I have a problem with native bridge sip. Asterisk sends an End Call to my billing server when it happens but the call stay up. Users continue to chat.
I would like to disable it native bridge sip. Any suggestions?
Running Asterisk 1.09
See msg below from asterisk:
– Attempting native bridge of SIP/28566260-cd09 and SIP/VIJAYSIP-4817
== Spawn extension (prepaid_outbound, 15613627889, 1) exited non-zero on ‘Local/15613627889@prepaid_outbound-f029,2’
– Executing SetCDRUserField(“Local/15613627889@prepaid_outbound-f029,2”, “”) in new stack
– Executing Hangup(“Local/15613627889@prepaid_outbound-f029,2”, “”) in new stack
== Spawn extension (prepaid_outbound, h, 2) exited non-zero on ‘Local/15613627889@prepaid_outbound-f029,2’
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Add the line
in the sip.conf file, you can either put it in the general section or in the user section.
I was told notransfer=yes is only for IAX.
Please confirm if it works with SIP.
if you want Asterisk to stay in the media path for a SIP call, then
canreinvite=no is what you need.
All my users in sip.conf contains canreinivite =no
still asterisk is attempting native bridge sip.
I want to disable native bridge sip.
I am sending my PSTN calls to this carrier:
i’ve now re-read your OP. what is it you’re trying to do ? terminate a call on request from your billing server ?
Asterisk is sending and END to the billing server when native bridge sip is attempted. But the call stays active and users are not billed.
I am tring to disable the following:
– Attempting native bridge of SIP/22.214.171.124-006436d0 and SIP/35219204-4d80
I dont want asterisk to attempt native bridge sip.
If you do not want asterisk to perform native bridging, then you have to set canreinvite=yes. Even then, asterisk will use native bridging if:
- If one of the clients is configured with canreinvite=NO, Asterisk will not issue a re-invite at all.
- If the clients use different codecs, Asterisk will not issue a re-invite.
- If the Dial() command contains t, ''T", “h”, “H”, “w”, “W” or “L” (with multiple arguments) Asterisk will not issue a re-invite.
This information was taken from this voip-info page: