Asterisk 11 And Pulseaudio Setup As Local User


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

/etc/dahdi and all that stuff.
The console channel gets errors on opening. Thanks,

Asterisk Users 2 days ago 6 Answer

For A Failed Retransmission - What Were The IP Addresses?


Hi Guys

Given these occassional errors on my Asterisk CLI:

[Jul 2 10:23:36] WARNING[2060]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 17bb3a993ad10f8818970ae952b81e73@ for seqno 102 (Critical Request) -- See Packet timed out after 32000ms with no response [Jul 2 10:23:49] WARNING[2060]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 2f6fa425581373c11c2ae58a276751bb@ for seqno 102 (Critical Request) -- See 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? is my Asterisk

Asterisk Users 3 days ago 0 Answer

Dell Portability



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…

Asterisk Users 3 days ago 2 Answer

Same PJSIP Username With Differents Domains



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.

Asterisk Users 9 days ago 2 Answer

Asterisk Dialplan Best Practices Syntax



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.


Asterisk Users 9 days ago 4 Answer