NO VOICE BETWEEN TWO REMOTE ENDPOINTS
Hi, I have configured FS like this.but voice is not transmitting between two endpoints.Can you help me where i did wrong ? Here is my configuration: - <?ui=2&view=bsp&ver=1qygpcgurkovy#11afef97be4488a0_11afef267e7502b2_> <profile name="internal" domain="$${domain}"> <aliases> <alias name="$${domain}"/> <alias name="default"/> </aliases> <gateways> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/> </gateways> <domains> </domains> <settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="rfc2833-pt" value="101"/> <!-- port to bind to for sip traffic --> <param name="sip-port" value="60"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="aggressive-nat-detection" value="true"/> <!-- ip address to use for rtp --> <param name="rtp-ip" value="$${local_ip_v4}"/> <!-- ip address to bind to --> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="$${hold_music}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5061"/> <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <param name="tls-version" value="tlsv1"/> <param name="inbound-no-media" value="true"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings> </profile> -- Regards ---------- M Emran Inspiration Software Ltd Web: www.inspiresoftbd.com
It could be many things... Firewall? Nat being a nazi... what does your dialplan look like? So many variables but from the info provided we need more to make a guess. You'll need to provide a sip trace. You can get that via starting FreeSWITCH like this TPORT_LOG=1 ./freeswitch If you're on windows you can set the ENV variable to collect the same output Then you can place that up on our paste bin located at http://pastebin:freeswitch@pastebin.freeswitch.org Thanks, /b PS: service iptables stop On Jul 7, 2008, at 4:03 PM, M.Emran wrote:
Hi,
I have configured FS like this.but voice is not transmitting between two endpoints.Can you help me where i did wrong ? Here is my configuration:
- <profile name="internal" domain="$${domain}"> <aliases> <alias name="$${domain}"/> <alias name="default"/> </aliases> <gateways> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/> </gateways> <domains> </domains>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="rfc2833-pt" value="101"/> <!-- port to bind to for sip traffic --> <param name="sip-port" value="60"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="aggressive-nat-detection" value="true"/> <!-- ip address to use for rtp --> <param name="rtp-ip" value="$${local_ip_v4}"/> <!-- ip address to bind to --> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="$${hold_music}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="record-template" value="$${base_dir}/recordings/$ {caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5061"/> <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <param name="tls-version" value="tlsv1"/>
<param name="inbound-no-media" value="true"/>
<param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/>
</settings> </profile>
-- Regards ---------- M Emran Inspiration Software Ltd Web: www.inspiresoftbd.com _______________________________________________ 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
Brian West sip:brian@freeswitch.org
Thanks for your instant reply. i m using CentOS5.1 <section name="dialplan" description="example_curl_dialplan"> <context name="public"> <extension name="echo test"> <condition field="destination_number" expression="999"> <action application="answer" /> <action application="echo"/> </condition> </extension> <extension name="curl_test"> <condition field="destination_number" expression="(.*)"> <action application="set" data="proxy_media=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=NO_ROUTE_DESTINATION,NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER"/> <action application="bridge" data="sofia/internal/$1%X.X.X.X" /> <action application="bridge" data="sofia/internal/$1@X.X.X.X" /> <action application="hangup" /> </condition> </extension> </context> </section> My dialplan is like this: TH SIP trace are as follows: *********************************************************************************** 2008-06-30 18:28:42 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/internal/73814@8.14.147.7:60[3981c0a5-52ae-4d42-b53b-144417411041] 2008-06-30 18:28:42 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Emran->999@public send 1082 bytes to udp/[202.56.7.133]:6060 at 01:28:42.850929: ------------------------------------------------------------------------ SIP/2.0 200 OK Content-Length: 494 v=0 o=- 0 2 IN IP4 202.56.7.133 s=CounterPath X-Lite 3.0 c=IN IP4 202.56.7.133 t=0 0 m=audio 6062 RTP/AVP 107 119 100 106 0 97 105 98 8 101 a=alt:1 2 : +kaK6fdy AO7bhX9b 10.130.5.144 6062 a=alt:2 1 : 1HGX4kCQ 8AY7lwQk 202.56.7.133 60622008-06-30 18:28:42 [NOTICE] switch_channel.c:1381 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/internal/73814@8.14.147.7:60! 