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| 06:51.08 | binary109 | hi anyone here use pbxes.org? |
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| 08:44.16 | binary109 | anyone here use pbxes.org? |
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| 10:06.40 | snmp | greetings |
| 10:06.51 | snmp | iax realtime anyone? |
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| 10:09.12 | snmp | no? |
| 10:09.17 | snmp | perche? |
| 10:10.12 | skrusty | http://www.voip-info.org/wiki/view/Asterisk+RealTime+IAX |
| 10:10.38 | snmp | yep |
| 10:10.43 | snmp | thanks alot |
| 10:10.48 | snmp | btdt |
| 10:11.35 | snmp | the question is |
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| 10:11.59 | snmp | drumroll |
| 10:12.23 | snmp | iaxpeers and iaxusers |
| 10:15.23 | snmp | in the iax.conf single record corresponds to trunk, type friend, with a specified username |
| 10:15.40 | skrusty | type varchar(6) NOT NULL, ? |
| 10:16.19 | snmp | but in realtime i cant make it single |
| 10:16.44 | snmp | i need a record for user and record for trunk? |
| 10:17.08 | snmp | two separate tables? |
| 10:17.32 | skrusty | could use views? tbh, i dont use iax |
| 10:18.02 | snmp | hwy not |
| 10:19.23 | wdoekes | what kind of question is that? |
| 10:20.05 | snmp | how to correctly fill the iax peers |
| 10:20.07 | skrusty | a poorly typed one? :) |
| 10:20.26 | snmp | srry for broken lang) |
| 10:20.30 | skrusty | hehe :) |
| 10:21.27 | snmp | or how to move trunk record from iax config to db |
| 10:21.33 | wdoekes | snmp: don't ask why people don't use module X. the obvious answer is that they don't need module X. as for the iaxpeers/iaxusers. you might get away with defining one table and setting that same table (e.g. "iaxfriends") for both tables in extconfig |
| 10:23.38 | snmp | thats trunk im talking about |
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| 10:24.48 | snmp | in other words i want to transfer single trunk record to a single row in a table |
| 10:29.42 | wdoekes | let me quote myself: defining one table and setting that same table (e.g. "iaxfriends") for both tables in extconfig |
| 10:31.18 | skrusty | assuming the type field doesn't cause an issue wdoekes suggestion sounds spot on to me |
| 10:31.42 | skrusty | most examples i've seen for sippeers does that |
| 10:34.34 | snmp | wdoekes: that was the first thing ive done |
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| 10:39.18 | snmp | looks like im still unclear.. well thaks for the patience and things) |
| 10:39.34 | skrusty | what have you tried? |
| 10:39.58 | skrusty | what did you put in extconfig ? |
| 10:40.29 | snmp | 1st try single record for peers |
| 10:40.44 | snmp | asterisk shows peer not user |
| 10:40.52 | wdoekes | skrusty: sippeers is actually a special case. sipusers is never used since 1.8(?), so you *must* use sippeers for both |
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| 10:41.28 | snmp | 2nd try two record pointing to a single table |
| 10:41.30 | skrusty | wdoekes: good point :) |
| 10:41.45 | skrusty | what exactly _isn't working_ |
| 10:42.29 | snmp | simultaneous presence of peer and user |
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| 10:43.53 | wdoekes | snmp: you're not showing us anything |
| 10:44.22 | snmp | did you get my problem?) |
| 10:45.59 | wdoekes | did you get ours? |
| 10:48.01 | snmp | this isnt polite to answer with question |
| 10:51.15 | Chainsaw | snmp: wdoekes is one of our most polite helpers. |
| 10:51.27 | Chainsaw | snmp: I really suggest that you reconsider your approach. |
| 10:51.35 | snmp | appreciate it |
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| 11:40.16 | davlefou | hi, i look for an sample code beet asterisk and python to get arg from asterisk. |
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| 13:32.17 | file | wobbles |
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| 13:42.23 | binary109 | anyone got pbxes.org? |
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| 13:58.51 | morsing | JonathanD! |
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| 14:12.24 | Katty | my cup says All American Refreshment. |
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| 14:31.19 | saint_ | hi . can someone tell me how many END events (E bit = true) a SIP client is supposed to send when sending DTMFs ? |
| 14:34.55 | [TK]D-Fender | [09:42]binary109anyone got pbxes.org? <- as in "uses their services?" |
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| 14:40.08 | [TK]D-Fender | binary109: If so... maybe some people.. then again this is #asterisk ... the land where people tend to install Asterisk themselves and control their own system rather than pay for some hosted service they don't have real control over. |
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| 14:40.48 | file | control? does anyone really ever have CONTROL mannnn? |
