After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP
calls are being destroyed after 1 minute and 20 seconds (80 seconds). Asterisk is sending a BYE message - I have no idea why. http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug. can anyone suggest how i can further deal with this?
I have had (what I consider) an odd request. The installation I'm working on
now is an office on a multi-floor building. They 're looking for some kind
of solution with the phone system to provide door control. We are a
non-profit so of course I'm looking for something VERY inexpensive. I'm sure /someone/ has done something like this. I'd appreciate any ideas. Cassius Smith
I originally thought I should post to the biz list but I am not looking
for offers of DID's, I am looking for actual user
experiences/information on obtaining a DID for an Office I am working
with in Hyderabad, India.
Can anyone offer recommendations based on personal experience of where I
might be able to obtain said DID? This will be 90% inbound traffic and
only within India.
If anyone feels strongly that I should have indeed posted this to the
biz list, please accept my apologies but, I felt…
Has anyone had a problem setting up two registrations (on the same Asterisk server) on one Polycom phone?
When the user tries to make a call on the 2nd line, it works fine. But when they try the first line, the CLI says:- Using INVITE request as basis request - href="mailto:firstname.lastname@example.org">email@example.com
Found peer client _202' <--- Which is incorrect, it should be client_201.
[2010-11-15 10:46:29] WARNING: chan_sip.c:9063 check_auth: username mismatch, have < client _202>, digest has < client _201>
[2010-11-15 10:46:29] NOTICE: chan_sip.c:15079 handle_request_invite: Failed to authenticate user " client…
I'm trying to create a link between two PBXs. One is Asterisk 1.4.15,
the other is an unknown 3rd party PBX. In my internal testing, beween two A*k servers, I found that if I
created two sip accounts from the same IP, one as peer and one as user
(intending to give an -IN and -OUT setup), then inbound calls always
seemed to route via the -OUT account and failed. My fix was to use
type=friend, which seemed to make sense and be ok. Now with the 3rd party PBX, if I set type=friend,…