[Asterisk-Dev] SS7 for *

[mawali at news.icns.com]

Hello Steve
      It would have been good if you took a little more time to tell what 
do you mean by "SS7 for * is now working.", since there has been no such 
announcement.

 Can you please explain (you do know there is a bounty on  it).

Regards

On Sat, 2 Oct 2004, Hadi Jadallah wrote:

> Hi Steve,
> 
> Steve Wrote:
> >If you want a GPL one, carry one. However, if you are prepared to pay, 
> >SS7 for * is now working.
>  
> Is there anybody I can contact regarding this? I am prepared to pay.
> 
> 
> Yours,
> Hadi.
> 
> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of asterisk-dev-request at lists.digium.com
> Sent: Friday, October 01, 2004 1:59 PM
> To: asterisk-dev at lists.digium.com
> Subject: Asterisk-Dev Digest, Vol 3, Issue 1
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> Today's Topics:
> 
>    1. qcall CDR problem (VoIP)
>    2. Re: ast_waitstream_full: Wait	failed	(Resource	temporarily
>       unavailable) (Dinesh Nair)
>    3. Re: Call control problems from Java (Miroslav Nachev)
>    4. Re: Use the Meetme application with another module	than
>       USB-UHCI (Arkadi Shishlov)
>    5. RE: Call control problems from Java (Zac Wolfe)
>    6. Re: Wish List / Brain Storm from AstriCon (Steve Underwood)
>    7. Re: Wish List / Brain Storm from AstriCon (Steve Underwood)
>    8. How is DTMF relayed? (Andreas Sikkema)
> 
> 
> ----------------------------------------------------------------------
> 
> Message: 1
> Date: Fri, 1 Oct 2004 12:16:41 +0800
> From: "VoIP" <voip at er21.com>
> Subject: [Asterisk-Dev] qcall CDR problem
> To: "'Asterisk Developers Mailing List'"
> 	<asterisk-dev at lists.digium.com>
> Message-ID: <ER-DNScKVOz9b1tvmC20000001a at er-dns.er21.com>
> Content-Type: text/plain;	charset="us-ascii"
> 
> Hi, I am trying to use qcall to do callback service. And put the qcall file as follows,
> ------------------------------------
> SIP/2001 at b.com 1001 at a.com SIP/3001 at c.com 0
> ------------------------------------
> Qcall first initiate a call to 2001 at b.com and wait for his answer, when 2001 at b.com answers the call, Qcall connects to 3001 at c.com. So 2001 and 3001
> can talk to each other. Here we set the caller id to 1001 at a.com.   
> 
> Actually there are 2 legs in this call. One is 2001 at b.com, and the other is 3001 at c.com. I found CDR only recorded call from 1001 at a.com to 3001 at c.com but no such CDR from 1001 at a.com to 2001 at b.com.
>  
> If 2001 at b.com and 3001 at c.com are both PSTN numbers, then it should be a big problem to charge the callback users. 
> 
> Although this is a user level questions, I doubt the CDR mechanism should have problem. 
> 
> Regds,
> KK 
> 
> 
> 
> 
> ------------------------------
> 
> Message: 2
> Date: Fri, 01 Oct 2004 12:19:21 +0800
> From: Dinesh Nair <dinesh at alphaque.com>
> Subject: Re: [Asterisk-Dev] ast_waitstream_full: Wait	failed	(Resource
> 	temporarily unavailable)
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <415CDAC9.3000708 at alphaque.com>
> Content-Type: text/plain; charset=us-ascii; format=flowed
> 
> On 01/10/2004 02:55 Gil Kloepfer said the following:
> > I think I identified this problem a while back.  If you guys would
> > like me to repost this, I can...otherwise look for the message
> > I sent with subject: Oddities in asterisk/say.c
> 
> excellent, gil. my first observation of the problem was voicemail hanging 
> up on me as well, which i then tracked down to the SayNumber application, 
> and further back to ast_waitstream_full().
> 
> your solution mirrors the observation (and solution) i posted as well.
> 
> so, based on your experience, it'd be ok to modify that snippet of code to
> 
> if (audiofd > -1 && ctrlfd > -1) in all places that it occurs in say.c ?
> 
> (the rest of the functions are just different language handling functions, 
> so theoretically it should be alright)
> 
>