| 14:41.08 | [TK]D-Fender | file: "high-quality delusion" |
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| 14:44.07 | ghost75 | does an ata already produce 30ms on local channel? |
| 14:44.16 | ghost75 | of jitter |
| 14:46.07 | [TK]D-Fender | ghost75: ATA's don't cause jitter. |
| 14:46.56 | ghost75 | i was assuming that RTPAUDIOQOS is only local channel and RTPAUDIOQOSBRIDGED is channel to peer |
| 14:47.11 | ghost75 | is that right? |
| 14:51.30 | ghost75 | was wondering about local jitter rx, where it comes from: https://onedrive.live.com/redir?resid=B371043B95253312%21110 |
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| 16:53.20 | dan_j | Hi, at midnight every night without fail, asterisk marks all sip peers as Unavailable and drops all calls. I run Nagios which also notices the problem with asterisk. However Nagios ping tests to that server are fine and don't have any problems. |
| 16:54.11 | [TK]D-Fender | Nothing in * has a timer like that on it so something youa re doing is causing a drop-off. |
| 16:54.15 | dan_j | This is the CLI which doesnt help. |
| 16:54.15 | dan_j | http://pastebin.com/8MCntmq6 |
| 16:54.40 | Kobaz | something i never actually dug into... what's the proper way to end a fastagi connect? |
| 16:54.42 | dan_j | I've tried to diagnose this in the past, but couldnt see anything. |
| 16:55.00 | Kobaz | [2014-03-17 12:43:53.967] -- <SIP/6860-00000108> AGI Script agi://127.0.0.1:20....etc completed, returning 4 ERROR[30781]: utils.c:1164 ast_carefulwrite: write() returned error: Broken pipe |
| 16:55.16 | Kobaz | how do you avoid the broken pipe |
| 16:55.22 | Kobaz | when you're done with the fastagi |
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| 16:57.29 | [TK]D-Fender | Same rules as regular AGI.... don't write until you've read |
| 16:58.21 | [TK]D-Fender | danWhat about debug from beofre that poitn... where we can see if modules are reloading, etc.... |
| 17:05.36 | Katty | TOILET PAPER ROLLS. |
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| 17:08.09 | dan_j | [TK]D-Fender: ok, i'll enable debug logging tonight and see what the results are. |
| 17:08.28 | dan_j | The only thing ive found is that I run ntpdate from crontab at midnight. |
| 17:10.24 | [TK]D-Fender | Perhaps a clock-skew (VM by any chance?) could be completely nuking your "last checked in" timeouts |
| 17:10.47 | Katty | WHOEVER MOVED CALL PARKING TO res_parking.conf......... |
| 17:10.53 | Katty | thank you for adding the note into features.conf |
| 17:11.07 | Katty | that was super sweet of you. |
| 17:11.41 | Kobaz | [TK]D-Fender: the read buffer is definitely cleared when i kill the socket |
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| 17:17.04 | Technodrome | Good day |
| 17:18.03 | [TK]D-Fender | Kobaz: And 1st use <- |
| 17:23.05 | Katty | is there some fancy call parking thing like meetme list, but not |
| 17:26.58 | [TK]D-Fender | You could parse out "dialplan show MEETMECONTEXT" |
| 17:27.10 | [TK]D-Fender | Since that's generated dynamically |
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| 17:30.49 | mjordan | Katty: np :-) So you're playing around with 12? |
| 17:32.21 | mjordan | Katty: there's Parkinglots, which gives you a list of parking lots (https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerAction_Parkinglots) - and ParkedCalls, which gives you information about the calls (https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerAction_ParkedCalls) |
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| 18:02.14 | Hobbes` | is it possible to disable/enable xmpp using the console? (the google talk outage is flooding my screen.) |
| 18:02.46 | WIMPy | module unload it |
| 18:03.02 | Penguin | Google talk outage? I thought I was the only one having an issue with it. |
| 18:03.54 | Katty | mjordan: <3 |
| 18:04.19 | Katty | i heard google was going to stop supporting xmpp, at some point |
| 18:04.27 | Katty | i think drmessano told me that. |
| 18:04.59 | Hobbes` | WIMPy thanks |
| 18:07.26 | Hobbes` | Katty afaik they've not issued any timeframe for that. They've stopped using it internally, we are using some kind xmpp gateway/translation service. |
| 18:07.50 | Katty | ah. |
| 18:08.10 | file | Google Voice will terminate support for XMPP in May 2014, which will stop service for 3rd party apps and devices that use the XMPP signalling protocol including Talkatone, GrooveIP and Obihai. |
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| 18:21.34 | mjordan | file: and Asterisk. |
| 18:29.37 | [TK]D-Fender | file: Oh, that's a global memorandum? |
| 18:29.47 | [TK]D-Fender | file: I don't recall seeing a hard cuto-off date before |
| 18:30.10 | file | I can't find where Google said it, but that's the date everyone knows (Obihai, me, etc) |
| 18:30.44 | [TK]D-Fender | file: Guess the dat is out there then... I remember that they were phasing it out, not it looks like a big-axe date |
