[Asterisk-Users] SIP -> PSTN calls not connecting properly

[Brian Swan]

I had this same problem.  For me, the Cisco phone wasn't detecting  
that the call was connected.  Turn on VAD, and maybe bump up the rx  
gain on the PSTN.

Hope that helps,
Brian

On Jun 23, 2006, at 8:04 AM, Ronan Mullally wrote:

> Hi,
>
> I've got a problem with my asterisk set up which has been going on  
> for a
> while (months).  I'm currently running 1.2.7.1 on a gentoo box with  
> the
> topology below:
>
>
>                      +-----+
>        PSTN ---------+  *  +------------- Service Provider
>        (wctdm400p)   +-+-+-+     IAX
>                        | |
>                        | |
>                  FXS --+ +-- SIP (cisco 7940)
>
>
> I can make calls from the FXS port to the PSTN or my IAX service  
> provider
> without any problems.
>
> I can make calls from my SIP phone to my IAX service provider, also  
> without
> any problems.
>
> I can receive calls to the FXS port and SIP phone without any  
> problems.
>
> However, when I call from my SIP phone to the PSTN my calls die,  
> repeatedly,
> after 2-3 minutes.  The display on the phone shows 'Session  
> Progress (in
> 183)' for the duration of the call, rather than 'Connected', so it  
> looks
> like the SIP phone is not recognising call connection on the PSTN.
>
> Output from the console is as follows:
>
>     -- Executing Dial("SIP/ronan-5e0e", "Zap/4/xxxxxxxxxxx") in new  
> stack
>     -- Called 4/xxxxxxxxxxx
>     -- Hungup 'Zap/4-1'
>   == Spawn extension (default, xxxxxxxxxxx, 1) exited non-zero on  
> 'SIP/ronan-5e0e'
>
> A packet trace from the * box shows:
>
>  ...
>
>  16.758516  192.168.2.9 -> 192.168.2.30 UDP Source port: 12230   
> Destination port: 31042
>  16.758595 192.168.2.30 -> 192.168.2.9  UDP Source port: 31042   
> Destination port: 12230
>  16.778540  192.168.2.9 -> 192.168.2.30 UDP Source port: 12230   
> Destination port: 31042
>  16.779004 192.168.2.30 -> 192.168.2.9  UDP Source port: 31042   
> Destination port: 12230
>  16.790884 192.168.2.30 -> 192.168.2.9  SIP Request: CANCEL  
> sip:xxxxxxxxxxx at 192.168.2.9;user=phone
>  16.791266  192.168.2.9 -> 192.168.2.30 SIP Status: 487 Request  
> Terminated
>  16.791477  192.168.2.9 -> 192.168.2.30 SIP Status: 200 OK
>
> (192.168.2.9 is the * box, .30 is the phone)
>
> This has been going on for some time, but I've put up with it as the
> majority of my calls are short so it's not a big issue.  As a  
> result I'm
> unsure when the problem started, so I've no idea what change I made  
> to the
> config that caused it.  I'm fairly sure the change is on asterisk  
> as I've
> not touched the config on the 7940 in a long time.
>
> My zaptel.conf, zapata.conf and sip.conf files are below, any  
> suggestions or
> clue transfer would be much appreciated.
>
>
> -Ronan
>
> # zaptel.conf
> loadzone=uk
> defaultzone=uk
> fxsks=4
> fxoks=1-3
>
> # zapata.conf
> [channels]
> group = 0
> context = incoming-POTS
> signalling = fxs_ks
> rxgain=10.0
> txgain=6.0
> echocancel=yes
> echocancelwhenbridged=no
> echotraining=300
> immediate=no
> busydetect=no
> busycount=5
> answeronpolarityswitch=yes
> hanguponpolarityswitch=yes
> callprogress=yes
> callwaiting=yes
> relaxdtmf=no
> progzone=uk
> useincomingcalleridonzaptransfer = yes
> usecallerid=no
> callerid=asreceived
> cidsignalling=v23
> cidstart=polarity
> ukcallerid=yes
> channel => 4
>
> # sip.conf
> [general]
> allow=ulaw
> allow=alaw
> allow=gsm
> allow=g723.1
> context=incoming
> recordhistory=yes
> port=5060
> bindaddr=0.0.0.0
> srvlookup=yes
> tos=lowdelay
> defaultexpirey=120
> nat=no
> localnet=192.168.0.0/255.255.252.0
>
> [ronan]
> regextension=ronan
> regcontext=4L
> mailbox=100 at default
> callerid=Ronan Mullally <100>
> restrictcid=no
> callgroup=1,2
> pickupgroup=1,2
> host=dynamic
> language=en
> type=friend
> context=default
> username=ronan
> secret=xxxxxxxxx
> fromdomain=4L.ie
> canreinvite=no
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> allow=g723.1
> qualify=100
> accountcode=ronan
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