search for: u102

...multiple extension dialing - ie dial 1 number and it rings at a number of sources. For the most part its worked.... Now if someone dials 107 it rings Sip phones at 102 and 107, then goes to voicemail after 40 seconds. exten => 107,1,Dial(SIP/102&SIP/107,40|r) exten => 107,2,Voicemail(u102@pstn) exten => 107,3,Hangup exten => 107,102,Voicemail(b102@pstn) exten => 107,103,Hangup The problem I'm running into is when I add my cell phone in exten => 107,1,Dial(SIP/102&SIP/107&Zap/2/11235551212,40|r) exten => 107,2,Voicemail(u102@pstn) exten => 107,3,Hangup...

...s rec'd to a number, Asterisk needs to dial several SIP extensions at the same time. The SIP extensions are for Cisco 7960s and each have multiple line appearnces. For example, exten => 9043442342,1,DIAL(SIP/102&SIP/103&SIP/104&SIP/105,,20) exten => 9043442342,1,Voicemail(u102) The issue I have is that I need each user of these extensions to have multiple line appearances ("roll over" lines). In a traditional PBX, usually this is accomplished by setting up a roll over lines... i.e my extension is 100, my roll over extension is 200, and next roll over exte...

...again to ring twice, ... If I pickup I do not hear on extension 601, and on the PSTN it is still signaling to ring. Can anybody enlighten me, please? extension.conf [incoming_88097074] exten => s,1,Wait(1) ;wait to get caller ID in. exten => s,2,Dial(SIP/102,20) exten => s,3,Voicemail(u102) exten => s,102,Voicemail(b102) exten => s,103,Hangup *CLI> Starting simple switch on 'Zap/4-1' Executing Wait("Zap/4-1", "1") in new stack Executing Dial("Zap/4-1", "SIP/601|20") in new stack Called 601 SIP/601-06ae is ringing WARNING[19...

...s made, it appears as though the call "completes" so it never rolls to asterisk voicemail. Here is my current config: exten => 102,1,Dial(${sipura},10,) exten => 102,n,playback(pls-wait-connect-call) exten => 102,n,Dial(IAX2/asterisk1/9139275900,10,r) exten => 102,n,VoiceMail(u102@default) exten => 102,107,VoiceMail(b102@default) Here is the log from asterisk: -- Executing [102@internal:2] Playback("SIP/101-0a1178c0", "pls-wait-connect-call") in new stack -- Playing 'pls-wait-connect-call' (language 'en') -- Executing [102@...

...ail(b100) [pbx_config] '101' => 1. Dial(SIP/mthomas|20|Tt) [pbx_config] 2. Voicemail(u101) [pbx_config] 102. Voicemail(b101) [pbx_config] '102' => 1. Dial(SIP/dli|20|Tt) [pbx_config] 2. Voicemail(u102) [pbx_config] 102. Voicemail(b102) [pbx_config] '105' => 1. Dial(SIP/nmartin|20|Tt) [pbx_config] 2. Voicemail(u105) [pbx_config] 102. Voicemail(b105) [pbx_config] '600' => 1. VoiceMailMain() [pbx...

...=> 100,1,Dial,Zap/2|20|m exten => 100,2,Voicemail,u100 exten => 100,102,Voicemail,b100 exten => 101,1,Dial,Zap/2|20|m exten => 101,2,WaitMusicOnHold,15 exten => 101,3,Voicemail,u101 exten => 101,102,Voicemail,b101 exten => 102,1,Dial,Zap/2|20|m exten => 102,2,Voicemail,u102 exten => 102,102,Voicemail,b102 exten => 103,1,Dial,Zap/2|20|m exten => 103,2,Voicemail,u103 exten => 103,102,Voicemail,b103 exten => 8500,1,VoicemailMain exten => i,1,Playback,pbx-invalid exten => t,1,Playback,demo-thanks exten => t,2,Hangup include => extensions [...

...nel module. The smbfs client software is version 2.0.1, and was installed with Redhat 5.0. The problem that I have is when I mount the share, I get _NO_ rights to it at all! This happens regardless of which server I use: the NT server, or the Solaris server: viz: smbmount //solaris/share ~/mnt -u102 -g102 -f777 -d777 gives this listing when you type `ls -la ~/mnt': ls: /home/jtullett/mnt: Permission denied The command `ls -lad ~/mnt' gives the result d--------- 1 jtullett root I _can_ access the shares as I would expect (with read/write permissions) via `smbclient'. S...

