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

[Ivan Stepaniuk]

listmail at websage.ca wrote:
> 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)...]

I was staring at the SIP transcript and I don't see anything wrong, I'm
out of suggestions, except that you could analyze and compare the
packets when your phone is connected directly (if it's physically
possible). I hope someone throws some light over this.

-- 
Iván Stepaniuk
Alba Fotónica S.L.