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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

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?

-----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 -

...ticipated 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....

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 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

...wall) open then would the firewall not allow two way communication without incoming port mapping/NAT (providing that the SIP phone started "talking" first)? I'm not sure about that - I'm being hopeful though :) STUN would be very acceptable to me if it worked though ;) Derek razza wrote: >Derek, >I'm not an expert in these area's hence the offer to play, but in answer >to your questions to the best of my ability - > >1. I don't see any reason the outbound proxy cant be in the public >domain although this is where the NAT issues start kicking in...

All, before I start, I have read the readme and realise the dial syntax has changed since 3.5.....(and I really wish it hadn't).... anyway can somone please help me round a config issue? Previously I had a capi.caonf as follows. This config allowed me to have multiple MSN's on the same ISDN/Fritz! card under seperate contexts.

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

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

Razza, I have a MP114 FXO/FXS that I have never got to work , even as an FXS, even though I have several other FXS's that work fine ie Linksys PAP2 etc.. would you put up your config? PDE

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

...in 1.6.2. Does it only work for G722 or any WB codec ? I am working with another 16k WB codec that I can transcode to 722 and vice versa. I was curious if the 1.6.2 MeetMe bridge can also mix 722 with any other WB codec natively(without downscaling). Thanks, Serhad Doken ------------------> Razza wrote: 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 in your re...

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

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

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)