search for: messagenet_incoming

...mode=rfc2833 host=172.16.34.132 context=anika_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493513333333 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [messagenet] type=peer defaultuser=4444444444 secret=MYSECRET dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.it port=5061 fromuser=4444444444 fromdomain=sip.messagenet.it usereqphone=yes canreinvite=no insecure=invite Here my extensions.conf: [stdexten] include => luca_incoming include => fax_incoming include => anika_incoming include => messagenet_incoming [lu...

...172.16.34.132 > port=5060 > fromuser=00493513333333 > fromdomain=172.16.34.132 > usereqphone=yes > canreinvite=no > insecure=invite > > [messagenet] > type=peer > defaultuser=4444444444 > secret=MYSECRET > dtmfmode=rfc2833 > host=sip.messagenet.it > context=messagenet_incoming > outboundproxy=sip.messagenet.it > port=5061 > fromuser=4444444444 > fromdomain=sip.messagenet.it > usereqphone=yes > canreinvite=no > insecure=invite > > > Here my extensions.conf: > > [stdexten] > include => luca_incoming > include => fax_incoming...

...=rfc2833 host=172.16.34.132 context=anika_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493513333333 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [messagenet] type=peer defaultuser=4444444444 secret=MYVERYSECRET dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.it port=5061 fromuser=4444444444 fromdomain=sip.messagenet.it usereqphone=yes canreinvite=no insecure=invite ===================================== extensions.conf on Ubuntu-PBX: [default] exten => s,1,Verbose(2,DEFAULT) include => luca_incoming include => fax_...

...P/00491773333333-0000000b' I tried to remove ALL includes in my [default], leaving just a log, but it calls, too... My [default] exten => _X.,1,Verbose(2,DEFAULT) include => internal_calls include => luca_incoming include => fax_incoming include => anika_incoming include => messagenet_incoming include => myproxy What's wrong, now? Many thanks for your help! Luca Bertoncello (lucabert at lucabert.de)

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

...ontext=anika_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493513333333 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [messagenet] type=peer defaultuser=4444444444 secret=MYVERYSECRET dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.it port=5061 fromuser=4444444444 fromdomain=sip.messagenet.it usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=60 Am I right if I say, that I just have to change "defaultuser", "host", "secret", "outboundproxy&q...

...nreinvite=yes insecure=port,invite nat=force_rport,comedia qualify=yes qualifyfreq=600 disallow=all allow=alaw allow=ulaw and the settings for MessageNet are: [messagenet] type=peer defaultuser=<mylogin> secret=<myveryverysecretpassword> dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.it port=5060 fromuser=<mylogin> fromdomain=sip.messagenet.it usereqphone=yes canreinvite=yes insecure=invite qualify=yes qualifyfreq=60 disallow=all allow=alaw allow=ulaw allow=gsm Any idea? Thanks a lot Luca Bertoncello (lucabert at lucabert.de)

Hi! I have a Asterisk installation to manage my phones at home (provider is Deutsche Telekom). It works, but very often the voice is "broken"... Yesterday during a call it was very difficult to understand what my partner sayd... It can NOT be a problem of other downloads/uploads, since in that moment there were no ones... I already had the problem in the past, solved it enabling the

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone