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[Freeswitch-users] FS - MjSip no voice [SOLVED] SIP 200 / 183 problem


 
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can_man at gmx.de
Guest
PostPosted: Tue Mar 31, 2009 9:17 am    Post subject: [Freeswitch-users] FS - MjSip no voice [SOLVED] SIP 200 / 18 Reply with quote
Hello,

I have found the problem. FS on my local network sends "SIP/2.0 200 OK"
after an invite and FS on the net through the external profil sends
SIP/2.0 183 Session Progress. But MjSip doesn't know how to deal with
183, so it just ignores the message. For testing I have changed
the 183 header to the 200 one and now it works.

Thank you for your help and the quick response time.
Best wishes,
Phil


Quote:
From FS on the net through the external profil:

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 90.181.59.141:5090;rport=60315;branch=z9hG4bK256321;received=78.105.17.88
From: <sip:puli@90.181.59.141:5090>;tag=z9hG4bK40977269
To: <sip:2345@90.181.59.141:5090>;tag=vgg3Zja8pNQcg
Call-ID: 507347917247@90.181.59.141
CSeq: 1 INVITE
Contact: <sip:mod_sofia@90.181.59.141:5090;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12839M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 267

v=0
o=FreeSWITCH 1072777625698755085 8893522831081357051 IN IP4 90.181.59.141
s=FreeSWITCH
c=IN IP4 91.121.59.148
t=0 0
m=audio 26722 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20



Quote:
From FS in my local network:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.143:5060;rport=5060;branch=z9hG4bK423233;received=192.168.1.102
From: <sip:brian@192.168.1.143>;tag=z9hG4bK42598163
To: <sip:1000@192.168.1.143>;tag=Q0X494ZUNaKHH
Call-ID: 961142687222@192.168.1.143
CSeq: 2 INVITE
Contact: <sip:mod_sofia@192.168.1.143:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12712M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Session-Expires: 120;refresher=uas
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 267

v=0
o=FreeSWITCH 5195745633884389954 8941954824002056485 IN IP4 192.168.1.143
s=FreeSWITCH
c=IN IP4 192.168.1.143
t=0 0
m=audio 22680 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20




Quote:
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@gmx.de> wrote:

Quote:
Hallo,

thank you for your answer Anthony.

Quote:

starting at line 192 you seem to be sending yourself a notify, not
sure
Quote:
Quote:
how you did that.

That is indeed strange, I have looked at the MjSip code but haven't
found
Quote:
the cause yet.

Quote:
you are not by any chance trying to call a registered endpoint using
the
Quote:
Quote:
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@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
Quote:
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'''\
Quote:
'''<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
Quote:
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

Quote:


On Sun, Mar 29, 2009 at 5:09 PM, <can_man@gmx.de> wrote:

Quote:
Hello everyone,

I am trying to get FS working with the MjSip Java Sip-stack, the
SipToSis
Quote:
source and the normal one. Everything works well within my own
network
Quote:
Quote:
and
Quote:
when using x-lite, but when it comes to making calls from MjSip to
an
Quote:
Quote:
Quote:
outside FS server I don't hear any voice - seems to be a NAT problem
or
Quote:
Quote:
some
Quote:
kind of other MjSip problem. Registration works fine though and SIP
messages
Quote:
get through ok, but non of the UDP RTP ones. Would be great if
someone
Quote:
Quote:
could
Quote:
advice me on how to do the setup correctly.

The whole FS trace can be found here:
http://pastebin.freeswitch.org/8029
Quote:

The settings for MjSip are:

"via_addr=91.101.58.142 (changed in the whole
trace)","host_port=5090",
Quote:
Quote:
Quote:
"transport_protocols=udp tcp","from_url=<sip:puli@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",
Quote:
Quote:
Quote:
"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

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
Quote:
Quote:
http://www.freeswitch.org




--
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale@hotmail.com
<MSN%3Aanthony_minessale@hotmail.com><
MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com>
<PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com>
Quote:
Quote:

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
<sip%3A888@conference.freeswitch.org><
sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org>
Quote:
Quote:

iax:guest@conference.freeswitch.org/888

googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org>
<googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org>
Quote:
Quote:

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

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

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

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

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

--
Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger01

_______________________________________________
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://www.freeswitch.org
Back to top
anthony.minessale at g...
Guest
PostPosted: Tue Mar 31, 2009 10:17 am    Post subject: [Freeswitch-users] FS - MjSip no voice [SOLVED] SIP 200 / 18 Reply with quote
like i said:
Quote:
maybe that phone does not support early media

try adding the answer application to your dialplan

early media == 183
answer = 200

it depends on your dialplan in FS



On Tue, Mar 31, 2009 at 9:06 AM, <can_man@gmx.de (can_man@gmx.de)> wrote:
Quote:
Hello,

I have found the problem. FS on my local network sends "SIP/2.0 200 OK"
after an invite and FS on the net through the external profil sends
SIP/2.0 183 Session Progress. But MjSip doesn't know how to deal with
183, so it just ignores the message. For testing I have changed
the 183 header to the 200 one and now it works.

