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

[Ivan Stepaniuk]

listmail at websage.ca wrote:
> After running for months without issue I've got a situation where
> incoming SIP calls to my asterisk server are failing while outbound
> calls appear to be working as expected.
>
> The server is a gateway between my home LAN and a broadband cable
> connection with a dynamic IP. The gateway runs FreeBSD 7.1 and Asterisk
> 1.6.0.15 (built from ports) and registers to my ISTP no problem.
> Outgoing calls can be made successfully and no error messages or
> warnings are reported by Asterisk.
>
> However, incoming calls appear to enter my dialplan as desired and go so
> far as to start ringing my SIP phone (Grandstream GXP-2000) but drop
> after two rings. The caller gets a busy tone and that's it. If I answer
> the call before the two rings I just get a moment of dead air and it
> drops in the same way.
>
> In the asterisk console (and log file) I see these messages at the fail
> point:
>
> [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)
>
> Okay, so I verified that my firewall is properly accepting traffic on
> the range of SIP and RTP ports as specified by my ITSP.
>
> After sending them a sip debug trace my provider said this:
>
> "It appears that your machine is not receiving replies when it tries to
> acknowledge the incoming call back to our server.  This could be a
> firewall issue or potentially something else that changed without your
> knowledge."
>
> Furthermore, they suggested I might try registering and connecting
> directly to their Asterisk using only the Grandstream phone. I tried
> this and...surprise! Both inbound and outbound calls work fine but
> leave me without voicemail or any other services my PBX would be
> providing.
>
> Right, so now I'm thinking there must be something wrong with my
> Asterisk configuration yet I've made no config changes that would
> account for the sudden (and consistent) incoming call failures.
>
> Here's the relevant portions of my sip.conf if it helps (with
> credentials and ips replaced by Xs):
>
> [general]
> alwaysauthreject=yes
> dtmfmode=auto
> disallow=all
> allow=ulaw
>
> register => XXXX:XXXX at XXX.XXX.XXX.XXX:5060
> register => XXXX:XXXX at XXX.XXX.XXX.XXX:5060
>
> [101]
> type=friend
> context=websage
> host=dynamic
> deny=0.0.0.0/0
> permit=XXX.XXX.XXX.XXX/24
> qualify=yes
> secret=XXXX
> mailbox=101 at default
> accountcode=101
>   

Does your asterisk server have two network interfaces, one with a 
private IP address and another one with the public one?
Did you try adding "canreinvite=no" to your 101 friend sip entry? What 
does the SIP debug say?

-- 
Iván Stepaniuk
Alba Fotónica S. L.
www.albafotonica.com