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| 18:51.47 | *** join/#asterisk PHunter (~Phunter@wsip-68-14-213-118.ph.ph.cox.net) |
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| 18:53.05 | PHunter | I am having an issue with select endpoints, where a call will come in and they have one way audio. After 2 attempted calls, the audio will work properly. Is this NAT issue or is there something up with RTP? |
| 18:53.42 | [TK]D-Fender | NAT issue |
| 18:54.06 | [TK]D-Fender | possibly due to where you are trying to send RTP to in the first place |
| 18:55.09 | PHunter | Okay, probably one of those silly carrier provided modem/router combos being terrible. |
| 18:57.41 | PHunter | Or not, looks like just a modem.. we use the same routers all over and never have this issue.. |
| 19:03.00 | PHunter | Any tips for resolving NAT issues? |
| 19:03.09 | PHunter | besides force_rport |
| 19:03.10 | PHunter | ? |
| 19:03.20 | rrittgarn | comedia |
| 19:03.39 | PHunter | I am doing comedia as well |
| 19:04.23 | PHunter | nat=force_rport,comedia |
| 19:04.33 | PHunter | btw im on Asterisk 11 |
| 19:05.24 | PHunter | we do have Keep-Alive on the phones to make sure they stay connected to the servers as well. |
| 19:06.35 | PHunter | Oh wow. Nevermind. That phone did not have Keep-Alive enabled.. |
| 19:11.04 | *** join/#asterisk fling_ (~fling@fsf/member/fling) |
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| 19:17.38 | *** join/#asterisk deranged (Bit@lolnerd.net) |
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| 19:29.11 | mjt | exten => _XXX,1,Dial(SIP/${EXTEN}) -- this is just one rule in the dialplan. When I call a number which is listed in sip.conf but not registered, my phone says "User not found", but when I dial an unrecognized number, nothing happens at all, my phone pretends I didn't call. Using linphone for now. Debug shows "Auto fallthrough, channel 'SIP/146-0000001c' status is 'CONGESTION'". How to make the phone to show something useful? |
| 19:30.13 | [TK]D-Fender | Stop using patterns that cover too much |
| 19:30.59 | [TK]D-Fender | or at least chec the results of your dial and then send something else |
| 19:31.15 | [TK]D-Fender | hangup(cause) |
| 19:31.25 | mjt | and which cause it will be? |
| 19:31.52 | mjt | it might be "no answer" or "not registered" or anything else' |
| 19:32.11 | mjt | I just can't understand something very basic, it looks like |
| 19:32.46 | [TK]D-Fender | NoOp(${DIALSTATUS}) |
| 19:32.53 | [TK]D-Fender | gO LOOK AT WHAT YOU GET FOR EACH |
| 19:32.54 | *** join/#asterisk fergus (~sergei@mail.55h.by) |
| 19:32.58 | mjt | NoOp? |
| 19:33.03 | [TK]D-Fender | yees |
| 19:41.47 | *** join/#asterisk areski (~areski@80.174.128.5.dyn.user.ono.com) |
| 19:44.30 | mjt | linphone just does nothing, it quietly terminates the call. zoiper says "Temporarily unavailable (408)". In both cases it is "NoOp("SIP/146-0000002c", "CHANUNAVAIL"). I've added "Hangup()", debug shows it too. Is linphone buggy? |
| 19:44.47 | *** join/#asterisk airjump (~Thunderbi@p5B0A3AB0.dip0.t-ipconnect.de) |
| 19:45.09 | mjt | Dial() results in "Everyone is busy/congested at this time (1:0/0/1)" |
| 19:45.09 | [TK]D-Fender | pass hangup something else |
| 19:45.25 | [TK]D-Fender | that variablee says what you need |
| 19:45.45 | [TK]D-Fender | So check thee result and send Hangup another code to pass back |
| 19:46.03 | mjt | Executing [144@office-phone:3] Hangup("SIP/146-0000002d", "CHANUNAVAIL") in new stack |
| 19:46.27 | mjt | grrm |
| 19:47.06 | mjordan | Hangup takes a cause code, not a text string of a dial status. Dial status != Hangup cause code |
| 19:47.29 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings |
| 19:47.42 | [TK]D-Fender | and you want to pass it something DIFFERNT |
| 19:48.02 | [TK]D-Fender | Clearly "chanunavail" is not the right thing... it's what you STARTED with |
| 19:48.14 | mjordan | you want the number in that table for whatever you want to map it to. And probably a call to Busy, Congestion, or Playtones (or something along those lines) if you want to play a tone back to the caller that says "Hey, that didn't work." |
