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| 00:20.11 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0-rc1 (2018/07/03), Standard: 15.5.0-rc1 (2018/07/03); 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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| 01:30.45 | eqw | Maliuta_: thank you! |
| 01:56.52 | Maliuta_ | eqw: you mean I got something right? ;) |
| 01:57.10 | Maliuta_ | well colour me not surprised :D |
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| 02:33.49 | eqw | I use 3 softphones. When I call to some numbers there's no sound in CSipSimple though all is fine with another softphones or numbers. Codec is ulaw. I see via sniffer that 172-length udp packets are sent to CSipSimple. What else should I check? |
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| 06:30.08 | scientes | is there cheap usb POTs adapters? |
| 06:30.16 | scientes | what is the cheapest hardware for POt support? |
| 06:30.54 | scientes | cause this is so expensive https://www.amazon.com/OBi200-1-Port-Adapter-Support-Service/dp/B00BUV7C9A/ |
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| 09:22.38 | sibiria | scientes: i know that linksys has (or had) a POTS-to-ethernet model for like $20 |
| 09:22.47 | sibiria | but maybe that's not a viable alternative |
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| 11:32.52 | Sminty | Hello there |
| 11:33.24 | Sminty | someone around that could help a little troubleshooting? |
| 11:34.31 | n3t | Sminty: what's your problem? |
| 11:34.45 | n3t | Maybe someone will be able to help. |
| 11:35.04 | Sminty | the good ol' "I can dial out, but calls won't come in" |
| 11:35.25 | Sminty | and I am not getting very much info asterisk is throwing at me when the call gets in |
| 11:35.37 | Sminty | atleast google didn't help that much |
| 11:46.42 | sibiria | as in "i get no audio" or "asterisk can't see SIP signaling coming in"? |
| 11:47.08 | sibiria | if the latter, it sounds very much like a firewall miss resulting in external traffic just not reaching through |
| 11:49.58 | Sminty | it's more like ERROR[1841]: pjproject:0 <?>: sip_parser.c Error parsing '4294967295': String value was greater than the maximum allowed value. |
| 11:50.10 | Sminty | when the call comes in |
| 11:51.38 | Sminty | and repeats until the automated message kicks in that the number is currently unavailable or I hang up |
| 11:53.07 | file | you'd need to provide a trace of the SIP traffic |
| 11:55.40 | Sminty | ok |
| 12:01.20 | Sminty | alrighty, server running and got it to talking, sip client registered |
| 12:02.12 | Sminty | okay what part of it could be of use? |
| 12:02.50 | file | the SIP traffic itself? the SIP parser is upset at something they are sending |
| 12:03.35 | Sminty | PJSIP invalid value error exception when parsing 'Contact' header on line 3 col -1: |
| 12:04.25 | Sminty | thats the other message I got |
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| 16:04.21 | DaveH | Hi all |
| 16:05.21 | DaveH | Am in the UK and having the dreaded no caller ID issue |
| 16:06.23 | DaveH | I have cidsignalling to v23 and cidstart to polarity, signalling to fxs_ks and callerid = asreceived, but still no caller id showing in the logs |
| 16:11.33 | DaveH | ok, that is weird.. it's now working |
| 16:11.38 | DaveH | you guys are awesome, lol |
| 16:17.37 | danjenkins | Hey all |
| 16:17.52 | danjenkins | Will devcon be on the 9th? |
| 16:18.37 | file | cresl1n is working on details |
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| 16:31.35 | *** join/#asterisk simbalion (~simba@45.77.108.203) |
| 16:31.53 | simbalion | Hi, I have a configuration that was working before but now calls sit in a queue and never get routed to phones? |
| 16:32.16 | [TK]D-Fender | is that ... a question? |
| 16:32.54 | simbalion | There is a call in the queue and a registered extension which shows not in use, strategy is ringall, but it doesn't ring. I'm connected with two different devices |
| 16:33.49 | [TK]D-Fender | Show us the whole status of what's going on along with the queue config |
| 16:33.53 | simbalion | [TK]D-Fender: Perhaps a better question for you might be, what should I look at to diagnose the problem? |
| 16:34.38 | [TK]D-Fender | queue status, memeber status, device status. queue definition. All the obvious usual stuff |
| 16:34.43 | Samot | 1) Is the extension a member of the queue? |
| 16:34.54 | Samot | 2) Do you see the queue try to send the call to said agent? |
