similar to: Remote-Party-ID and CallerID

Hi to all, i saw that in chan_sip there is the possibility to let the * to take the number from the Remote-Party-ID header field on incoming calls from gateway. What about to let the * to generate the Remote-Party-ID on outgoing calls? this is is useful for us to let the users to have their outgoing number hidden but let our switch to get the correct record for accounting. I think that If i hide

Hello ppl, Sorry to re-post it, but kinda these issues are getting on my nerves. I tried Set(CALLERID(num)=7329) on 1.2.12, which works fine, but not on 1.4.4. The problem : 1. I receive call from caller 'AAA' on my number, 'BBB' which is on my Asterisk box. 2. I have to redirect the call to some other number, say, my cell num - 'CCC'. 3. My PSTN provider wants the

We've been using Asterisk 16 for a while now, and tried turning on send_rpid = yes in my pjsip config for end points.  This solves a problem we're having where attended transfers aren't updating the CallerID when the transfer is complete (it would show the callerID of the party attempting the transfer, and never update after the transfer happened). The side effect of this change

Has anyone noticed that Asterisk seems to always set the remote-party-id in a SIP invite to be the same value as the From: field? In most cases that isn't a problem. However, in the case of an attended transfer it IS a problem. The remote-party-id should be the party who initially called and the From: should be the party doing the attended transfer. This seems like a bug. - Doug.

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

Hi... I just started working with Asterisk and found something that looks like an error, but i want to be sure, so that's why I'm asking you. When i make a call from "A" to "B" (both SIP clients), I don't see the name of the called party in the phone that initiated the call, just the dialed number. I made an ethereal trace and found out, that there is no name

We would like to present a toll free CallerID when making outbound toll calls. In the past, when our PRIs were directly connected to a Nortel CS1000 we could do this, without issue. Now that the PRIs are front ended by a mediagateway facing asterisk, we can no longer do this. Is it possible to set the billing number via a SIP header and set what should be presented as callerid as another header

Hello Guys, after an update to cvs head (thanks oej!) my CiscoGW can now flag unkown caller's to Number AND Name "Unkown". Before i again open a new bug (which isn't a bug :-)), can someone confirm this: - PrivacyManager does not recognize this as an unknown number - it's not possible to set ANY CID with SetCallerID, it allways stays on Unknown (with chan_capi i had

I am using AMI to Originate a call. I have been able to get the caller id number to be passed through. However, I can't get the name to be passed through. A person I'm working with has a Freeswitch that is able to pass the caller id name and number through for their call. Comparing the Asterisk SIP messages to the Freeswitch SIP messages, I have narrowed the problem down to a single

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

We have a scenario where we wish to present a toll-free caller id, yet have our calls rated based on our billing-telephone-number. Is it possible to present a number in the sip header for billing and another number in the header for jurisdicional call rating? Whereas today, all of our calls are billed at the highest rate (intra-state) because we're presenting a number that isn't in the

Dear All, I am a newbie about asterisk. I have 1x X100P card 3x Sip phone I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I got no caller id, even my direct PSTN service operator. So at that moment I was using *1.0.9. than I changed to asterisk@home 1.3(1.0.9). I got same

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension that matches my number with them, in the

For INBOUND calls, does Asterisk support P-Asserted-Identify or Remote-Party-ID, or does it support both? Again, this is for INBOUND only. I know how to add those headers for outbound calls. My guess from what I have seen is that it supports both, but I wanted to check with the list. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Heres a weird one... Call comes in on mISDN channel. Little bit of dialcode (in a macro) looks up the number in the astdb and puts an name to it. No real magic there, and it works well. Same macro also has parameter passed in to put a prefix on the name - this is set in the DDI handling and is dependent on the number called and allows phone users to see which number was called (company

Can you confirm that currently there is no way to update the caller ID via the manager interface once the B leg is ringing or connected? Looks like this would be feasible with the functions introduced in https://issues.asterisk.org/view.php?id=8824 ("[patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation"). Such functionality could be desirable in

Hello list, is it possible to extract the Remote-Party-ID from an incoming call in the dialplan ? Is there some kind of function for this ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110316/bda02b4d/attachment.htm>

Hello, Does anyone know how to set the remote party id? Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091102/e27bedb8/attachment.htm

I'm finding that I'm a bit disappointed that Asterisk doesn't naturally forward the Remote-Party-ID from inbound SIP calls (where trustedrpid=yes) to outbound SIP calls. I guess this is going to be something we have to use SER for, unless we make our own custom build (which I'm reluctant to do). Is there a good reason that the main version doesn't do it?

> -----Original Message----- > From: Michiel van Baak [mailto:michiel@vanbaak.info] > Sent: 26 May 2005 20:22 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID) > > > Anton, > > My script is not connecting to the manager interface. > The php script is run as agi script as first when a call > comes in. The