May 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Termination Providers (OT..Somewhat)
by Yogesh Patel 15 May '08

15 May '08

Hello Matt, can you please add International Traffic Providersor something specific to wholesale carriers looking to buy routes from Providers. Thanks, Yogesh. From: Matt Klein <mklein(a)nmedia.net> Date: 5/14/08 21:10Subject: Re: [Freeswitch-users] Termination Providers (OT..Somewhat) Jonathan et al: I have started a page for this: http://wiki.freeswitch.org/wiki/Providers Please modify to suit your needs. Thanks! Matt On Thu, 15 May 2008, Jonathan Palley wrote: > Hello - > We are currently very unhappy with our termination partner and are looking > for someone else. We do calls to China, US, H.K. and Canada. Quality is > key. Our traffic is doubling month over month. > > I realize this is somewhat off topic for this list, but this is a very savvy > community and I think it would be beneficial to start a wiki page guiding > people in the right direction on this subject so its not brought up on this > list and those in this business can let the rest of us know. > > Thanks! > Jonathan > _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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FS, snom 370 V7.1.30 and SRTP
by Helmut Kuper 15 May '08

15 May '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I installed today FS and connected some snom 370 V7.1.30 phones with it. I tryed to check snom's SRTP feature with FS. The call is established and FS sends "Secure Music on hold" back to snom, but I can't hear anything. Call was established with 722@16000, MOH file was loaded and transcoded by FS (at least FS console said so). With RTP-Enc deactivated in snom phone it worked. Any ideas or experiences ? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkgsZkMACgkQ4tZeNddg3dxSgACfYt6qIi8eWiomBrVay8xw3tc/ 83sAn00qD9uhBqKR4JSeSxWJiDmdGUJp =rJKZ -----END PGP SIGNATURE-----

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Calls
by Czaderna 15 May '08

15 May '08

Hello I've got problem with creating call between two mobile phones. In freeswitch console mode I'm writting something like this: originate sofia/gateway/voipdiscount.com/0048xxxxxxxxx1@sip.voipdiscount.com &bridge(sofia/gateway/voipdiscount.com/0048xxxxxxxx2(a)sip.voipdiscount.com) Console output: 2008-05-15 15:55:25 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/outbound/0048xxxxxx(a)sip.voipdiscount.com [91e866c4-2286-11dd-b71a-d1af1d3a3d85] 2008-05-15 15:55:26 [NOTICE] switch_channel.c:1380 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/outbound/004833xxxxxx(a)sip.voipdiscount.com! 2008-05-15 15:55:26 [NOTICE] sofia_glue.c:1758 sofia_glue_tech_media() Pre-Answer sofia/outbound/0048xxxxxxx(a)sip.voipdiscount.com! API CALL [originate(sofia/gateway/voipdiscount.com/0048xxxxxx@sip.voipdiscount.com &bridge(sofia/gateway/voipdiscount.com/0048yyyyy(a)sip.voipdiscount.com)) output: +OK 91e866c4-2286-11dd-b71a-d1af1d3a3d85 2008-05-15 15:55:26 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/outbound/0048yyyyy(a)sip.voipdiscount.com [9279a044-2286-11dd-b71a-d1af1d3a3d85] freeswitch@m9> 2008-05-15 15:55:27 [NOTICE] switch_channel.c:1380 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/outbound/0048yyyyyy(a)sip.voipdiscount.com! 2008-05-15 15:55:27 [NOTICE] sofia_glue.c:1758 sofia_glue_tech_media() Pre-Answer sofia/outbound/0048yyyyy(a)sip.voipdiscount.com! 2008-05-15 15:55:32 [NOTICE] sofia.c:1894 sofia_handle_sip_i_state() Channel [sofia/outbound/0048xxxxx(a)sip.voipdiscount.com] has been answered 2008-05-15 15:55:32 [NOTICE] sofia.c:1898 sofia_handle_sip_i_state() Channel [sofia/outbound/0048yyyyyy(a)sip.voipdiscount.com] has been answered 2008-05-15 15:55:39 [NOTICE] sofia.c:1992 sofia_handle_sip_i_state() Hangup sofia/outbound/0048yyyyy(a)sip.voipdiscount.com [CS_LOOPBACK] [NORMAL_CLEARING] 2008-05-15 15:55:39 [NOTICE] switch_core_state_machine.c:156 switch_core_standard_on_execute() Hangup sofia/outbound/0048xxxxxx(a)sip.voipdiscount.com [CS_EXECUTE] [NORMAL_CLEARING] 2008-05-15 15:55:39 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 4 (sofia/outbound/0048yyyyyyy(a)sip.voipdiscount.com) Ended 2008-05-15 15:55:39 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/outbound/0048yyyyyy(a)sip.voipdiscount.com [CS_HANGUP] 2008-05-15 15:55:39 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 3 (sofia/outbound/0048xxxxxx(a)sip.voipdiscount.com) Ended 2008-05-15 15:55:39 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/outbound/0048xxxxxxx(a)sip.voipdiscount.com [CS_HANGUP] After receive calls I speake to one phone but in the second I hear nothing (silence). When I try to take a call from local account evertyhing was ok: originate sofia/default/1001%192.168.2.113 &bridge(sofia/gateway/voipdiscount.com/0048xxxxxxxx2(a)sip.voipdiscount.com) Where is the problem. Can anyone help me ? Thanks

