search for: calledids

This is one of the chronic problems. Try this option in sip.conf: match_auth_username=yes Carefully read the description, it is better to test in "after hours". 02.04.2015 2:50, Andrew Galdes ?????: > Hello all, > > I have an Asterisk server (Asterisk 10.12.4) with multiple sip > accounts with the same service provides. We have 8 phone numbers in > total. > >

Maybe hidecallerid=yes in Zapata.conf Thanks, Steve Totaro http://www.asteriskhelpdesk.com > -----Original Message----- > From: Rich Adamson [mailto:radamson@routers.com] > Sent: Thursday, April 13, 2006 12:14 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Display "Confideltial" or "unknown" on > calledid

Hello all, I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts with the same service provides. We have 8 phone numbers in total. Incoming calls from the public are all correctly directed to appropriate office handsets. However, the display on the reception phone (the only one i care about) is always showing the same "SIP/Account1_0843214321" rather than the account

Hi, Andrew. You are trying to solve two tasks: definition through what line the call came and a beautiful display of this information. 1. definition through what line the call came. If the username and password for inbound and outbound registration the same, then try the following: a) delete "register" lines. b) add option "callbackextension=Company1" to Company1 friend

I would like to hear stories from people using TAPI, CTI or CallerID software with Asterisk. What are you guys using, setup examples, etc. Has anybody sucessfully integrated SugarCRM with Asterisk and how did you do it. Are you running callerid software? Did you stumble into problems like using tapi and callerid software returned both the callerid and calledid? Hope you can help me out with

Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but it does work. For prosperity, the SIP service is through Internode. Here is my "extensions.conf" file: exten => s,1,Set(thedid="${SIP_HEADER(TO)}"); ignore this one exten => s,2,Set(pseudodid=${SIP_HEADER(To)}) exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)}) exten =>

When receiving inbound calls from a Vonage Softphone extension, I'm unable to view/maniupulate calledid data. but it shows up in the CDR records and on called handsets.. any ideas? exten => asda,n,NoOp(callerID is ${CALLERID}) exten => asda,n,NoOp(CallerID is ${CALLERIDNAME}) exten => asda,n,NoOp(CallerID is ${CALLERIDNUM}) -- Executing [asd at pstn-in:2]

Hi, i have the same problem. Trying to use the dialplan function CONNECTEDLINE() this way Set(CONNECTEDLINE(name)=${SIPPEER(${EXTEN},callerid_name)}) Set(CONNECTEDLINE(num)=${EXTEN}) ends with [Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function CONNECTEDLINE not registered Same happens trying function CALLEDID. I am using Asterisk 1.6.1.20. What do i

Guys. Anybody knows why this is happening? Seems every time I make an internal call, the manager shows this and I don't get the callerid on my identapop but rather the calledid.. Event: Dial Privilege: call,all Source: SIP/intruder1-85f0 Destination: SIP/test-f037 CallerID: 201 CallerIDName: Anton Krall SrcUniqueID: 1117038116.7 DestUniqueID: 1117038116.8 Event: Newchannel Privilege:

Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I

Hi Dmitriy and others and thanks for your help so far. The option "match_auth_username=yes" seems to have had no effect. From my reading, this option will try to match the username of the incoming SIP account to a section heading. If that is how it must work then i can see a big problem. I'm trying to present the receptionist with a nice display of which line the call came in on.

I don't suppose anybody has an updated CalledID patch for Asterisk 1.4.26: https://issues.asterisk.org/view.php?id=8824 I've been running 1.4.21.1 for a while and have the need to update one of our Asterisk Conference/Faxserver/switches and wanted to see if the patch available would apply. It failed in several places. This keeps me from updating my other installs. Doug -- Ben

Hi Guys! After 1 week or looking for answers about CTI and Asterisk, I havent been able to find the necessary applications to do what I want to do. Maybe you guys have more insight on this. I tried installing asttapi, and works great! Can make outbound calls from outlook, etc.. Nice work! For incoming calls.. Ive been trying software like ascendis callerid, call alert and identapop pro, but

...ss 1, then Ring group will be activated. All are working fine. My Problem: I want to avoid IVR for some phone numbers depends on their called IDs. If my common users will call to my DID number, my ring group will be activated directly without playing IVR. We will do it by adding one by one calledIDs in inbound route and redirect it to ring group. This solution is suitable for 5 or 10 caller IDs. But, I have 200 standard caller IDs. Its difficult to add all these one by one. Is there any script or any other way to handle to do this at a time? Thank you. Regards, Chandra....

Hi, I have two * box. One is box1 and the other is box2. And I have two iax clients A and B. A is registed with box1 and B is registed with box2. If I make a call from A to B using the following method: IAX/[<user>[:<secret>]@]<peer>[:<portno>][/<exten>[@<context>][/<options>]] The CalledID that B got is <user> but not A's CallerID. I

Hello, for a long time i am using asterisk 1.6 with astgui. but for production system i intend to use asterisk 1.4 which i think might be more robust. And for a more developed service options i preferd to install with freepbx. But still there are big plusses and minusses for both system. My complain about astgui+1.6 was.. For example there were no backup trunk config running on that version.Even

Hi everybody, Every time callers reach my FXO port, asterisk produces one ring tone just before it executes Answer(). How to remove this? I have commented "#define RINGBEGIN" on zconfig.h, but it does not help. Thanks in advance for your help. Cheers, Anto -------------- next part -------------- An HTML attachment was scrubbed... URL:

Skipped content of type multipart/alternative-------------- next part -------------- ############ # amd BOX # ############ ## Step 1 ## Bob(ext. 6202) place a remote IAX2 call to the operator (ext. 6302) ## Reminder : _62XX are register on 'amd' and _63XX on 'dell' -- Executing SetGroup("SIP/6202-d193", "IAX") in new stack -- Executing

Hi, I just upgraded by 1.0.x home server to 1.2.2. Overall the upgrade went fine, but a strange problem has cropped up with the CALLERID name value of incoming calls from the X101P card. When an incoming call is presented (via Vonage ATA), the calledid value getting double quotes up from: -- Executing NoOp("Zap/1-1", """WIRELESS CALLE" <1404xxxxxxx>") in

Hi! I am new to Asterisk too, I got the similar problem and I would like to know how to get * to work behind NAT. When I have the SIP Debug turn on, I got the error 479 from FWD when * try to register with FWD, it looks like * is using the local IP (192.168.x.x) in the Contact field. I have put the nat=yes in the [FWD.Pulver.com] content, but it does not seems to make Asterisk aware the