similar to: SIP caller id

We recently switched my wife's business over to an Asterisk setup using Cisco IP phones (7940s and 7960s) with chan_sccp. They didn't use any kind of "office-style" phone system before, they had one phone in the office with a built in answering machine that would display the Caller ID of the person who left the message while playing the message. I know in the Asterisk

Hi all, Today I've stumbled upon a very strange behaviour with an analog fxs/fxo gateway (AddPac AP1002, http://www.addpac.com/english/AP1002.html) connected to a CVS HEAD(from today) Asterisk server. This manifested itself after enabling the CallerID on the pstn lines connected to the FXO ports of the module. Both FXO modules have their own sip username/passwords and are registered to the

hi, i can get callerid in my phone directly connected to the pstn line. when i cannoect it o * it doen't give me callerid. i have set usecallerid=yes in zapata.conf file/ whaat could have happened? cm ===== Designs __________________________________ Do you Yahoo!? Free Pop-Up Blocker - Get it now http://companion.yahoo.com/

My first post here, so a brief intro: I'm somewhat new to Asterisk, but have been working with SIP in depth for about 3 years. I studied Asterisk for about a year and have been experimenting with it hands-on for the past month or so. I've done 6 Asterisk installs in wildly different roles/applications, some of them test systems, others in semi-production, so I know a little bit about

I have a question - if a user calls up w/ blocked caller id I get the following on my phone Incoming call from asterisk This is the same on my Cisco 7940s and Polycom phones. For average users this is not intuitive at all.. I'd like to configure this so if I deploy this at a customer site it says "caller id unavialable". With the spelling done right.... Any ideas on how this

Stefan Gofferje wrote: > Hi folks, > > I have a rather nasty problem. I have set up an asterisk test system > with a Cisco phone, a X-Lite client and so on and did some testing. > To the developers: great work! Hell of great! > However, it seems to me like asterisk puts the Caller ID Number into the > SIP Display Name and Called ID into the Caller ID Number. That is kinda

Hello list: I'm not sure what is going on, but I am using: asterisk -stable cvs (november cvs download) SIP channel Cisco ATA186 Zaptel 4-port PRI for PSTN Caller ID is enabled on the cisco ATA and seems to work fine. We do not get caller name at this time over the PRI ... Caller name works fine VoIP to VoIP and for PSTN calls, it displays the calling number like caller ID. But for some

hi, how to set caller ID for internal SIP users when dialing out on telco ISDN p2p (hfc card) line? I need to setup numbers from 0 - > 9 (10 sip users).internal caller id is working correctly .. from 0 to 9 ,but when I dial on isdn telco line -> gsm show only our prefix number ( xxxxxxY ) , only 6 (x) numbers of 7 (Y). here is extension 1 dialing on isdn line but on gsm is only 6

I can't seem to get consistant outbound caller ID working correctly. I have set the fromuser and callerid field in my sip.conf for my TelaSIP peer, but half the time it shows up as "No Caller ID" on my cell phone, other times it shows it correctly. Using asterisk CVS. Any ideas? Doug

Hi I'm using the latest 1.4 asterisk, when I get an incoming call from sipgate ( my only sip trunk) the variable Noop(${CALLERID(num)}) is populated with the ower channel ID not the callerid is this correct? the correct callerid show on the internal phones though!! if so how do I get the callerid from an incoming sip trunk? Thanks for your help Robb

Hello all, My pbx server is connected to a sip gateway, when I call an originate command from the asterisk console, to establish a sip connection, the gateway doesn't accept URL with white spaces, for example: * Via: SIP/2.0/UDP 10.10.1.10:5060;branch=z9hG4bK387d772e;rport * * From: "PBX SERVER" <sip:PBX SERVER at 10.10.1.10>;tag=as2512881b * * To:

Hi all, This may be a really stupid question, but, what preset global dialplan variables can I use to determine the calling leg when using Dial()? Say I have phones (SIP peers) originating calls out of the same context, and I need to set the ANI differently depending on who is calling out in order to make it consistent with their inbound DIDs? Asterisk appears to provide a wealth of variables

Hi, We have a system with both ISDN trunks and SIP. We receive incoming calls on both but always dial out via SIP. When dialing out the caller id is set like this: exten => _X.,1,Set(CALLERID(num)=${CC_ORIGNUM}) exten => _X.,n,Set(CALLERID(name)=${CC_ORIGNAME}) exten => _X.,n,Dial(${CC_DIALSTRING}, 60, em) This always works fine on SIP and on ISDN as well when the number is not hidden.

Hello, I'm wondering why asterisk is not transferring the callerid to the sip device. Scenario as follows: sangoma <---> zaptel <---> asterisk <---> sip <---> SIP-Device zaptel is reporting the callerid, but in the sip packages the sip-address shows unknown as user part, as this sip debug package shows: Executing Dial("Zap/62-1",

Just to bug you all (feel free to rant at me), a client wants to set his caller*ID number for outbound calls though us to PSTN. the client is using SIP to us, he can set the caller*ID name fine. if he sets his caller*ID number to anything other than his account number (8440101), the call is dropped into the default context (and then hung up by our dial plan). To get around this i

> > To get around this i updated CVS HEAD and changed the sip entity from > > type=user to type=peer (yes peer!) (type=friend works too but im making > > a point) the client now must register to set his outbound caller*ID > Number. > > Yes, that is normal. SIP has difficulty separating the remote party > identification from the authentication identification

Hi all I have 2 snom 100's and an ix66 (sip aware firewall) set up with asterisk. I needed to register a number of lines so what I've done is make asterisk register all the lines i need (attaching them to an extention eg 1000) and then register each phone with asterisk. so for example in sip.conf: register => andy@sip.mydomain1.org/1000 register => andy@sip.mydomain2.org/1000

Is there a way to append a 91 in front of all incoming caller id numbers? What I am interested in is this - when a call comes in and the caller ID comes across, it is in the format 9165551212. That would be fine, but if I want to call them right back, and I choose that call and 'dial' on my phone, it fails because it hasn't dialed a 9 first. I assume also that if there was a 9, it

Hi Setup : Asterisk 1.6 on Fedora Core 9 with TE410P.. 1. I;ve noticed that whenever during "background(menu-filename)" method - i try to press any key for selection like 1 for some prompt, 2 for another prompt etc...Asterisk takes a while before it takes me to the respective option..Is that normal behaviour ? by the time the caller waits to listen to the appropriate prompt on selecting

thanks a lot for the reply. i thought of that and i did try to send *exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT)exten => _9X.,n,Hangup(${HANGUPCAUSE})* but the provider replies back that it is a wrong number. Then i inserted the sim to an ordinary mobile phone and dialed #31# and the number, then the call progressed fine and it restricted the number. What am i doing wrong