[Asterisk-Users] SIP to SIP sadness

[Anders Svensson]

Look in rtp.conf. You must have the same udp-ports open as the settings in
rtp.conf

 

Anders

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michael Furdyk
Sent: den 17 oktober 2005 21:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SIP to SIP sadness

 

Okay so it seems like it was the firewall, someone just suggested that we
disable it (On Redhat server) and it's working fine... so does anyone know
clearly what ports (other than 5060) SIP uses for these calls?

 

-- Mike

 

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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michael Furdyk
Sent: October 17, 2005 2:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] SIP to SIP sadness

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;received=19
2.168.1.24;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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