Stewart Nelson wrote:
Well, provider is now sending a different tag, so Asterisk does not
find a match, assumes that this response is for a call it does not know
about, and discards it.
Yes, that is what is happening here.
That makes sense, but since Asterisk always generates a unique Call-ID for
e
> The next step would to be turn pedantic=yes back on, then generate a
> failing call with 'sip debug', 'set verbose 255' and 'set debug 255' in
> place. Capture all the output (there will be a lot) and then post a bug
> in Mantis describing the situation and attaching the output file.
Kevin, than
Stewart Nelson wrote:
How should I proceed? IMO, this provider offers an excellent combination
of price, reliability, quality, and support, and I believe that many in
Asterisk community would want to use them. AFAICT, their SIP/SDP does
not actually violate any RFCs.
The next step would to be tur
>> I'm running [EMAIL PROTECTED] (Asterisk 1.0). Is it possible that this bug
>> has already been fixed in a later version (I can't find anything that
>> seems relevant at bugs.digium.com)?
> This issue (multiple c= lines) has already been fixed in CVS HEAD (if
> 'pedantic' SIP parsing is enable
Stewart Nelson wrote:
I'm running [EMAIL PROTECTED] (Asterisk 1.0). Is it possible that this bug
has already been fixed in a later version (I can't find anything that
seems relevant at bugs.digium.com)?
This issue (multiple c= lines) has already been fixed in CVS HEAD (if
'pedantic' SIP parsing i
have you tried to set the
externip=xxx.xxx.xxx.xxx
localnet=xxx.xxx.xxx.0/255.255.255.0
settings in your sip.conf? it corrected similar problems for me.
michael
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Hi,
I'm testing Asterisk with a new provider. On calls to US
toll-free numbers, there is no audio (calls to normal numbers
are ok).
In response to a valid INVITE from Asterisk, something like
this is received:
SIP/2.0 183 Session Progress
v:SIP/2.0/UDP [my public IP]:5060;branch=z9hG4bK62d91cea
