[Asterisk-Users] SIP to SIP sadness

[Michael Furdyk]

Wow, after getting the O'Reilly book delivered last week along with two
Digium TDM400P's, I'm really getting the hang of this. But the SIP to
SIP issue is still a problem... and it seems silly because everything
else (should have been?) so much harder but is working pretty
flawlessly. Basically I get no audio either way, and it tries to do a
"native bridge" (handoff?)
 
So when I dial another SIP extension, I get:
 
 ---
    -- SIP/324-ab4d answered SIP/322-7e8d
We're at 192.168.1.195 port 16874
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (NAT) to 192.168.1.24:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.24:5060;branch=z9hG4bK8E845F95F34044ACA77C54EF28288C32;receive
d=192.168.1.24;rport=5060
From: Michael Furdyk <sip:322 at 192.168.1.195>;tag=411158625
To: <sip:324 at 192.168.1.195>;tag=as6606adb1
Call-ID: 28E78AC1-5FDE-414E-8059-68B393A24F60 at 192.168.1.24
CSeq: 30931 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:324 at 192.168.1.195>
Content-Type: application/sdp
Content-Length: 239
 
v=0
o=root 3348 3348 IN IP4 192.168.1.195
s=session
c=IN IP4 192.168.1.195
t=0 0
m=audio 16874 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 
---
    -- Attempting native bridge of SIP/322-7e8d and SIP/324-ab4d
 
<-- SIP read from 192.168.1.24:5060: 
ACK sip:324 at 192.168.1.195 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.24:5060;rport;branch=z9hG4bKB644BADE71EF4422878597A96BE8D613
From: Michael Furdyk <sip:322 at 192.168.1.195>;tag=411158625
To: <sip:324 at 192.168.1.195>;tag=as6606adb1
Contact: <sip:322 at 192.168.1.24:5060>
Call-ID: 28E78AC1-5FDE-414E-8059-68B393A24F60 at 192.168.1.24
CSeq: 30931 ACK
Max-Forwards: 70
Content-Length: 0
 
Here is my default in SIP.conf. Each SIP config has canreinvite=no
 
[general]
disallow=all
allow=gsm
allow=ulaw
nat=no
canreinvite=no
externip=(real external IP is here)
localnet=192.168.1.195/255.255.255.0
srvlookup=yes
sipdebug=yes

I have tried nat=no and nat=yes
 
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