search for: _8xxx

Hello, I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI as channels. A call comes in via IAX2 and should be redirected to CAPI. So I wrote the following dialplan: [fromiax] exten => _8XXX,1,Answer exten => _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r) [fromcapi] exten => 265,1,Answer exten => 265,2,Dial(IAX2/PoC/11@from-lw) exten => 265-BUSY,1,Busy exten => 265-NOANSWER,1,Busy [default] exten => s,1,Answer exten => s,2,Congestion The asterisk on the described side i...

I have my * and polycom system setup to do Auto-Answer for internal SIP/Staff calls, and I am running into an issue with this and the polycom call transfer feature. * is seeing a new call come through from the polycom and is then transferring the call over. I need to know if there is some way I can grab a message from the SIP header or something to determine if I should not set the ALERT_INFO tag

Hello, with asterisk 1.6 i am trying to make a dialplan Which i have such entry in extensions.conf exten => _8XXX,1,Dial(SIP/${EXTEN}) But some of my clients have both IAX and SIP accounts, to use iax clients while outside of my Local Area, and SIP clients (or hardware phones) in local area. But with such rule, i can only dial SIP accounts. Is there a parameter to find how the user connected? such as exten =...

Hi list, I set up two asterisk servers , 1001 is the first asterisk server's sip user, and 2001 is the second asterisk server's sip user. Each of them work well, but I don't konw how to connect them. I want to let the user in 1th Asterisk can call the user in 2nd Asterisk. First asterisk server ip : 192.168.3.101 Second asterisk server ip : 192.168.3.102 can someone

...:18". In features.conf, I have: [featuremap] blindxfer => #1 ; Blind transfer disconnect => *0 ; Disconnect ;automon => *1 ; One Touch Record atxfer => *2 ; Attended transfer My extensions.conf is like this: exten => _8XXX,1,Dial(SIP/${EXTEN},30,Ttm) Another problem is, when caller started the transfer, no dial tone is given. The log said "NOTICE[11245]: app.c:67 ast_app_dtget: Huh....? no dial for indications?". Anybody has the same problem as I do? BTW, can I have more precise control of transfer beha...

...here 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 -e 's/://'`) echo ${array[0]} = ${array[1]} >>$LOG_FILE export ${array[0]}=${array[1]} done echo &...

Hi, Have looked around for info about this: <http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the extension), where can we put something so that "102*" goes straight to voicemail without waiting while the

...EXTEN:1}]?5:4) exten => _8X,4,MeetMe(${EXTEN}|sM) exten => _8X,5,MeetMe(${EXTEN}|asM) exten => _8XX,1,Answer exten => _8XX,2,Wait(1) exten => _8XX,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5:4) exten => _8XX,4,MeetMe(${EXTEN}|sM) exten => _8XX,5,MeetMe(${EXTEN}|asM) exten => _8XXX,1,Answer exten => _8XXX,2,Wait(1) exten => _8XXX,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5:4) exten => _8XXX,4,MeetMe(${EXTEN}|sM) exten => _8XXX,5,MeetMe(${EXTEN}|asM) exten => _8XXXX,1,Answer exten => _8XXXX,2,Wait(1) exten => _8XXXX,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5...

...UMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:3},60,r) exten => _393.,3,Congestion ; Local echo test exten => 611,1,Answer() exten => 611,2,PlayBack(demo-echotest) exten => 611,3,Echo() exten => 611,4,PlayBack(demo-echodone) exten => 611,5,Hangup() ; Manage Voicemail exten => _8XXX,1,Answer() exten => _8XXX,2,VoiceMailMain(${EXTEN:1}) ; Outbound via PSTN [outbound-local] exten => _9XXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9XXXXXX,2,Congestion() exten => _9XXXXXX,102,Congestion() exten => 999,1,Dial(${OUTBOUNDTRUNK}/999) exten => 9999,1,Dial(${O...

