FS - MjSip no voice
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@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
You should press f8 to get more detailed output from FS. you also should capture more of the call, starting at line 192 you seem to be sending yourself a notify, not sure how you did that. 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@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. otherwise, enable debugging with f8 and reproduce your issue and capture *all* the output. On Sun, Mar 29, 2009 at 5:09 PM, <can_man@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@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
_______________________________________________ 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
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@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@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@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
_______________________________________________ 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
-- Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate + Telefonanschluss für nur 17,95 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K11308T4569a
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:
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@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@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@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
_______________________________________________ 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>< 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>
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>
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>
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
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
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
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
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:
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@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@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@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
_______________________________________________ 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><
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>
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>
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>
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
like i said:
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> wrote:
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
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
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 <sip%3Abrian@192.168.1.143>
;tag=z9hG4bK42598163 To: <sip:1000@192.168.1.143 <sip%3A1000@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
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:
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@1.2.3.4you 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@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@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
_______________________________________________ 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> <MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com> <
MSN%3Aanthony_minessale@hotmail.com<MSN%253Aanthony_minessale@hotmail.com> <MSN%253Aanthony_minessale@hotmail.com<MSN%25253Aanthony_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>
<PAYPAL%3Aanthony.minessale@gmail.com<PAYPAL%253Aanthony.minessale@gmail.com> <PAYPAL%253Aanthony.minessale@gmail.com<PAYPAL%25253Aanthony.minessale@gmail.com>
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> <
sip%3A888@conference.freeswitch.org<sip%253A888@conference.freeswitch.org> <sip%253A888@conference.freeswitch.org<sip%25253A888@conference.freeswitch.org>
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>
<googletalk%3Aconf%2B888@conference.freeswitch.org<googletalk%253Aconf%252B888@conference.freeswitch.org> <googletalk%253Aconf%252B888@conference.freeswitch.org<googletalk%25253Aconf%25252B888@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
_______________________________________________ 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>< 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>
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>
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>
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 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
participants (2)
-
Anthony Minessale
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can_man＠gmx.de