[Asterisk-Users] iConnectHere - calls dropping out?

Shaun Ewing shaun at ewing.dropbear.id.au
Fri May 23 11:33:28 MST 2003


Well, I have done some playing.

The calls won't drop out as long as I have the 't' at the end of the dialing
line, eg:
exten => s,1,Dial,SIP/${ARG1}@iconnecthere|60|t

But this isn't very desirable, because if the person I'm calling happens to
press their # key, they get given the call transfer prompt.

Is it possible to disable the actual transfer prompts but still allow that
transfer permission to be there? I'm guessing that the iconnecthere switches
transfer me a few moments into the call causing the disconnect (very
uneducated guess).

Any ideas? I'd be interested to see how other people interface with the
iConnect system.

Regards,
Shaun


----- Original Message -----
From: "Shaun Ewing" <shaun at ewing.dropbear.id.au>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, May 24, 2003 12:21 AM
Subject: [Asterisk-Users] iConnectHere - calls dropping out?


> Hi all,
>
> This is my first post here - I started with Asterisk a few days ago and
have
> "fallen in love" - fantastic product. I've only got softphones connected
at
> the moment - I'll probably order the FXO/FXS cards in about a month (and
> then think about getting some hardware SIP phones). Our current phone
system
> is quite a few years old and isn't growing with us (when a single line
> telephone port is a $600 add-on - you know it's time to change) - so
> Asterisk will be excellent.
>
> Anyway, I'm writing to see if anybody else has experienced problems with
> calls to the iConnectHere gateway dropping out.
>
> I've got the following setup:
>
> Anybody wishing to call via iConnectHere dials 82 followed by the number.
>
> Calls seem to be connecting fine - voice passes for 5-10 seconds but then
> the call disconnects.
>
> The following shows up in the console:
>     -- Executing Dial("SIP/6012-fbc2", "SIP/xxxxxxxxx at iconnecthere") in
new
> stack
>     -- Called xxxxxxxxx at iconnecthere
>     -- SIP/iconnecthere-a5fb is making progress passing it to
SIP/6012-fbc2
>     -- SIP/iconnecthere-a5fb answered SIP/6012-fbc2
>     -- Attempting native bridge of SIP/6012-fbc2 and SIP/iconnecthere-a5fb
> WARNING[9226]: File chan_sip.c, Line 409 (retrans_pkt): Maximum retries
> exceeded on call 564c3b1f518e29b81f48ff716e141443 at 203.217.xx.xx for seqno
> 104 (Request)
>   == Spawn extension (default, 82xxxxxxxxx, 2) exited non-zero on
> 'SIP/6012-fbc2'
>
> Sometimes I'll also get a whole heap of:
> WARNING[150546]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect
> process 2 frames
> WARNING[150546]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect
> process 2 frames
>
>
> Appropriate sections of the configuration files are below:
>
> sip.conf
>
> [iconnecthere]
> type=friend
> username=xxxxxxxx
> secret=xxxx
> host=sipauth.deltathree.com
>
> extensions.conf
>
> exten => _82.,2,Dial,SIP/${EXTEN-2}@iconnecthere
>
>
> The same config works fine on Free World Dialup - using 81 as the prefix,
so
> I'm quite perplexed as to why this happens on iConnectHere.
>
> Any help or insight would be greatly appreciated.
>
> Regards,
> Shaun Ewing
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users




More information about the asterisk-users mailing list