...EXTEN:1}]?5:4) exten => _8X,4,MeetMe(${EXTEN}|sM) exten => _8X,5,MeetMe(${EXTEN}|asM) exten => _8XX,1,Answer exten => _8XX,2,Wait(1) exten => _8XX,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5:4) exten => _8XX,4,MeetMe(${EXTEN}|sM) exten => _8XX,5,MeetMe(${EXTEN}|asM) exten => _8XXX,1,Answer exten => _8XXX,2,Wait(1) exten => _8XXX,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5:4) exten => _8XXX,4,MeetMe(${EXTEN}|sM) exten => _8XXX,5,MeetMe(${EXTEN}|asM) exten => _8XXXX,1,Answer exten => _8XXXX,2,Wait(1) exten => _8XXXX,3,GotoIf($[${CALLERIDNUM} = ${EXTEN:1}]?5...

As usual, this is most likely a easy question, but here it goes any way: How can I change the caller id on a transferred call so the called party knows the call has been transferred from a colleague and it's not coming directly from our outside lines? The story goes like this: 1) Client calls. All phones ring. 2) Someone picks up the phone. 3) The phone gets transferred to someone. 4) The

...using IAX? I have two Asterisk boxes behind a NAT and one of them use the default port 5036 for IAX, the second one use 5038. To call an extension of the first one, the line in extensions.conf is: exten => _9XXX,1,Dial(IAX/user:pass@195.3.32.191/${EXTEN:1}) and for the second one: exten => _8XXX,1,Dial(IAX/user:pass@195.3.32.191:5038/${EXTEN:1}) The first one works great, the second one give me this error in the Asterisk console: -- Executing Dial("SIP/351-24f4", "IAX/user:pass@195.3.32.191:5038/500") in new stack WARNING[1200825920]: File chan_iax.c, Line 1550 (...

Hi, I'm back with my question, maybe someone can help me: I want to use IAX on another UDP port (not the default 5036), because I have 2 Asterisks behind the same NAT. Changing the default port in iax.conf file from 5036 to 5038 and then calling using the syntax: exten => _8XXX,1,Dial(IAX/user:pass@195.3.32.191:5038/${EXTEN:1}) I get the follwing error in the Asterisk console: -- Executing Dial("SIP/351-24f4", "IAX/user:pass@195.3.32.191:5038/500") in new stack WARNING[1200825920]: File chan_iax.c, Line 1550 (create_addr): No such host: 195.3.32....

Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first rings at my phone and in the second i can see the callers number. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib

...But not on the same time. This is how my extensions.ael look like now: context internal { 100 => Playback(tt-monkeys); 101 => Dial(SIP/cgm); 102 => Dial(SIP/bluecommand); _9XX => ?????????? 500 => Agi(agi://localhost/internal.agi); _8XXX => { NoOp("Calling ${EXTEN}"); Dial(SIP/${EXTEN}); } }; context database { eswitches { Realtime/ContextShouldBeHere@${CURSERVER}; }; } I want it to jump from _9XX to the context database, passing...

...vpb.conf [interfaces] echocancel=on board=1 context=internal mode=fxo channel=1 channel=2 channel=3 channel=4 channel=5 channel=6 channel=7 channel=8 mode=dialtone channel=9 channel=10 channel=11 channel=12 extensions.conf [default] include =>internal include=>incoming [internal] exten =>_8XXX,1,Dial,(vpb/1-4/{EXTEN}/BYEXTENSION,20); the fxo port exten => 9120,1,Dial(vpb/1-9/{EXTEN:1}/BYEXTENSION,20);the dialtone port exten => 9121,1,Dial(vpb/1-10/{EXTEN:1}/BYEXTENSION,20) exten => 9119,1,Dial,vpb/1-12/{EXTEN:1}/BYEXTENSION,20 exten => 9123,1,Dial,vpb/1-11/{VPB},20 [incoming...

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

...33 sent. Dec 13 13:13:05 DEBUG[-1285764176]: chan_oh323.c:3111 setup_h323_connection: Setting channel's native format to ALAW! any ideas? Thanx extensions.conf exten => 2005,1,NoOp( call for ${EXTEN}) exten => 2005,1,Dial(SIP/${EXTEN},60,tr) exten => 2005,3,Congestion exten => _8XXX,1,Dial,OH323/h323:${EXTEN}@192.168.204.130,tr [voip-h323] include => default __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment...

Hi, I wonder if you guys can recomend a good SIP phone. A phone thats works great with * has a lot of features, and is cheap. Actually all kind pf VoIP hardware is of interesst. Is there a really good site for VoIP harware ? /Mike