Note: forwarded message attached.
--- Begin Message ---SIP requires RTP connections in addition to the signaling connection which normally happens on UDP 5060. The RTP connections vary in port usage (the range is configurable through rtp.conf) and are nearly impossible to get going without some "man in the middle" help when you have two Asterisk servers that are both behind NAT firewalls.If that's the case, you're much better off here with IAX where your signaling and media stream can be consolidated into a single stream.
On 10/17/05, Michael Furdyk <[EMAIL PROTECTED]> wrote: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
From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] 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=192.168.1.24;rport=5060
From: Michael Furdyk < sip:[EMAIL PROTECTED]>;tag=411158625
To: < sip:[EMAIL PROTECTED]>;tag=as6606adb1
Call-ID: [EMAIL PROTECTED]
CSeq: 30931 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: < sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 239v=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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5060;rport;branch=z9hG4bKB644BADE71EF4422878597A96BE8D613
From: Michael Furdyk <sip:[EMAIL PROTECTED]>;tag=411158625
To: < sip:[EMAIL PROTECTED]>;tag=as6606adb1
Contact: <sip:[EMAIL PROTECTED]:5060>
Call-ID: [EMAIL PROTECTED]
CSeq: 30931 ACK
Max-Forwards: 70
Content-Length: 0Here 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=yesI have tried nat=no and nat=yes
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