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...in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) On Feb 13, 2008 4:21 PM, Razza <razza30 at gmail.com> wrote: > I am aware there is a SIP over TCP patch. Will this ever become part of > a release, if so are there any timelines? > Thanks in advance. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital....

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

I am aware there is a SIP over TCP patch. Will this ever become part of a release, if so are there any timelines? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080213/1f05009a/attachment.htm

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

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)

-----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 all, Has anyone succesfully installed an AVM Fritz! card on Fedora 7 using the drivers at ATrpms recently? http://atrpms.net/dist/f7/fcpci/ I tried with a clean F7 build on my EPIA 5000 yesterday, after modifying /etc/capi.conf (removing the coment # in front of fcpci line) I received the following error when executing 'capiinit' - FATAL: Error inserting fcpci

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

I have quite an old version of Chameleon Mail, currently the prompts played when leaving a message are ? -- Executing VoiceMail("SIP/209-3b0e", "u5") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing

Hi all, I have been running Asterisk for years (CVS-HEAD on 2005-08-24) with no problems save a failed harddrive. I have decided to build a new box and have Asterisk 1.6.2.2 playing nicely with mISDN after lots of changes to dialplan syntax etc. I am struggling with SIP trunks to sipgate.co.uk and dualtalk.com. Does anyone have a working examples? When I make an outgoing call I get... [Feb 5

All, I have not found/seen a resolution to the issue where my TDM400P seems to cause problems, as outlined in the "mISDN (HFC-S) and TDM400P - mISDN: ISAC XDU no TX_BUSY" thread. I have also not found/seen a simple 'how to' on patching DAHDi with ZAPHFC as outlined in the "HFC-S card" thread. Do I have anyother options with ISDN? Thanks in advance!

Hi all, I have two sipgate accounts (numbers), if I have both accounts register only one will work for incoming calls (which is all i'm interested in). However if I disable either account the other account will work perfectly. Am I missing something obvious? Thanks in advance, Ray. Excerpts from sip.conf - [general] 8<---- SNIP! ---->8 Register => 1212121:aaaaaaaa at

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 have 7965 and am trying to convert the firmware to SIP (SIP45.8-3-4SR1S). Does anyone have a valid XMLDefault.cnf.xml they could share? I have tried the version at voip-info<info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP&view_comment_id=14768#Troubleshooting>for the 7941/7961 but unfortunately /var/log/messages shows in.tftp stops sending after