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| 00:18.05 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); 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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| 07:00.53 | Haris | hello all |
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| 08:16.58 | nullr0ute | hi |
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| 08:40.33 | Haris | does a dialplan or asterisk reload disrupt active calls ? |
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| 09:05.25 | Haris | for a Q, what does it mean when moh is set to inherit ? Where is it going to inherit it from ? |
| 09:05.40 | Haris | for e.g., I have only one Q |
| 09:05.49 | Haris | one moh class i.e., default |
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| 10:01.27 | catphish | i'm using odbc adaptive cdr with mysql and it appears to regularly issue "LOCK TABLES cdr WRITE", does anyone know why this is, and whether it can be prevented? |
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| 10:11.21 | Rac-on | Haris: a dialplan reload wont disrupt active calls |
| 10:11.35 | Haris | awesome |
| 10:11.40 | Haris | how about core reload ? |
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| 10:11.48 | Haris | I think fpbx reload would do a core reload too ? |
| 10:12.01 | Haris | not sure |
| 10:12.14 | Rac-on | i dont know, this is #asterisk, no #freepbx ;) |
| 10:12.20 | Haris | =) |
| 10:14.31 | Rac-on | i am not sure about a core reload, reloads are mostly save, but i would sugest testing it |
| 10:16.20 | Samot | Haris: Stop asking questions in here. |
| 10:16.23 | Samot | You have FreePBX |
| 10:21.18 | catphish | ignore my question, was a 3rd party app locking the table :) |
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| 11:24.35 | TandyUK | help, somethign has gone very wrong with the gui we use wrapped around asterisk, not interested in that right now tbh |
| 11:25.17 | TandyUK | does anyone know of a quick way to find and replace the line "transport = udp" with "transport = udp,tcp,tls" in any file under /etc/asterisk/**/sip.conf |
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| 11:34.20 | TandyUK | sed -i 's/= udp$/= udp,tcp,tls/g' **/sip.conf ftw :) |
| 11:34.57 | TandyUK | now to bang some heads lol |
| 11:40.33 | tzafrir | ** is zsh, right? |
| 11:40.44 | TandyUK | ** means any subfolder |
| 11:40.57 | tzafrir | Right, but zsh-specific |
| 11:40.58 | TandyUK | as in /etc/asterisk/<tenant>/sip.conf |
| 11:41.02 | TandyUK | no thats bash |
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| 11:42.18 | Haris | questions regarding asterisk have nothing to do with Freepbx, even if I have it |
| 11:42.35 | Haris | how asterisk functions is not dependent on fpbx |
| 11:42.36 | tzafrir | The 'g' there is rather pointless, unless you have a comment in that line and you want to replace in that comment as well |
| 11:42.37 | Haris | all the time |
| 11:42.47 | Haris | or in every case |
| 11:42.56 | Haris | I can have a customized setup even with fpbx install |
| 11:44.07 | TandyUK | tzafrir: its was a quick and dirty hack, its not expected to be neat or perfect |
| 11:44.14 | TandyUK | but it did the job |
| 11:44.58 | Haris | I'm not always working on fpbx for asterisk things |
| 11:45.56 | Haris | The tendency of americans to show strength via anger is not constructive |
| 11:46.11 | Haris | in every case |
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| 11:48.14 | Haris | ah well |
| 11:48.23 | Haris | goes quiet |
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| 12:14.11 | adeel | on asterisk 11, is there a way to reload the asterisk.conf file, without restarting? |
| 12:18.21 | dadrc | don't think so |
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| 12:37.47 | adeel | hmm...note to self, don't define maxcalls in asterisk.conf |
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| 13:11.29 | TandyUK | adeel: 'module reload' doesnt do it? |
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| 13:38.16 | adeel | TandyUK: what module would i be reloading? |
| 13:38.31 | adeel | and nope |
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| 14:54.07 | skirmisha | hi guys |
| 14:54.30 | skirmisha | is there variable that can give me call duration while user is on phone? |
| 14:54.48 | skirmisha | probably something with channels? |
| 14:58.09 | [TK]D-Fender | what do you mean "variable"? |
| 14:58.17 | [TK]D-Fender | Where do you expect the get this information? |
| 15:00.06 | skirmisha | from agi script |
| 15:00.34 | skirmisha | i want to request live channel and to get duration of the call since the call is answered |
| 15:01.25 | [TK]D-Fender | How do you expect to be running an AGI while ON the phone? |
| 15:01.48 | [TK]D-Fender | Your description or sense of timing seems off |
| 15:02.17 | skirmisha | nope, it will be while ($call) whatever, meaning while call is up |
| 15:02.43 | [TK]D-Fender | how do you expect to PROCESS it during that call? |
| 15:03.03 | skirmisha | i want to create custom dialplan if call duration is more than 1 hour |
| 15:03.12 | igcewieling1 | skirmisha: what you need is not possible. re-thiink it. |
| 15:03.17 | [TK]D-Fender | AGI does not normall continue processing in the background |
| 15:03.47 | FrozenFire | https://wiki.asterisk.org/wiki/display/AST/CDR+Variables |
| 15:03.58 | [TK]D-Fender | FrozenFire, that's after the fact |
| 15:04.16 | FrozenFire | Ah |
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| 15:04.41 | igcewieling1 | Once Dial() happens no further processing in the dialplan will happen until the call ends. |
| 15:04.51 | skirmisha | well if for example i have IVR in agi i can manipulate the call |
| 15:05.28 | igcewieling1 | skirmisha: not the same. there is no Dial involved in an IVR |
| 15:05.40 | skirmisha | in that case i can run script in crontab giving me all live calls |
| 15:05.42 | [TK]D-Fender | and "manipulate" is VAGUE |
| 15:05.45 | [TK]D-Fender | there is no definition |
| 15:05.46 | FrozenFire | That seems odd. I'm absolutely by no means an expert here, but my company's application executes AGI commands while a call is active. |
| 15:05.51 | igcewieling1 | skirmisha: that is the only way. |
| 15:06.00 | [TK]D-Fender | you can effectively determine dialplan things to do. That eos not mean they arent BLOCKING |
| 15:06.04 | [TK]D-Fender | does* |
| 15:06.29 | [TK]D-Fender | <FrozenFire> That seems odd. I'm absolutely by no means an expert here, but my company's application executes AGI commands while a call is active. <- not while BRIDGED |
| 15:07.26 | skirmisha | what about channel answertime? |
| 15:07.41 | FrozenFire | Hrmm. I definitely need to learn more about Asterisk. I'm just troubleshooting an existing application, and I'm realizing how little I know about it. :P |
| 15:08.37 | seanbright | skirmisha: please describe your call flow. is this channel bridged to another channel or is it just floating around in dialplan? |
| 15:08.44 | skirmisha | probably i can store answer time in db and then using another script checking local time epoch i can compare both values and get call duration |
| 15:08.48 | [TK]D-Fender | seanbright, BRIDGED |
| 15:09.19 | skirmisha | it is bridged channel |
| 15:09.25 | seanbright | this schtick never gets old |
| 15:09.30 | [TK]D-Fender | AGI is NOT a background thing |
| 15:09.35 | [TK]D-Fender | it is not magic |
| 15:10.05 | seanbright | skirmisha: you will probably want to interact with AMI to determine how long the call has been live |
| 15:10.18 | [TK]D-Fender | that'd be my 1st choice |
| 15:10.24 | seanbright | BRIDGED |
| 15:10.31 | skirmisha | ok got this part, but can i store the answer time in variable when channel is bridged? |
| 15:11.33 | [TK]D-Fender | yes |
| 15:11.36 | [TK]D-Fender | Look at your dial |
| 15:12.08 | seanbright | skirmisha: if you use the CoreShowChannels AMI action, the responses will contain the duration of the call |
| 15:12.55 | seanbright | so there is no need to store anything |
| 15:14.32 | skirmisha | great, thanks guys |
| 15:14.50 | seanbright | you're welcome |
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| 16:53.05 | Merlin | is it possible to configure asterisk to write ACD data to _both_ mysql and a text file (i.e. /var/log/asterisk/queue_log) in extconfig.conf ? |
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| 17:01.08 | igcewieling1 | ACD data? |
| 17:01.33 | igcewieling1 | Asterisk should be able to log CDRs to mysql and a text file. Just configure them both. |
| 17:05.08 | Merlin | not CDR |
| 17:05.20 | Merlin | i'm talking about queue_log, queue data |
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| 18:16.46 | syadnom_ | hi all. anyone running siren7 on asterisk13? I'm running it in 'default' configuration (just loading the modules). Are there any tunables? Like tuning the FEC ? |
| 18:18.45 | [TK]D-Fender | codecs.conf <- if thre's an option it's in the obvious looking config |
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| 19:20.50 | syadnom_ | [TK]D-Fender, there aren't any entries there for siren |
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| 19:29.58 | syadnom_ | it seems impossible to find any documentation. the siren7 codec is a binary distribution from polycom and comes with no info, and I'm coming up completely empty when searching for siren7 and asterisk. There is a note in pjsip (im on chan_sip still) to set PLC on.. This seems to be an odd thing to have nearly no information available on... |
| 19:32.36 | igcewieling1 | syadnom_: that tells me almost nobody uses it. |
| 19:33.36 | syadnom_ | igcewieling1, I find that hard to believe. |
| 19:34.18 | drmessano | You're the first person I've seen ask about Siren 7 in years |
| 19:35.00 | igcewieling1 | I found g722 good enough for my needs. |
| 19:35.18 | syadnom_ | I'm kinda stuck with it. Polycom doesnt support opus in <vvx500 phones and I have customers with less-than-ideal internet connections. |
| 19:35.40 | syadnom_ | siren7 is vastly better than g711/722 when there is a little packet loss. |
| 19:36.52 | drmessano | I would imagine the number of people that know that or even care about that are very few. Pretty much G711 G722 and G729 are the three anyone ever asks about |
| 19:37.11 | drmessano | The lack of documentation is probably a really good indication of its usage |
| 19:37.29 | syadnom_ | idk, maybe polycom didn't make any configurable options :/ |
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| 19:38.35 | syadnom_ | I do hosted pbx, so codec matters much more for me than for the more common on-prem boxes. |
| 19:39.36 | drmessano | Hosted boxes are a lot more common than you would think |
| 19:40.33 | syadnom_ | right, but there are issues with hosted and I take business from 8x8 and ringcentral weekly because they still use g711/722 |
| 19:40.54 | drmessano | Most of the time I see people using G711, and if bandwidth is a consideration its G729 |
| 19:41.33 | syadnom_ | those have nothing for PLC for FEC, so there will be audio issues on consumer grand connections. inevitable. |
| 19:41.45 | drmessano | Shrug. |
| 19:42.00 | syadnom_ | drifted off topic I suppose |
| 19:42.32 | syadnom_ | basically I'm trying to stretch a little more out of siren7 because polycom's lower level phones don't support opus. |
| 19:42.50 | syadnom_ | siren7 is pretty good, but the defaults in the polycom asterisk module leave some on the table. |
| 19:43.05 | drmessano | I'm not doubting or arguing any of the technical merits of what you were saying, but the statement about siren7 being incredibly uncommon is not inaccurate. |
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| 19:44.53 | drmessano | You're like the first person I've seen on IRC ask about it since the modules were put up years ago. I've looked into siren7 and siren14 myself on occasion and never have found anyone bragging about using it |
| 19:45.41 | syadnom_ | does digium do any analytics on codec usage ? |
