similar to: Asterisk does not bridge zap channels on outgoing calls

Hi I am trying to use SetCDRUserField in an agi script but with no success. I am using the CDR mysql addon, however I can't see it being at fault as my attempt is not doing anything to the CVS CD either. has anyone used this, any hints guidence would be greatly appreciated. The syntax I am using is like so .. res=DoExec('SetCDRUserField','12345'); and then dialing the

I have A@H 2.1, running * 1.2.1. I am trying to put information into the userfield with SetCDRUserField and AppendCDRUserField. However, the field is never populated in the cdr - I've checked the csv files and the MySQL asteriskcdrdb table. The field is defined in the MySQL table, but is always empty. The csv files that get created don't have a userfield at all, that is, there

We are using DISA with local SIP users. The user enters in a 2 digit code then they get a dialtone and the phone dials out. The problem is that the calls waits 10 seconds after the outgoing number is dialed, no matter what I put for the timeout values. Anyone else using DISA that has run into this? exten => _2X,1,Answer exten => _2X,2,DigitTimeout(2) exten =>

Question: I'm trying to put a double quote into the CDRUserField. What I end up with is a pair of double quotes. Example: exten => s,n,SetCDRUserField(data) exten => s,n,AppendCDRUserField(") exten => s,n,AppendCDRUserField(moredata) My record will look like this: data""moredata What I want is: data"moredata The wiki mentions using a backslash in order to

I would like to send the CDR records from all our machines around the world to a single database. But I need the hostname included with each record for monitoring purposes. Is there a better way than using the userfield and adding SetCDRUserfield for every call to set the userfield to the name of the host? Thanks...

Hi all, I'm trying to add custom billing info into the CDR records. I did a SetCDRUserField from my agi. Asterisk seems to acknowledge the call, but the value is not anywhere in the CDR record. I checked the CSV and Mysql CDR table. The field is always blank. Anyone had any success with this? Cheers and TIA.

Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing

Hi, I write the CDR of my Asterisk 1.2.17 server in MySQL database using cdr_addon_mysql.so. Now I'm trying to write some custom values to userfield column by the SET(CDR(USERFILED)=SOME_TEXT) sintax, but nothing gets writeen in MySQL cdr table!! Why? I'm I skeeping something or what? Taking a look at the URL:

Paul, Thanks - it worked! For the record, this is exactly what I did: cd/usr/src/asterisk/cdr grep -in "userfield" cdr_csv.c (to find the line that had "#define CSV_LOGUSERFIELD 1" commented out) Opened cdr_csv.c and removed the /* and */ comment marks Saved & exited After shutting down * I went to /usr/src/asterisk and did the usual: Make clean Make Make install I also

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SetCDRUserField doesn't appear to be working in the 5/25/05 CVS - Anyone else having this problem?

Hello, I have an Asterisk box with a TE410P connected to a PRI line and agents with X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I make outbound calls it hangs up as soon as other party tries to picks up the call. Does someone ever experienced this situation? On X-Lite, only G711-ulaw is enabled and here is what i put in sip.conf: [4001] type=friend username=4001

Hi all, I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P (EuroISDN cpe) connected to another similar asterisk box B acting as EuroISDN master. I'm performing some load tests by contiously feeding up to concurrent 30 call files to /var/spool/asterisk/outgoing/ on box A which inititate via a dialplan context/extension a outbound call (redirected via chan_local) to

Ok... I am reluctant to ask this question as I believe that it may be like asking what someones favorite linux distribution is... but I need to make an informed decision. We are getting ready to upgrade from a TE210P to a quad T1 card with echo cancellation. I am trying to decide between the Sangoma card and the Digium card. I need this to have great quality and I need it to work well. I would

Gang, I'm having this error pop up when I do a ForkCDR, and I'm not sure how to get around it. Here are a few log lines: Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing ForkCDR("Zap/49-1", "") in new stack Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a CDR The scenario occurs like this: I use a .call file to generate a call on

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

Hello everybody, I have started using Asterisk in a call center with ACD. I have noticed something and I wonder if anyone knows whether it is a bug or a feature! I am using Queue application to ring a number of agents that have logged on using AgentCallbackLogin. Now, while an agent receives a call from the Queue they cannot logoff using AgentCallbackLogin. Instead the Agent is asked for

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I want to implement a one way announcement and paging facility using Asterisk and Sipura phones. The wiki says Sipura phones only support Auto Answer using the Call-Info header which is no lone shipped with asterisk stable since 1.0.4. I would like to ask if anyone has implemented a similiar facility using Sipura SPA-841 or any other SIP phones. If I could take a look at how