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| 04:52.07 | karelk | res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Can't open lib '/usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so' : file not found |
| 04:52.18 | karelk | which package do I need to install ? |
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| 06:25.47 | *** topic/#asterisk is Join us at AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.29.0 (2019/10/08) 16.6.0 (2019/10/08); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 07:25.45 | snuff-work | myodbc.. maybe mysql? odbc driver |
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| 08:12.46 | jamesc | Anyone know why PJSIP_HEADER( update/add cannot find modify the Contact: <sip:asterisk@x.x.x.x:5060> before a Dial? |
| 08:23.53 | jamesc | how do I set contact_user per call? |
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| 11:16.12 | asttdk | Hello everyone, newbie question. Can I limit CPS through the asterisk configuration? If so, is there any manual/wiki that could point me to the right direction? Thanks a bunch :) |
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| 13:52.42 | sibiria | is there a way to make chan_sip still put the IP address in the From: header when using an anonymous caller ID, instead of "@anonymous.invalid"? |
| 13:53.15 | sibiria | it's correct in the Contact header (sip:anonymous@1.2.3.4:5060> |
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| 14:10.25 | bf_ | Is there a difference between use of = and => in the dial plan? |
| 14:10.34 | bf_ | I tried to google it but I can't find an explanation |
| 14:11.10 | bf_ | My intuition is that both are the same, and => is the older, more verbose way to write things. Similar to n which can be used in modern asterisk versions |
| 14:11.32 | bf_ | But holy moly is the documentation for asterisk fragmented all over the place %) |
| 14:12.17 | file | they are the same. |
| 14:13.27 | bf_ | Thank you file |
| 14:13.59 | bf_ | Is there a recommended OSS module for good-enough quality TTS? |
| 14:14.14 | bf_ | I found https://www.voip-info.org/text-to-speech-tts/ but most vendors seem to be commercial |
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| 14:15.22 | sibiria | festival is probably the most common one |
| 14:16.32 | Samot | Well he said good quality. |
| 14:16.37 | sibiria | :D |
| 14:16.41 | sibiria | yeh, can't say that about festival... |
| 14:16.43 | Samot | For me, Festival doesn't fall in that category. |
| 14:17.08 | sibiria | the speech synth from 1985 in AmigaOS competes very well with festival... |
| 14:18.00 | bf_ | Hm %) |
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| 14:44.13 | sibiria | setting "fromdomain" in sip.conf appears to get rid of the @anonymous.invalid problem in the To header on outbound call invitations, but i wonder if it's the right way to solve it |
| 14:44.35 | sibiria | in the From header* |
| 14:59.51 | Samot | What are you trying to solve exactly? |
| 15:02.54 | sibiria | legacy code base does anonymous outbound calls by setting the caller ID to "" <> and including the "Privacy: id" header. with chan_sip this turns the From header's number into <sip:anonymous@anonymous.invalid> instead of <sip:anonymous@1.2.3.4>, which confuses one particular trunk provider |
| 15:04.01 | Samot | Do they accept PAI or RPID? |
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| 15:07.20 | sibiria | not sure, haven't tested yet |
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| 15:13.32 | sibiria | the main kludge with their trunk appears to be that it absolutely needs the correct IP address in the From header. they can't seem to settle for the Contact header (which chan_sip presents correctly) |
| 15:25.22 | Samot | Because you generally don't auth via the contact header. |
| 15:25.57 | Samot | I require CallerID to be in PAI/RPID format while the from user/domain must match their account details. |
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| 18:41.34 | jjrh | Is there a page I might be missing that better describes configuration and usage of the asterisk realtime database? https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration is the best I have found. |
| 18:45.09 | file | using it with what? |
| 18:49.23 | jjrh | Well initial use is just storing sip accounts with odbc, but I would also be interested in doing things like having dialplan configuration happening there for some things. |
| 18:56.35 | jjrh | use case of dialplan would be stuff like letting a end user add a second number to ring,changing voicemail, forwarding, etc. Also for use cases where you have a failover/redundant server without copying or syncing configuration. |
| 18:59.08 | jjrh | Mostly trying to find a doc (if one exists) that goes into a bit more detail. |
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| 19:12.31 | igcewieling | jjrh: you don't want to put dialplan into a database. |
| 19:13.07 | igcewieling | I'd store the settings in a database, then handle everything in the dialplan, get the data from the database with the dialplan ODBC functions. |
