search for: _9xxxxxxxxxx

...extentions,f) exten => 2,1,Directory(voicemail,extentions) exten => 1234,1,Playback(abandon-all-hope) include => extentions exten => i,1,Playback(vm-goodbye) exten => i,2,Hangup() exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup() [outbound] ignorepat => 9 exten => _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9XXXXXXXXXX,2,Congestion() exten => _9XXXXXXXXXX,102,Congestion() exten => _91800NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _91800NXXXXXX,2,Congestion() exten => _91800NXXXXXX,102,Congestion() exten => _91888NXXXXXX,1,Dial(${...

...nd calls [localcalls] ; For local calls ignorepat => 9 exten => _9XXXXXXX.,1,SetCIDNum(9523*2****|a) exten => _9XXXXXXX.,2,Dial(Zap/g1/${EXTEN:1}) exten => _9XXXXXXX.,3,Congestion exten => _9XXXXXXX.,103,Congestion exten => _9XXXXXXX.,104,Hangup exten => _9XXXXXXXXXX.,1,SetCIDNum(9523*2****|a) exten => _9XXXXXXXXXX.,2,Dial(Zap/g1/${EXTEN:1}) exten => _9XXXXXXXXXX.,3,Congestion exten => _9XXXXXXXXXX.,103,Congestion exten => _9XXXXXXXXXX.,104,Hangup exten => _91XXXXXXXXXX.,1,SetCIDNum(9523*2****|a) exten => _91XXXXXXXXXX.,2,Dial(Zap/g1/...

...o specific outbound trunks. I have tried to find appropriate commands in the Asterisk documentation, but could find nothing so far that would help me to find an answer for this problem. Could one potential solution involve the following? (not sure if this is the correct way to do it): exten => _9XXXXXXXXXX,1,Goto(catchallvoip|BYEXTENSION|2)exten => _9XXXXXXXXXX,1,Goto(thisEXTdialsadiffplan|304|1)...where [catchallvoip] and [thisEXTdialsadiffplan] are defined trunks for outbound calling. The above is not meant to be specific to international calling, but instead uses one of two defined trunks for n...

...ckground(our-voicemail-sound) exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup( ) exten => 0,1,Dial(SIP/${ATTENDANT},20) exten => 1,1,Directory(voicemail,internal,f) exten => 2,1,Directory(voicemail,internal) include => extentions [local-trunks] exten => _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9XXXXXXXXXX,2,Congestion( ) exten => _9XXXXXXXXXX,102,Congestion( ) exten => 911,1,Dial(${OUTBOUNDTRUNK}/911) exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911) [local-access] ignorepat => 9 include => local include => lo...

...ten => 866219xxxx,2,Wait,4 exten => 866219xxxx,3,Answer exten => 866219xxxx,4,Authenticate(/etc/asterisk/authenticate.txt|a) exten => 866219xxxx,5,GoTO(s|1) exten => s,1,BackGround(pls-entr-num-uwish2-call) exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten => _9XXXXXXXXXX,1,Playback(pls-wait-connect-call) exten => _9XXXXXXXXXX,2,AbsoluteTimeout(3600) exten => _9XXXXXXXXXX,3,ResetCDR(w) exten => _9XXXXXXXXXX,4,Dial(H323/${EXTEN}@10.10.10.10,90) exten => _9XXXXXXXXXX,5,Congestion exten => _9XXXXXXXXXX,105,Busy 1. Isn't there be a better way to coll...

...ry(voicemail,extentions) include => meetme-ext include => extentions exten => i,1,Playback(pbx-invalid) exten => i,2,Goto(incoming,s,1) exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup() [outbound] ignorepat => 9 include => parkedcalls exten => _9XXXXXXXXXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T) exten => _9XXXXXXXXXX,2,Congestion() exten => _9XXXXXXXXXX,102,Congestion() exten => _91900NXXXXXX,1,Congestion() exten => _91976NXXXXXX,1,Congestion() exten => _91[123456789]XXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/ww${EXTEN:1},30,T) e...

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

...=remote driver=i4l dialtype=tone mode=immediate context=s0bus group=1 ; group=1,2,3,9-12 msn=0 incomingmsn=123456789,123456780 device => /dev/ttyI0 device => /dev/ttyI1 (as found on another post to the list) In extensions.conf I have: [globals] TRUNK=Modem/ttyI0 [trunk] xten => _9XXXXXXXXXX,1,Dial(${TRUNK}/${EXTEN:1}||Ttm) exten => _9XXXXXXXXXX,2,Congestion [sip] exten => 7201,1,Dial(SIP/phone1,20,Ttr) exten => 7205,1,Dial(SIP/phone2,20,Ttr) exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr) [s0bus] exten => s,1,Wait,1; exten => s,2,Answer exten => s,3,DigitT...

