Native Bride Sip

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/ 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: