similar to: Weird CallerID question

Is it possible to allow distinctive rings work for FXS ports as well? I need a certain FXS extension to ring a distinctive double ring. I modified zapata.conf appropriately for dring1,dring2 and it just Seems to ignore my updates. Do distinctive rings only work for FXO ports? Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul Seniuk.vcf

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

Hello all, I have a WildCard TE410P setup and working with a full PRI with the latest CVS. SIP and IAX2 gateways are accessing the PRI without issue, however; echo is very prominent in some calls and is only heard by the IAX2/SIP client. The echo Is not present in calls to cell phones because they are digital, centrex land Lines have a barely noticable echo, but analog lines aint so pretty.

I was looking at the wiki on 'Asterisk as a voice/fax switch' And was wondering if the extension 'fax' is global to extensions.conf Or just to the context it is in? The reason I ask, is that my PRI might have 5 channels that will be scrictly Fax, and to be functional, I need multiple 'fax' extensions in my various Contexts. Hope that makes sense, Paul Seniuk

Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. NOC at GT is telling us

Hello all, I am trying to setup a local asterisk server with SIP & ZAP extentions with And IAX2 'switch' to another Asterisk gateway with a PRI. I have managed to Get it working correctly for calls coming in and out. However, CallerID only Seems to work when I dial out. However, everytime I receive a call, it shows Up as 'IAX Guest User' on a SIP client. How can I

Hello all, I updated from CVS a few days ago and noticed that my calls just cut out without reason. The CLI says this: -- Hungup 'Zap/3-1' It occurs without error or warning. Is their a bug in CVS asterisk or libpri? This never occurred before. Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul Seniuk.vcf Type:

I am attempting installation of spandsp on to an Asterisk installation on Linux RH9 the distribution i am using is that are URL http://ftp2.tootai.net - the README for which i have followed verbatim - my only issue on this was the target for the port.h / tif_dir.h / tiffiop.h files in the 'headers' folder of the distribtion i put these in the /usr/include folder based simply on the

Hello everyone. Could you please help me? I need to do some sales analysis and i'm stuk. I've got a dataset that contains next information: Sales_department; sales_manager; Client; transaction ID; Product (SKU), Cost, Gross income, Profitability Ineed to perfom an ABC analysis (by the Cost of sold products to this client) to find the A clients of each manager, i'm doing it this

Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of

Hi All, I have a server setup with an incomming PSTN line and a bunch of Grandstream BT100 phones. Is there a way for asterisk to foward an incomming callerID from the PSTN to the SIP phone that is setup as an extension? We have a Voice menu setup for incomming calls and I would like to recieve the caller ID of the calls we are recieving after the incomming caller reaches their final

I never heard of Asterisk before today, but from what i'm looking at on the website and hearing, it sounds pretty incredibly. If I understand correctly with a 1,500.00 Wildcard TE410p T1 card, a good BSD or Linux Server, and a couple IP phones or Netmeeting on a few workstations, and of course, Asterisk which is free; I call have a small call center. This can't be? I was looking at

Hi, This is really strange (but probably simple solution). The CALLERID(all) setting doesn't seem to work when the incomming callerid is 'unknown'. Dialplan looks like this: exten => _3072,1,Answer exten => _3072,n,Set(CALLERID(all)=DIRECT <0850553072>) exten => _3072,n,Dial(SIP/2001&SIP/2002&SIP/2003&SIP/2004&SIP/2201&SIP/2202&SIP/2

Hello, Can Asterisk perform as a H323 Gatekeeper? Here is my scenario: I have a customer that has a calling card program that will be transmitted as VOIP from a Cisco 5300 in Hong Kong and terminated here in North America. The catch is that, the termination is being handled by a third party company and we still need be the 'buffer' in which we pass the VOIP traffic to the

Hi there, I was wondering how I can use setgroup and checkgroup for perfom incoming and outgoing limitation checks. I've have some users that doesn't what to be able to recieve more than 1 call at a time, and I also want to limit a users outgoing call abilities. Any help would be greatly appreciated. Kind regards Cf --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus

Hello, I need some form of a "permutative" operation on a numeric vector x x = (v1, v2, v3, ..., vN) that produces x.r = (v1, v1+2, v1+v2+v3, ... v1+v2+...+vN) If the operation is sum() I can run x <- 5:8 m <- matrix(rep(x, length(x)), ncol=length(x)) (x.r <- rowsum(m * upper.tri(m, diag=TRUE), rep(1, length(x)))) But there's two things I don't know and kindly

Hi folks, We've been having a weird issue... It is happening more often in the last few months... Most inbound calls, we have in our dialplan before Queue(): Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL}); So when the call rings a member, softphone will show this string .... The issue is that sometimes the string showing in the softphone is not the same. Its a

Hello, I am experiencing a weird problem on asterisk when I place an outbound call, park it and then retrieve it. I am using extensions.ael with macro and switch and I get something as SW_456_... that is autogenerated by asterisk when compiling the extensions.ael This doesn't happen when the call comes from outside. The bad CallerID is displayed only on Cisco 504G phones and it is

Well, Once upon a time, I had problems receiving callerid, and then one day, Mark was logged into my asterisk box helping with something else, and I asked him about this, and he showed me a nice tweak to some source file that made it work. Some time later, I must have done hundreds of CVS updates, and along the way, lost the above patch (one liner) and so callerID hasn't worked for a long

Hi Chris, Yes the call was from PSTN and your solution worked great! I've read about SetCallerPres earlier but I didn't connect the dots this time. Thanks alot! :) Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Christoph F?rstaller Skickat: den 29 mars 2007 15:29 Till: Asterisk Users