Native bridging

Asterisk Asterisk Support
1

Dear all

I am testing local call with asterisk.
I setup 1111 and 1112

I can hear ringing 1111 to 1112 or 1112 to 1111 but I not hear voice when pick up the handset.

BUT sometime I can hear also voice ,so I check in asterisk when calling

I got two status

– Packet2Packet bridging SIP/1111-0000006c and SIP/1112-0000006d
I can hear all voice when Packet2Packet bridging

– native bridging SIP/1111-0000006c and SIP/1112-0000006d
I can`t hear voice when native bridging

How can I disable the native bridging?
I want only – Packet2Packet bridging
please help me

Thanks

2

Funny, I had the opposite problem.

I think you put

canreinvite=no

in sip.conf for at least one of the phones to disable native bridging.

3

yes

My setting follow

[1111]
type=friend
callerid=1111
username=1111
secret=1111
canreivite=no
host=dynamic
context=TECH
allow=alaw
allow=gsm
nat=yes
dtmfmode=info

[1112]
type=friend
callerid=1112
username=1112
secret=1112
canreivite=no
host=dynamic
context=TECH
allow=alaw
allow=gsm
nat=yes
dtmfmode=info

How can I fix that?

4

You’re missing an “n” from canreinvite

5

hi

now I changed the canreinvite :smile:

but same problem how to disable the native bridging?

6

No idea. That worked for me, although as I said I had the opposite problem and set canreinvite=yes to get it to work.

I also have nat=no (everything is on 1 side of the router) but otherwise similar to yours.

I also specified bindport on mine. 5060 for the phones. 5061 for my SPA3102 to PSTN connection.

7

canreinvite=no WILL disable native bridging.

Native bridging failures are likely the result of bad NAT configurations, or configurations where NAT is incompatible with routing RTP directly between the phones.

8

[quote=“please”]hi

now I changed the canreinvite :smile:

but same problem how to disable the native bridging?[/quote]
Have you done a sip reload since the above modification?