| 19:45.47 | drmessano | And when I say I've looked into it I mean I have googled for information on it and not found forum posts or blogs or anything where people talk about the merits of it and actually having to put it anywhere |
| 19:46.35 | Samot | How long has opus been out? |
| 19:46.40 | syadnom_ | not long. |
| 19:46.50 | Samot | OK. |
| 19:47.00 | drmessano | I'm sure they have download statistics and from that you could extrapolate some usage data. Surely they have done some analysis to determine whether or not they should continue supporting those codecs. |
| 19:47.14 | drmessano | Since they have to build them for every major release |
| 19:47.16 | syadnom_ | a few months really. came with v14.0.1 I think, then backported to 13 |
| 19:47.25 | Samot | Well Polycom is probably working on it. |
| 19:47.46 | syadnom_ | polycom builds the models for asterisk. |
| 19:47.54 | Samot | ? |
| 19:48.09 | syadnom_ | digium doesn't handle the siren7/14 codecs, polycom does. |
| 19:48.15 | drmessano | Oh ok |
| 19:48.20 | syadnom_ | polycom owns it and gives a royalty free license... |
| 19:48.28 | drmessano | Right |
| 19:48.56 | drmessano | Well then, that gives me even less Hope |
| 19:48.57 | syadnom_ | blows my mind, siren7 is massively better than g.711 on a consumer connection. I did a ton of testing and I can loose as much as 20% of packets (if they are randomly distributed) on siren7. |
| 19:49.14 | syadnom_ | if it's sequential, 5-7% loss has no effect on audio. |
| 19:49.23 | syadnom_ | That's impressive. |
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| 19:49.29 | Samot | I'm stuck on "polycom makes models for asterisk" |
| 19:49.36 | drmessano | Modules |
| 19:49.42 | drmessano | They make the modules |
| 19:49.46 | Samot | Ahhh |
| 19:49.49 | Samot | Yes. |
| 19:50.02 | syadnom_ | oops, typo |
| 19:50.10 | Samot | So we're upset that Polycom hasn't instantly released firmware with opus? |
| 19:50.21 | drmessano | Uh no? |
| 19:50.25 | Samot | OK |
| 19:50.29 | syadnom_ | no, they will never release opus firmware for <VVX500 |
| 19:50.32 | drmessano | that wasn't at all what he was saying |
| 19:50.34 | Samot | I'm catching up. |
| 19:50.41 | syadnom_ | they already released it for >=VVX500 |
| 19:50.49 | syadnom_ | and it works like a champ. |
| 19:51.00 | Samot | Did the 500's get a new model upgrade as well? |
| 19:51.28 | syadnom_ | no, but it has a fast enough CPU to handle OPUS, the 4x0 and lower don't |
| 19:51.43 | Samot | OK, then I'm shocked. |
| 19:51.48 | syadnom_ | why so? |
| 19:52.01 | Samot | Because the 500 generally gets good stuff. |
| 19:52.15 | syadnom_ | ? maybe we aren't on the same page here. the 500 DOES support OPUS |
| 19:52.19 | Samot | It's an "executive" phone so it generally gets all the trimmings. |
| 19:52.40 | Samot | I could see the 310/410's never supporting it. |
| 19:53.01 | Samot | But the 411 was put out to solve CPU issues the 410's had. |
| 19:53.03 | drmessano | I'm so confused now |
| 19:53.06 | syadnom_ | not that you'd ever actually do this (maybe?) but a 24kbps VBR OPUS stream with 50% PLC can truly drop 50% of packets |
| 19:53.24 | syadnom_ | Samot, I have confirmation from Polycom that the 310/410 will never support opus for lack of CPU power. |
| 19:53.32 | Samot | Oh yeah. |
| 19:53.38 | Samot | I totally believe that. |
| 19:53.43 | Samot | That's why there's the 411 now. |
| 19:53.52 | Samot | It was a hardware upgrade for the 410. |
| 19:53.56 | Samot | Memory, CPU.. |
| 19:54.02 | Samot | And they added a USB port. |
| 19:54.42 | Samot | That's why I'm shocked about the 500 not getting opus. |
| 19:54.48 | Samot | It does have the power to handle it. |
| 19:55.04 | syadnom_ | again, the 500 DOES support opus right now |
| 19:55.12 | Samot | OK |
| 19:55.18 | Samot | So I was the confused one. |
| 19:55.19 | Samot | Sorry. |
| 19:56.04 | syadnom_ | idk if the 401/411 supports opus though, googling now |
| 19:58.06 | Samot | I think it's based on the firmware it can support. |
| 19:58.18 | Samot | 5.4.1 introduced opus. |