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rtp multicast paging
by Brian Snipes 15 May '08

15 May '08

Would I be correct in assuming that you can change the default paging ip and port in this manner: <action application="esf_page_group" data="mcast_ip mcast_port"/> Thanks, Brian

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FS as SBC
by mashudi 15 May '08

15 May '08

Dear all, is it possible to configure FS as SBC ? so it can connect two different network ?

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bridge command example
by Cavalera Claudio Luigi 15 May '08

15 May '08

Hello, how can I modify the following example <action application="bridge" data="iax/guest@somebox/1234,sofia/test-int/1000@somehost"/> so that both locations are able to answer and not just the first one? I've taken it from http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Mutiple_destina tions_in_outbound_calls I'm testing something very similar in a php page towards freeswitch socket api, it was nice to have a one shot command to do everything :-) Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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Termination Providers (OT..Somewhat)
by Jonathan Palley 14 May '08

14 May '08

Hello - We are currently very unhappy with our termination partner and are looking for someone else. We do calls to China, US, H.K. and Canada. Quality is key. Our traffic is doubling month over month. I realize this is somewhat off topic for this list, but this is a very savvy community and I think it would be beneficial to start a wiki page guiding people in the right direction on this subject so its not brought up on this list and those in this business can let the rest of us know. Thanks! Jonathan

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Problem with TLS (call setup)
by Krzysiek 14 May '08

