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| 01:20.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.2 (2018/02/21), Standard: 15.2.2 (2018/02/21); 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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| 05:18.28 | elcontrastador | i have a main asterisk system using pjsip that uses multicast for paging. A satellite site has a path to it that doesn't support multicast. It has a server there running chan_sip. My plan is to trunk these systems and just sip call the satellite and then multicast from that satellite box. It worked when chan_sip to chan_sip. The main box is now pjsip and i'm new to it. What is the simplest config to get a simple trunk between the main pjsip box |
| 05:18.28 | elcontrastador | <PROTECTED> |
| 06:32.54 | elcontrastador | https://gist.github.com/elcontrastador/7af35458958204a5f476dd4e50fcf322 |
| 06:34.03 | elcontrastador | Any help with this would be GREATLY appreciated...i'm about to blow my cock off over here...missing some basic knowledge of pjsip here... |
| 06:36.28 | elcontrastador | calls only need to go from main server to remote...remote only has one exten |
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| 09:32.18 | elcontrastador | https://gist.github.com/elcontrastador/7af35458958204a5f476dd4e50fcf322 |
| 09:33.18 | elcontrastador | Need to trunk between two Ast systems (pjsip to chan_sip)... Unable to create request with auth. No auth credentials for realm(s) 'asterisk' in challenge. |
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| 10:07.41 | ddalf | Hi folks |
| 10:08.12 | roswell | yello |
| 10:20.15 | ddalf | i've got a question, on which is the best way to do that on Asterisk. I have an incoming call on * which have to be "intercepted" and splitted into two calls. One is the real destination and another one is a fake number for recording purpose. Of course the other channel of the recording platform should be muted |
| 10:20.25 | ddalf | it's basically a lawfull interception system |
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| 10:24.49 | roswell | ddalf, i guess the Monitor app is what you're looking for |
| 10:25.09 | roswell | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Monitor |
| 10:27.34 | ddalf | @roswell thankyou. Correct if i'm wrong byt the Monitor app is to record locally a file. I can't record locally but i have to do something like a 3-way call with one endpoint on mute |
| 10:29.21 | roswell | ddalf, got it. google for ChanSpy then |
| 10:30.14 | ddalf | Chanspy should be fine, mixed with an Originated channel. Right? |
| 10:31.11 | roswell | ddalf, chanspy merely 'mirrors' all traffic, both in- and outbound, to another channel, so yes |
| 10:31.44 | ddalf | Perfect thankyou! |
| 10:31.53 | roswell | np mate |
| 10:32.19 | ddalf | :) |
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| 10:58.21 | ddalf | reload |
| 11:01.29 | elcontrastador | I need page to fall thru the dial plan after call is complete like Dial does: same => n,Page(PJSIP/20@hw-page-gw&PJSIP/22@hw-page-gw&PJSIP/23@hw-page-gw&MulticastRTP/basic/239.255.255.1:30180//${MULTICAST_OPTIONS},,F(hw_page,${EXTEN},stop)) |
| 11:01.39 | elcontrastador | see F() option at end |
| 11:02.32 | elcontrastador | The intent is that I need to run a cleanup script to stop the cisco phones from listening after the page |
| 11:03.11 | elcontrastador | The following works perfectly: same => n,Dial(MulticastRTP/basic/239.255.255.1:30180//${MULTICAST_OPTIONS},,F(hw_page,${EXTEN},stop)) |
| 11:03.54 | elcontrastador | but i need to bridge the 3 paging interfaces in too |
| 11:11.42 | elcontrastador | What are my options? |
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| 11:38.53 | Samot | Well, one is not using Dial() options in Page(). |
| 11:39.04 | Samot | There's no F option for Page() |
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| 13:25.26 | joshua62493 | Hey. I have a serious problem with my Asterisk server running 11.7 on Ubunt |
| 13:25.37 | joshua62493 | Ubuntu 14.04 |
| 13:29.14 | joshua62493 | Is there anyone in this group that can help me or do I need to go to #asteridk-dev? |
| 13:30.46 | file | if you ask a question someone may answer, but Asterisk 11 is no longer supported so people may not want to help with such an old version |
| 13:31.28 | file | and #asterisk-dev isn't a tier 2 support channel, it's for developer discussions - and same applies there, people may not want to help with such an old unsupported version |
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| 13:32.44 | joshua62493 | Basically I think its a bug in the system. I would like to upgrade the software without taking all my customers offline |
| 13:34.09 | joshua62493 | on Ubuntu 14.04 what is the easiest way to upgrade Asterisk? |
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| 14:50.57 | rwb | Anyone using analog FXS (DAHDI) having ring voltage issues? I have a DIGIUM TDM2400P w/24 FXS cards, I'm using 1PWR2401B External Power Adapter. Many of the analog phones are still having a bit of trouble ringing. Should I up the voltage a bit going into the card? |
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| 14:55.33 | rwb | Most of my lines are under 75ft. I'm hoping not to have to boost the voltage on each line. I would rather boost it at the card but only if that's safe. |
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| 16:47.46 | rrittgarn | join #asterisk-dev |
| 16:47.56 | rrittgarn | sorry... mt |
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| 16:50.41 | rrittgarn | while I'm here though... going to guess that without app_voicemail, the m option in app_mixmon won't work? |
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| 19:28.25 | Demon_VoIP | asterisk 15. pjsip. The aim: Deny anonymous calls if the field "from" contains a local domain. Grateful for the help. |
| 19:29.20 | Demon_VoIP | After studying the code, I understood how to make anonymous calls with the local domain and prohibit all others. But the reverse problem can not be solved. |
| 19:31.32 | Demon_VoIP | It would help if the code looks for something like anonymous@other endpoint (for foreign domains). But.. |
| 19:33.20 | Demon_VoIP | anonymous calls from the local domain. What for? :( |
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| 20:32.27 | Samot | What do you mean by a "local domain" |
| 20:32.30 | Samot | That's vague. |
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| 20:42.30 | Demon_VoIP | Samot, domains from domain_alias |
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| 20:45.48 | Samot | Uhm. |
| 20:45.53 | Samot | What is the actual goal here? |
| 20:46.01 | Demon_VoIP | Of course, you can register aliases for foreing domains. It's unclear why. But I did not see in pjsip settings similar to chan_sip to specify the "native" domains of the server. |
| 20:46.42 | Demon_VoIP | goal: forbidden anonymouse calls with "local" domains in "from" header. |
| 20:46.43 | Samot | domain_alias in PJSIP is for AORs |
| 20:47.01 | Samot | So you want to blacklist domains? |
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| 20:47.43 | Demon_VoIP | Samot, blacklist only for anonymous endpoint. |
| 20:47.56 | Samot | OK |
| 20:48.00 | Samot | anonymous how? |
| 20:48.07 | Samot | You mean in the FROM USER? |
| 20:48.24 | Samot | anonymous@1.1.1.1:5060? |
| 20:49.05 | Demon_VoIP | Samot, no. from: anyusername@localdomain.com:5060 |
| 20:49.40 | Demon_VoIP | endpoint_identifier_order : ip,username,anonymous |
| 20:50.36 | Demon_VoIP | if endpoint for username exists than anonymous identifier not used |
| 20:51.22 | Demon_VoIP | the goal: if client set my domain in from header it must be identified. |
| 20:53.24 | Demon_VoIP | now i add one code line in res_pjsip_endpoint_identifier_anonymous.c. That lets me i want. |
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