Thank you for your help and the quick response time.
Best wishes,
Phil


Quote:
From FS on the net through the external profil:

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 90.181.59.141:5090;rport=60315;branch=z9hG4bK256321;received=78.105.17.88
From: <sip:puli@90.181.59.141:5090>;tag=z9hG4bK40977269
To: <sip:2345@90.181.59.141:5090>;tag=vgg3Zja8pNQcg
Call-ID: 507347917247@90.181.59.141 (507347917247@90.181.59.141)
CSeq: 1 INVITE
Contact: <sip:mod_sofia@90.181.59.141:5090;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12839M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 267

v=0
o=FreeSWITCH 1072777625698755085 8893522831081357051 IN IP4 90.181.59.141
s=FreeSWITCH
c=IN IP4 91.121.59.148
t=0 0
m=audio 26722 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20



Quote:
From FS in my local network:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.143:5060;rport=5060;branch=z9hG4bK423233;received=192.168.1.102
From: <sip:brian@192.168.1.143 ([email]sip%3Abrian@192.168.1.143[/email])>;tag=z9hG4bK42598163
To: <sip:1000@192.168.1.143 ([email]sip%3A1000@192.168.1.143[/email])>;tag=Q0X494ZUNaKHH
Call-ID: 961142687222@192.168.1.143 (961142687222@192.168.1.143)
CSeq: 2 INVITE
Contact: <sip:mod_sofia@192.168.1.143:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12712M
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Session-Expires: 120;refresher=uas
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 267

v=0
o=FreeSWITCH 5195745633884389954 8941954824002056485 IN IP4 192.168.1.143
s=FreeSWITCH
c=IN IP4 192.168.1.143
t=0 0
m=audio 22680 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20




Quote:
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@gmx.de (can_man@gmx.de)> wrote:

Quote:
Hallo,

thank you for your answer Anthony.

Quote:

starting at line 192 you seem to be sending yourself a notify, not
sure
Quote:
Quote:
how you did that.

That is indeed strange, I have looked at the MjSip code but haven't
found
Quote:
the cause yet.

Quote:
you are not by any chance trying to call a registered endpoint using
the
Quote:
Quote:
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@1.2.3.4 (1000@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
Quote:
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'''\
Quote:
       '''<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
Quote:
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

Quote:


On Sun, Mar 29, 2009 at 5:09 PM, <can_man@gmx.de (can_man@gmx.de)> wrote:

Quote:
Hello everyone,

I am trying to get FS working with the MjSip Java Sip-stack, the
SipToSis
Quote:
source and the normal one. Everything works well within my own
network
Quote:
Quote:
and
Quote:
when using x-lite, but when it comes to making calls from MjSip to
an
Quote:
Quote:
Quote:
outside FS server I don't hear any voice - seems to be a NAT problem
or
Quote:
Quote:
some
Quote:
kind of other MjSip problem. Registration works fine though and SIP
messages
Quote:
get through ok, but non of the UDP RTP ones. Would be great if
someone
Quote:
Quote:
could
Quote:
advice me on how to do the setup correctly.

The whole FS trace can be found here:
http://pastebin.freeswitch.org/8029
Quote:

The settings for MjSip are:

"via_addr=91.101.58.142 (changed in the whole
trace)","host_port=5090",
Quote:
Quote:
Quote:
"transport_protocols=udp tcp","from_url=<sip:puli@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",
Quote:
Quote:
Quote:
"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

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
Quote:
Quote:
http://www.freeswitch.org




--
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
<MSN%3Aanthony_minessale@hotmail.com ([email]MSN%253Aanthony_minessale@hotmail.com[/email])><
MSN%3Aanthony_minessale@hotmail.com ([email]MSN%253Aanthony_minessale@hotmail.com[/email])<MSN%253Aanthony_minessale@hotmail.com ([email]MSN%25253Aanthony_minessale@hotmail.com[/email])>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])<PAYPAL%3Aanthony.minessale@gmail.com ([email]PAYPAL%253Aanthony.minessale@gmail.com[/email])>
<PAYPAL%3Aanthony.minessale@gmail.com ([email]PAYPAL%253Aanthony.minessale@gmail.com[/email])<PAYPAL%253Aanthony.minessale@gmail.com ([email]PAYPAL%25253Aanthony.minessale@gmail.com[/email])>
Quote:
Quote:

IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
<sip%3A888@conference.freeswitch.org ([email]sip%253A888@conference.freeswitch.org[/email])><
sip%3A888@conference.freeswitch.org ([email]sip%253A888@conference.freeswitch.org[/email])<sip%253A888@conference.freeswitch.org ([email]sip%25253A888@conference.freeswitch.org[/email])>
Quote:
Quote:

iax:guest@conference.freeswitch.org/888

googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])<googletalk%3Aconf%2B888@conference.freeswitch.org ([email]googletalk%253Aconf%252B888@conference.freeswitch.org[/email])>
<googletalk%3Aconf%2B888@conference.freeswitch.org ([email]googletalk%253Aconf%252B888@conference.freeswitch.org[/email])<googletalk%253Aconf%252B888@conference.freeswitch.org ([email]googletalk%25253Aconf%25252B888@conference.freeswitch.org[/email])>
Quote:
Quote:

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

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email]) <MSN%3Aanthony_minessale@hotmail.com ([email]MSN%253Aanthony_minessale@hotmail.com[/email])>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])<PAYPAL%3Aanthony.minessale@gmail.com ([email]PAYPAL%253Aanthony.minessale@gmail.com[/email])>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email]) <sip%3A888@conference.freeswitch.org ([email]sip%253A888@conference.freeswitch.org[/email])>
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])<googletalk%3Aconf%2B888@conference.freeswitch.org ([email]googletalk%253Aconf%252B888@conference.freeswitch.org[/email])>
pstn:213-799-1400

--
Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger01

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
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FreeSWITCH Developer Conference
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