| 19:58.18 | syadnom_ | polycom docs suck, they are so non-specific. |
| 19:58.36 | Samot | Sorry, 5.4.0 |
| 19:58.42 | Samot | 5.4.1 was for the new line of phohnes. |
| 19:58.47 | Samot | 5.4.1 was for the new line of phones |
| 19:58.52 | syadnom_ | 5.4.1 'implies' that it's enabled on all phones, but it's not and it doesn't give a list. |
| 19:59.19 | syadnom_ | I'm going to upgrade one right now just to check... |
| 19:59.55 | Samot | This feature is currently supported on VVX 500 and 600 phones only. |
| 20:00.02 | Samot | That was as of the 5.4.0 release. |
| 20:00.08 | Samot | Before the hardware upgrades. |
| 20:01.10 | syadnom_ | really, the 400 and lower phones are really slow anyway so I don't know why i'd expect them to handle it |
| 20:02.07 | syadnom_ | *but* opus isn't much more complex than g722, and g.722.1c (siren14) is less complex than g.722. They support g.722 but not siren14 and opus... so it almost seams like it's just a decision, not truly a hardware limitation. |
| 20:03.47 | Chainsaw | It's probably DSP code rather than main CPU code. |
| 20:04.17 | Chainsaw | You make cheap handsets cheap by putting in less hardware. That includes DSP resources. |
| 20:04.54 | syadnom_ | sure, but if it can do g.722 maybe it could handle the less complex g.722.1c. |
| 20:05.42 | Samot | Well.. |
| 20:05.58 | Samot | The answer on that is no. The VVX411 does not support opus. |
| 20:07.02 | syadnom_ | i don't get why polycom isn't supporting opus across the whole line ('x01/x11' models) |
| 20:07.30 | syadnom_ | yealink supports it on all their new phones (well...they say they support it) and grandstream and snom are soon to follow. |
| 20:08.23 | Samot | Yeah, I a bit surprised the 411 doesn't support it. |
| 20:08.35 | syadnom_ | I just verified on the 411, opus is there but in grey. |
| 20:08.48 | Samot | Yeah me too. |
| 20:10.04 | syadnom_ | have you guys every messed with opus? |
| 20:10.07 | drmessano | syadnom_: If Polycom is building the binary for Asterisk what about reaching out to them regarding the tuning or options? |
| 20:10.18 | syadnom_ | drmessano, you ever asked polycom for support? |
| 20:10.19 | drmessano | For siren7 |
| 20:10.27 | drmessano | Yes. It's horrible |
| 20:10.30 | syadnom_ | yeah |
| 20:10.32 | syadnom_ | exactly |
| 20:10.42 | drmessano | I just didn't know if you had even attempted |
| 20:10.48 | drmessano | Miracles do happen sometimes |
| 20:10.48 | syadnom_ | 'have you asked your distributor for support?' |
| 20:10.58 | syadnom_ | yes, I did, months ago. |
| 20:11.46 | syadnom_ | opened a ticket, it's rejected saying to through my vendor :/ posted on the forums but our good friend Stephen rejected me with his cut'n'paste 'open a ticket with your vendor' line too. |
| 20:13.18 | drmessano | What about asking someone at Digium who their point of contact is? In this case you're not asking for general Polycom hardware support, you're asking for support of this module, which is probably outside of the wheelhouse of standard Polycom support, so even if Digium can't help you maybe they can put you together with a dev. |
| 20:14.29 | syadnom_ | I find that digium recoils when discussing hardware support for other vendors |
| 20:14.46 | drmessano | It's not hardware support, it's support of an asterisk module that they are obviously distributing |
| 20:14.59 | syadnom_ | digium isn't distributing it |
| 20:15.49 | drmessano | Really, because I've only ever downloaded it from the digital download site |
| 20:15.52 | drmessano | Digium |
| 20:16.48 | syadnom_ | where is this? because I download it from Polycom's site. |
| 20:17.15 | drmessano | http://downloads.digium.com/pub/telephony/ |
| 20:17.23 | drmessano | It's always been there |
| 20:18.00 | syadnom_ | huh, I get it through my polycom portal downloads page. |
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| 20:18.29 | drmessano | I don't see the validity of arguing Digium won't help you when they clearly distribute it.. so they have some contact at Polycom... which is who could be helping you |