14 May '08

Hi I try to setup TLS SIP signalling between Freeswitch and Minisip softphone, but it doesn't work. I think I did everythnig ok in freeswitch configuration because 'sofia profile' command printed that profile with tls is running on port 5061. Minisip softphones registered properly(First I had to import a CA certificate). Problem appears when I try to setup a call between them (the same subnet). When one softphone try to setup a call, the second won't ring. Here is a trace: ####################################### freeswitch@trixswitch03> recv 669 bytes from tls/[192.168.1.3]:1072 at 20:33:52.506179: ------------------------------------------------------------------------ INVITE sip:1001@192.168.1.2 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.3:1072;rport;branch=z9hG4bK28253 Route: <sips:192.168.1.2:5061;lr=true;transport=tcp> From: <sip:1002@192.168.1.2>;tag=30769 To: <sip:1001@192.168.1.2> Call-ID: 5206(a)192.168.1.3 CSeq: 501 INVITE Max-Forwards: 70 Contact: <sip:1002@192.168.1.3:1072;transport=tcp>;expires=1000 User-Agent: Minisip Supported: 100rel,sdp-anat Content-Type: application/sdp Content-Length: 218 v=0 o=- 3344 3344 IN IP4 192.168.1.3 s=Minisip Session t=0 0 m=audio 32616 RTP/AVP 0 8 101 c=IN IP4 192.168.1.3 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 268 bytes to tls/[192.168.1.3]:1072 at 20:33:52.509158: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.1.3:1072;rport=1072;branch=z9hG4bK28253 From: <sip:1002@192.168.1.2>;tag=30769 To: <sip:1001@192.168.1.2> Call-ID: 5206(a)192.168.1.3 CSeq: 501 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8086 Content-Length: 0 ------------------------------------------------------------------------ send 728 bytes to tls/[192.168.1.3]:1072 at 20:33:52.520721: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 192.168.1.3:1072;rport=1072;branch=z9hG4bK28253 From: <sip:1002@192.168.1.2>;tag=30769 To: <sip:1001@192.168.1.2>;tag=127yN23jBU4HF Call-ID: 5206(a)192.168.1.3 CSeq: 501 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8086 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 Proxy-Authenticate: Digest realm="192.168.1.2", nonce="fef2ab8e-483f-45d6-8b13-59f383bd75fb", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 316 bytes from tls/[192.168.1.3]:1072 at 20:33:52.528439: ------------------------------------------------------------------------ ACK sip:1001@192.168.1.2 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.3:1072;rport;branch=z9hG4bK28253 Max-Forwards: 70 Route: <sips:192.168.1.2:5061;lr=true;transport=tcp> From: <sip:1002@192.168.1.2>;tag=30769 Call-ID: 5206(a)192.168.1.3 CSeq: 501 ACK To: <sip:1001@192.168.1.2>;tag=127yN23jBU4HF Content-Length: 0 ------------------------------------------------------------------------ recv 863 bytes from tls/[192.168.1.3]:1072 at 20:33:52.568375: ------------------------------------------------------------------------ INVITE sip:1001@192.168.1.2 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.3:1072;rport;branch=z9hG4bK6868 Route: <sips:192.168.1.2:5061;lr=true;transport=tcp> From: <sip:1002@192.168.1.2>;tag=30769 To: <sip:1001@192.168.1.2> Call-ID: 5206(a)192.168.1.3 CSeq: 502 INVITE Max-Forwards: 70 Contact: <sip:1002@192.168.1.3:1072;transport=tcp>;expires=1000 User-Agent: Minisip Supported: 100rel,sdp-anat Proxy-Authorization: Digest algorithm=MD5,username="1002",realm="192.168.1.2",nonce="fef2ab8e-483f-45d6-8b13-59f383bd75fb",uri="sip:1001@192.168.1.2",response="cdf7016d5fe00d8c715047a9edc094e8" Content-Type: application/sdp Content-Length: 218 v=0 o=- 3344 3344 IN IP4 192.168.1.3 s=Minisip Session t=0 0 m=audio 32616 RTP/AVP 0 8 101 c=IN IP4 192.168.1.3 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 267 bytes to tls/[192.168.1.3]:1072 at 20:33:52.570791: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.1.3:1072;rport=1072;branch=z9hG4bK6868 From: <sip:1002@192.168.1.2>;tag=30769 To: <sip:1001@192.168.1.2> Call-ID: 5206(a)192.168.1.3 CSeq: 502 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8086 Content-Length: 0 ------------------------------------------------------------------------ 2008-05-14 22:33:52 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1002(a)192.168.1.2 [025c4667-55eb-4db8-932a-c4117c810dd7] 2008-05-14 22:33:52 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1002->1001@default 2008-05-14 22:33:52 [INFO] switch_ivr_async.c:1357 