[Asterisk-Users] SIP to SIP sadness

[Mojo with Horan & Company, LLC]

just 5060/udp and the rtp ports (beginning at 10000 and incrementing) 
should be free and clear.  I open 10000-10500 for my inter-office phones 
(and update rtp.conf to match)

Michael Furdyk 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:* 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=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 
> <mailto: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 
> <mailto: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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-- 
Mojo <mojo at horanappraisals.com>
Office Manger, Horan & Company, LLC
(907) 747-6666 x112