search for: _xxxxxxx

Hey, all. I'm trying to set my CID such that, internally, I see a four-digit extension (which is also handy when checking VM), but externally, I see the full 10-digit number. So I plugged these lines into my extensions.conf: exten => _XXXXXXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2) exten => _XXXXXXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1}) exten => _XXXXXXX,3,NoOp(${CALLERIDNUM}) exten => _XXXXXXX,4,Dial(${OUTBOUNDTRUNK}/${EXTEN}) (I wanted to test against my own extension, "1625"; if that worked, I wanted to stri...

I have for each phone such a paragraph in my dialplan. I would like to save this by using a Macro. How can I do that? exten => 8863959,1,Dial(SIP/8863959,60,r) exten => 8863959,2,NoOp(${DIALSTATUS}) exten => 8863959,3,Voicemail,u8863959@Customers exten => 8863959,104,Voicemail,b8863959@Customers exten => 8863959,105,hangup

...and outgoing calls work with Asterisk except the caller ID for the outgoing calls. I think I have exhausted all possible options regarding setting a caller ID and it still doesn't work. The recipients will get "private number". The incomings caller ids are work just fine. exten => _XXXXXXX.,1,Set(DIAL_NUMBER=...) exten => _XXXXXXX.,2,Set(CALLERID(num)=...) exten => _XXXXXXX.,3,Set(CALLERID(name)=...) exten => _XXXXXXX.,4,Set(CALLERID(all)= ?...? <...>) exten => _XXXXXXX.,n,Dial(SIP/${EXTEN}@gsm1) Also there is 1 setting in the Goip device itself to set "SIM Nu...

...sions by the dialplan itself. The problem is that, if I don't declare the incoming extension, Asterisk hangs up the call by default. So I want to know if there's some kind of "ManagerControl() application to do this, so my dial plan will look like: [incoming_extensions] exten => _XXXXXXX,1,ManagerControl(....) Thanks a lot for your help. -- Atly. Alvaro Palma

Hi there, I'm using asterisk meetme function like: exten => 9070,n,MeetMe(|dcM) and everything works pretty well. But I would like to add a review of the entered conference number before the user jumps into the conference. Somthing like: *:"Please enter the conference number followed by the hash key" (works) U: 123456# (works) *: "You are entering conference number

I'm dialing DTMF's in a SIP channel using the options: [sip.conf] dmtfmode=info [extensions.conf] exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN})) (this is a custom SIP gateway, which receives the DTMF's sent from softphones through Asterisk, and based on them, build the destination PSTN number). My problem is that Dial send the DTMF's to the SIP/gateway user at a rate of about 1 DTMF each 300ms....

I have couple of questions about the following. Currently I have 2 phone lines going into my house, and I would like to have both of those coming into asterisk. I also want to have a backup asterisk, so here are the main questions (I am knew to this so I apologize if I ask something stupid): - Is there a dual FXO card available from digium or do I need 2 x single FXO (if this is the case then

...=> 49-72 immediate=yes Extensions.conf: [from-cb] exten => s,1,DISA,no-password|internal [internal] include => sip-stations include => iax-trunks include => outbound [outbound] exten => _1XXXXXXXXXX,1,Dial(DAHDI/g1/${EXTEN}) exten => _XXXXXXXXXX,1,Dial(DAHDI/g1/${EXTEN}) exten => _XXXXXXX,1,Dial(DAHDI/g1/${EXTEN}) When I pickup a line, and hit any key I get: -- Starting simple switch on 'DAHDI/49-1' -- Hungup 'DAHDI/49-1' -- Starting simple switch on 'DAHDI/49-1' -- Hungup 'DAH...

Hello everyone, I'm sure someone had an experience arranging hunt-group setup for incoming calls on T1 PRI channels of Digium TE110P card. For instance, I have main DID channel associated with number (555) 222 0001. And I have whole bunch of other DID channels on same T1 card like (555) 222 0090, (555) 222 0091, (555) 222 0093. My goal is when a call comes to the main number which is

Hi Group, i have a MeetMe Question. I use "MeetMe(,Ms)" in the Dialplan and if a Conference Room does't exist Asterisk play (conf-invalid.slin) If i use "MeetMe(${room},Ms)" (value from DTMF Read) and the Conference Room doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk Hangup the Call. there is a solution for the kind my problem? Thanx and

...et(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)}) exten => 106,n,Monitor(wav,${CALLFILENAME},m) exten => 106,hint,SIP/106 exten => 106,Macro(stdexten,106,${HINT}) When I called x106 this was logged: -- Executing [106 at voicemenu-custom-4:1] Set("DAHDI/7-1", "CALLFILENAME=_xxxxxxx") in new stack -- Executing [106 at voicemenu-custom-4:2] Monitor("DAHDI/7-1", "wav|_xxx-xxx- xxxx|m") in new stack == Auto fallthrough, channel 'DAHDI/7-1' status is 'UNKNOWN' -- Hungup 'DAHDI/7-1' When I don't have the first two line...

