Posted: Fri Apr 04, 2003 3:55 pm Post subject: [Asterisk-Dev] Patch to add Record-Route handling for SIP, a
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Attached is my patch that adds proper Record-Route handling for Asterisk's
SIP channel. Well, "proper" may be too strong, but I did try.
With this patch applied Asterisk now works with Free World Dialup.
Asterisk should now play much better with proxies in general.
I suspect that this patch will also fix some of the problems people have
been reporting on the -users list, because previously * didn't honour
Contact: headers when sending followup requests on a call.
This meant that INFO, BYE etc for open calls may have been sent to the
wrong place (ie a proxy rather than the end-point). So that would account
for calls not clearing properly, and maybe for lost DTMF signalling?
Anyway - would you apply the patch - I can then chase up on -users for
feedback.
Posted: Fri Apr 04, 2003 8:21 pm Post subject: [Asterisk-Dev] Patch to add Record-Route handling for SIP, a
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Here's try 2 at this patch. Jim Dennis pointed out my misuse of strncpy.
Steve
On Fri, 4 Apr 2003, Stephen Davies wrote:
Quote:
Hi Mark, others,
Attached is my patch that adds proper Record-Route handling for Asterisk's
SIP channel. Well, "proper" may be too strong, but I did try.
With this patch applied Asterisk now works with Free World Dialup.
Asterisk should now play much better with proxies in general.
I suspect that this patch will also fix some of the problems people have
been reporting on the -users list, because previously * didn't honour
Contact: headers when sending followup requests on a call.
This meant that INFO, BYE etc for open calls may have been sent to the
wrong place (ie a proxy rather than the end-point). So that would account
for calls not clearing properly, and maybe for lost DTMF signalling?
Anyway - would you apply the patch - I can then chase up on -users for
feedback.
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