Re: [Asterisk-Dev] Asterisk on Cygiwn
Time Bandit wrote: >>Boot linux and use vmware when windows is needed. I know several laptop >>owners that do this. >> >> >I do this on my desktop. It is the only way I found to run windows >without ever rebooting my machine :) > > Same here and I also find that vmware suspend of the windows virtual works better and faster than most laptops running windows natively. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Delphi ActiveX component
On 11/16/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > Hi everybody. > > I need develop a IAX softphone with Delphi, but i didnt find a OCX component. > Anyone know how can I > find this component ? > > Tomas I don't believe one exists as part of the standard distribution. You're welcome to roll your own though around IAXLib. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Delphi ActiveX component
Hi everybody. I need develop a IAX softphone with Delphi, but i didnt find a OCX component. Anyone know how can I find this component ? Tomas ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Asterisk on Cygiwn
> Boot linux and use vmware when windows is needed. I know several laptop > owners that do this. I do this on my desktop. It is the only way I found to run windows without ever rebooting my machine :) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Call Admission control
Dear All, Sorry I am newbie in this forum but I wanted to check out if someday we can see Asterisk perform CAC on contexts and limit the number of concurrent calls a context can make to another context, I can see it performed on iax and zap trunks but with my working scenario I really need to have such an implementation in order to ensure the quality of the calls between different sites, here is a scenario I am having : I am having 1 Asterisk Server I have phones in 3 locations Each Location are using a different extensions.conf context to seperate the dialing. I tried reaching out to the users list but it seems we can limit the incoming but the outgoing is not in reach of limiting, do u have any ideas or suggestions? Thx MAG ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Getting caller id
On 11/16/05, Jacob Tinning <[EMAIL PROTECTED]> wrote: > On Wed, 16 Nov 2005, ast guy wrote: > > > In ast_channel (channel.h) structure there is ast_callerid variable, > > but when I use that variable in code it gives error: > > > > structure has no member named `cid' > > How do I get caller id string ? > > I think you can read the caller in chan->cid.cid_num and chan->cid.cid_name. > > Mvh. Jacob > -- mentioned 'ast_callerid cid' is in ast_channel structure but any way found it under ast_channel->callerid but mentioned in docs ast_channel.ast_callerid.cid_num ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Getting caller id
On Wed, 16 Nov 2005, ast guy wrote: > In ast_channel (channel.h) structure there is ast_callerid variable, > but when I use that variable in code it gives error: > > structure has no member named `cid' > How do I get caller id string ? I think you can read the caller in chan->cid.cid_num and chan->cid.cid_name. Mvh. Jacob -- ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Getting caller id
In ast_channel (channel.h) structure there is ast_callerid variable, but when I use that variable in code it gives error: structure has no member named `cid' How do I get caller id string ? ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Changing Port 5060 with 81
On 11/16/05, Abdul Lateef Khan <[EMAIL PROTECTED]> wrote: > Hi friends, > > I want to change the standard 5060 sip port to our any defined port. i made > some change in sip.conf but it is not working, I have 2 softphone which are > able to register with 81 port but the any kind of hardphone is not able to > register using 81 port. > here is my sip.conf configuration > > [general] > port=5060 > > > [123456] > type=friend > username=123456 > host=dynamic > port=81 ;the hardphone should be register with 81 port > context=voip > allow=g729 > allow=alaw > allow=g723.1 > > Please help me how i can register with 81 port? > > Thank You > Abdul Lateef > did you try changing to other port ? ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] ooh323 cause asterisk reload broken
OS:Red Hat Linux 8.0 3.2-7 gcc version 3.2 20020903 asterisk-1.2.0 rc2 asterisk-ooh323c when I reload in CLI, asterisk crash. (gdb) bt #0 0x4018ef89 in free () from /lib/libc.so.6 #1 0x406e84d2 in delete_peers () from /usr/lib/asterisk/modules/chan_ooh323.so #2 0x406e5d64 in ooh323_do_reload () from /usr/lib/asterisk/modules/chan_ooh323.so #3 0x406e7f37 in do_monitor () from /usr/lib/asterisk/modules/chan_ooh323.so #4 0x4002bae0 in pthread_start_thread () from /lib/libpthread.so.0 ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Re: mtp-2 ([EMAIL PROTECTED])
Hi Jan, I believe that MTP2 (Q.703) is availabe for Zaptel devices in source/library form in the newly released GPL SS7 drivers by Sifira A/S http://www.sifira.dk/. It is described as being mostly complete last time I checked. I don’t know if its in a from that is usable for your purposes. You can check http://www.voip-info.org/wiki/view/Asterisk+ss7+channels for further info. With Regards, Hadi Jadallah > Date: Tue, 15 Nov 2005 21:15:32 +0100 > From: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]> > Subject: [Asterisk-Dev] mtp-2 > To: Asterisk Developers Mailing List > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > hi All, > > I would be interested to know if anyone have done any mtp-2 > work on the Digium boards. I just bought a TE110P for this > purpose, but I have difficulties verifying if the board will > support mtp-2 or not as long as there is no hardware manual, > or register description available. I simply dont know how > much of the Falc56 that are available and what limitations > this board have. > > Any guidance would be highly apreaciated. > > Jan > > ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] mtp-2
Ahhhr, Just realized that there even is a separate ss7 listing thanks. Jan Jacob Tinning wrote: On Tue, 15 Nov 2005, [EMAIL PROTECTED] wrote: I would be interested to know if anyone have done any mtp-2 work on the Digium boards. chan_ss7 is reading mtp2 data from timeslot on a Digium board. At this time, it has been tested on TE110P and TE410P. See http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+channels Download the channel-module and read the code to get the details. Any guidance would be highly apreciated. I hope this will help you - at least to get started. Mvh. Jacob ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] mtp-2
On Tue, 15 Nov 2005, [EMAIL PROTECTED] wrote: > I would be interested to know if anyone have done any mtp-2 work on the > Digium boards. chan_ss7 is reading mtp2 data from timeslot on a Digium board. At this time, it has been tested on TE110P and TE410P. See http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+channels Download the channel-module and read the code to get the details. > Any guidance would be highly apreciated. I hope this will help you - at least to get started. Mvh. Jacob -- Jacob Tinning System Developer SIFIRA ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] mtp-2
The duplicated messages was actually an accident, I did not believe the first had arrived due to a 1 hour delay. I have asked Digium the question, but had no reply on it. I do however know there are members in this comunity that are working on mtp-2 that might share some info, but I have a feeling they use different boards. Thanks anyway Jan Jared Smith wrote: On Tue, 2005-11-15 at 21:15 +0100, [EMAIL PROTECTED] wrote: I would be interested to know if anyone have done any mtp-2 work on the Digium boards. I just bought a TE110P for this purpose, but I have difficulties verifying if the board will support mtp-2 or not as long as there is no hardware manual, or register description available. I simply donæt know how much of the Falc56 that are available and what limitations this board have. This is something you'll probably want to ask Digium directly, as they're the only ones that know all the details on the boards themselves. FYI, repeatedly asking the same question on the mailing list makes people less likely (not more likely) to answer your question. -Jared ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