...---------------- [osaka] type=friend context=sip-outgoing host=192.168.0.139 disallow=all allow=alaw [6672019] type=friend context=sip-outgoing canreinvite=no host=dynamic nat=no Under extensions.conf : -------------------------------- [sip-outgoing] include=sip_outgoing [sip_outgoing] exten => _XXXXXXX,1,Dial(SIP/osaka/${EXTEN}) [line-incoming] exten => _6XXXXXX,1,Dial(SIP/${EXTEN}) Please be informed that the sip outbound toward the external sip server is quite ok , but sip incoming is not working . Can you please let me know why my incoming route is not working properly ? Thank you --------...

...conf" and "h323.conf" files are: *************************************** extensions.conf >>>> *************************************** [default] ..... ..... [outgoing] exten=>_7XX,1,Goto(voip-h323|${EXTEN}|1) exten=>_*XX,1,Goto(servicios|${EXTEN}|1) exten=>_XXXXXXXXX,1,Dial(Zap/1/${EXTEN}|30) exten=>_XXXXXXXXX,2,Playback(invalid) exten=>_XXXXXXXXX,3,Hungup() exten=>_X,1,Playback(invalid) exten=>_X,2,Hungup exten=>_XX,1,Playback(invalid) exten=>_XX,2,Hungup exten=>_XXXX,1,Playback(invalid) exten=>_XXXX,2,Hungup exten=>_XXXXX,1,Playb...

....1 with Asterisk CVS-1-01/09/05-01:41:37. Our provider uses Mera MVTS softswitch and supports only H.323. We don't use gatekeeper for connection but provider requires SOURCE PHONE NUMBER for route out calls and I don't know how I can specify this number. Call with this string exten => _XXXXXXX,1,Dial,OH323/${EXTEN}@mvts_ip_addr returns -- H.323 call 'ip$localhost/12715' cleared, reason 11 (Gatekeeper could not find user) Please help! How can I supply source phone number for oh323? -- Alexander Averyanov

...st last 2 digits. [invalid] exten => _X.,1,Playback(pbx-invalid) exten => _X.,2,Hangup() [internal] include => invalid exten => _XX,1,Dial(SIP/12345${EXTEN}) ; Short alias for internal lines exten => _12345XX,1,Dial(SIP/${EXTEN}) ; Long alias for internal lines [local] exten => _XXXXXXX,1,Dial(Zap/g1/${EXTEN}) ; Local area calls [full] include => internal include => local (As it's defined, local also include invalid context) My idea is that, if any user authorized to make only internal calls dial any number different than 12345XX, he/she'll receive the invalid m...

...te=no [1111] type=friend username=1111 secret=<secret> host=dynamic context=tutorial nat=never insecure=invite qualify=yes -------------- next part -------------- [general] static=yes writeprotect=no clearglobalvars=no [globals] CONSOLE=Console/dsp [tutorial] exten => _XXXXXXX.,1,Dial(SIP/${EXTEN}@provider,,r) -------------- next part -------------- A non-text attachment was scrubbed... Name: asterisk.logs.example1.txt.bz2 Type: application/x-bzip2 Size: 17606 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070503/78996c74/a...

...Outgoint fax is the problem, when IAXMODEM dial out using Dahdi channel, dahdi answers and start to dial the outside number however Iaxmodem thinks that dahdi is the remote fax machine and starts sending fax data eventually giving up. Here is my dialplan in extensions.conf [fax-out] exten => _XXXXXXX,1,Set(__SIP_CODEC=alaw) exten => _XXXXXXX,2,Dial(dahdi/g3) If the remote machine answers within the first ring, the outgoing fax works fine, but if remote fax machine does not answer within the first ring, I get the fallowing erro: "no carrier found" Hylfafax and Iaxmodem are workin...

Hi all, I have just published my last few weeks of hard work: IPSwitchBoard BETA. Please let me know what you think and post comments on the Wiki. http://www.voip-info.org/wiki-IPSwitchBoard+BETA Thank you

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am trying to set the CDR(userfield) to a certain vaule using the application map of features.conf but I am not able to do it. When I receive a call I would like to tag it with a client code (3 digit numeric) so I can referenci it later from the CDR. I have edited features.conf with something like: code => #111,self,SET(CDR(userfield(111)) or

...0) [default] include => demo exten => 300,1,Macro(stdexten,1234,SIP/300) exten => 301,1,Macro(stdexten,1234,SIP/301) exten => s,1,Dial(SIP/300,10) exten => s,2,Dial(SIP/301,10) exten => s,3,Dial,ZAP/2/XXXXXXX,10 exten => s,4,Voicemail,1234 exten => s,5,Hangup exten => _XXXXXXX,1,Dial,ZAP/1/${EXTEN} exten => _1XXXXXXXXXX,1,Dial,ZAP/2/${EXTEN} Thanks, David -- This message has been scanned for viruses and dangerous content by KE6UPI, and is believed to be clean. KE6UPI thanks MailScanner for their support. Please contact support@ke6upi.com if you have questions ab...