[Freeswitch-users] FS - MjSip no voice

[Anthony Minessale]

maybe that phone does not support early media

try adding the answer application to your dialplan


On Mon, Mar 30, 2009 at 3:33 PM, <can_man at gmx.de> wrote:

> Hallo,
>
> thank you for your answer Anthony.
>
> >
> > starting at line 192 you seem to be sending yourself a notify, not sure
> > how you did that.
>
> That is indeed strange, I have looked at the MjSip code but haven't found
> the cause yet.
>
> > you are not by any chance trying to call a registered endpoint using the
> > FS
> > ip together with @ are you?
> > say you fs box is 1.2.3.4 and the phone is registered as 1000
> >
> > If you want to call 1000 you don't use sofia/internal/1000 at 1.2.3.4 you
> > would
> > use sofia/internal/1000%1.2.3.4
> > The % tells it to resolve the domain as a locally hosted domain and
> > translate it to the registered contact instead of using dns.
> >
>
> For testing I at the moment send the incoming call to the voicemail of user
> 1000 with this code:
>
> return '''<?xml version="1.0" encoding="UTF-8" standalone="no"?>\n'''\
>        '''<document type="freeswitch/xml">\n'''\
>        '''<section name="dialplan" description="RE Dial Plan For
> FreeSwitch">\n'''\
>        '''<context name="public">\n'''\
>        '''<extension name="voicemail%s">\n'''\
>        '''<condition field="destination_number" expression="^(%s)$">\n'''\
>        '''<action application="voicemail" data="default $${domain}
> %s"/>\n'''\
>        '''</condition>\n'''\
>        '''</extension>\n'''\
>        '''</context>\n'''\
>        '''</section>\n'''\
>        '''</document>''' % (didNumber, didNumber, id)
>
>
> Works fine with a normal SIP client.
> I have captured more output with debug enabled and have also captured the
> SIP messages originating from MjSip.
>
> FS: http://pastebin.freeswitch.org/8045
> MjSip: http://pastebin.freeswitch.org/8046
>
> Thank you very much for your help.
> Best wishes,
> Phil
>
> >
> >
> > On Sun, Mar 29, 2009 at 5:09 PM, <can_man at gmx.de> wrote:
> >
> > > Hello everyone,
> > >
> > > I am trying to get FS working with the MjSip Java Sip-stack, the
> > SipToSis
> > > source and the normal one. Everything works well within my own network
> > and
> > > when using x-lite, but when it comes to making calls from MjSip to an
> > > outside FS server I don't hear any voice - seems to be a NAT problem or
> > some
> > > kind of other MjSip problem. Registration works fine though and SIP
> > messages
> > > get through ok, but non of the UDP RTP ones. Would be great if someone
> > could
> > > advice me on how to do the setup correctly.
> > >
> > > The whole FS trace can be found here:
> > http://pastebin.freeswitch.org/8029
> > >
> > > The settings for MjSip are:
> > >
> > > "via_addr=91.101.58.142 (changed in the whole trace)","host_port=5090",
> > > "transport_protocols=udp tcp","from_url=<sip:puli at 91.101.58.142:5090
> >",
> > >
> > >
> >
> "username=puli","realm=91.101.58.142","passwd=1234","debug_level=8","do_register=yes",
> > >
> > >
> >
> "#do_unregister=yes","#do_unregister_all=yes","keepalive_time=8000","audio=yes","audio_port=21068",
> > >
> > >
> >
> "audio_avp=0","audio_codec=PCMU","audio_sample_rate=8000","audio_sample_size=1","audio_frame_size=500",
> > > "bin_rat=rat","bin_vic=vic"
> > >
> > >
> > > Thank you very much.
> > > Best wishes,
> > > Phil
> > >
> > > --
> > > Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate +
> > > Telefonanschluss für nur 17,95 Euro/mtl.!*
> > > http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a
> > >
> > > _______________________________________________
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> > >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
>
> --
> Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate +
> Telefonanschluss für nur 17,95 Euro/mtl.!*
> http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a
>
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>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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