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

listmail at websage.ca listmail at websage.ca
Tue Oct 13 19:25:41 CDT 2009


On Sat, 10 Oct 2009 18:02:04 -0700
listmail at websage.ca wrote:

> 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...
> 


[Bumping this in the hope that someone might have some new insight or
suggestions since I posted this on a holiday weekend (in my part of
the world anyway)...]



-- 
   
Greg Maruszeczka




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