Hi ;)
I'm a french newbie and i installed asterisk 1 day ago.
I've got an ATA186 and a computer with Sjphone installed.
If i want to call the sjphone from the ata or call the ata from de sjphone
everything is ok.
My problem is ,that i can't call the voicemail or any other phone number
..as 600 for exemple from the ata or the jphone.
I don't know why but i looked after a long time..
here a copy of my extension.conf , sip.conf and voicemail.conf.
Thanks for your help.
Julien.
Extension.conf
[general]
static=yes
writeprotect=yes
[bogon-calls]
exten => _.,1,Congestion
[from-sip]
exten => 1943,1,Dial(SIP/1943,5)
exten => 1943,2,Voicemail(u1943)
exten => 1943,102,Voicemail(b1943)
exten => 1943,103,Hangup
exten => 1945,1,Dial(SIP/1945,6)
exten => 1945,2,Voicemail(u1945)
exten => 1945,102,Voicemail(b1945)
exten => 1945,103,Hangup
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
-----------------------------
sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
allow=all
context = bogon-calls
[1943]
type=friend
username=1943
secret=1943
host=dynamic
context=from-sip
mailbox=1943
[1945]
type=friend
username=1945
secret=1945
host=dynamic
context=from-sip
mailbox=1945
-----------------------
voicemail.conf
[general]
format=wav
[local]
1943 => 1943,Essai 1,xxx@yy.com
1945 => 1945,Essai2,rrr@ttt.bil
At 15:13 10-8-2003 +0200, you wrote:>If i want to call the sjphone from the ata or call the ata from de sjphone >everything is ok. >My problem is ,that i can't call the voicemail or any other phone number >..as 600 for exemple from the ata or the jphone. >I don't know why but i looked after a long time.. > >[from-sip] >exten => 1943,1,Dial(SIP/1943,5) >exten => 1943,2,Voicemail(u1943) >exten => 1943,102,Voicemail(b1943) >exten => 1943,103,Hangup > >exten => 1945,1,Dial(SIP/1945,6) >exten => 1945,2,Voicemail(u1945) >exten => 1945,102,Voicemail(b1945) >exten => 1945,103,Hangup > >exten => 2999,1,VoicemailMain(${CALLERIDNUM})Call me silly, but this does mean your Voicemailbox is at extension 2999, not 600. Or did I misunderstand you ? BTW, any other phonenumber not being callable would make sense, since you simply don't have anything else in the [from-sip] context. You could include the IAXtel gateway or add a device to connect to your phone line... Florian
Fabia, The only numbers you should be able to dial from that config are 1945 1943 2999 and nothing else... The entry under bogon-calls (isn't it bogus calls?) should read exten => s,1,Congestion rather that using the _. ... HTH Andy *********** REPLY SEPARATOR *********** On 10/08/2003 at 15:13 Fabia wrote:>Hi ;) > >I'm a french newbie and i installed asterisk 1 day ago. >I've got an ATA186 and a computer with Sjphone installed. > >If i want to call the sjphone from the ata or call the ata from de sjphone >everything is ok. >My problem is ,that i can't call the voicemail or any other phone number >..as 600 for exemple from the ata or the jphone. >I don't know why but i looked after a long time.. > >here a copy of my extension.conf , sip.conf and voicemail.conf. > >Thanks for your help. >Julien. > >Extension.conf > >[general] > >static=yes >writeprotect=yes > >[bogon-calls] >exten => _.,1,Congestion >[from-sip] >exten => 1943,1,Dial(SIP/1943,5) >exten => 1943,2,Voicemail(u1943) >exten => 1943,102,Voicemail(b1943) >exten => 1943,103,Hangup > >exten => 1945,1,Dial(SIP/1945,6) >exten => 1945,2,Voicemail(u1945) >exten => 1945,102,Voicemail(b1945) >exten => 1945,103,Hangup > >exten => 2999,1,VoicemailMain(${CALLERIDNUM}) > > >----------------------------- >sip.conf > >[general] > >port = 5060 >bindaddr = 0.0.0.0 >allow=all >context = bogon-calls > >[1943] > >type=friend >username=1943 >secret=1943 >host=dynamic >context=from-sip >mailbox=1943 > >[1945] > >type=friend >username=1945 >secret=1945 >host=dynamic >context=from-sip >mailbox=1945 >----------------------- >voicemail.conf > >[general] > >format=wav > >[local] > >1943 => 1943,Essai 1,xxx@yy.com >1945 => 1945,Essai2,rrr@ttt.bil > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users
Senad Jordanovic
2003-Aug-10 08:57 UTC
[Asterisk-Users] Registering SIP with FWD and ICONNECTHERE
Hi, I am trying to register a SIP service(s) with FWD and ICONNECTHERE. Here is my sip.conf: ------------------------------------------------ [general] port =5060 bindaddr =0.0.0.0 context =default calledid=No CallID register => 400703:PASSWORD@fwd.pulver.com/400703 register => USERNAME:PASSWORD@sipauth.deltathree.com/400703 [fwd] type=friend username=USERNAME secret=PASSWORD host=fwd.pulver.com dtmfmode=inband [iconnect] type=friend username=USERNAME secret=PASSWORD host=sipauth.deltathree.com dtmfmode=inband ---------------------------------------------- Here is error output i get: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on call 23bb9b484917dd803090b59678ee9a4c@212.159.81.202 for seqno 102 (Request) NOTICE[81926]: File chan_sip.c, Line 2476 (sip_reg_timeout): Registration for '400703@192.246.69.223' timed out, trying again WARNING[81926]: File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on call 23bb9b484917dd803090b59678ee9a4c@212.159.81.202 for seqno 102 (Request) NOTICE[81926]: File chan_sip.c, Line 2476 (sip_reg_timeout): Registration for '400703@192.246.69.223' timed out, trying again It shows "212.159.81.202" IP, which is one of old IPs i used when astrisk was originally installed. Now the asterisk is behind NAT. Anyone had the same problem? Senad