[asterisk-users] outgoing sip calls work; incoming calls fail

[listmail at websage.ca]

On Sun, 11 Oct 2009 02:11:47 +0200
Ivan Stepaniuk <ivan at albafotonica.com> wrote:

> listmail at websage.ca wrote:
> >
> > On the LAN side I can see the INVITE and OKAY messages which end
> > with a CANCEL, apparently initiated by the Asterisk gateway.
> >
> > On the WAN side I can see that my Asterisk gateway is repeatedly
> > sending OKAY messages in response to the INVITE from my ITSP. I
> > assume the trouble is that these messages are either not getting
> > back to my provider or something is blocking the confirmation from
> > them. This more or less confirms what was seen in the sip debug
> > trace as well.
> Post that SIP message from the CLI (sip debug), try adding 
> "externip=XXX.XXX.XXX.XXX" (your external/public IP address) to your 
> sip.conf global section, asterisk may be including it's private
> address in the OKAY sent to your provider.
> 
>


Here's the last message in sip debug before it gives up:

...

Retransmitting #6 (no NAT) to 66.51.127.173:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
66.51.127.173;branch=z9hG4bKee03.6a63373.0;received=66.51.127.173 Via:
SIP/2.0/UDP 66.51.127.163:5080;rport=5080;branch=z9hG4bKpae7KcSeB2Bte
Record-Route: <sip:66.51.127.173;lr;ftag=9Z5N4eayXp3Qm> From:
"2508864577" <sip:2508864577 at 66.51.127.163>;tag=9Z5N4eayXp3Qm To:
<sip:12504129568 at 66.51.127.173>;tag=as32af6364 Call-ID:
b734bd26-2fae-122d-91a5-653b331e358a CSeq: 121440337 INVITE
User-Agent: Asterisk PBX 1.6.0.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: -1;refresher=uas
Contact: <sip:12504129568 at 96.50.76.138>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 992672626 992672626 IN IP4 96.50.76.138
s=Asterisk PBX 1.6.0.15
c=IN IP4 96.50.76.138
t=0 0
m=audio 15550 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Oct  9 12:42:47] WARNING[1056]: chan_sip.c:2917 retrans_pkt: Maximum
retries exceeded on transmission b734bd26-2fae-122d-91a5-653b331e358a
for seqno 121440337 (Critical Response) -- See doc/sip-retransmit.txt.
[Oct  9 12:42:47] WARNING[1056]: chan_sip.c:2944 retrans_pkt: Hanging
up call b734bd26-2fae-122d-91a5-653b331e358a - no reply to our critical
packet (see doc/sip-retransmit.txt). Scheduling destruction of SIP
dialog '4e8ef1b977bf0e062212334634080665 at 192.168.11.1' in 6400 ms
(Method: INVITE)

...

66.51.127.173 is my provider's SIP server
66.51.127.163 is my provider's RTP server

I even check DNS to make sure both forward and reverse records jive. 

Externip was a good suggestion, and worth a try, though because I'm
registering with my provider and using dynamic=yes, wouldn't they just
reply to that anyway, especially given that the registration works
fine? 

Anyway, after adding externip=<my-external-ip> to [general] and doing a
sip reload in the console the problem remains...

GM

-- 
   
Greg Maruszeczka