switch_ivr_bind_dtmf_meta_session() Bound: 1 execute_extension::dx XML features 2008-05-14 22:33:52 [INFO] switch_ivr_async.c:1357 switch_ivr_bind_dtmf_meta_session() Bound: 2 record_session::/opt/freeswitch/recordings/1002.2008-05-14-22-33-52.wav 2008-05-14 22:33:52 [INFO] switch_ivr_async.c:1357 switch_ivr_bind_dtmf_meta_session() Bound: 3 execute_extension::cf XML features 2008-05-14 22:33:52 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001@192.168.1.4:1065;transport=tcp [3f4e1eba-46bd-4475-a3a6-bd37ce43821b] ####################################### and few seconds later something like this appears: ####################################### send 656 bytes to tls/[192.168.1.3]:1072 at 20:34:22.011100: ------------------------------------------------------------------------ SIP/2.0 500 No session set by user Via: SIP/2.0/TLS 192.168.1.3:1072;rport=1072;branch=z9hG4bK6868 From: <sip:1002@192.168.1.2>;tag=30769 To: <sip:1001@192.168.1.2>;tag=2B1QQXmp83t4a Call-ID: 5206(a)192.168.1.3 CSeq: 502 INVITE Contact: <sip:mod_sofia@192.168.1.2:5060;transport=tcp> User-Agent: FreeSWITCH-mod_sofia/1.0.pre4-8086 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 Content-Length: 0 ------------------------------------------------------------------------ recv 510 bytes from tls/[192.168.1.3]:1072 at 20:34:22.021547: ------------------------------------------------------------------------ ACK sip:1001@192.168.1.2 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.3:1072;rport;branch=z9hG4bK6868 Max-Forwards: 70 Route: <sips:192.168.1.2:5061;lr=true;transport=tcp> From: <sip:1002@192.168.1.2>;tag=30769 Call-ID: 5206(a)192.168.1.3 CSeq: 502 ACK Proxy-Authorization: Digest algorithm=MD5,username="1002",realm="192.168.1.2",nonce="fef2ab8e-483f-45d6-8b13-59f383bd75fb",uri="sip:1001@192.168.1.2",response="cdf7016d5fe00d8c715047a9edc094e8" To: <sip:1001@192.168.1.2>;tag=2B1QQXmp83t4a Content-Length: 0 ------------------------------------------------------------------------ 2008-05-14 22:34:22 [NOTICE] switch_ivr_originate.c:1154 switch_ivr_originate() Hangup sofia/default/1001@192.168.1.4:1065;transport=tcp [CS_HOLD] [NO_ANSWER] 2008-05-14 22:34:22 [ERR] switch_ivr_originate.c:813 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2008-05-14 22:34:22 [INFO] mod_dptools.c:1551 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2008-05-14 22:34:22 [NOTICE] mod_dptools.c:438 answer_function() Channel [sofia/default/1002(a)192.168.1.2] has been answered 2008-05-14 22:34:22 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 4 (sofia/default/1001@192.168.1.4:1065;transport=tcp) Ended 2008-05-14 22:34:22 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/1001@192.168.1.4:1065;transport=tcp [CS_HANGUP] 2008-05-14 22:34:22 [NOTICE] sofia.c:1992 sofia_handle_sip_i_state() Hangup sofia/default/1002(a)192.168.1.2 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2008-05-14 22:34:22 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 3 (sofia/default/1002(a)192.168.1.2) Ended 2008-05-14 22:34:22 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/1002(a)192.168.1.2 [CS_HANGUP] ####################################### Thanks for help Chris

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folder Structure
by Bill O'Neal 14 May '08

14 May '08

I'm trying to wrap my mind around the folder structure, Can someone fill in the blanks conf/sip_profiles: 1. External.xml: a. Profile settings for all Providers found in Conf/sip_profiles/external/ b. Uses the "public dialplan" 2. Internal.xml a. Profile setting for internal devices b. Includes users (with parse=true) of the directory file conf/directory ( ie all the default 1000 to 1019 extensions) c. This is the default and $${domain} alais d. Used if handling register for sip clients e. Uses the "default dialplan" 3. Nat.xml: a. Conf/sip_profiles/External.xml: 1. External.xml: * location of Voip_Providers Conf/sip_profiles/Internal.xml 1. internal.xml: conf/directory: 1. Includes internal sip phone user info. i. Different from sip_profiles in the way of

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Agenda for meeting today at 1PM central standard time.
by Brian West 14 May '08

14 May '08

http://wiki.freeswitch.org/wiki/Wiki_meet_2008_05_14 Add anything you think we need to add to that list. Brian West sip:brian@freeswitch.org

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