| 19:13.22 | igcewieling | This is how FreePBX and any other sane Asterisk GUI does these sorts of things. |
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| 21:13.01 | jjrh | igcewieling: what's the downside of doing it in the database? I see https://www.voip-info.org/asterisk-realtime-extensions/ and it looks fairly sane to me? |
| 21:14.03 | jjrh | Only thing I see is pattern matching can be a problem. |
| 21:14.19 | file | tightly coupling realtime applications, such as a PBX, to a database has repercussions |
| 21:14.19 | igcewieling | Yup, sure looks easy. It isn't. |
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| 21:15.11 | igcewieling | FreePBX uses dialparties.agi to do the database lookups and dialplan do processes the results. |
| 21:15.13 | file | you end up introducing points at which an external thing (the database) can block large parts of the system |
| 21:15.43 | jjrh | it will block /all/ of asterisk's process? |
| 21:16.03 | file | it can result in blocking key operations |
| 21:16.11 | file | with chan_sip it can block the handling of UDP traffic |
| 21:16.18 | jjrh | oh geez |
| 21:16.23 | file | with the dialplan it can result in channels getting blocked as they progress through the dialplan |
| 21:17.13 | igcewieling | doesn't matter if the rest of Asterisk is working fine, if the dialplan isn't working, nothing else will either. |
| 21:17.19 | file | database usage with realtime applications is something you have to think about intelligently |
| 21:17.30 | jjrh | other channels unrelated to it? Aka 1111 blocks and my channel to totally unrelated 2222 is blocked? |
| 21:17.50 | file | depends on what is going on at the time |
| 21:18.11 | file | if both are executing dialplan and the database is blocked for a period of time, then both will be impacted |
| 21:18.31 | file | if it's SIP traffic in chan_sip then all traffic is handled by 1 UDP thread, so all traffic would be impacted |
| 21:18.50 | jjrh | ahh okay. |
| 21:19.32 | file | you can certainly do these things, but you have to be aware and think about consequences |
| 21:19.38 | file | something like the web world it matters much less |
| 21:19.47 | jjrh | didn't know that about chan_sip - explains why recently my agi script that had a bug in it caused everything to break. |
| 21:20.38 | jjrh | ended up putting in a timeout inside that exits if for any reason it gets stuck. |
| 21:26.18 | jjrh | On a mostly unrelated note - does anyone use the LUA or AEL dialplan languages? |
| 21:26.30 | seanbright | i use AEL |
| 21:26.48 | seanbright | although slowly migrating back to normal dialplan |
| 21:26.55 | jjrh | Why? (the migration) |
| 21:26.58 | file | I think I've seen... 3 people... use LUA... over the past few years |
| 21:27.10 | seanbright | well. |
| 21:28.19 | seanbright | dialplan is more concise. all ael is doing is generating dialplan, and it's not always obvious why it does the things that it does. |
| 21:30.26 | seanbright | we put as much logic as reasonable outside of asterisk anyway, so not using ael means one less dependency |
| 21:31.18 | jjrh | Yeah fact no one knows LUA or AEL is a pretty big reason for me not to be different and allow others to maintain stuff. |
| 21:35.24 | MLC | Is anyone using any tools to measure call quality (rather than relying on subjective user comments)? |
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| 21:43.52 | igcewieling | MLC recent Asterisk versions (Asterisk 16+?) have some extra channel variables which indicate ringing time. Asterisk can also log RTCP info. I think Homer SIP Capture has some call metrics, but I don't recall for sure. |
| 21:44.11 | jjrh | sip3.io maybe? |
| 21:44.15 | MLC | how to enable RTCP logging? |
| 21:44.43 | igcewieling | I personally check our Asterisk boxes with "pjsip show channelstats" and investigate endpoints with packet loss or high jitter, but that is not automated. |
| 21:45.06 | jjrh | My employer uses some slick expensive call monitoring solution that sits as a network tap |
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| 21:45.29 | igcewieling | MLC "core show function CHANNEL" and note the qos related items |
| 21:45.54 | igcewieling | I generate a CEL User Event with the info I want. |
| 21:47.51 | MLC | cool |
| 21:48.16 | jjrh | https://github.com/sip3io/tapir never tried this myself but it looks slick. |
| 21:48.22 | igcewieling | $(CHANNEL(rtcp,all)} would be a good start 8-| |
| 21:49.05 | seanbright | > This repository is no longer supported. Our team will push brand new version of VoIP Monitoring and Troubleshooting platform under the same organization but with another name (SIP3) in the nearest time. |
| 21:49.59 | jjrh | Yeah - they are under sip3.io now I guess. |
| 21:56.45 | jjrh | I have voipmonitor.org written down never tried it either but is partially opensource I guess their sniffer is open, but the webportal isn't. |
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