| 18:31.15 | file | the time counts down |
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| 18:40.47 | Penguin | Will the SMS service continue after the XMPP support is removed? |
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| 18:56.24 | hexanol | while doing some tests with the native bridging functionality (with chan_sip) in asterisk 12.1.1, after doing an attended transfer, it looks like it's not possible to stay in native bridge |
| 18:56.42 | hexanol | is it a known problem ? could it be a problem with my configuration ? |
| 18:57.12 | hexanol | before the transfer, both calls are bridged using the native bridge |
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| 19:18.28 | [TK]D-Fender | hexanol: You don't seem to be looking at any of the details of where your call was originally VS after... |
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| 19:34.47 | hexanol | hum, not sure I understand your point |
| 19:34.53 | hexanol | this is something that was working fine with asterisk 11 |
| 19:35.13 | hexanol | for a reason I don't know yet, this doesn't seem to work with asterisk 12 |
| 19:35.26 | mjordan | hexanol: there's a lot of 'it depends' in there |
| 19:35.41 | hexanol | a do a first call from a SIP phone to another phone, and when I answer, calls are bridged using the native bridge |
| 19:35.49 | mjordan | hexanol: first, the CLI should tell you how you were bridged. But more importantly, it all depends on how the Dials were performed |
| 19:36.25 | mjordan | If any channel - at any point in time - ends up with some feature attached to it (feature being only loosely related to features.conf) - that requires DTMF or audio to go through the core, it won't natively bridge |
| 19:36.43 | hexanol | well the Dial are always the same, i.e. with the hHtT options |
| 19:36.51 | mjordan | k, you just answered my question |
| 19:36.53 | mjordan | :-D |
| 19:36.58 | mjordan | you can't native bridge if you are using features |
| 19:37.01 | hexanol | but what's weird is that |
| 19:37.03 | mjordan | and you weren't doing that in Asterisk 11 either |
| 19:37.10 | mjordan | trust me. It may even be lying to you. |
| 19:37.13 | mjordan | But you aren't natively bridged. |
| 19:37.21 | hexanol | well that's a bit surprising |
| 19:37.39 | hexanol | let me show you what I see in the asterisk CLI |
| 19:37.42 | mjordan | hHtT are DTMF features. If the DTMF is inband or rfc2833, then the media *must* be decoded |
| 19:37.54 | hexanol | I'm actually using SIP INFO |
| 19:38.03 | mjordan | ah |
| 19:38.15 | mjordan | so there *might* be a chance then :-) |
| 19:38.35 | file | hum... |
| 19:38.37 | hexanol | here's a paste |
| 19:38.38 | hexanol | http://pastebin.com/URyjr8UM |
| 19:38.58 | hexanol | where the hHtT options are used and it says "switching from simple_bridge tech to native_rtp" |
| 19:39.18 | hexanol | and with an "rtp set debug on", I do not see any rtp frame going by |
| 19:39.20 | mjordan | yeah, SIP INFO DTMF would be one of those cases where there's an exception to the rule |
| 19:39.29 | hexanol | and the sound works fine between the phones |
| 19:39.33 | hexanol | ok |
| 19:39.34 | mjordan | (which is applicable everywhere) |
| 19:39.49 | hexanol | I'll try without the hHtT options |
| 19:40.03 | [TK]D-Fender | TT prevents direct bridging |
| 19:40.16 | [TK]D-Fender | Anything where * has to sit in the middle |
| 19:42.40 | hexanol | I actually have the same results when using no options to Dial |
| 19:45.06 | hexanol | this is what I'm getting in the CLI when I complete an attended transfer (using chan_sip) from two calls bridged using the native rtp bridge http://pastebin.com/8eirfSwL |
| 19:46.18 | mjordan | We need to see what happened with SIP/tovawi-0000002f |
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| 19:48.47 | hexanol | ok I'll try to get a more complete trace |
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| 19:56.57 | hexanol | mjordan: well here's one http://pastebin.com/raw.php?i=MG0EHnNj |
| 19:57.12 | hexanol | a bit on the verbose side |
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| 20:00.21 | hexanol | in any way, I was just wondering, because direct media works fine when calling from A to B and also works fine when calling from B to C, but when doing an attended transfert, it doesn't seem to work, which seems a bit unexpected |
| 20:00.33 | hexanol | so I was wondering if someone already met the same problem |
| 20:00.38 | mjordan | it would if it dropped back into native_rtp |
| 20:01.07 | mjordan | hexanol: if you don't mind, please do open a bug |
| 20:01.08 | hexanol | and yes, that's the question, why is it not dropping back into native_rtp ? |
| 20:01.14 | mjordan | dunno! But it should in that case. |
| 20:01.17 | hexanol | ok but this isn't something that you are aware of ? |
