similar to: Delay in Zap Calls?

Any body used patch, http://bugs.digium.com/bug_view_page.php?bug_id=0001719 to get the callerid for BT Line. I applied the patch successfully but could not get it to work. Any help. Here are the logs: -- Starting simple switch on 'Zap/1-1' Jun 17 18:22:31 NOTICE[426000]: chan_zap.c:4811 ss_thread: Got event 2 (Ring/Answered)... Jun 17 18:22:34 NOTICE[426000]: chan_zap.c:4811

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

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

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, Below if the error message which I got from asterisk. I was trying to fax to asterisk from my fax machine. I really dunno what is the problem. I use alaw & ulaw codec only through my ATA 186. Can anyone help me what could be the problem. -- Executing Goto("SIP/-080ef9a0", "13732|s|1") in new stack -- Goto (13732,s,1) -- Executing

Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing

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

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

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

When I dial a SIP phone which is specified in the sip.conf, but the phone is not connected, Asterisk gives the message "The user at Extension XXX is on the phone ...." Shouldn't the message be the unavailable message? Is there something wrong with my set up or is this a "bug" with Asterisk? Simon Brown

Greetings, This is probably some configuration issue, but for some reason my system has stopped playing a ringing sound when an extension is dialed. The phone rings but there is no ring sound in the ear piece. Gene Kochanowsky

I have coded chan_sip.c so that you can have // sip.conf register => username:password@somedomain.com/redirectconfig [redirectconfig] redirect=yes redirecturi=sip:12345@domain1.com redirecturi=sip:34556@domain2.com redirecturi=sip:87877@domain3.com .... so when you receive a call it will redirect to the alternating uri's with a SIP 300 Message. It works with the following sequence,

Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at

Assume you have the messages button on your Cisco phone set to dial 3009. Here's an sample dialplan entry that will make the "DND" and "ToVM" and "Messages" button work as expected. This should work for both -stable and -head. exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) exten => 3009,2,VoicemailMain() exten => 3009,3,Hangup exten =>

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

I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up fine on my 7960... W/ the name on top and the number below that. -- Executing NoOp("SIP/614-3ede", "Caller*ID is Matthew Marlowe <6092521155>") in new stack When the phone rings, only 'Matthew Marlowe' would display. When I answer, both the Name & Number will show.

I saw several threads requesting an Asterisk forum to complement the email list. i.e. http://lists.digium.com/pipermail/asterisk-dev/2004-February/003103.html I recently created an Asterisk forum within TMC's popular VoIP forums for everyone to use. http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15

I've managed to use GhoustScript (gs) to take a postscript file and convert it to tiffg3, but I CANNOT seem to make it merge multiple files. Here is the output from tiffinfo on the file that SG generates: fteTYGeh2v.tif: TIFF Directory at offset 0x8 Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 1056 Resolution: 204, 96 pixels/inch Bits/Sample: 1

>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