| 19:49.34 | *** join/#asterisk Demon_VoIP (~demon@ip253.net222.n37.ru) |
| 19:50.02 | mjt | oww.. ;) |
| 19:50.49 | mjt | btw, why most of the dialplan examples in the wiki starts with "exten => <something>,1,NoOp()" ? |
| 19:50.56 | cuzner | got a rookie dialplan question, is this the correct logical not operator syntax to evaluate ${REGEX()}? |
| 19:50.59 | cuzner | same => n,ExecIf($[${REGEX("1|2" ${this_Could_Be_One_Or_Two})} AND !${REGEX("3|4" ${this_Could_Be_Three_Or_Four})}]?Set(LOCAL(Foo)=Bar)):Set(LOCAL(Bar)=Foo)) |
| 19:52.47 | [TK]D-Fender | mjt: as a beginner's reminder of where you are in thee dialplan |
| 19:54.04 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 19:54.34 | mjt | (I found it easier to start them with exten=>xx,1,NoOp() too, to be able to use same=>n,yyy() in subsequent lines and be able to comment out any of them at any time without touching the first one) |
| 20:00.10 | cuzner | [TK]D-Fender: c'mon, you know you want to check my dialplan syntax :P |
| 20:11.06 | cuzner | damnit, that AND should be a & :P |
| 20:18.01 | *** join/#asterisk nix8n82 (~AndChat56@2601:1:9780:76d:bd93:703c:2147:dea8) |
| 20:18.51 | mjt | [TK]D-Fender: thank you very much for your help and patience. Hopefully it will be for good! |
| 20:19.07 | [TK]D-Fender | Hopeefully |
| 20:27.03 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
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| 20:52.01 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 20:55.19 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
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| 21:35.51 | *** part/#asterisk mjordan (mjordan@nat/digium/x-ykqgdrrviyppstbv) |
| 22:17.25 | *** join/#asterisk infobot (ibot@rikers.org) |
| 22:17.25 | *** 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 |
| 22:50.57 | *** join/#asterisk chris349 (~office@c-50-140-101-134.hsd1.fl.comcast.net) |
| 22:52.19 | chris349 | I am suddenly having a strange issue with NAT and one way audio only when using the g729 codec. When I enable icesupport=yes, however, it is still not sending the public IP address to the SIP peer. What would be the best way to get audio working both ways with the g729 codec? |
| 22:53.35 | *** join/#asterisk fireglow (fireglow@unaffiliated/fireglow) |
| 22:54.32 | fireglow | Hello. Asterisk 11.16.0. I use DAHDI to receive calls. When the caller hangs up, my phones continue to ring for about 5 seconds afterwards. How can I shorten this time, and improve detection of the hangup event? |
| 22:55.09 | WIMPy | By getting rid of your analog lines. |
| 22:55.24 | fireglow | I would if I could. |
| 22:55.29 | fireglow | any other options? |
| 22:55.40 | WIMPy | No |
| 22:55.56 | fireglow | Maybe someone else has an idea, I'll stick around |
| 23:00.33 | chris349 | My carrier tells me the reason why I have one-way audio when using the g729 codec is because the SIP debug shows this line: c=IN IP4 172.21.10.200 Is there any way to have asterisk put its extrnal IP there, would this even solve the issue? |
| 23:00.56 | chris349 | If I switch the codec to ulaw the audio works both ways. |
| 23:02.04 | WIMPy | I don't ssee how the CODEC could make a difference, but have you set it up for NAT correctely? (extern*) |
| 23:03.50 | chris349 | I dont see either, but that is what is hapening. I cant hardcode the IP address with externip= because it is dynamic. |
| 23:04.24 | WIMPy | Use ddns or STUN. |
| 23:07.51 | chris349 | I have enabled stun but it still shows the private IP in the main lines, it does put the public IP somewhere else but the main ones still have the private ip |
| 23:08.21 | [TK]D-Fender | chris349, "sip show settings" <- if you don't see your public IP ip top you've done something wrong... |
| 23:10.49 | chris349 | [TK]D-Fender, It shows Externaddr: (null), how would I get it to recognize the correct IP via STUN? |
| 23:16.07 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 23:16.45 | [TK]D-Fender | I'd confirm that you set that up properly... |
| 23:27.20 | *** part/#asterisk kharwell (kharwell@nat/digium/x-eblxdqdrnayxyalv) |
| 23:27.27 | chris349 | It is setup properly. |