search for: somecontext

Hello list, i have one question related to call forwarding. i have 2 number for the inbound and i want to configure asterisk like that. When the customer call the first number 0522XXXXXX the call will be forwarding automatically to anther number 0520xxxxxx Does anybody have a solution to this problem. Thanks and Regards. -------------- next part -------------- An HTML attachment was

...n and not match immediately. No luck though - dialing still starts immediately when one digit past 011 is received. Any thoughts on how to get around this? Right now the best I have (and that's not saying much) is to have something like: [initialcontext] exten => _9011,1,DISA(no-password|somecontext) [somecontext] exten => _X.,1,Dial(Zap/R1/011${EXTEN}) But that's ugly, not to mention confusing to the users because the amplitude of the dialtone generated by the GXP is lower than the dialtone generated by *, so they notice the bump when they've dialed 9011. Any suggestions appreci...

Hi list I'm trying to implement sourcerouting on a distributed installation, but I can't get contexts to work right. My goal is to do a Dial(whatever@somecontext) and vary the somecontext based on different criteria. This is going on over trunked IAX2 links. How do I set up my IAX-accounts to manage this? I have tried to play around with 'context' and 'peercontext' on the server being dialed, but no luck. Is it legal to have multiple 'c...

Hi I have noticed new behaviour of asterisk 10.0 realtime. In 1.6 when I was using realtime: """ [somecontext] exten => someexten1...... exten => someexten2...... exten => someexten3...... exten => someexten4...... switch => Realtime/${CONTEXT}@extensions """ switch statement was executed after lines above (so there was a precedence of the lines declared in a extensions...

Hi, With Asterisk 1.8 is it now possible to register the same SIP account at multiple endpoints and for both to ring when the associated extension is dialed ? -- Thanks, Phil -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110309/fe9d7bc7/attachment.htm>

...call: 1183671934|1183671934.5745|emerg_nccc_ld_ts|NONE|ENTERQUEUE||427 1183671937|NONE|NONE|Local/491 at agents/n|PAUSEALL| 1183671940|1183671934.5745|emerg_nccc_ld_ts|Local/491 at agents |CONNECT|6|1183671934.5746 1183672005|1183671934.5745|emerg_nccc_ld_ts|Local/491 at agents |TRANSFER|7777|from-somecontext|6|65 If I use /n when adding the channel to the queue: AddQueueMember(queuename,Local/XXXX at agents/n) Then my 'h' extension is not executed until the bridged call is actually over. I do the same transfer, but it doesn't show up in the queue log - the call appears to have been term...

...3 Application: GotoIf AppData: 0?blockclid Uniqueid: 1183582822.104759 |CONNECT|12|1183582816.104754 1183582833|1183582485.104605|queuename|SIP/XXXX|AGENTATTEMPT| ---END--- --START-- 1183762515|1183762515.18034|queuename|SIP/XXXX|REMOVEMEMBER| 1183762518|1183762485.18025|queuename| Context: from-somecontext Extension: XXXX Priority: 6 Application: Hangup AppData: Uniqueid: 1183762515.18034 |CONNECT|5|1183762513.18033 ---END--- --START-- 1183762659|1183762631.18061|queuename|NONE|EXITWITHTIMEOUT|1 1183762661|1183762485.18025|queuename|: macro-singlequeue Extension: s-TIMEOUT Priority: 1 Application:...

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, I got my * working fine with FWD at office with 2 extensions, i receive calls and i can make calls thru FWD. I got also my * at home, and i connected it using auth=rsa. From my home, i can make calls using my office iax, but if i try to redirect incomming calls from FWD to my * at home, it rejects the call. I created the pub/key pairs for rsa and its working ok and i just pasted the

Hi All, Anyone know if theres a way to share a Cisco 7940 between hot-desk users? My phones get their setup via SIP .cnf files, that load at boot via tftp, so I'm assuming the configs a failry static. However if I want a phone to be hot-desked, I could have different users sitting there. Is there any concept of "logging on" in these environments? Cheers, Adrian

I'm trying to clarify contexts and their uses. I do have a good general understanding of them. My question is about "undeclared" and "non-existant" contexts. If I have a block somewhere (in sip.conf, for example), and it has no "context=thiscontext" field, does it just automatically use the "default" context? Or is this settable? (I see there is an

I am wanting to front-end a legacy PBX with an asterisk box. I have done plenty of asterisk work over the last 6 months to PRI circuits, but not with a PBX being involved. I know I can use asterisk and digium cards in this manner, but do I need separate cards for the PRI -> Asterisk side to the Asterisk -> PBX side, or will a 4-port PRI card do the job? (I already have a spare one of