Hi all, I have Asterisk running on FreeBSD 4.x and I have made configurations to sip.conf, extensions.conf and voicemail.conf. I have also setup SIP phones on two different PCs. My problem is that when one of the SIP phones logins in, the other won't. My sip.conf has: [101] type=friend host=dynamic username=101 secret=test dtmfmode=rfc2833 context=from-sip mailbox=201

...herinquiries) exten => s,14,Background(press4) exten => s,15,Wait,1 exten => s,16,Background(repeatoptions) exten => s,17,Background(pressstar) exten => s,18,WaitExten(10) exten => t,1,Hangup exten => *,1,Goto(s|7) exten => 102,1,Dial(SIP/femi,15) exten => 102,2,Voicemail(u102 at office) exten => 102,3,Hangup exten => 1,1,Dial(SIP/femi&Gtalk/asterisk/ggggggggggg at gmail.com,15) exten => 1,2,Voicemail(u1 at office) exten => 1,3,Hangup exten => 2,1,Dial(SIP/femi,15) exten => 2,2,Voicemail(u2 at office) exten => 2,3,Hangup exten => 4,1,Dial(SIP/...

..."100" for users to reach the switchboard as they would from outside: [internal-extensions] exten => 100,1,Goto(mainmenu,s,10) exten => 101,1,Dial(SIP/101,30) exten => 101,2,Voicemail(u101) exten => 101,3,Hangup() exten => 102,1,Dial(SIP/102,30) exten => 102,2,Voicemail(u102) exten => 102,3,Hangup() dialing 100 then hits "mainmenu" [mainmenu] exten => s,10,Answer exten => s,11,Wait(1) exten => s,12,Background(buddy-cloud/welcome2) exten => s,13,WaitExten(15) exten => s,14,NoOp(Number dialed ${EXTEN}) include => internal-extensions exte...

Hi all. I am new to the list and new to asterisk. I have asterisk installed and running. I am using it as a voicemail server only. What I would like to do is send users to a general mailbox that will be addressed as <companyname>@asterisk and give them the option to wait for the tone and leave a message, or press 9 to dial by name. My questions are: 1. What is the best way to do

...= 500 cid_name = Cisco1 cid_num = 104 line => 104 extensions.conf [internal] include => outbound-local include => outbound-long-distance ; Software phone exten => 101,1,Dial(SIP/test-softphone,,r) exten => 102,1,Dial(SIP/bob,20) exten => 102,2,Voicemail(u102) exten => 102,102,Voicemail(b102) exten => 102,103,Hangup() exten => 103,1,Dial(SIP/bill,20) exten => 103,2,Voicemail(u103) exten => 103,102,Voicemail(b103) exten => 103,103,Hangup() exten => 104,Dial(SCCP/SEP00036BC3852B,20) exten => 104,2,Voicemail(u104) exten => 104,...

...ntents of SIP.CONF: [102] type=friend username=102 secret=chandra host=dynamic context=tutorial [109] type=friend username=109 secret=ravi host=dynamic context=tutorial Contents of EXTENSIONS.CONF: [tutorial] exten => 102,1,Dial(SIP/102,10) exten => 102,2,Voicemail(u102) exten => 102,3,Voicemail(b102) exten => 102,4,Hangup exten => 109,1,Dial(SIP/109,10) exten => 109,2,Voicemail(u109) exten => 109,3,Voicemail(b109) exten => 109,4,Hangup Contents of VOICEMAIL.CONF: [default] 102 => chandra,Chandramouli,chandra@xyz.com,chandra@xyz...

Folks, Apologies ahead of time if this has already been asked (read the list for the last month looking for something similar). I have been trying to get the Background command to work with no joy yet. Here is what I am trying to do: 1. Answer the call. 2. Play the message in the background, while waiting on DTMF from user. 3. If I get a "1", then interrupt the message and dial the

...et=test1 dtmfmode=rfc2833 context=from-sip mailbox=202 callerid="102" <2135> nat=yes My extensions.conf has: exten => 101,1,Dial(SIP/101,20,tr) exten => 101,2,VoiceMail,u101 exten => 101,102,VoiceMail,b101 exten => 102,1,Dial(SIP/102,20,tr) exten => 102,2,VoiceMail,u102 exten => 102,102,VoiceMail,b102 My voicemail.conf has: 101 => 2348,Emma, kidjue@yahoo.co.uk 101 => 2348,juki, juki@one2net.co.ug How do I proceed from here then? > This is not a -dev question. It should only be posted to -users. > > On Thu, 2005-02-10 at 22:15 -0700, Juki wro...