Asterisk 1.8 Local Bridging issues

Asterisk Asterisk Support
1

I seem to be having a very strange problem on 1.8.3.2. Bridging doesn’t seem to be working for a call which comes inbound on our DID numbers and is then diverted to an extension, which is further instructed to dial an outbound number. There is no ringtone, and no audio available.

However, dialing the extension directly works, as does inbound DID to a SIP phone (or IVR).

This setup worked on our previous (1.4) installation.

extensions.conf

[from-reception]
include => ext-internal

[ext-internal]
exten => 107,1,Dial(SIP/sipcall/00353000000000,60)

sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=g729
allow=ulaw
allow=alaw
srvlookup=yes
callerid = Unknown
alwaysauthreject=yes
dtmfmode=rfc2833

register => 41610000007:xxxxxx@myvoipprovider.com/107

[sipcall]
type=peer
defaultuser=41610000000
secret=xxxxxxx
context=from-reception
host= myvoipprovider.com
fromuser=41615111100
qualify=yes
fromdomain=myvoipprovider.com
insecure=port,invite
caninvite=no
canreinvite=no
nat=no

that’s literally it. I even stripped this right down, so these are the only config files running. I get the following result:

== Using SIP RTP CoS mark 5
– Executing [107@from-reception:1] Dial(“SIP/sipcall-00000023”, “SIP/sipcall/00353000000000,60”) in new stack
== Using SIP RTP CoS mark 5
– Called sipcall/00353000000000
– SIP/sipcall-00000024 is making progress passing it to SIP/sipcall-00000023
– SIP/sipcall-00000024 answered SIP/sipcall-00000023
– Locally bridging SIP/sipcall-00000023 and SIP/sipcall-00000024

but no audio. Have also tried using another provider on the outbound, no luck.

this is pretty serious. We divert our incoming landlines to our mobiles using this method, and nothing is working.

note that all of our other features are working with no issues, i.e. SIP to PSTN, DID to SIP, Skype4Asterisk to PSTN, etc, etc, etc.

2

is anyone able to replicate this?

Thanks

3

Replace that caninvite and canreinvite with a directmedia=no

Cheers.

4

no joy - same result.

5

even worse now, it’s working sporadically. mind you, it might have been doing that all along. i’ve had 2 calls connect in about 20 attempts. there is no dialtone on anything. Like i say, setup worked on 1.4 with no issues, and everything else not involving this particular type of bridge works fine.

6

help!

7

same problem…
did you solved ?

PS: didn’t happen in 1.6.0.1

8

I detected the same exact issue, which boils down to: There is no way to force “Local bridging”. Version 1.8 simply goes to Remote Bridging. No matter what you do, the behavior is random. It did not happen in 1.6.2.

Any ideas how to get around this?
All my customers that are sending calls from behind a NAT have one way audio.

9

A little bit necropost, but Answer before Dial helps me.