...cal Response) Feb 15 14:10:28 WARNING[25664]: chan_sip.c:1227 retrans_pkt: Maximum retries exceeded on transmission 7E760C00-C443-6205-772B-88682E2484DE-5060@10.1.1.40 for seqno 1 (Critical Response) -- Hungup 'Zap/1-1' Here's a part of my dialplan for outside calls: exten => _9XXXXXXXXXX,1,Set(CALLERID(all)=<450-655-****>) exten => _9XXXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) And here's a Macro that I use for incoming call for VoiceGenie: [macro-voicegenie] exten => s,1,Answer exten => s,2,SIPAddHeader(X-Asterisk-DID: ${ARG1}) exten => s,3,SIPAddHeader(...

Hi everyone, I am a new one for that lists....actually i have final year project on VOIP & IMS ...so i want to install asterisk on my pc ...IS it possble that ...we can call on small LAN network without buying any card...i will clear my point as that...suppose i have a linux machine on which i want to install asterisk and HOW it will install...and second point is that ..i have 2

...[ContextC] exten => 30,1,Dial(SIP/30,20) exten => 31,1,Dial(SIP/31,20) exten => 32,1,Dial(SIP/32,20) include => outbound [default] exten => _1X,1,GoTo(ContextA,${EXTEN},1) exten => _2X,1,GoTo(ContextA,${EXTEN},1) exten => _3X,1,GoTo(ContextA,${EXTEN},1) [outbound] exten => _9XXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@192.168.1.100) So each user registers and they can call each other and they can dial 9xxxxxxxxxx to dial local and ld. The issue arises when they want/need to call the other companies in the other contexts. I want the call to go direct to the other user instead of out o...

...ault]\par exten => 1000,1,Dial,Zap/1,20\par exten => 1000,2,Voicemail,u1000\par exten => 1000,3,Hangup\par exten => 1000,102,Voicemail,b1000\par exten => 1000,103,Hangup\par exten => _9XXXXXXXX,1,Dial(Zap/4/$\{EXTEN:1\})\par exten => _9XXXXXXXX,2,Congestion\par \par exten => _9XXXXXXXXXX,1,Dial(Zap/4/$\{EXTEN:1\})\par exten => _9XXXXXXXXXX,2,Congestion\par ; Extension 2000 Sipura line 1\par exten => 2000,1,Dial,sip/spa2000|30|t\par exten => 2000,2,Voicemail,u2000\par ;Extension try for X-Lite\par exten => 12345,1,Dial(Zap/4/$\{EXTEN:1\})\par exten => 12345,2,Congesti...

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can make call outside and exten 2006 to 2010 can not make call outside. heres my dial plan. ? sip.conf ? [general] port = 5060 bindaddr = 0.0.0.0 context = others [2000] type=friend context=outside secret=1234 host=dynamic [2001] type=friend context=outside secret=1234 host=dynamic [2002] type=friend context=outside

...faxdetect=both faxdetect=incoming faxdetect=outgoing faxdetect=no context=default ; Points to the default context of your extensions.conf channel => 1-15,17-31,32-46,48-62; for E1 i've configured the outgoing calls [outrt-001-9_outside] include => outrt-001-9_outside-custom exten => _9XXXXXXXXXX,1,Macro(dialout-trunk,1,${EXTEN:1},) exten => _9XXXXXXXXXX,2,Macro(outisbusy) ; No available circuits if i try to call i get: Feb 13 06:19:44 DEBUG[3637] chan_sip.c: Setting NAT on RTP to 0 Feb 13 06:19:44 DEBUG[3637] chan_sip.c: Stopping retransmission on '6EE9B5C0-2043-9AC4-9F18-76F84A25...

...faxdetect=both faxdetect=incoming faxdetect=outgoing faxdetect=no context=default ; Points to the default context of your extensions.conf channel => 1-15,17-31,32-46,48-62; for E1 i've configured the outgoing calls [outrt-001-9_outside] include => outrt-001-9_outside-custom exten => _9XXXXXXXXXX,1,Macro(dialout-trunk,1,${EXTEN:1},) exten => _9XXXXXXXXXX,2,Macro(outisbusy) ; No available circuits if i try to call i get: Feb 13 06:19:44 DEBUG[3637] chan_sip.c: Setting NAT on RTP to 0 Feb 13 06:19:44 DEBUG[3637] chan_sip.c: Stopping retransmission on '6EE9B5C0-2043-9AC4-9F18-76F84A25...

I have a question on Asterisk and whether it will work with the following design. Install ASTERISK on the external side of the Network. Purchase an AudioCodes 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here is the twist. The company currently has Cisco Call Manager 3.3 which does not support SIP Trunking. But it does have a VG248. I would like to place 4 lines

...is possible. I have test it with Asterisk and oh323. We have routed some calls thru a second h323 gateway (like Vegastream and Cirilium). Following is the configuration: ; Vegastream ------------ exten => _01XXXXXXXXXX,1,Dial(OH323/BYEXTENSION@xxx.xxx.xxx.xxx) ; Crilium --------- exten => _9XXXXXXXXXX,1,Dial(OH323/BYEXTENSION@xxx.xxx.xxx.xxx) Shimul > > > > --__--__-- > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Aste...

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client