search for: 12345,1

...the Asterisk-server) in my living room, when I'm not at my desk, the sip-phone is switched off. I would like to be able to accept calls at both phones (when available) and have Voicemail kick in if I don't answer. The 'normal' extension would be something like this: exten => 12345,1,Dial(SIP/me,30) exten => 12345,2,VoiceMail(su12345) Works fine as long as the sip-phone is available, if it is not, it is flagged congested/busy, so the next extension would be 102, if I wanted VoiceMail to kick in in that case, this works: exten => 12345,1,Dial(SIP/me,30) exten => 12...

...rings and then an irritating timeout with H.323 message 'no user responding' instead of 'user busy'. Asterisk knows the user is busy and jumps to the prio+101 extension. The CDR also logs the call as 'busy'. With Zap ISDN channels the following works nicely: exten => 12345,1,Dial(SIP/${EXTEN},45,r) exten => 12345,102,SetVar(PRI_CAUSE=17) exten => 12345,103,Hangup Can we do something similar with chan_oh323? Many thanks, Jan

...y unwieldy. That's why you would stick the members into a global variable [globals] DIYCALLGROUP => SIP/111&SIP/112&SIP113 etc. then dial using Dial(${DIYCALLGROUP},...) Also, you can use the callgroup feature in sip.conf [111] ... callgroup=1 callerid="Member 1"<12345> [112] ... callgroup=1 callerid="Member 2"<12345> [113] ... callgroup=1 callerid="Member 3"<12345> then in your dialplan exten => 12345,1,Dial(SIP/111) ; dialling one member rings them all this should call the entire call group. There have been some issu...

...d the difference between working and non-working, and I realized that other people were experiencing the same problem as I was. The trick is the *order* in which you put your hint priorities in your dialplan. My non-working sidecar dialplan had all the hint priorities grouped together: exten => 12345,hint,SIP/12345 exten => 12346,hint,SIP/12346 Which would register the hint, but it wouldn't work on the Snom. The way to make it work, for sure, is to make sure your hint priority is the last priority underneath the *related* priority for the extension. So, this will work: exten => 1234...

Hi! I use the following configuration to register my asterisk server to my SIP provider: register => 12345:passwd@sip.provider.com/12345 sip.conf: [sipout-test] type=peer username=12345 fromuser=12345 fromdomain=provider.com secret=passwd insecure=very host=sip.provider.com qualify=yes context=test-incoming extensions.conf: exten => 12345,1,Dial(SIP/10) exten => _0NXZXXXXXX,1,Dial(SIP/${EXTEN}@s...

...eneral] bindaddr=192.168.0.160 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asterisk1] type=peer username=asteriskm auth=plaintext secret=asgard host=192.168.0.161 qualify=yes **extensions.conf** [general] [1ST-T1] exten => _XXXXX,1,AGI(rexx.agi) exten => 12345,1,Dial(IAX2/asterisk1/80483) exten => 12345,n,Hangup() Asterisk1 config: **iax.conf** [general] bindaddr=192.168.0.161 bandwidth=low disallow=all allow=gsm jitterbuffer=no tos=lowdelay authdebug=no [asteriskm] type=user context=incoming-iax auth=plaintext secret=asgard host=192.168.0.160 quali...

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

Hello, I have a noticed strange behavior in Asterisk 1.6.18.2 with ReceiveFax Digium FAX Driver: 1.6.2.0_1.3.0 (optimized for i686_32). I use a context [capi-in] for icoming ISDN calls: ====== [capi-in] ; Faxe fuer Ruben exten => 12345,1,Macro(faxin,ruben.roegels at jumping-frog.org,${EXTEN}) ====== My macro for the fax receiving looks like that: ====== [macro-faxin] ; Faxe ; ARG1 = eMail-Adresse exten => s,1,Verbose(${BOUNDARY} Eingehender Ruf von ${CALLERID(num)}) exten => s,n,Verbose(${BOUNDARY} BCHANNELINFO ${BCHANNEL...

...got the error 603 Declined. Below the sip.conf: *[gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* extension.conf: *[tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten => 12345,1,Dial(SIP,giusy*) Below the output of SIP debug of IP caller (192.168.1.116) in asterisk *dhcppc0*CLI> <--- SIP read from 192.168.1.116:14862 ---> INVITE sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-...

