- FreeSWITCH-users - FreeSWITCH Mailing Lists

Coredump after update to r8458
by Luis Jimenez 17 May '08

17 May '08

I got core dump when my phone register. http://pastebin.freeswitch.org/4460 Luis Jimenez

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Need to get help on sofia_reg.c:558 sofia_reg_handle_register() send challange
by sye 16 May '08

16 May '08

Hi, I set a sip service from a sip vendor. For any PSTN call to 1800-number, that sip vendor will forward to 1800-number(a)my.domain. This domain is the freeswitch instance I set to forward to a local phone number. When I test it --- make PSTN call to 1800-number, I see the sip vendor sends sip request, but freeswitch log file shows this message: sofia_reg.c:558 sofia_reg_handle_register() send challange for [18669304954(a)208.74.66.11] 208.74.66.11 is freeswitch ip even if I set acl.conf.xml <list name="strict" default="deny"> <node type="allow" cidr="<sip_vendor_ip_address>/32"/> </list> How to configure it? Thanks, Stanley

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Event socket: api vs. bgapi
by Michael Collins 16 May '08

16 May '08

Question: Are there compelling reasons *not* to use api for simple commands like "show channels" or "oz list"? Since api is blocking and bgapi is non-blocking I just wondered if there are recommendations for when to use each one. I can definitely see why you'd use bgapi for an "originate" command. I didn't see anything on the wiki that discussed when to use api vs. bgapi so I'll be happy to write up a blurb on the recommended use of each one. Thanks, MC

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XML RPC
by Cavalera Claudio Luigi 16 May '08

16 May '08

Hello! I've installed latest svn trunk on a dedicatet machine, so vmware issues are gone. Music on hold now sounds pretty good! :-) I would like to issue some commands to freeswitch via web browser. I think that XML-RPC is the way to go but I've just found this documentation http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC which is not enough for me at this point. Is there any XML-RPC quickstart as the Javascript quickstart? I would like to try some click to dial from a web browser, maybe with a bit of javascript too :-) Thanks a lot! Ciao, 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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Re: [Freeswitch-users] calls
by Czaderna 16 May '08

16 May '08

Helo >> Before I go any farther let me point something out and I really would like to know where on the wiki this is so we can FIX IT. >> sofia/gateway/voipdiscount.com/xxxxx <- is plenty... you have already set the sip.voipdiscount.com in the gateway. There is no error on the wiki. I made a mistake. >> If you need to set the domain like that then make sure you set from-domain and from-user in your gateway settings. >> Try doing it that way and if it still gives you problems please reply and let me know so I can test it out also, Please do "console loglevel debug" that >> might give a bit more info on what is going on also. Everything is ok. I used following command originate {ignore_early_media=true}sofia/gateway/voipdiscount.com/0048xxxxxxxx1 &bridge(sofia/gateway/voipdiscount.com/0048xxxxxxxx2) and it's working. Thanks for help. Paul

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Multicast volume
by Brian Snipes 16 May '08

16 May '08

Is it possible to control the volume of a call leg? Specifically lowering. What I am experiencing is that with the multicast paging, the volume level appears to the end user to be higher than a normal call held on speakerphone on the same phone. Brian

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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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