search for: huntgroups

I'm developing asterisk to work in a small Call Center for a Mobile Communications Provider. I'm looking for references for Agent Logins and Huntgroups. If any of you have already configured Asterisk for such a Task and would be willing to let me take a look at your config files could you please email them to me at mailto:jhelmich@bluesky.as Thanks, Jason Helmich MIS, Blue Sky Communications PGP Key ID: 0x4CF71E92 jhelmich@bluesky.as 011.684....

...Hangup > exten => s,104,Voicemail(b810920) > exten => s,105,Hangup > > > Trouble is, this appears to the office and living room phones as two > separate calls, so, if I miss this call, I have "2 missed calls" in my > display. > Is there a way to make real huntgroups where I can say "first this and > this, after 15 secs that also"? > > Regards, > Stefan > I've worked on this off and on. I've been thinking about writing a patch to 'Dial/RetryDial' (why isn't retry just a flag for the Dial app?) that would do...

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

...) exten => _*1XXX,11,SetVar(LOOPER=$[${LOOPER} + 1]) exten => _*1XXX,12,Goto(9) exten => _*1XXX,13,Goto(4) exten => _*1XXX,14,Hangup exten => 1,1,VoiceMail(b${CALLEDEXTEN}) exten => 1,2,Hangup exten => 2,1,SetCallerID("Camped on ${CALLEDEXTEN}") exten => 2,2,Goto(huntgroups,101,1) exten => 2,3,Hangup [mainmenu] exten => s,1,Set(LOOPER=1) exten => s,2,ResponseTimeout(6) exten => s,3,Background(entagroup/mainmenu) exten => s,4,Background(entagroup/silence) exten => s,5,Set(LOOPER=$[${LOOPER} + 1]) exten => s,6,GotoIf($[${LOOPER} < 4]?mainmenu,...

I have about 5 incoming USA SIP lines, but my provider does not have any sort of roll-over or huntgroup feature. Does anybody have an idea on how I can create a general number that will ring to the next available, non-busy SIP line that I have? Is there a provider out there that would do this? Any suggestions would be greatly welcome. Thank you. -------------- next part --------------

Dear All I want to integrate sugarcrm and asterisk , so when customer call the call center the agent or extension which answers the call , before pickup the phone and talk to customer , view his/her information if it is available. I do this as described below : 1-Setup login username for sugarcrm for each extension 2-Extension Users will login to the sugarcrm. 3-Develop php script to handle new

Hi all, I regret that I don't know much about telephony as I'm a networking bod, but here goes... We are thinking about implementing a VoIP service so that staff and students can make VoIP calls from home or using our wireless LAN on campus. Clearly, we would like it to integrate with our PBX so VoIP users can talk to the PSTN as well. We don't actually control the University's

Hello all, Quick description of scenario: Would like to be able to plug in an analog line to a Digium Wildcard X100P FXO card in an * server. Whenever that line receives a call, we would like it to automatically ring a hunt group of VoIP soft phones that are connected to the * server. Is this possible, and if it is, what's the easiest way to go about it? My thanks in advance. Gordon

...aded and initialized type leap gtc: challenge = "Password: " gtc: auth_type = "PAP" rlm_eap: Loaded and initialized type gtc mschapv2: with_ntdomain_hack = no rlm_eap: Loaded and initialized type mschapv2 Module: Instantiated eap (eap) Module: Loaded preprocess preprocess: huntgroups = "/etc/raddb/huntgroups" preprocess: hints = "/etc/raddb/hints" preprocess: with_ascend_hack = no preprocess: ascend_channels_per_line = 23 preprocess: with_ntdomain_hack = no preprocess: with_specialix_jetstream_hack = no preprocess: with_cisco_vsa_hack = no Module: Inst...

I'm having problems with my dial plan. I have the following in my extentions.conf: [incoming] ; ; Incoming calls via the PSTN land here ; exten => s,1,Answer exten => s,2,DigitTimeout(5) exten => s,3,ResponseTimeout(10) exten => s,4,BackGround(dial-exten) exten => s,5,Wait(15) exten => s,6,Congestion exten => s,7,Wait(10) exten => s,8,Hangup ; ; Let incoming calls

Nuthin beats an Atlas: http://www.adtran.com/adtranpx/Doc/0/TUA2HMOPDK3KN6S9LM1FH91169/61200305L2-8 A.pdf Telephony Swiss army knife. You can make it do anything. Be prepared to crap your pants when you see the price, though. -----Original Message----- From: Time Bandit [mailto:timebandit001@gmail.com] Sent: Friday, May 26, 2006 9:23 AM To: Asterisk Users Mailing List - Non-Commercial

Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN <--> Asterisk <--> E1 cable <--> Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I