| 20:01.21 | mjordan | nope! |
| 20:01.33 | hexanol | alright |
| 20:01.41 | hexanol | well I don't have time right now but I'll do this tomorrow |
| 20:01.44 | hexanol | thanks for your help |
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| 20:02.35 | mjordan | no worries, thanks for catching it. That's something we'll want to get fixed up |
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| 20:25.35 | brittel | Anyone here experienced with sangoma gsm modules? |
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| 20:34.52 | jeffspeff | I'm experiencing the same issue as I did a while back, thought it was a fluke so I didn't write down the fix. Installing asterisk on a vm. asterisk fails to start giving an error about libasteriskssl.so.1 no such file or directory. I remember the problem had something to do with asterisk not pointing the proper directory. it was using /usr/lib instead of /usr/lib64 even though this is a 64 bit system |
| 20:34.52 | jeffspeff | . if i remember correctly, there was a switch put on ./configure or something... any ideas? |
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| 20:48.51 | Katty | who wants to come over for a nerf gun party when it gets warmer. |
| 20:51.48 | drmessano | Sounds good to me |
| 20:51.51 | drmessano | I will bring my speedo |
| 20:52.17 | Qwell | There needs to be a new device, a speedo-o-meter. |
| 20:52.59 | drmessano | My speedo goes from 0 to 60 in, oh nevermind |
| 20:54.17 | Katty | oh, water gun party |
| 20:54.20 | Katty | that might be even more fun |
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| 21:18.44 | ghost75 | https://onedrive.live.com/?cid=b371043b95253312&id=B371043B95253312%21110&sff=1 <- any idea why i have jitter on local channel? |
| 21:20.56 | Qwell | ghost75: You should replace your tires. |
| 21:21.08 | ghost75 | :> |
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| 21:26.32 | ghost75 | Qwell: data is from rtpaudioqos and rtpaudioqosbridged |
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| 22:35.33 | fireman_biff | Hi all, I have DUNDi setup between around 20 PBXs. One of them (asterisk 1.8) is no longer responding to lookups, though it can still perform lookups on other PBXs. We have rebooted the problematic PBX and regenerated new keys for that PBX, but no luck. There are a lot of ENCREJ errors. Any idea where I should look next? |
| 22:36.16 | WIMPy | Have you verified the correct keys are loaded? |
| 22:36.43 | WIMPy | Compare the hashes with 'keys show'. |
| 22:37.04 | fireman_biff | I ran "keys show" and the sums matched, is that good enough? |
| 22:37.10 | fireman_biff | oh ok, then yeah |
| 22:38.38 | WIMPy | Are the requests going to the correct box? |
| 22:39.54 | fireman_biff | they should, cause nothing's been changed afaik, but let me doublecheck |
| 22:40.26 | WIMPy | Do you see that box sending the rejects? |
| 22:42.57 | fireman_biff | hmm... the logs are showing "pbx_dundi.c: Max retries exceeded to host '<IP ADDRESS>:4520' msg 0 on call XXX" for the IP I'm testing from |
| 22:43.04 | fireman_biff | I've never noticed that message before |
| 22:44.45 | fireman_biff | The test box is getting ENCREJ from the problematic PBX, but although the problematic PBX is sending several ENCREJ's message I'm not actually seeing any of them destined to my test box right now... I'll look into that a bit |
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| 23:08.20 | fireman_biff | WIMPy: the requests are definitely going to the correct box. When my test box sends the problematic one a request, the problemmatic one shows the Rx-Frame (type=ENCRYPT) and then a Tx-Frame (type=ENCREJ) |
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| 23:10.17 | fireman_biff | hmm... hold on... |
| 23:12.43 | fireman_biff | got it, somehow the problematic box ended up with the incorrect eid for the test box |
| 23:12.49 | fireman_biff | stupid problem :| |
| 23:12.59 | fireman_biff | thanks for the help WIMPy |
| 23:15.11 | WIMPy | Would be nice if they were names... |
| 23:16.17 | fireman_biff | hmm... I've always used MAC addresses cause I saw somewhere that they were supposed to be... what happens if you give it descriptive names or use an IP address instead? |
| 23:16.36 | WIMPy | doesn't work. |
| 23:16.44 | fireman_biff | shame |
| 23:17.19 | WIMPy | But I'm always annoyed by any log messages, as I can't make any sense of them without looking up the MAC in the configuration first. |
| 23:20.24 | fireman_biff | I wrote a freepbx module so the other guys in the office could manage DUNDi from the web interface, it includes a page that lets you see the status of the peers along with the name of the peer (since the name is defined in the freepbx module), which is how I got around dealing with EIDs all the time. But apparently its not very helpful if you enter the wrong EID |
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