similar to: CDR on Transfer...

On 8/27/10, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am trying to set the CDR(userfield) to a certain vaule using the application map of features.conf but I am not able to do it. When I receive a call I would like to tag it with a client code (3 digit numeric) so I can referenci it later from the CDR. I have edited features.conf with something like: code => #111,self,SET(CDR(userfield(111)) or

Is there any way to know if a call was transferred from reading the CDR? Any relation in fields like UNIQUEID? Something that can be scripted to make a special report? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Sometimes I get the following error on the console: [May 31 11:14:01] ERROR[23502]: cdr_addon_mysql.c:230 mysql_log: mysql_cdr: Failed to insert into database: (1062) Duplicate entry '1180628004.3214' for key 1 -- Zap/38-1 answered UniCall/11-1 Why would the UNIQUEID be duplicated? Isn't is supposed to be UNIQUE? This worries me a bit because the customer is losing calls that

I'm trying to use the MySQL CDR records. According to dialplan show, the line in the dialplan is: 11. Set(CDR(userfield)=${billing_code}) [pbx_ael] It looks like the value is being set when I watch the console during the call: -- Executing [s at restphone_event_loop:11] Set("SIP/icall-0075a2e0", "CDR(userfield)=boatmenu") in new stack But the record that's

Folks: does anybody have an idea? how to capture the DTMF digits to a file, after an extn asnwer? then POST it to a url? Regards, JR

Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks

Ia had a server overload today because someone did a call forward to their own extension. To do a call forward I write a key called CFWD with the extensi?n number and number to dial . The main script tests if the key/value exists and dials the number stored in the database. What is an easy way to prevent dumb people from creating a loop? -- Telecomunicaciones Abiertas de M?xico S.A. de

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come over the other lines. -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director

Hi, I have a working asterisk 1.4.0 with Mysql Realtime configuration, and today I encountered this error. Now, I have no acces to any information in mysql realtime, so nothing work now !!!!! [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Everything is fine. [Jan 24 10:32:40] DEBUG[31070] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions

I am having a bit of a problem implementing the pickup command in my dial plan. I have setup this rule: exten => _*8XXX,1,Pickup(${EXTEN:2}) This works as expected when someone dials an extensions number and I can get the call. The problem I have is that when a call enters my welcome menu and does not press anything there is a timeout that sends them to the recepcionist. The rule is:

I am trying to set up Astunicall 1.4.16 with a link from Alestra in Mexico City. I have done everything I usually do for other links in Mexico but this one simply will not send or receive calls. I just get Protocol error. Anyone has any experience with R2 and Alestra? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001

Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not

Guys. Can you dial 800 and 888 toll free numbers using FWD? how do you dial them cause I tried using 1800xxxxx and 1888xxxxx and I simply get a "nobody can asnwer the call" signal on asterisk. Can you dial 800 toll free from FWD?

Hi iam using 1.2.17 does any one have information meetme in realtime and store in mysql i dont see any document could some one help me is this possible ? ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070606/36d236c2/attachment.htm

Using Digium's "Asterisk Developer's Kit (TDM)", I've been trying to make an outside call by copying sample.call to /var/spool/asterisk/outgoing. I want the VoiceMailMain to run when the call is answered. The call is made correctly but, as you probably know, the application starts as soon as the call is made. I see there are two solutions: Using callprogress=yes in

Does anyone know if you can use softphones on thin clients? I have a new customer that wants to use Eyebeam (about 10 users) on a thin client platform. Each user has a little box on their desk that has a USB port, mic and headphone jacks and monitor. I am worried about conflicts when running 10 softphones on the same server since they will all try to use por 5060. -- Telecomunicaciones

I am using Asterisk 1.4.4 on a CentOS 5 machine for a small call center with 6 agents. I am using realtime for queues and sip and I am also trying to use realtime static to load agents.conf. The only problem I am having is that no agents are loaded when I start Asterisk. I have to manually do a "module reload chan_agent.so" so the agents get loaded from the database. Obviously

Hi to everybody, i tried to find an asnwer before posting this, but most astcc billing issues i searched refer to the case when no billing occurs at all. In my case i get only initial charges and any subsequent minute does not count for billing. In my iax.conf i entered the "notransfer = yes" but nothing changed. My test card and test calls are the following TEST-CARD en N/A

This question appears to have been asked previously, but not answered. the last response I can find to this previous thread is here: http://tolstoy.newcastle.edu.au/R/help/04/07/0126.html. The asnwer was to provide debugging info, not an answer. So the problem is that I'm trying to use lda on my dataset. You can download my data here: http://northstar-www.dartmouth.edu/~jgilbert/nolda, I used