similar to: determining what number was dialed?

In my * server I have a T100P, connected with a T1 crossover cable to an Adtran TA750 with 2 FXO cards. After I do 'modprobe wct1xxp;ztcfg' the alarm on the card starts blinking red. Running zttool confirmed that yes, it is a red alarm. I've read a few previous posts on this type of setup and I'm not too much closer. What should I be checking next? A different cable? The

I've got a need to do something like the following: [foo-context] exten => _.,1,SetCIDNum(123) exten => _.,2,SetCIDName(XYZ) include => local include => tollfree But of course, this example won't work. The goal here is this: if a call ends up being handled by the "local" or "tollfree" contexts, I want those SetCID*** commands executed. Otherwise, I

The CDR, both the csv file and in MySQL does not contain the dialed number (src) in case of a call placed using .call files. "","","302","extensions","","Zap/4-1","SIP/7777-82c8","Dial","SIP/7777|40|whtWHT","2006-12-29 15:48:56","2006-12-29 15:48:59","2006- 12-29

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 145 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040818/d0d64775/attachment.gif

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 145 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040923/d37c237a/attachment.gif

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 145 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040922/ae05773c/attachment.gif

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 145 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050623/3460cdea/attachment.gif

Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul pmahler@signate.com

I've put a t100p in our * server and I'm having trouble configuring it. It is directly connected to an Adtran TA 750 channel bank with two FXO cards (8 analog incoming lines total). I'm able to insmod and modprobe both zaptel and wct1xxp with no trouble, but when I start * with /usb/sbin/asterisk -c I get the following output: [root@rosella root]# /usr/sbin/asterisk -c Asterisk

Hi, I've just initiated a new bounty for the above; http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL Any takers or any contributors please respond to me privately. I do not know exactly how the bounty process works, but I can coordinate on this ? SW -------------- next part -------------- An HTML attachment was scrubbed... URL:

Just a little pun there! I've been mostly lurking for a couple of weeks and realize how little I know and understand about this PBX and phone stuff. I did a little looking about and came across a glossary but they terms are -- for me -- kind of out of context. I'm wondering if there is (much as I hate the term "Dummy") a "PBX for Dummy's" or similar. I've

Hi, I'm trying to make an apparently simple thing work, but I don't see how it is possible with Asterisk. This is my extensions.conf: exten => 1234,1,Dial(SIP/123456/555-555-5555|20|D(7777)) ;After call connects, send DTMF 7777 exten => 1234,2,VoiceMail(1234@context); What I obviously want is that if nobody answer the call, go to voicemail. Basic stuff. Problem is Asterisk

Hi Hekuran I have installed asterisk-1.0-RC2.tar.gz, asterisk-prepaid-0.3.1 and postgresql. When I tried to call from any IAX client to another IAX client and also sip client to sip client it worked fine. And also the cdr table filled properly. Now I tried to configure asterisk-prepaid-0.3.1 with asterisk. I have compiled asterisk-prepaid-0.3.1 and also copy the configure file. I

Hi, Both under the current R-devel (r66456) and a version from about 3 months ago, I experience the following behavior: > parallel::detectCores(TRUE) Error in system(cmd, TRUE) : error in running command > traceback() 3: system(cmd, TRUE) 2: gsub("^ +", "", system(cmd, TRUE)[1]) 1: parallel::detectCores(TRUE) > This is on Ubuntu 14.04. Does anybody else see this? [I

Hi, Thank you for your feedback, Andrew. Since Samba is not the only application making use of the TLS_CIPHER_SUITE negotiation rules in ldap.conf, I would like to ensure that all of them still use the highest encryption possible. Currently I had to remove "TLS_CIPHER_SUITE" as a workarrou d in order to let Samba work wirh LDAP in TLS mode. Does anyone have a suggestion how I can apply

Has anyone gotten CID from Bell Canada to work properly with *? We have our * box down at our datacentre in St Louis, and whenever we call it from a Bell Canada Telephone line, all we see is '' for the CID. I did some digging on google and the mailing lists and couldn't find anything pertaining directly to Bell-Canada and * CID, but didn't find much. I did however find :

Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. Here's a link to the product: http://www.linksys.com/products/product.asp?prid=652&scid=29 -Ron -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, We've released another update to our Asterisk GUI Client suite: http://astguiclient.sf.net/ Screen shots: http://astguiclient.sourceforge.net/screenshots.html The client suite runs on both Windows and UNIX and includes a dialer (the suite is not an asterisk configuration tool) In addition to the usual bug fixes, this is mostly an update for the VICIDIAL dialer application.

Hi, When I setup a user, I give them an extension like 570xxxxxxx. This is fine and dandy while in one area code, but we've since gone to other area codes. I'd like the user's to retain the ability to dial 7 digits no matter what number they have. Any thoughts on how to do that? EXAMPLE: User has number 7175551212. I want that when they dial 3235555 it dials 717-323-5555.

Hello, I have a CAC Channel Bank I with FXS cards. I've the system up and running, with just 1 issue. When I make an inbound call, Asterisk says "Zap/26 is ringing", however, the phone never rings. No lights are lit on the CAC during the calll. Outbound call works no problem, and the CAC lights up correctly. Any ideas what could be the problem? -- Richard Cook