| 20:18.48 | drmessano | That's the contact you need.. not Tier 1 hardware support |
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| 20:24.29 | malcolmd | the modules that we provide are just the ITU code. the convenience factor is that we can provide them for your use with asterisk without your having to go and get an individual usage license from polycom yourself. |
| 20:25.06 | malcolmd | i'm not aware of any native plc or fec baked into the ITU code for the siren codecs |
| 20:25.15 | malcolmd | also, opus requires substantially more cpu time than g.722 |
| 20:25.29 | malcolmd | and g.722.1(c) require more than g.722, but less than opus |
| 20:26.06 | malcolmd | one of the things that makes g.722.1c extra hungry is the resampling in asterisk between 32kHz and other rates, in order to get a call going with non siren14-capable endpoints |
| 20:26.25 | malcolmd | but, they're both lightweights compared to opus |
| 20:29.39 | malcolmd | digium's support department won't handle support queries against the siren codecs we distribute, because they're not sold, commercial products. if you have an issue with one of the siren downloads, you can open an issue on issues.asterisk.org |
| 20:31.36 | syadnom_ | malcolmd, there is reference in the pjsip driver of enabling PLC on siren7 |
| 20:32.04 | malcolmd | in chan_pjsip for asterisk or in pjproject's pjmedia? |
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| 20:32.19 | syadnom_ | i closed my tab, gotta find it again.. |
| 20:32.38 | malcolmd | to that though, it's just "i'm not aware" :) |
| 20:35.15 | syadnom_ | basically, if I make a direct call between 2 siren7 endpoints, and I drop packets in between, I can drop as many as 2x the packets as I can running this same call through asterisk. I believe that this is because I can't set the PLC behavior in the asterisk module. |
| 20:36.19 | malcolmd | the implementation in the endpoint may have baked in plc. the implementation in asterisk may not. you'll see something similar if you take a digium phone and make a phone to phone call using g.722 (not siren) as compared to running it through asterisk. the implementation in the phone is better. |
| 20:36.38 | syadnom_ | sure. |
| 20:37.22 | syadnom_ | if the call is siren7<>siren7 (ie passthrough) it sounds great. unfortunately, most of my calls are g711<>Siren7 |
| 20:37.26 | malcolmd | there's a native plc patch for asterisk floating around on the issue tracker |
| 20:38.20 | malcolmd | are you really doing passthrough (ye olde direct media) or are we just talking about what has to happen when g.711 has to be upsampled and downsampled? |
| 20:40.15 | syadnom_ | direct media, ie no transcoding. |
| 20:40.46 | syadnom_ | I'm dropping packets between asterisk and the phone so asterisk (or the other endpoint) isn't in the mix. |
| 20:43.31 | syadnom_ | I think I'm looking at that patch (25629 ) |
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| 20:46.14 | syadnom_ | the ugly solution here is to put in an SBC and run opus from asterisk to the SBC :/ then I can just use g.722 and be done with it. |
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| 20:53.44 | zaf | hey malcolmd, couple dpma questions: 1- is there a way to have the built-in voicemail app require a password from the user to access messages? |
| 20:55.11 | malcolmd | 1 - yes |
| 20:55.46 | malcolmd | see require_password on https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration |
| 20:55.54 | zaf | and 2 - i know d80 doesn't support host_alternate sip registry, but any good workarounds if i have a failover asterisk server? |
| 20:56.08 | zaf | thx |
| 20:56.08 | malcolmd | create a type=application, application=voicemail, require_password=yes and apply it to a phone |
| 20:56.10 | malcolmd | np |
| 20:56.49 | malcolmd | 2 - no good ones. if you're doing address takeover, rather than something on another ip, i'm going to assume something's going to break |
