similar to: variables from before call entered queue

Does anybody know if there is an app that will cause similar to occur on users PC? I have a scenario where users will have snom phones on their desks. Ideally when their phone receives a call I need to popup a web browser with a specific url. Any ideas appreciated. Neil on 5/7/05 10:52 PM, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> wrote:

Hi all, In order to use the "open url" function of zoiper (it opens an url based on the asterisk $callerid(dnid)), I need rewriting of the dnid. In my dialplan I have: exten => 1000,3,Set(CALLERID(dnid)=newdnid) exten => 1000,4,Noop(${CALLERID(dnid)}) exten => 1000,5,Queue(test-queue) but the callerid(dnid) shows the extension called (the member of the test-queue) and not

I have both the PJSIP add and the chan_sip way of adding SIP headers in there. The Verbose is showing the variable value is there. The INVITE to PJSIP/Agent1 does not include either X-My-DNID or X-My-DNID2 headers. exten => 1234,1,Verbose(X-My-DNID:${MY_DNID}) same => n,Set(X-My-DNID=${MY_DNID}) same => n,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID}) same => n,Dial(PJSIP/Agent1)

Hi Guys I have just installed a call center onto Suse 10. I have managed to do a DBget (astdb) and extract the DNID numbers to play a DNID specific greeting. We have installed Snom 320 and the customer would like us to Send the DNID(nam) to the phone screens so that the agent will be able to answer in the correct language and with the specific customer company name (ie. Agent says

Question: Naturally there are times when need to I reformat an extension in a context as such: ;Reformat add CC1 exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) -or- ;Reformat 011 with with +CC exten => _011X. ,1,Goto(+${EXTEN:3},1) It's a helpful trick, BUT there are times when I want to send the call to another context in its original un-reformatted state. Naturally the ${EXTEN}

I am having a problem with the Call Queue Feature. If I let a user into the Queue prior to an agent being available for them to take the call, they experience the following: 1) they hear that they are the first in line 2) when the agent finally logs in (the caller on hold in the queue is sent to the users phone) 3) the AGENT is still in the login phase hearing that they are "successfully

Hi, We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian Sarge) and the behaviour of our Call Centre queues has changed slightly. Before the upgrade, when a caller was waiting in the queue, the estimated hold time was announced as expected ("estimated hold time is less than 2 minutes ..."). Now the caller gets an announcement of their sequence in the queue

hi, I hope someone can help me:-) we’ve got a freepbx server. there are 2 special extensions (2001, 2002). if someone calls this extensions (or a call is forwarded to these extensions) and these extension hangup (not the caller party), then we’d like to put the calls back into a queue (1000) and wouldn’t like to hangup. I read your description about hangup hooks:

Dear Asterisk-Users: Question: ======== How do I get asterisk to pass DNID/RDNIS information between asterisk machines using iax2, in a Dial(IAX2...) command ? Setup: ===== I have two asterisk boxes, MASTER and SLAVE. MASTER is running 1.2.0 and SLAVE is running 1.2.1. The main box handles incoming calls on a multiple lines (both via hardware connection to our internal PBX and calls

Greetings. I have next feature in features.conf : send => *9,peer/both,AGI,/etc/asterisk/agi/map_mail.pl What it does is parsing CALLERID and DNID from AGI input, performing some actions in MySQL with these values, and then running application for peer (for example, PlayBack) Sounds simple, and it really is. When my user is receiving a call (we are the B leg) and presses *9, everything

Please has anyone experienced a bug with queues in Asterisk? No matter what settings my queue always thinks it got agents available in it. Plus if I take all the members out, then calls don't join the queue even though I've specified join-empty. Any advice? Neil

Hi, Does anyone know anything about the following? In a hosted environment where several area DIDs are provisioned on a single server, how do most carriers establish the origination DID, number. Asterisk allows us to modify the CallerID, name, number and DNID channel variables before dialling out via SIP. Most carriers will allow us to spoof a callerID when placing a call, and pass it forward.

Hi, We encounter a problem with the variable CALLERID(dnid) We use E1 lines where we can make an inbound call or an outbound call on the same channel (not at the same time) If the CALLERID(dnid) is not used, than the CALLERID(dnid) will be the CALLERID(dnid) of the previous call For example: - First we get a inbound call on channel DAHDI/11-1 with CALLERID(dnid) = '655871460' We read

I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs

> -- Started music on hold, class 'default', on > SIP/193.111.200.67-0815c790 > -- outgoing agentcall, to agent '1001', on 'Local/201@sip-0add,1' > -- Called Agent/1001 > -- Executing Dial("Local/201@sip-0add,2", "SIP/101|20|tr") in new > stack > -- Called 101 > -- Agent/1001 is ringing > --

> -----Original Message----- > From: Robert Webb [mailto:rwebb@ropeguru.com] > Sent: Saturday, March 05, 2005 5:24 PM > To: 'Asterisk Users Mailing List - Non-Commercial > Discussion'; 'leandro_tenorio' > Subject: RE: [Asterisk-Users] IAX2 (Variables) > > > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com

Hi all, I?m using * with a Suse 8.1 (kernel 2.4.19-4GB). My hardware are a Eicon Diva 2.01 PCI and a Eicon Diva Server 4BRI; I have installed the drivers wich came with the distribution using the distro installer (yast2). I?m experiencing problems with outgoing calls so I?ll try with chan_capi instead of chan_modem. The point is that I?m having problems with the compilation:

Hi, Since Carl has kindly provided us with fax support for CAPI based cards, we have been using it with much success. Today I have modified app_capiFax so that it now supports a dynamic CSID. The following example uses the DNID created by chan_capi on an AVM Fritz! card. * Receive a fax with CAPI API. * Usage : capiAnswerFax2(path_output_file.SFF|stationID) * * This function can be

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Hi. I've discovered a bug in the Dial() string processing (for Asterisk 13.14.1 at least). According to the documentation in channels/chan_sip.c the Dial() string syntax is: * SIP/devicename * or SIP/username at domain (SIP uri) * or SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] * or SIP/devicename/extension * or SIP/devicename/extension/IPorHost * or