Archives : October-2014
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all, Im setting up a couple of test boxes and Im running into a problem. What I need help with is determining whether Im going something wrong or if I need to post a bug report. I have two asterisk 13.0-beta 3 machines set up with extensions connec..
Running 11.13.1 on Fedora.This is a new install, but a copy of a previous – working -install.module load chan_sip Unable to load module chan_sip Command module load chan_sip failed. SIP channel loading…[Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31..
with the below defined in logger.conf on 11.6 cert 6I am not getting any log message other than notice and warning in any fileswhen doing module reload logger – queue log is the only one that says it restarts*CLI> module reload logger== Parsing /etc/asterisk/logger.co..
I use a simple scheme:SIP video phone A (h264/Asterisk 1.8.11)SIP video phone B (h264/Asterisk 11.7.0)When calls from A to B and vice versa drop on pickup.On B side:[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker ..
What should the PJSIP configuration be if your external IP address is dynamic, as is common with most home networks, and probably a lot of small business networks as well?The external_media_address and external_signaling_address transport settings ..
I am struggling to have a SPA504G to auto answer (for intercom/paging). Ihave tried the following SIP headers (not all together), but without luck:SIPAddHeader(Call-Info:\;answer-after=0);SIPAddHeader(Call-Info: answer-after=0);SIPAddHeader(Alert-In..
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there, I have an issue , I want to make a video call to a streaming source using Asterisk . Someone can help me in this issue please?Thanks i..
The suggestion that Asterisk should consider deprecating A..