I am trying to get Asterisk 11 to co-exist with a CentOS 7 box that has pulse audio running as a local user. Has anyone done that? What is the trick? I changed directories
/var/run/asterisk, /var/spool/asterisk, /var/log/asterisk, /usr/lib/asterisk, /etc/dahdi and all that stuff.The console channel gets errors on opening. Thanks,
Given these occassional errors on my Asterisk CLI:
[Jul 2 10:23:36] WARNING: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission firstname.lastname@example.org:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [Jul 2 10:23:49] WARNING: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission email@example.com:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response
E. g. I have the transmission numbers
on which packets timed out.
Is there any way I can determine the IP addresses where those packets referred to by these errors timed out on?
What is Sequence 102?
192.168.11.31 is my Asterisk…
I built an LXC container with an "image" of asterisk 11.18 precompiled and installed. It runs fine on the dev platform, which is a Dell R320 running Ubuntu 14.04LTS. I shutdown the container, tarred it up, and untarred on a Dell PE1850, also running Ubuntu 14.04LTS. The container itself is Ubuntu 14.04LTS. Both platforms as far as I know are amd64.
The container boots fine on the 1850, but trying to run asterisk segfaults. The source tree was still in the container, so I just did a make clean; make; make install. It now runs fine.
Is there some compile flag I…
In PJSIP configuration, I thought that "from_domain" parameter in a endpoint permits to have two SIP peers with the same usernames with a different domain.
I've tested at the transport level, I see no changes.
I've also tested with realm parameter in auth configuration, it seems to change only the digest auth value during registration.
I'm pretty new with PJSIP, however, I have the impression that you can't do that with Asterisk, each username in PJSIP must be unique accross an Asterisk instance. Is it correct ?
Thanks for your answers.
I've two yocto questions about the syntax of dialplan:
1. What's the "official" notation of each line: "=>" or "=" ? In the wiki of Asterisk, I see very often "=>", however, what's the reason for both syntaxes authorized ? Historical ?
2. To write info in logs/console, you have two commands: NoOp and Verbose. Verbose seems to be better, because you can define a log level. Are you agree or another command fits better for logs ?
Thanks for your responses.