similar to: Participant unable to hear other participants in ConfBridge

Thanks for sharing the link. Actually I'm looking for a different approach without installing/using third party i.e. a user sends an email to Asterisk (which is also running mail service), as Asterisk receives the mail where the mail contains attachment and subject contains destination number, Asterisk will download the file and capture the number and later send fax to destination number just

[root at abc asterisk]# lsof -u 50771 | wc -l 0 BTW, I'm using CentOS 6.5 > > Date: Thu, 13 Oct 2016 10:20:19 -0400 >> From: Dovid Bender <dovid at telecurve.com> >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <asterisk-users at lists.digium.com> >> Subject: Re: [asterisk-users] Openfile Issue >> Message-ID:

I did radius client status testing with radius server, able to access the radius server. However, still getting radius CDR issue after setting debug level 8 even granting 666 access to radiusclient-ng config files. message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR not recorded! Please advise if I missed out anything. Date: Mon, 26 Sep 2016 12:09:34 +0200 > From:

Is there a way to make ConfBridge hang up on the final participant in a conference (obviously after some sort of initial grace period)? Background - I have just moved all of the phones in my house to separate extensions. As a replacement for the POTS-style shared line, I have implemented a "barge in" feature; any internal extension is able to join the call of any other internal

Hi Andrew and Willy, Thanks for sharing the info. As for enabling radius server debugging 'radiusd -X', made some test calls don't see the radiusclient sending data to radius server. However, using radtest or radiusclient testing, able to send data to radius server (after enabling debug). For further testing, on my other server using OpenSIPs, setup the radiusclient and data was

Hi, I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see any issues until today. The setup I configured for inbound fax is quite simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38 protocol and later Asterisk stores/forwards the fax to specific end user. The configuration I made in sip.conf for enabling T38 is listed below; t38pt_udptl =

Hi, Currently I'm experiencing crashes on Asterisk more recently, see messages below (crashed reason: segfault signal 6). abrt-hook-ccpp[19864]: Process 7082 (asterisk) of user 0 killed by SIGABRT - dumping core asterisk: ERROR[15373][C-0004e304]: astobj2.c:131 in INTERNAL_OBJ: FRACK!, Failed assertion bad magic number 0x0 for object 0x7fbd2c 00d170 (0) After running the backtrace for the

Hi all, I'm getting one way audio when calling over the SIP trunk i.e. end device B (remote end of SIP trunk) can hear device A (softphone registered with Asterisk) but device A can't hear device B. Even though I configured same NAT configurations on other servers and they are working good. The NAT configuration is listed below; localnet=130.0.0.0/130.0.0.0 externhost=12.131.12.13

Is there a way of temporarily suspending participants in a conference? Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the confbridge session that D and E can't hear, is there a way to suspend D and E for a while (whilst they are played music or whatever) and later join them back in? Failing that, I was considering kicking them and using an AGI script to rejoin

Hi All, I am working on Asterisk 12.6.0 with ConfBridge module, when there are multiple user like admin and normal participant running with conference. When I try to kicked 2 user (Normal User), it play file "conf-kicked" and again join conference My scenario in confbridge like. 1] Admin User (e.g. SIP/8484-00000000) 2] Normal User (e.g. SIP/8484-00000001) 3] Admin User (e.g.

Hi all, I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently my AGI is working fine in my two servers but not in my other four servers. When I tried execute an AGI (as a user asterisk) in command line it works fine (even I also declare environmental variables in user profile and in my AGI), but when I tried to call my AGI (perl) in dial plan, it don't get

Hi, Last year, I installed Asterisk 10.4.2 and enabled logrotate on daily basis which was working perfect. Now in couple of months back, the logrotate feature is not working at all but simply appending the logs in 'messages' file. Listing down down the configuration for logrotate below; /var/log/asterisk/messages { missingok rotate 5 daily postrotate /usr/sbin/asterisk -rx 'logger

Hi, I'm currently using Asterisk 16.4.0 cert version and working on webrtc. For configuration perspective, I'm pretty much done with it but here the real issue I'm currently facing i.e. when setting parameters max_audio_streams & max_video_streams to any positive greater than 0 integer value in realtime (DB) of any endpoints. After running command "pjsip show endpoint

Hi everyone, I haven't used Asterisk in many years, but in searching for a good podcasting solution that will allow me to record three or four participants to individual tracks (which would allow me to go in and do noise removal on each participant individually, giving a higher quality), I came up with the idea to use Asterisk. Now I've installed it and got it all set up and did a test

Hi All, We are currently evaluating the confbridge application while we prepare to upgrade our environment to asterisk v1.6.2.x. We have run in to two issues using it to kick/mute participants in a bridge and would like to ask for the experience of others running the application for any work-arounds. Firstly for kicking participants, would it be possible to use the softhangup application

Trace with 3 clients. We can hear each other but no video. https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz On 11/14/17 5:06 PM, Joshua Colp wrote: > On Tue, Nov 14, 2017, at 07:03 PM, Carlos Chavez wrote: >> On 11/14/17 4:27 PM, Joshua Colp wrote: >> >>> On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: >>>> On 11/14/17 3:55

Hi all, I'm getting compilation error as trying to install latest version of dahdi on CentOS box 5.9 which I now updated from 5.6. I also installed the dependencies but still not getting the clue to get install the driver. Listing down the errors below; CC [M]

Hi, I'm getting error: ' FAX session '9' is complete, result: 'FAILED' (FAX_FAILURE_PARTIAL), error: '3RD_T2_TIMEOUT', pages: 1, resolution: '204x196', transfer rate: '9600', remoteSID: '' ' when I tried send fax more than 2 pages to Asterisk using T.38. First I set speed rate to 14400 which I was getting same error message while

Hi, Few months back I configured Asterisk 11.6.0 for an outbound fax using T.38 protocol as listing down the flow below; Asterisk Fax server -> (IP) -> Cisco VGW ->(IP) -> Carrier The issue I'm currently getting when Asterisk receives warnings as listed below, it is overloading the Cisco VGW, therefore need to restart Asterisk service or sometimes reboot VGW to clear these

/**************************************************************************** * * * Always Be Conferencing (ABC) * * * * Creator: chris @ Penguin PBX Solutions * *