similar to: How to send Caller ID Extension to Trunk?

Hi Guys I wonder if its possible to set a different MoH based on groups, I mean if one of the Admin group put on hold the call play music 1, if another from Technical Support put on hold the call play music 3, something like this Admin - Music1 Contrallors - Music 2 Technical Support - Music 3 Thanks -- -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

I have an installation which is currenly using 14 DID Trunk Lines. I need to be able to use Caller ID information and currently it is not available on these lines. Is there another way to access this information? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/62559e22/attachment.htm

Hi, I'd like to terminate IAX call on PRI interface. What conditions should be met to be able to send arbitrary caller numbers to called party, so he can call back to supplied ISDN number (that is different for every IAX user) - not through PRI interface, but plain ISDN call !! Thanks in advance, regards, Rob.

I've looked through the archives - and see questions similar to mine, but no answers. What, if anything, can be done to get the incoming Caller ID to be presented on the Budgetone's Caller ID display? In all other respects the phone+Asterisk seem to be extremely happy with each other.

Assuming that you are in the US (don't know about elsewhere), the Calling Party Name is obtained by a database lookup performed by the PSTN switch terminating the call. Calling Party Name is the name assigned to your line or location by the phone company. You can't "send" it. You can only receive it, if the phone company has it to transmit in the FACILITY message. Display Name

Hi, I'm setting up system with repeated calling of outside extension. When it answers, local extension will ring. Supplied caller id displays correctly on outside phone, but on local extension it's empty. Can I somehow supply proper caller id to local extension too ? Regards, Rob.

I'm trying to copy the functionality of something like this in extensions.conf [extension matching on callerid]: exten => _NXXNXXXXXX/7065557230,1,NoCDR exten => _NXXNXXXXXX/7065557230,2,Dial(Zap/g1/${EXTEN}) exten => _NXXNXXXXXX/7065557230,3,Busy in realtime: | 16 | fb | _NXXNXXXXXX/7065557230 | 1 | NoCDR | | | 17 | fb | _NXXNXXXXXX/7065557230 | 2 |

I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling

Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) & Portech MV-370, my problem is that when arrived an external call I don't view (on my internal phone) the phone number but I have the number extension that is configured on MV-370. The MV-370 configuration is: Mobile to Lan Table : 0 * 192.168.1.1 Lan to Mobile Table: 0 * # SIP Setting: Display Name: Portech User Name: 1005

Greetings fellow list members, I am trying to add some tricky functionality to Asterisk dialplan and I was curious if anyone else has come up with a solution to something like this. Basically I have phone representatives that log into one of several queues (not using chan Agent, we log in by the extension), and frequently these agents have to make attended transfer calls to outside numbers.

On 6/18/2015 1:27 PM, Greg Woods wrote: > My provider claims that I am somehow sending an old number that > doesn't appear anywhere <snip> > (I just moved from a POTS provider Century Link to a VOIP provider). Set(CALLERID(number)=${var}) works fine for me. Perhaps some debugging on the channel could help? Set sip debug if it's a sip trunk? You'll at least get to

Hello, We have two asterisk PBXs connected. PBX 1 has SIP trunks connected to our provider. PBX 2 is a remote PBX and SIP Trunk connected to PBX 1. We are able to dial extensions either way and PBX 2 is able to dial out using PBX 1 SIP trunks connected to our provider. We would like to use a separated Caller-ID for PBX 2 and cannot figure out how to do this. Any suggestions

Hi, I would like to use the from caller id, to allow calls yes or no. 101, and 111 should be allowed to use the Trunk, the rest of the phones are not. Is this even possible? So if the from caller id is 101 or 111, then allow the call, otherwise hangup. Thanks, Peter -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am not sure how this works in other countries, but in Australia - according to Telstra - the caller id is managed by the terminating device which is the telephone. If the caller id has been flagged as a silent number then generally the caller id is not display, but rather display with an alternative message 'Number Withheld' or 'Private' or whatever. Right now every

Hello, We use Asterisk and as per book we use MAC addresses as user names. So, when call coming in from outside (SIP trunk) - caller id is good. But when users calling each other on extensions - they see MAC addresses. How would I make it so we see actual names instead of MAC addresses? Without changing users.. Thank you, Ivan

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an

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

You should run the 4ESS protocol ("a" on the Definity command "change DS1 board#) as the Definity may not send Display Name for the NI-2 protocol (settings "b" or "d") Remember to make the Asterisk zapata settings consistent with the Definity. On the Definity trunk group form Page 1, change "Codeset to send Display" to "0 (zero)", also make

Thanks all, I did contact Callcentric about it and their tech support helped meget those headers established. They even helped to troubleshoot Asterisk dialplan. A the end all works as it should. Thank you,Ivan Message: 2 Date: Mon, 15 Oct 2018 23:39:31 +0200 From: Daniel Tryba <daniel at tryba.nl> To: Asterisk Users Mailing List - Non-Commercial Discussion     <asterisk-users at