Hi list!
Today I replaced my old Asterisk 1.8.30.0 on a OpenWRT switch with Asterisk
13.14.1 running on a Banana PI.
Well, I'm trying since hours to connect my mobile phone to the new Asterisk,
but I can't...
I can register it, and I can see it with "sip show peers".
If I try to call my mobile phone from my desk VoIP-phone I see that:
[Feb 17 19:54:10] NOTICE[15630]: chan_sip.c:24457 handle_response_peerpoke: Peer
'00491777654321' is now Reachable. (16ms / 2000ms)
== Using SIP RTP CoS mark 5
> 0xb44ce478 -- Strict RTP learning after remote address set to:
192.168.200.10:41000
-- Executing [4 at default:1] Dial("SIP/00493511234567-0000001a",
"local/4 at luca_mobile") in new stack
-- Called local/4 at luca_mobile
-- Executing [4 at luca_mobile:1] NoOp("Local/4 at
luca_mobile-0000000f;2", "") in new stack
-- Executing [4 at luca_mobile:2] Verbose("Local/4 at
luca_mobile-0000000f;2", "2,Call for Mobile - [00493511234567]")
in new stack
== Call for Mobile - [00493511234567]
-- Executing [4 at luca_mobile:3] Set("Local/4 at
luca_mobile-0000000f;2", "available=NOT_INUSE") in new stack
-- Executing [4 at luca_mobile:4] GotoIf("Local/4 at
luca_mobile-0000000f;2", "0?unavailable") in new stack
-- Executing [4 at luca_mobile:5] Dial("Local/4 at
luca_mobile-0000000f;2", "SIP/00491777654321,,RxX") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/00491777654321
-- Local/4 at luca_mobile-0000000f;1 is ringing
-- SIP/00491777654321-0000001b is ringing
-- Local/4 at luca_mobile-0000000f;1 is ringing
== Spawn extension (default, 4, 1) exited non-zero on
'SIP/00493511234567-0000001a'
== Spawn extension (luca_mobile, 4, 5) exited non-zero on 'Local/4 at
luca_mobile-0000000f;2'
but my mobile phone does NOT ring...
If I try to call another VoIP-phone in my Asterisk from my desk-phone it
works:
== Using SIP RTP CoS mark 5
> 0xb44ce478 -- Strict RTP learning after remote address set to:
192.168.200.10:41000
-- Executing [2 at default:1] Dial("SIP/00493511234567-00000022",
"local/2 at anika_voip") in new stack
-- Called local/2 at anika_voip
-- Executing [2 at anika_voip:1] Verbose("Local/2 at
anika_voip-00000013;2", "2,Call for Anika - [00493511234567]") in
new stack
== Call for Anika - [00493511234567]
-- Executing [2 at anika_voip:2] Dial("Local/2 at
anika_voip-00000013;2", "SIP/00493511234765,20,RxX") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/00493511234765
-- Local/2 at anika_voip-00000013;1 is ringing
-- SIP/00493511234765-00000023 is ringing
-- Local/2 at anika_voip-00000013;1 is ringing
== Spawn extension (default, 2, 1) exited non-zero on
'SIP/00493511234567-00000022'
== Spawn extension (anika_voip, 2, 2) exited non-zero on 'Local/2 at
anika_voip-00000013;2'
The desk-VoIP-phone is in the network "phone0" (192.168.200.0/24) and
the
mobile phone in the network "intlan0" (192.168.10.0/24). The BananaPI
hat IPs
on bot networks and I configured Asterisk to bind to 0.0.0.0.
And, as I said, the mobile phone CAN register in Asterisk...
Can someone help me to understand WHY using my mobile phone on my new
Asterisk doesn't work?
And, of course, to solve my problem... ;)
Thank you very very much for your help!
Luca Bertoncello
(lucabert at lucabert.de)