search for: kannaiyan

Have anyone tried to interface BT's Broadband Voice with asterisk? Kannaiyan

...s transfer=yes cancallforward=yes usecallerid=yes ukcallerid=yes echotraining=yes echocancel=yes echocancelwhenbridged=yes jitterbuffers=4 rxgain=0.0 txgain=0.0 group=1 pickupgroup=1-4 immediate=no context=default signalling=fxs_ks callerid=asreceived channel=1 Am I missing anything here? Kannaiyan

....85 R8x15.4lines/mm OK Inch-based resolution preferred: no Metric-based resolution preferred: no Minimum scan line time for higher resolutions: T15.4 = T7.7 >>> DIS: 80 00 c6 f0 80 80 01 NOTICE[278543]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec 100 received Thanks in advance, Kannaiyan

I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this

...rq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Can anyone please guide me how to compile zaprtc. Thanks in advance. Best Regards, Kannaiyan

I have this line in my extensions.conf, exten => _393.,1,Dial(ZAP/${EXTEN:3},20,tr) when I make a zap call, it gives me two rings and then makes the zap call. Is there is a way I can make the call immediate? Kannaiyan

Is anyone selling G729 License elsewhere other than Digium? Anyone allowed to sell a similar License as a reseller? -Kannaiyan

Hi. I've searched the archives and found nothing regarding collaborating Asterisk with a Panasonic PBX (TD1232 to be exact) Here's my question: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? On the hardware page for the X100P card is says it's great for

Does anyone know if the E1 cards that digium sells work in India. Also are there any distributers for those cards in India. By E1 cards I mean E100P, TE410P or TE405P -- regards Vikram (http://www.vicramresearch.com)

Hi all, Is there a possibility to set the codecs Asterisk will choose in the dialplan ("exten=>" statements or their contexts) instead of sip.conf? My problem is that I connect my SIP phone with several providers (Nikotel, Sipgate, Stanaphone) for icoming and outgoing calls. Not all of these providers offer the same set of codecs. I'd like Asterisk to use the same codec for the

Hello. I've been going through the archives, but can't discern the state or future direction of IAX on the SNOM100. The most recent image appears to be from September 2002. There was a message on this list stating that SNOM was coming to visit Digium last April with the intention of adding IAX support themselves. For a while there was reference to the I100E on the asterisk and/or

...kup your existing chan_sip.c 3. replace the chan_sip.c with the current one 4. Type, "make install" when you receive a call, it should now pass the SIP 300 message to the caller which you can see with sip debug. Can anyone please help me, what could be the problem. Thanks in advance. Kannaiyan

>When I call a SIP user, the phone should ring in more than one >extentions. Also more than one phone should be able to register with >asterisk. Right now it is not the case. There is no issue here. You seem to be confused, that's all. A SIP account is a SIP account and an extension is an extension. You can assign an extension to an account (or to multiple accounts) and the tool for

...13731]/1 -- Attempting native bridge of IAX2[13731@13731]/1 and IAX2[provider]/4 -- Channel 'IAX2[provider]/4' unable to transfer -- Hungup 'IAX2[provider]/4' Can anyone please help me where could be the problem and how to avoid native bridging. Thanks for your help. Kannaiyan

I've just noticed that if you start a call by writing a file to /var/spool/asterisk/outgoing the cdr created on termination logs the call placed to the local extension - not to the destination in the PSTN. Hence there is no record of the PSTN number dialled. I guess most people want to log the outgoing portion not the local call leg? Anyone know of a setting that changes this? Iain

I downloaded asterisk and was trying to compile fresh, It end up in error, Any help appreciated. cvs checkout asterisk cd asterisk make clean make END UP with following error, (Previously I was able to compile without any errors. After a make clean stopped compiling.) gcc -shared -Xlinker -x -o chan_iax2.so chan_iax2.o iax2-parser.o -lmysqlclient -lz /usr/bin/ld: cannot find -lmysqlclient

Anyone here have experience with these devices? They would ppear to be an affordable alternative to multiple X100Ps. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com "Kick at the darkness 'till it bleeds daylight" - Bruce

Hello everybody, I am trying to setup asterisk to redirect international calls via a carrier which uses a fixed price tel number. The scenario is Dial 087..something (UK number) Pause for answer at the other end Send required telephone number 003..etc followed by # What is the easiest way of doing this? I have trouble with both the pause and adding the # at the end of the number. Best

hi I want to compress and trunk 2E1 capacity over a satellite SCPC link using asterisk. I am nw to asterisk and I need suggestions on how to implement this. Best regards Arinze --------------------------------- ALL-NEW Yahoo! Messenger - all new features - even more fun! -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've set-up a mirror at ftp.gbnet.net/pub/digium It's (sort of) available via www.gbnet.net/public Mirror's the whole of ftp.asterisk.org/pub Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19