How do I get * to take an incoming oh323 call and let it dial a number? I.E. if my boss sets up netmeeting with the gateway as my.pabxbox.com, whenever he enters a number to dial it always just dials into the pabx rather than calling that number i.e. he wants to call 12345 he types it in and presses dial but it just goes to the message.... also I have developed an H323 client app which I thought I would dial the numbers as XXX@pabx.com where XXX is the number and pabx.com is the asterisk machine.... Peace out, Matt Riddell

Good day all I want to put the openline4 card into a box that will support 3 different companies I read the caller ID id fixed but now HOW DO I: If a call come in for 12345 it plays company 1's welcome message If a call come in for 98765 it plays company 2's welcome message ens.. Does This make sense Thanks Altus

...SY" <103> mailbox=103 channel => 2 and here is my extension.conf file [cda] exten =>s,1,Answer ;Operators exten => 110,1,SetLanguage(fr) exten => 110,2,AgentLogin exten => 15,1,SetLanguage(fr) exten => 15,2,VoicemailMain ;Clients campagne natural mystic exten => 12345,1,SetLanguage(fr) exten => 12345,2,Answer exten => 12345,3,Queue(110) ; Pour recevoir les faxes exten =>fax,1,RxFax(/var/spool/asterisk/incoming/lamine.tif) ; Pour envoyer le fax exten =>8236331,1,TxFax(/var/spool/asterisk/incoming/lamine.tif) But when I launch Asterisk i get the foll...

Hi there, Is there an accepted way of running an AGI script on answering of a channel? Is it even possible? I don't need to execute AGI commands, I just need to know a channel has been answered. Thanks, David. -- "One world, one web, one program" -- Microsoft promotional advert. "Ein Volk, ein Reich, ein Fuehrer" -- Adolf Hitler.

Hello, Asterisk Server A makes an outbound call, and upon connect: exten =>1,n,RetryDial(/var/lib/asterisk/sounds/connecting,0,3,SIP/${connectto},,tT ) (${connectto} most of the time happens to be 12345 at 66.xx.xx.66 or 54321 {IP masqueraded ofcourse}) ..transfers it to * Server B (i.e 66.xx.xx.66) via SIP. (Background info, Server B registers on Server A as 1000, and Server A registers on Server B as 1000. Both of them are on direct IPs, and not behind a hardware firewall. Server A has no...

Hello, While running Asterisk version 1.6.1.0 I've discovered that playing sounds on local channels does not work anymore as it used to in Asterisk 1.6.0.5 The sounds are played poorly or not at all ... A simple test context would be [default] exten => 12345,1,Dial(Local/s at test) exten => 12345,2,Hangup [test] exten => s,1,Playback("1-for-am-2-for-pm") Activating the jitter buffer on the local channel did not help (i.e. Dial(Local/s at test/nj) ) Has anybody encounter the same problem? Is there any solution to this? Thank you! --...

...activated threewaycalling in the zapata.conf file: ; internal S0 bus (first hfc/s card): context=local signalling = bri_net_ptmp group = 2 channel => 1-2 threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes I laso added the t and the T options to the extension lines: exten => 12345,1,Ringing exten => 12345,2,Dial(Zap/g2/${EXTEN},,Ttr) exten => 12345,3,Hangup _________________________________________________________________ Schlu? mit Spam! http://www.msn.de/antispam/prevention/junkmailfilter Wir helfen Ihnen, unerw?nschte E-Mails zu reduzieren.

Hi, I have a simple asterisk application that bridges a call to an operator. Now I want to detect if this operator has picked up his phone or not. Ass John Todd told me on the irc channel, that cannot easily be done because Asterisk understands this internally, but does not present these data to the dial plan... Is there a solution to my problem or do I have to hack the code... I'm sure

...erisk]# vi sip.conf [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial* *[giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial* *[root at dhcppc0 asterisk]# vi extensions.conf [tutorial] exten => 1234,1,Dial(SIP,gianca)* *exten => 12345,1,Dial(SIP,giusy) * Here the XLITE user data: *Display Name: gianca* *Username: 1234* *Password: pwd_gianca* *Authorization User Name: 1234* *Domain: 192.168.1.100* ** Here the output of wireshark in between Xlite client and asterisk server: ** *0040 2e 31 30 30 20 53 49 50 2f 32 2e 30 0d 0a 56...

Hi Brian, many thanks to you for your answers in the past! The always gave me the little bit of mising information... My Asterisk box is running fine now so I want to try the "next step"... And now to all of you .... What I want to implement is to use 1 button of my snom-360 phone for following purpose: If I leave my desk I press this button. A light should show up as an

Hi Here is part of my extensions.conf exten => 8661231234,1,agi,dtmf.php When I dial this number, this is what I see in my asterisk console: -- Accepting AUTHENTICATED call from 198.22.67.70: > requested format = gsm, > requested prefs = (), > actual format = gsm, > host prefs = (gsm|ilbc|speex), > priority = mine -- Executing