similar to: SIP over TCP

I am working on a project that requires shared extension. Where shared line looks at the status of a line/trunk, shared extension would look at a series of channels as the same "extension". The users would like to add destination channels on the fly, to provide roaming extensions, but maintaining fixed channels as well. If a call comes in on an extension, the system needs to honor the

I've been failing to get chan_mobile working, so am looking to the experts to help :o) I have followed this guide - http://www.voipphreak.ca/2008/10/30/installing-and-configuring-chan_mobile-for-bluetooth-presence-support-in-asterisk-16/ and this guide - http://www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html and tried hybrids of the two which is

What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's

Hi, I'm trying to implement a Voice Drop service within Asterisk dial-plan. The service is supposed to work as following: 1. A initiates a call to B 2. The call is answered by B's answering machine 3. A hears the answering machine's greeting and the recording beep 4. A speaks a few words into the recording to personalize the message 5. A presses some DTMF keys (say, '##') to

Hi all, does anyone know of an application that will run in Windows (in my case users PC's) and behave in a similar fasion to chan_mobile? I'd like the app to register with asterisk, then talk to a (or a number of) mobiles over bluetooth thus creating an FXO port? I'm not interested in SMS etc. just voice. Thanks in advance Ray. -------------- next part -------------- An HTML

I had Asterisk 1.6.2.2 running fine with a mISDN using a HFC-S based card. I installed my TDM400P into the PC, it's really slow to boot now, when it finally does I gets stuck in a loop of reporting "isac xdu no tx_busy". Anyone able to assist? Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does any one put a HFC-S card working in nt ptp mode? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100221/ca845019/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News -

Hi list, i'm keen to move to Asterisk 1.6, so really need to update my system which is running Mandrake 9.2 although it has been solid for years, fo Fedora 8. I have a Fritz! card for ISDN BRI, I have installed the drivers/kernel drivers from ATrpms (http://dl.atrpms.net/all/fcpci-03.11.07-14.fc8.i386.rpm and http://dl.atrpms.net/all/fcpci-kmdl-2.6.23.14-115.fc8PAE-03.11.07-14.fc8.i686.rpm)

2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <russell at digium.com> * CREDITS, include/asterisk/http.h, main/tcptls.c (added), main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), main/Makefile, main/http.c, include/asterisk/tcptls.h (added), configs/sip.conf.sample, CHANGES: Merge changes from team/group/sip-tcptls This set of changes

Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? Thanks. -- Rgds, -- Rgds, Hans Yin Web: homeofhans.homeip.net Email: hansyin at gmail.com MSN: hansyin at hotmail.com Skype: hans_yin_vancouver

Hi all, I?ve read that meetme works at G711 (ulaw), so asterisk would down-mix a number of parties using G722, is that still correct? If so, i?ve also read that Joshua Colp was/is working on a replacement (conf_bridge?) that works with G722. If this is this still in active development are there any planned timelines? If it?s in 1.6.0.6, and i?ve just missed it or it?s been renamed please be nice

Asterisk supports SIP-T? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090217/7cc67800/attachment.htm

Hi all, I just upgraded to Asterisk 1.4.18 a few days ago and I don't use Broadvoice TOO often, however I have a Vermont number with them and so my mother in law calls it to talk to my wife once in a while, so that's why it took me so long to notice it wasn't working. Anyway, when she calls she gets a busy signal (as I've tested when calling it from my cell). When I enable

Is anyone using a cisco router as an ISDN gateway with Asterisk? As you might have seen from a couple of my threads, I have been looking at Fritz! and Cologne cards, both of which require development against a specific version of asterisk/zaptel (e.g. chan_capi), which is intrusdive and causes a lag in deployment. I was thinking a better approach might be to use a seperate gateway, such as a Cisco

I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML

Can any one tell me how to install CAPI for Fritz cards on a Mandrake 9.2 system. I have been working through http://www.voip-info.org/tiki-index.php?page=Asterisk%20How%20to%20conne ct%20with%20CAPI and just cant seem to get it working. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work?

I have a standard BT home DSL, which means I cannot have a static IP address, therefore i'm forced to use NAT, I subscribe to a DDNS service and have written a VB app which polls the router every 10 seconds and updates the DDNS if appropriate. This is fine but I need to be able to modify my sip.conf (externip = w.x.y.z) and reload sip, does anyone know of a script/app which does an nslookup

> what change to what API, what magic? The changes I'd suggested to fishead_from_ogg and friends to return an int rather than a packet, so the magic check can return an error if it doesn't match. The packet/header pointer would be an added parameter to the functions. Actually I think I'd sent this to you only rather than the list. And it looks like skeleton is not used widely