search for: checkmwi

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

...e' timed out, trying again (Attempt #n) I looked at the sip debug stuff, and all I can see is my asterisk sending the registration packets, but no answer is received. Here's the relevant parts of my sip.conf: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes tos=0x18 checkmwi=10 videosupport=yes allow=all relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 register => XXXXX:pass@sipgate.de/XXXXX ;XXXXX == sipgateid [XXXXX] type=friend insecure=very nat=yes username=XXXXX fromuser=XXXXX fromdomain=sipgate.de secret=pass host=sipgate.de qualify=yes -- Michiel van Baak...

...est=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 dtmfmode=rfc2833 ;defaultip=10.10.60.246 mailbox=15 ;subscribemwi=yes notifymimetype=text/plain checkmwi=10 vmexten=9998 at internal disallow=all allow=ulaw All of my sip phones exist in the "internal" context and I have the appropriate extensions (I think) set up. In my voicemail.conf I have an [internal] section with the mailbox 15 defined with password 1234. I can...

I've configured asterisk 1.4 to use imap storage for voice-mail and while I'm happy with it generally speaking it really seem to hammer the IMAP server. It appear, from the IMAP server logs that it's polling the imap server every *second* for mailbox updates for the users' voice-mail folders. Is it really necessary to do this once a second? Is this tunable anywhere? Thanx, b.

...V5060-0-138966241-538634340-1288768121281- Max-Forwards: 70 Content-Length: 0 Here's the configs: subscribecontext = device-hints allowexternaldomains = yes allowguest = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progr...

Hi Everyone, I am using real time asterisk architecture and have placed the following in sip.conf: [general] notifymimetype=text/plain checkmwi=10 rtcachefriends=yes but the MWI doesn't work?! Can anyone give me any pointers as to what the problem could be? Thanks ramin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/200...

...fter voicemail is recorded? I'm guessing the configuration isn't too far off because the phones do periodically receive NOTIFY messages that correctly reflect the state of the voice-mailbox. I double checked sip.conf to ensure that each phone has a 'mailbox' parameter and that 'checkmwi' is set to 10. I should note that I also tried configuring one of my phones (Polycom IP601) to SUBSCRIBE to Asterisk, and while Asterisk 200 OK'ed the subscription, it didn't seem to have an effect on the issue. In short, how can I configure Asterisk so that new voicemail will trigger...

...vice load no result. What I did not configured? My sip.conf [general] context = default allowguest = no bindport = 5060 bindaddr = 0.0.0.0 allowexternaldomains = no allowoverlap = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = no checkmwi = 10 compactheaders = no defaultexpiry = 120 domain=sop-korniychuk domain=172.30.8.13 domain=172.30.8.13:5060 dumphistory = no externrefresh = 10 g726nonstandard = no notifyringing = yes srvlookup = yes t1min = 100 t38pt_udptl = no ;tos_audio = none ;tos_sip = none ;tos_video = none trustrpid = no...

...lt ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes ;domain=mydomain.tld ;domain=mydomain.tld,mydomain-incoming ;domain=1.2.3.4 ;allowexternalinvites=no ;autodomain=yes ;pedantic=yes ;tos=184 ;tos=lowdelay ;maxexpiry=3600 ;defaultexpiry=120 ;notifymimetype=text/plain ;checkmwi=10 ;vmexten=voicemail ;videosupport=yes ;recordhistory=yes disallow=all allow=g729 allow=gsm allow=ulaw jitterbuffer=yes maxjitterbuffer=1500 ;allow=ilbc ;musicclass=default ;language=en ;relaxdtmf=yes rtptimeout=60 ;rtpholdtimeout=300 ;trustrpid = no ;sendrpid = yes ;progressinband=never ;useragen...

...///////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes rtptimeout=60 rtpholdtimeout=300 useragent=PBX dtmfmode = rfc2833 checkmwi=20 promiscredir=no nat=yes autodomain=no domain=nxs.yi.org,sip allowexternalinvites=yes rtcachefriends=yes rtupdate=yes rtautoclear=yes ignoreregexpire=yes and extensions.conf [general] static=yes writeprotect=no autofallthrough=yes ////////////////////////////////////////////////////// [glob...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have an Asterisk box sitting between the PSTN and a legacy PBX. I have successfully configured Asterisk to use IMAP for voicemail and have written the necessary script to turn the MWI indicator (via a .call file to the PBX) on and off. I have two issues still outstanding: 1) When the user listens to his voice mail via the phone, it will be

...///////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes rtptimeout=60 rtpholdtimeout=300 useragent=PBX dtmfmode = rfc2833 checkmwi=20 promiscredir=no nat=yes autodomain=no domain=nxs.yi.org,sip allowexternalinvites=yes rtcachefriends=yes rtupdate=yes rtautoclear=yes ignoreregexpire=yes and extensions.conf [general] static=yes writeprotect=no autofallthrough=yes ////////////////////////////////////////////////////// [glob...

...ay,throughput,reliability,mincost,none > maxexpiry=3600 ; Max length of incoming registration we allow > defaultexpiry=120 ; Default length of incoming/outgoing registration > ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY > ;checkmwi=10 ; Default time between mailbox checks for peers > ;vmexten=voicemail ; dialplan extension to reach mailbox sets the > ;videosupport=yes ; Turn on support for SIP video > ;recordhistory=yes ; Record SIP history by default > disallow=a...

...risk.pem tlscafile=/etc/asterisk/keys/ca.crt callevents=no jbenable=no videosupport=yes allowguest=no srvlookup=no defaultexpiry=120 minexpiry=60 maxexpiry=3600 registerattempts=0 registertimeout=20 g726nonstandard=no maxcallbitrate=384 canreinvite=no rtptimeout=30 rtpholdtimeout=300 rtpkeepalive=0 checkmwi=10 notifyringing=yes notifyhold=yes nat=yes [1000] deny=0.0.0.0/0.0.0.0 secret=6ff108122cce3b0b45e0abf374c14ef4 dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=yes port=5060 qualify=yes qualifyfreq=60 transport=udp avpf=no icesupport=no...

...; lowdelay,throughput,reliability,mincost,none ;maxexpiry=3600 ; Max length of incoming registration we allow ;defaultexpiry=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;checkmwi=10 ; Default time between mailbox checks for peers ;vmexten=voicemail ; dialplan extension to reach mailbox sets the ; Message-Account in the MWI notify message ; defau...

I have /var/spool/asterisk/voicemail NFS mounted from another server. Everything is fine, until I simulate an NFS server failure, by shutting down the NFS server process. At this point, Asterisk becomes almost non-responsive. It won't even process a 'sip show peers' command correctly. It displays a few lines of text, pauses for several seconds, and then displays the rest. When a call

You are reading the instructions for the STABLE 1.0 version of asterisk and are using the CVS version. Goto the wiki and read the instructions for RealTime. -Matthew ----- Original Message ----- From: "Muhammad Rizwan Khan" <rizwan@advcomm.net> To: <Asterisk-Dev@lists.digium.com> Sent: Wednesday, January 05, 2005 12:42 PM Subject: [Asterisk-Dev] Asterisk with MySQL >