2008-06-30 18:28:42 [NOTICE] sofia_glue.c:1782 sofia_glue_tech_media() Pre-Answer sofia/internal/73814@8.14.147.7:60! 2008-06-30 18:28:42 [NOTICE] mod_dptools.c:620 answer_function() Channel [sofia/internal/73814@8.14.147.7:60] has been answered a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:97 SPEEX/8000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 1082 bytes to udp/[202.56.7.133]:6060 at 01:28:43.390886: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.130.5.144:6060 ;branch=z9hG4bK-d87543-ba226852c327c735-1--d87543-;rport=6060;received= 202.56.7.133 From: "Emran"<sip:73814@8.14.147.7:60>;tag=2130c455 To: "999" <sip:999@8.14.147.7:60>;tag=pBr0jSNFm2tDS Call-ID: NjMxZTAxOTlhNzg3ZmFkMDExZmJkYzM3NjRmODZmZTc. CSeq: 2 INVITE Contact: <sip:mod_sofia@8.14.147.7:60;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.rc6-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, precondition, timer Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 270 v=0 o=FreeSWITCH 8868993956311956419 6207280862328188583 IN IP4 8.14.147.7 s=FreeSWITCH c=IN IP4 8.14.147.7 t=0 0 a=sendrecv m=audio 25042 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ recv 322 bytes from udp/[202.56.7.133]:6060 at 01:28:43.392783: ------------------------------------------------------------------------ ACK sip:999@8.14.147.7:60 SIP/2.0 Via: SIP/2.0/UDP 10.130.5.144:6060 ;branch=z9hG4bK-d87543-cb5fdd7f0f480f4f-1--d87543-;rport To: "999" <sip:999@8.14.147.7:60>;tag=N2y7gy4BQS4tD From: "Emran"<sip:73814@8.14.147.7:60>;tag=2130c455 Call-ID: NjMxZTAxOTlhNzg3ZmFkMDExZmJkYzM3NjRmODZmZTc. CSeq: 1 ACK Content-Length: 0 ------------------------------------------------------------------------ send 1082 bytes to udp/[202.56.7.133]:6060 at 01:28:43.649996: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.130.5.144:6060 ;branch=z9hG4bK-d87543-ba226852c327c735-1--d87543-;rport=6060;received= 202.56.7.133 From: "Emran"<sip:73814@8.14.147.7:60>;tag=2130c455 To: "999" <sip:999@8.14.147.7:60>;tag=pBr0jSNFm2tDS Call-ID: NjMxZTAxOTlhNzg3ZmFkMDExZmJkYzM3NjRmODZmZTc. CSeq: 2 INVITE Contact: <sip:mod_sofia@8.14.147.7:60;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.rc6-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, precondition, timer Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 270 v=0 o=FreeSWITCH 8868993956311956419 6207280862328188583 IN IP4 8.14.147.7 s=FreeSWITCH c=IN IP4 8.14.147.7 t=0 0 a=sendrecv m=audio 25042 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ------------------------------------------------------------------------ recv 704 bytes from udp/[202.56.7.133]:6060 at 01:28:43.701995: ------------------------------------------------------------------------ ACK sip:mod_sofia@8.14.147.7:60;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.130.5.144:6060 ;branch=z9hG4bK-d87543-e225ef323f565677-1--d87543-;rport Max-Forwards: 70 Contact: <sip:73814@202.56.7.133:6060> To: "999"<sip:999@8.14.147.7:60>;tag=pBr0jSNFm2tDS From: "Emran"<sip:73814@8.14.147.7:60>;tag=2130c455 Call-ID: NjMxZTAxOTlhNzg3ZmFkMDExZmJkYzM3NjRmODZmZTc. CSeq: 2 ACK Proxy-Authorization: Digest username="73814",realm="8.14.147.7 ",nonce="cf982c06-49ad-4783-b66f-1b3fbc39eba6",uri="sip:999@8.14.147.7:60",response="d4216c57ffccdac32c0aad69a3e7edb4",cnonce="ad462cc0040fbaf128ab19ac726dd9fd",nc=00000001,qop=auth,algorithm=MD5recv 704 bytes from udp/[202.56.7.133]:6060 at 01:28:44.276254: ------------------------------------------------------------------------ ACK sip:mod_sofia@8.14.147.7:60;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.130.5.144:6060 ;branch=z9hG4bK-d87543-e225ef323f565677-1--d87543-;rport Max-Forwards: 70 Contact: <sip:73814@202.56.7.133:6060> To: "999"<sip:999@8.14.147.7:60>;tag=pBr0jSNFm2tDS From: "Emran"<sip:73814@8.14.147.7:60>;tag=2130c455 Call-ID: NjMxZTAxOTlhNzg3ZmFkMDExZmJkYzM3NjRmODZmZTc. CSeq: 2 ACK Proxy-Authorization: Digest username="73814",realm="8.14.147.7 ",nonce="cf982c06-49ad-4783-b66f-1b3fbc39eba6",uri="sip:999@8.14.147.7:60 ",response="d4216c57ffccdac32c0aad69a3e7edb4",cnonce="ad462cc0040fbaf128ab19ac726dd9fd",nc=00000001,qop=auth,algorithm=MD5 User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 ------------------------------------------------------------------------ recv 704 bytes from udp/[202.56.7.133]:6060 at 01:28:44.492472: ------------------------------------------------------------------------ ACK sip:mod_sofia@8.14.147.7:60;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.130.5.144:6060 ;branch=z9hG4bK-d87543-e225ef323f565677-1--d87543-;rport Max-Forwards: 70 Contact: <sip:73814@202.56.7.133:6060> To: "999"<sip:999@8.14.147.7:60>;tag=pBr0jSNFm2tDS From: "Emran"<sip:73814@8.14.147.7:60>;tag=2130c455 Call-ID: NjMxZTAxOTlhNzg3ZmFkMDExZmJkYzM3NjRmODZmZTc. CSeq: 2 ACK Proxy-Authorization: Digest username="73814",realm="8.14.147.7 ",nonce="cf982c06-49ad-4783-b66f-1b3fbc39eba6",uri="sip:999@8.14.147.7:60 ",response="d4216c57ffccdac32c0aad69a3e7edb4",cnonce="ad462cc0040fbaf128ab19ac726dd9fd",nc=00000001,qop=auth,algorithm=MD5 User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 ------------------------------------------------------------------------ recv 764 bytes from udp/[202.56.7.133]:6060 at 01:29:02.939495: ------------------------------------------------------------------------ BYE sip:mod_sofia@8.14.147.7:60;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.130.5.144:6060 ;branch=z9hG4bK-d87543-3d675138d06a9b12-1--d87543-;rport Max-Forwards: 70 Contact: <sip:73814@202.56.7.133:6060> To: "999"<sip:999@8.14.147.7:60>;tag=pBr0jSNFm2tDS From: "Emran"<sip:73814@8.14.147.7:60>;tag=2130c455 Call-ID: NjMxZTAxOTlhNzg3ZmFkMDExZmJkYzM3NjRmODZmZTc. CSeq: 3 BYE Proxy-Authorization: Digest username="73814",realm="8.14.147.7 ",nonce="cf982c06-49ad-4783-b66f-1b3fbc39eba6",uri="sip:mod_sofia@8.14.147.7:60 ;transport=udp",response="6b9abae5ffece0385d2e16d5114ab97d",cnonce="d763d24d7aa2fd4f617d61888f11232d",nc=00000002,qop=auth,algorithm=MD5 User-Agent: X-Lite release 1011s stamp 41150 Reason: SIP;description="User Hung Up" Content-Length: 0 ------------------------------------------------------------------------ send 537 bytes to udp/[202.56.7.133]:6060 at 01:29:02.941894: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.130.5.144:6060 ;branch=z9hG4bK-d87543-3d675138d06a9b12-1--d87543-;rport=6060;received= 202.56.7.133 From: "Emran"<sip:73814@8.14.147.7:60>;tag=2130c455 To: "999"<sip:999@8.14.147.7:60>;tag=pBr0jSNFm2tDS Call-ID: NjMxZTAxOTlhNzg3ZmFkMDExZmJkYzM3NjRmODZmZTc. CSeq: 3 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.rc6-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, precondition, timer Content-Length: 0 ------------------------------------------------------------------------ 2008-06-30 18:29:02 [NOTICE] sofia.c:2126 sofia_handle_sip_i_state() Hangup sofia/internal/73814@8.14.147.7:60 [CS_EXECUTE] [NORMAL_CLEARING] 2008-06-30 18:29:02 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 2 (sofia/internal/73814@8.14.147.7:60) Ended 2008-06-30 18:29:02 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/internal/ 73814@8.14.147.7:60 [CS_HANGUP] ********************************************************************* On Tue, Jul 8, 2008 at 3:09 AM, Brian West <brian@freeswitch.org> wrote:
It could be many things... Firewall? Nat being a nazi... what does your dialplan look like? So many variables but from the info provided we need more to make a guess. You'll need to provide a sip trace. You can get that via starting FreeSWITCH like this TPORT_LOG=1 ./freeswitch
If you're on windows you can set the ENV variable to collect the same output
Then you can place that up on our paste bin located at http://pastebin:freeswitch@pastebin.freeswitch.org
Thanks, /b PS: service iptables stop
On Jul 7, 2008, at 4:03 PM, M.Emran wrote:
Hi,
I have configured FS like this.but voice is not transmitting between two endpoints.Can you help me where i did wrong ? Here is my configuration:
-<http://?ui=2&view=bsp&ver=1qygpcgurkovy#11afef97be4488a0_11afef267e7502b2_>
<profile name="internal" domain="$${domain}"> <aliases> <alias name="$${domain}"/> <alias name="default"/> </aliases> <gateways> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/> </gateways> <domains> </domains>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="rfc2833-pt" value="101"/> <!-- port to bind to for sip traffic --> <param name="sip-port" value="60"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="aggressive-nat-detection" value="true"/> <!-- ip address to use for rtp --> <param name="rtp-ip" value="$${local_ip_v4}"/> <!-- ip address to bind to --> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="$${hold_music}"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5061"/> <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <param name="tls-version" value="tlsv1"/>
<param name="inbound-no-media" value="true"/>
<param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/>
</settings> </profile>
-- Regards ---------- M Emran Inspiration Software Ltd Web: www.inspiresoftbd.com _______________________________________________ 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
Brian West sip:brian@freeswitch.org
_______________________________________________ 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
-- Regards ---------- M Emran Managing Director InSpiration Software Ltd. Robin Tower 42-43, Purana Palton(4th) Dhaka-1000,Bangladesh. Phone# 880-2-9559461 Cell# 880-1713033045 E-mail:monemran@gmail.com <E-mail%3Amonemran@gmail.com> IM: mohammademran@hotmail.com Web: www.inspiresoftbd.com
participants (2)
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Brian West
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M.Emran