| 16:36.12 | simbalion | Here is sip peers and queue https://hastebin.com/raw/adafiyujew |
| 16:36.44 | Samot | Which queue is having the issue? |
| 16:37.01 | simbalion | Samot: Yes, the extensions are members. No, I don't see any log info about transferring the call. All queues have the issue |
| 16:37.20 | simbalion | it _was working_ before, like 2 weeks ago, and I didn't modify the queue settings at all |
| 16:37.34 | simbalion | The only thing I changed is my SIP trunk provider |
| 16:39.05 | Samot | Did you just do a sip reload or did you restart Asterisk after changing SIP providers? |
| 16:39.25 | simbalion | I did 'reload' which reloads everything I thought? |
| 16:39.36 | Samot | So you did a global reload? |
| 16:39.39 | Samot | Not just SIP? |
| 16:39.45 | simbalion | right |
| 16:39.47 | Samot | OK. |
| 16:39.50 | [TK]D-Fender | Stop talking about "It was working before". this doesn't offer anything to go on |
| 16:39.55 | Samot | Leads to the next question.. |
| 16:40.04 | Samot | Are you the only one with the ability to make changes? |
| 16:40.04 | simbalion | The calls are being answered by the PBX and routing properly to the queue, but they aren't being sent to member phones apparently |
| 16:40.11 | simbalion | Samot: yes |
| 16:40.22 | [TK]D-Fender | <[TK]D-Fender> queue status, memeber status, device status. queue definition. All the obvious usual stuff |
| 16:40.24 | Samot | And this has been happening since the reload? |
| 16:40.31 | [TK]D-Fender | And version info..... |
| 16:40.48 | simbalion | [TK]D-Fender: I respectfully disagree, having been diagnosing software problems for 20+ years I can say with certainty that knowing all the details helps narrow down the source of the problem. Knowing that the configuration used to work is important. Thanks :) |
| 16:41.00 | Samot | And this has been happening since the reload? |
| 16:41.07 | simbalion | Samot: I haven't had it working with the new SIP trunk yet, I just got my DID transferred today from the old provider. |
| 16:41.17 | [TK]D-Fender | saying "it was working before" doesn't tell us why it isn't working NOW' |
| 16:41.21 | Samot | When did the issue start? |
| 16:41.35 | [TK]D-Fender | Show the configs, show ALL of the status dumps |
| 16:41.50 | Samot | You added a new SIP trunk, you may have done some other dialplan changes.. |
| 16:41.54 | Samot | And you did a reload... |
| 16:42.01 | Samot | Did this issue start after the reload was done? |
| 16:42.13 | simbalion | Samot: Today, presumably. I'm being vague because I was seeing this issue with the old provider, but I don't know if it was the same cause or possibly related to the reasons I left their service |
| 16:42.36 | Samot | You're missing my point. |
| 16:42.39 | Samot | You did a reload. |
| 16:42.42 | simbalion | [TK]D-Fender: You need to be specific. There's like 20 config files, which ones do you want to see. |
| 16:42.53 | seanbright | "queue status, memeber status, device status. queue definition" |
| 16:42.55 | Samot | I'm trying to determine if the reload causes something to be changed. |
| 16:42.58 | [TK]D-Fender | I only stated ONE |
| 16:43.02 | [TK]D-Fender | QUEUE config |
| 16:43.15 | [TK]D-Fender | the rest were status dumps of clearly ever piece involved in this |
| 16:43.49 | [TK]D-Fender | And that also means device status dumps for all of those AND current active * channels, not just a peer dump |
| 16:43.58 | [TK]D-Fender | lok at the shit that is in PLAY here. |
| 16:44.18 | simbalion | Samot: Your question makes no sense. Of course I did the reload after changing the trunk settings. OF course the problem was noticed after the reload. |
| 16:44.34 | simbalion | [TK]D-Fender: One moment |
| 16:44.36 | Samot | Did the problem exist before the reload? |
| 16:45.08 | Samot | That was my question. |
| 16:45.12 | Samot | It makes perfect sense. |
| 16:45.17 | simbalion | queues.conf: https://hastebin.com/raw/ozokaturim |
| 16:46.14 | [TK]D-Fender | We should also see the call progress through the queue |
| 16:46.27 | simbalion | [TK]D-Fender: How can I view that? |
| 16:46.44 | [TK]D-Fender | verbose 10 at cli and watch the call enter |
| 16:47.06 | simbalion | it says no such command |
| 16:47.20 | Samot | core set verbose 10 |
| 16:47.36 | simbalion | Thanks! :) |
| 16:47.40 | Samot | He was just giving a verbosity level. |
| 16:48.12 | simbalion | hrm wait maybe its working, my zoiper app says I've missed 10 calls, but it never rang |