| 20:57.38 | zaf | guess dns failover isn't a good option? |
| 20:57.39 | malcolmd | but that breakage wouldn't be unique to the d80 |
| 20:57.53 | malcolmd | dunno; i don't do a lot of work in the high-availability area |
| 20:58.07 | zaf | any plans to add host_alternate to d80? |
| 20:58.26 | malcolmd | i can't give you any dates for anything that's not currently available |
| 20:58.30 | malcolmd | standard digium line :) |
| 20:58.37 | syadnom_ | zaf, for failover, if your phones support it, look at dns-srv. I'm using this with yealinks and if the phone loses it's registration, it will lookup dns-srv records to find the next server |
| 20:59.13 | zaf | malcolmd, is lack of alternate_server considered a bug? :P |
| 21:02.01 | zaf | syadnom_, interesting. i'll look at that |
| 21:05.54 | syadnom_ | zaf, I basically had to switch to it because I consider yealink to have a bug. specifying a primary and a failover server meant that once the retry count ran out, the phone would never try to register again until it was rebooted. |
| 21:06.11 | syadnom_ | setting just the primary sip server, it will retry forever. |
| 21:06.28 | syadnom_ | and set the transport to DNS-SRV that is. |
| 21:07.07 | syadnom_ | Also, this was the only way I could reliably get the phone to use UDP -or- TCP without reprovisioning it. Then I setup a UDP and a TCP SRV record for each server. |
| 21:18.49 | malcolmd | zaf, nice try ;) |
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| 22:01.16 | sunspots | I'm trying to debug a problem with asterisk crashing, I've enabled core dumps, verified they're working with a kill -3, but whenever I can reproduce the actual problem it doesn't dump a core :( |
| 22:02.09 | sunspots | The very last line it spits out is "VAL: tried to access phoneset in 3 type val", which is probably just gibberish due to some random memory error, but it's consistent. |
| 22:02.33 | sunspots | is there a different channel I should be asking for help in by chance? |
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| 22:12.01 | sunspots | Also, when I try to use valgrind to debug, I get an illegal instruction at start up. I'm attempting to launch with "valgrind --suppressions=/usr/src/asterisk-11.23.1/contrib/valgrind.supp --log-fd=9 asterisk -vvvvcg 9>valgrind.txt" |
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| 22:39.16 | ttbakiatwoam | I am trying to write an IF statement to SET a CallerID via a dialplan and I am having syntax issues it always returns false, the ${OUTQUEUE} variable is used earlier in a set statment and is functioning so I know the information is there, and it's numeric exten => 28,n,Set(CALLERID(all)=${IF($[ ${OUTQUEUE} = 1000 ]?["true" <3213213210>][:"false" <8908908900>])}) |
| 22:40.10 | ttbakiatwoam | I think the issue is somewhere in the evaluation portion of the if statement. Has anyone worked with these before? |
| 22:47.48 | [TK]D-Fender | You don't do {} after the ? in your IF |
| 22:47.50 | [TK]D-Fender | [] |
| 22:47.52 | [TK]D-Fender | rather |
| 22:48.50 | [TK]D-Fender | <PROTECTED> |
| 22:49.36 | [TK]D-Fender | IF($[evaluation if you feel like it]?true:false) |
| 22:49.46 | [TK]D-Fender | no extra stuff around the result part |
| 22:53.19 | ttbakiatwoam | Awesome! That totally worked thank you! |
| 22:54.05 | ttbakiatwoam | I understand that I can nest if statements to get and if else style result correct? |
| 22:55.07 | ttbakiatwoam | IF($[evaluation if you feel like it]?true:IF($[evaluation if you feel like it]?true:false)) would something like this work? |
| 23:03.04 | [TK]D-Fender | if you continue to use your functions properly, yes |
| 23:03.39 | ttbakiatwoam | Awesome thanks |
| 23:04.13 | [TK]D-Fender | You're welcome |
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| 23:56.09 | sunspots | I'm trying to debug an issue where Asterisk crashes after about a 1000 calls, can someone point me to where I should go to get some advice? |
| 23:56.39 | sunspots | I'll try asterisk-dev |
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