search for: vmexten

Hello list, is the field "vmexten" available when using SIP peers in a realtime Mysql-DB ? Thanks. Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110928/e905865b/attachment.htm>

...tackle the MWI. I can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a section from my sip.conf for my test phone: [general] context=internal allowguest=no allowoverlap=no allowtransfer=yes notifyhold=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes vmexten=9998 at internal ;vmexten=*97 disallow=all allow=ulaw allow=ilbc mohinterpret=default mohsuggest=default language=en useragent=TCTC PBX ;dtmfmode = info fromdomain=10.10.60.253 ;relaxdtmf=yes [15] username=15 host=dynamic type=friend context=internal secret=edited-out subscribecontext=internal dtm...

Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL. What would be the proper way to do this for Asterisk 13 and PJSIP? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

Hello. I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the sip.conf entry, I have mailbox=1234@default and vmexten=*98. The light on the snom 320 turns on when I have voicemail and the retrieve button dials the correct extensions. However, the light turns off immediately after making the call to voicemail, even if I do not check the voicemail. Any idea on how to get this to behave properly? Nabeel

...quot;The person you are calling is unavailable." I can get the MWIs to light up on the SIP phones when there's a voicemail waiting but there's no way to check it. SIP debugging through the CLI hasn't so far given any great hints. I've tried various settings of mailbox= and vmexten= combinations in the sip.conf. For someone with a mailbox of 126@default and extension of 126 in the 'internal' context, what should the vmexten be? The real phones and the sip phones are all within the same 'internal' context in extensions.conf, as is the *8,1,VoiceMailMain() fun...

Dear List, I'm having a blinking MWI light on the snom 320 even when there's no message waiting in Asterisk. We've managed to make the voicemail button work using fromdomain=192.168.0.1 in sip.conf vmexten=2500 (our VoicemailMain application extension in extensions.conf). We also added notifymimetype=application/simple-message-summary also in sip.conf to allow SIP simple MWI notifications. But the light is still blinking and there are no voicemail messages, any ideas about how to address this iss...

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

...t 10.5.7.21 Voice-Message: 0/0 (0/0) --- ----------------------------------------------------------------------------- from sip.conf: -------------- [29] insecure=port,invite context=ccme type=friend host=r2.home.misty.com qualify=yes dtmfmode=rfc2833 canreinvite=no nat=no mailbox=29 fromuser=777 vmexten=777 username=29 from voicemail.conf: ------------------- [home] ; testing 29 => 1234,Joe Test,joetest at localhost ------------------- [root at sylvester asterisk]# ls /var/spool/asterisk/voicemail/home/29/INBOX msg0000.gsm msg0000.WAV msg0001.wav msg0002.txt msg0003.gsm msg0003.WAV msg0...

...en => 701,hint,park:701 at parkedcalls-test exten => 702,hint,park:702 at parkedcalls-test In extensions.include in [from-inside-redir-test] and [from-inside-restricted-redir-test]: include => parkedcalls-test For each extension in sip.conf in [XXX-test]: In between the existing 'vmexten' and 'secret' lines parkinglot=parkinglot_test Output from files "sip show peer 101-test.txt" and "sip show peer 102-test.txt" shows that parking lot is set to "parkinglot_test" sip show peer 101-test CODE: SELECT ALL voice2*CLI> sip show peer 101-t...

...testers.com fromuser: NULL qualify: NULL defaultip: NULL outboundproxy: PU.BL.IC.IP contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/keys/ca.crt sippasswd: md5ofmypwd rpid: NULL...

...main: testers.com fromuser: 660 qualify: NULL defaultip: NULL outboundproxy: 1.1.1.1 contactpermit: NULL contactdeny: NULL fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/keys/ca.crt sippasswd: a84a4ddcda13d13c9573d5294047b6a2...

...xdetect: NULL buggymwi: NULL auth: NULL fullname: NULL trunkname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL parkinglot: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL autoframing: NULL rtpkeepalive: NULL call-limit: NULL g726nonstandard: NULL ignoresdpversion: NULL allowtransfer: NULL dynamic: NULL path: NULL supportpath: NULL sippasswd: my-md5-pwd rpid: NULL...

Hi, Trying to set up these two buttons on a snom 360. The message waiting key seems to send a call to it's own number, which is obviously engaged and where you are prompted to leave another message to yourself, and the conference key seems to do nothing. Anyone manged to overcome these problems so that the conference key actually conferences, and the message waiting "retrieve"

...dling which cause this problem, cause i used the same sip.conf just adding notifyringing=yes, limitonpeers=yes and allowsubscribe=yes. Here is my sip.conf with one client: [general] context=incoming realm=softpbx bindport=5060 bindaddr=0.0.0.0 srvlookup=yes useclientcode=yes defaultexpirey=3600 vmexten=voicemail disallow=all allow=alaw allow=ulaw allow=gsm ;qualify=no ;canreinvite=no musicclass=default language=de useragent=ipanlage callevents=yes nat=yes rtcachefriends=no rtupdate=no rtautoclear=no ignoreregexpire=yes amaflags=omit canreinvite=no subscribecontext=outcust limitonpeers=yes allow...

...bnets connected to an Asterisk appliance, yet be able to support Internet SIP calling. My config looks like this: [general] context=INVALID allowguest=yes autodomain=no ... domain=redfish-solutions.com,redfish-internet ... [internal](!) type=friend qualify=50 nat=yes host=dynamic canreinvite=no vmexten=voicemail [sipura](!) ; nothing for now, add SLA's later [redfish](!) context=redfish-internal [garage_1](internal,sipura,redfish) callerid=Redfish Solutions <117> secret=xyzzy port=5060 mailbox=117 at redfish,1234 the problem is this: if I comment out "domain=redfish-solutions...

Hi I have a big problems on my Asterisk systems : I have one Asterisk Server with static IP (no nat) and 6 Linksys SPA941. All SPA are after a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router * SPA-3, we have a pat 5062 => SPA-3 * SPA-4, we have a pat 5063 =>

Hi, Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly. Thank you. Jad Wauthier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070512/4c0387be/attachment.htm

Can anybody help me on configuring utstarcom F1000 on asterisk? Is there a way to do it or is it impossible? ___________________________________________________________________________ Este mensaje se dirige exclusivamente a su destinatario y puede contener información privilegiada o confidencial. Si no es vd. el destinatario indicado, queda notificado de que la utilización, divulgación y/o

...of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes limitonpeers=yes callcounter=yes vmexten=5199 nat=no ; SE registrations register => user1:password1 at sipgate.co.uk:5060/se2489 register => user2:password2 at sipgate.co.uk:5060/se1268 register => user3:password3 at sipgate.co.uk:5060/se0845 register => user4:password4 at callcentric.com:5060/se1777...

Long story short... Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Useragent : snom360/6.5.2 Function: F_RETRIEVE [root@pbx ~]# asterisk -rx "show version" Asterisk 1.2.13 built by root @ pbx on a i686 running Linux on 2006-11-17 16:35:22 UTC [gateway] exten => 201,hint,SIP/201 exten =>