Re: [Asterisk-Users] Dialling out with clone X100P board
Hi Ryan:
Christmas intervened!
Got it working. It turned out not to be the ww that did it, but the
toneduration parameter in the zapata.conf file.
Setting
toneduration=200
did the trick.
Thanks for the help, hope this tip helps someone else later on.
Happy New Year!
Roger
[EMAIL PROTECTED] wrote:
I had the same problem at first. Try adding a "w" or two before the
${EXTEN}. That makes it wait a little bit before sending the DTMF numbers.
Here is the dial() I'm using:
Dial(ZAP/1/ww${EXTEN})
Try it out and see. Let us know if it works.
Ryan
Hi all :
I need a little help please.
I have a clone X100P board. I have it all set up and working (just
testing so far) for incoming calls from PSTN.
For outgoing to PSTN I have a strange problem.
I dial out OK, the Zap channel answers the SIP channel ok, (But I do not
see a Call bridged message, and the call has some strange charateristics.
If I call 123, I can connect to and hear the time clock provided by BT
(I'm in the UK) Is this 'audio before answer'?)
If I call any other external number, eg my cellphone, it never rings,
and after 30 secs or so the Zap channel hangs up.
I have been testing this with a very simple Dial(ZAP/1/${EXTEN}) command.
What should I be looking for in my setup?
Many thanks, and happy Christmas to all.
Roger
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tired of doing the hard work you already did.
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Re: [Asterisk-Users] Dialling out with clone X100P board
I had the same problem at first. Try adding a "w" or two before the
${EXTEN}. That makes it wait a little bit before sending the DTMF numbers.
Here is the dial() I'm using:
Dial(ZAP/1/ww${EXTEN})
Try it out and see. Let us know if it works.
Ryan
> Hi all :
>
> I need a little help please.
>
> I have a clone X100P board. I have it all set up and working (just
> testing so far) for incoming calls from PSTN.
>
> For outgoing to PSTN I have a strange problem.
>
> I dial out OK, the Zap channel answers the SIP channel ok, (But I do not
> see a Call bridged message, and the call has some strange charateristics.
>
> If I call 123, I can connect to and hear the time clock provided by BT
> (I'm in the UK) Is this 'audio before answer'?)
>
> If I call any other external number, eg my cellphone, it never rings,
> and after 30 secs or so the Zap channel hangs up.
>
> I have been testing this with a very simple Dial(ZAP/1/${EXTEN}) command.
>
> What should I be looking for in my setup?
>
> Many thanks, and happy Christmas to all.
>
> Roger
>
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [Asterisk-Users] dialling out
Is it an error or is it a WARNING? On Fri, 24 Oct 2003 [EMAIL PROTECTED] wrote: > when I dial out from my Cisco phone I get this error > > > File channel.c, Line 2258 (ast_channel_bridge): Didn't get a frame from > channel: SIP/210.9.49.216-c26e > > > Regards Mick > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialling out
tar Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Pounder Sent: Thursday, 16 October 2003 8:16 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dialling out something in the dialplan has too course a filter and is matching just the first portion of the number At 06:39 PM 10/15/2003, you wrote: >When trying to dial out > >9 82420173 our main number > >I get the engaged signal before I finish entering the phone number > >Any ideas > > > > >Regards Mick > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialling out
something in the dialplan has too course a filter and is matching just the first portion of the number At 06:39 PM 10/15/2003, you wrote: When trying to dial out 9 82420173 our main number I get the engaged signal before I finish entering the phone number Any ideas Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialling out
No I am using a Cisco 7940 Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Besch Sent: Tuesday, 14 October 2003 11:13 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] dialling out Mick, If you're using the Grandstream, there appears to be a bug in early dial. After the 4th or 5th digit, * sends back an address incomplete message and the phone responds with the buzy - generated at the phone. A SIP debug trace indicates that * is sending the correct information and continues to waut for the next digit, which never arrives. I have sent Grandstream a bug report and they are working on the problem. As a temporary workaround turn the early dial option off. If you are not using the Grandstream, then there may be a similar problem in other phones, or, despite appearances from the SIP debug trace, there may be a problem with *; I'm betting on the phone in this instance. Turn on SIP debug in the CLI and study the responses after each digit. Ignoring the acknowlege packets, you should get a series of invites from the phone, each subsequent one with one additional digit, and each followed by an address incomplete response from *. If everything is working, this happens until the invite requests an extension that is actually in the dialplan. Stephen R. Besch [EMAIL PROTECTED] wrote: >When trying to dial out > >982420173 our main number > >I get the engaged signal before I finish entering the phone number > >Any ideas > > > > >Regards Mick > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialling out
Mick, If you're using the Grandstream, there appears to be a bug in early dial. After the 4th or 5th digit, * sends back an address incomplete message and the phone responds with the buzy - generated at the phone. A SIP debug trace indicates that * is sending the correct information and continues to waut for the next digit, which never arrives. I have sent Grandstream a bug report and they are working on the problem. As a temporary workaround turn the early dial option off. If you are not using the Grandstream, then there may be a similar problem in other phones, or, despite appearances from the SIP debug trace, there may be a problem with *; I'm betting on the phone in this instance. Turn on SIP debug in the CLI and study the responses after each digit. Ignoring the acknowlege packets, you should get a series of invites from the phone, each subsequent one with one additional digit, and each followed by an address incomplete response from *. If everything is working, this happens until the invite requests an extension that is actually in the dialplan. Stephen R. Besch [EMAIL PROTECTED] wrote: When trying to dial out 982420173 our main number I get the engaged signal before I finish entering the phone number Any ideas Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