| 16:48.23 | simbalion | however I also have a grandstream handset that also never rang |
| 16:48.41 | Samot | You said the logs showed no calls being sent. |
| 16:48.53 | simbalion | testing now |
| 16:52.25 | simbalion | So if I have Zoiper registered (android SIP app), I see this: https://hastebin.com/raw/ezevaqiquc |
| 16:52.51 | Samot | So Zoiper is returning a busy |
| 16:53.00 | simbalion | However if I unregister the zoiper app so only my grandstream handset is connected I see my extension as unavailable, and the "is busy" errors aren't happening. I'm just sitting in the queue indefinitely |
| 16:53.16 | simbalion | Maybe both my devices are screwed up lol |
| 16:53.23 | simbalion | But I don't know how that could be |
| 16:53.30 | seanbright | are they both registering with the same user/pass? |
| 16:53.52 | simbalion | seanbright: Yep |
| 16:53.56 | Samot | Nope. |
| 16:54.00 | Samot | Doesn't work that way |
| 16:54.07 | simbalion | It was working |
| 16:54.10 | Samot | Chan_SIP only supports one location. |
| 16:54.13 | simbalion | It would ring both devices |
| 16:54.21 | Samot | It can't. |
| 16:54.30 | Samot | Chan_SIP only supports one contact |
| 16:54.32 | simbalion | Okay I will try configuring the android with a different password, perhaps the previous behavior was a fluke |
| 16:54.52 | Samot | It would not ring both devices. It would only have the last location to be registered. |
| 16:55.05 | simbalion | Samot: but it would ring both devices, it did so for a couple of weeks |
| 16:55.32 | Samot | Chan_SIP does not support multiple contacts for an AOR/peer. |
| 16:55.55 | Samot | So you cannot and should not have 100 on both the Grandstream and Zoiper. |
| 16:56.30 | Samot | They will overwrite the locations of each other with each registration. |
| 16:56.50 | simbalion | Samot: What is '100' in reference to? |
| 16:56.57 | Samot | An "extension" |
| 16:57.00 | Samot | A peer name |
| 16:57.03 | Samot | the SIP Account. |
| 16:57.12 | seanbright | "simba" |
| 16:57.33 | [TK]D-Fender | moves on to more productive matters |
| 17:00.11 | simbalion | Samot & seanbright ty |
| 17:00.42 | simbalion | Zoiper is still rejecting calls as busy, possibly because I'm calling from the same handset, which used to work and is probably an issue with how android is configured |
| 17:03.18 | simbalion | Success! I was able to receive a call on the grandstream while calling from my mobile |
| 17:03.40 | simbalion | I had to re-register it because it had "unspecified" in the IP section, probably due to conflicting with the other device |
| 17:04.00 | simbalion | Thanks for the help fellas |
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| 18:50.10 | Demon_VoIP | Hi. Page https://www.asterisk.org/downloads/asterisk/all-asterisk-versions has wrong url for "Download - 15.5.0-rc1". Does nobody use it? :) |
| 18:51.15 | seanbright | checks with nobody |
| 18:51.37 | [TK]D-Fender | http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 18:51.42 | [TK]D-Fender | your link? Dunno... |
| 18:51.50 | [TK]D-Fender | I use the full folder... |
| 18:53.03 | Demon_VoIP | Where is no problem to find tar.gz for me :) But href is wrong.. |
| 18:54.12 | Demon_VoIP | And thanks.. i'll not file issue for fix :) |
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| 19:21.47 | kharwell | Demon_VoIP: thanks for catching that. The page has been updated with the correct links. (although might take a few minutes to fully sync) |
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| 20:06.23 | jeffspeff | is there still a need to use --with-pjproject-bundled with ./configure ? |
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| 20:17.15 | jeffspeff | I found the answer |
| 20:17.15 | jeffspeff | Beginning with Asterisk 15.0.0, it is enabled by default but can be disabled with the --without-pjproject-bundled option to ./configure. |
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| 21:02.07 | rfr__ | Hello All, I am still having problems getting a dhadi/Asterisk configuration ( based on an old Zaptel/Asterisk configuration that does work ) to work at all. Still no dial tone but I don't see any obvious errors from the CLI. Here's my CLI output and config files. Have experiments changing fxo_ks to fxo_ls, FXOKS to FXOLS in chan_dhadi.conf and system.conf, but that did not help and made my fxo channels disappear. https://pastebin.co |
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