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 $600 add-on - you know it's time to change) - so
Asterisk will be excellent.
Anyway, I'm writing to see if anybody else has experienced problems with
calls to the iConnectHere gateway dropping out.
I've got the following setup:
Anybody wishing to call via iConnectHere dials 82 followed by the number.
Calls seem to be connecting fine - voice passes for 5-10 seconds but then
the call disconnects.
The following shows up in the console:
-- Executing Dial("SIP/6012-fbc2",
"SIP/xxxxxxxxx@iconnecthere") in new
stack
-- Called xxxxxxxxx@iconnecthere
-- SIP/iconnecthere-a5fb is making progress passing it to SIP/6012-fbc2
-- SIP/iconnecthere-a5fb answered SIP/6012-fbc2
-- Attempting native bridge of SIP/6012-fbc2 and SIP/iconnecthere-a5fb
WARNING[9226]: File chan_sip.c, Line 409 (retrans_pkt): Maximum retries
exceeded on call 564c3b1f518e29b81f48ff716e141443@203.217.xx.xx for seqno
104 (Request)
== Spawn extension (default, 82xxxxxxxxx, 2) exited non-zero on
'SIP/6012-fbc2'
Sometimes I'll also get a whole heap of:
WARNING[150546]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect
process 2 frames
WARNING[150546]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect
process 2 frames
Appropriate sections of the configuration files are below:
sip.conf
[iconnecthere]
type=friend
username=xxxxxxxx
secret=xxxx
host=sipauth.deltathree.com
extensions.conf
exten => _82.,2,Dial,SIP/${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
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@ewing.dropbear.id.au>
To: <asterisk-users@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 $600 add-on - you know it's time to change) - so
> Asterisk will be excellent.
>
> Anyway, I'm writing to see if anybody else has experienced problems
with
> calls to the iConnectHere gateway dropping out.
>
> I've got the following setup:
>
> Anybody wishing to call via iConnectHere dials 82 followed by the number.
>
> Calls seem to be connecting fine - voice passes for 5-10 seconds but then
> the call disconnects.
>
> The following shows up in the console:
> -- Executing Dial("SIP/6012-fbc2",
"SIP/xxxxxxxxx@iconnecthere") in
new> stack
> -- Called xxxxxxxxx@iconnecthere
> -- SIP/iconnecthere-a5fb is making progress passing it to
SIP/6012-fbc2> -- SIP/iconnecthere-a5fb answered SIP/6012-fbc2
> -- Attempting native bridge of SIP/6012-fbc2 and SIP/iconnecthere-a5fb
> WARNING[9226]: File chan_sip.c, Line 409 (retrans_pkt): Maximum retries
> exceeded on call 564c3b1f518e29b81f48ff716e141443@203.217.xx.xx for seqno
> 104 (Request)
> == Spawn extension (default, 82xxxxxxxxx, 2) exited non-zero on
> 'SIP/6012-fbc2'
>
> Sometimes I'll also get a whole heap of:
> WARNING[150546]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect
> process 2 frames
> WARNING[150546]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect
> process 2 frames
>
>
> Appropriate sections of the configuration files are below:
>
> sip.conf
>
> [iconnecthere]
> type=friend
> username=xxxxxxxx
> secret=xxxx
> host=sipauth.deltathree.com
>
> extensions.conf
>
> exten => _82.,2,Dial,SIP/${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@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
Hi,> 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.I had the same problem calling from my cisco ATA (sip) to iconnect. As far as I understand it it hat something to do with astererisk trying to native bridge the call. I added the following line to the cisco client configuration of my sip.conf: [cisco-ATA] canreinvite=no Now it works. Hopefully this info helps you in some way. Good luck! Oliver
> Hi, > > Anybody wishing to call via iConnectHere dials 82 followed by thenumber.> > > > Calls seem to be connecting fine - voice passes for 5-10 seconds butthen> > the call disconnects. > > I had the same problem calling from my cisco ATA (sip) to iconnect. As > far as I understand it it hat something to do with astererisk trying to > native bridge the call. I added the following line to the cisco client > configuration of my sip.conf: > > [cisco-ATA] > canreinvite=noThat done the trick! Thanks Oliver! --Shaun> Now it works. Hopefully this info helps you in some way. > > Good luck! > > Oliver