search for: _10xx

Displaying 8 results from an estimated 8 matches for "_10xx".

2006 Mar 03
4
really need help with outgoing calls..PSTN errors
...it is supposed to be. I am thinking it is a problem between the zap interface and the PSTN. thanks extensions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] ATTENDANT=1001 OUTBOUNDTRUNK=ZAP/g1 [extentions] exten => _10XX,1,Ringing exten => _10XX,2,Dial(SIP/${EXTEN},20) exten => _10XX,3,Answer exten => _10XX,4,VoiceMail(u${EXTEN}@voicemail) exten => _10XX,5,Hangup [voicemail] exten => _910XX,1,Wait(1) exten => _910XX,2,VoiceMailMain(${EXTEN:1}@voicemail) [local] include => extentions include...
2006 Jan 31
3
ZAP <--> sip(polycom301) can not hear each other
...=password type=friend host=dynamic canreinvite=yes disallow=all allow=all extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] ATTENDANT=1001 OUTBOUNDTRUNK=ZAP/g1 [extentions] exten => _10XX,1,Ringing exten => _10XX,2,Dial(SIP/${EXTEN},20) exten => _10XX,3,Answer exten => _10XX,4,VoiceMail(u${EXTEN}@voicemail) exten => _10XX,5,Hangup [voicemail] exten => _910XX,1,Wait(1) exten => _910XX,2,VoiceMailMain(${EXTEN:1}@voicemail) [local] include =&gt...
2006 Jun 22
4
when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on
...priorityjumping=no [globals] ATTENDANT=SIP/1006&SIP/1002&SIP/1011&SIP/1009 OUTBOUNDTRUNK=ZAP/g1 [meetme-ext] exten => 600,1,MeetMe(1234|Mp|98765) [extentions] include => parkedcalls include => meetme-ext include => direct-to-voicemail exten => _10XX,1,Dial(SIP/${EXTEN},20,t) exten => _10XX,n,Answer exten => _10XX,n,VoiceMail(u${EXTEN}@voicemail) exten => _10XX,n,Hangup() [voicemail] exten => _910XX,1,Wait(1) exten => _910XX,n,VoiceMailMain(${EXTEN:1}@voicemail) [direct-to-voicemail] exten => _810XX,1,Voi...
2006 Mar 13
1
Scrolling messages
Several times a day I get this meesage scrolling on one of our asterisk boxes: Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our native format has changed to alaw Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our
2009 May 20
1
Queue and Dial operation - Common Variables?
...#39;t really find a way to relate the pair. I need to perform some DB operations for agentlogin instance and dial instance. Is there a variable that is common for both instance or is there a way that I can pass variables across. My context and AGI's are given below. [specqueuestat] exten => _10XX,1,AGI(agi_agentlogin.sh|${EXTEN}) exten => _10XX,2,AgentCallbackLogin(${agentno}||${sip_id}@specqueuestat) exten => _8XXX,1,AGI(agi_qdial.sh|${EXTEN}|${CALLERIDNUM}) --agi_agentlogin.sh *declare -a array while read -e ARG && [ "$ARG" ] ; do array=(` echo $ARG | sed...
2005 Mar 24
2
Fun with CAPI
Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension
2006 Oct 30
1
dealing with blind transfers to invalid extensions
...ng in my dialplan seems to be working well except for one problem. When calls are blind transferred to an invalid extension I would like the call to go to the operator on ext 1000? What is the best way to do this? Thanks in advance Here's a snippet of my extensions.conf [default] exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}) exten=>_11XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}) include=>record include=>parkedcalls include=>voicepulseoutgoing include=>conferences include=>voicemail [macro-stdexten] exten=>s,1,Dial(${ARG2},20,t) exten=>s,2,Goto(s-${DIALSTAT...
2007 Jul 17
2
2 PRI on asterisk
Dear all I am going to install 2 port pri card on asterisk but i dont know how to incomming call goes in to IVR and how to route call outside base on pattern match means if some one call on mobile phone then use PRI 1 and if call on landline phon call route through pri 2 how to make dission base on pattern number Rgds satish patel