May 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Snom360 softphone and Freeswitch
by Krzysiek 19 May '08

19 May '08

Hi I have recently tested snom 360 softphone (www.snom.com/download/snom360-5.3.exe) with Freeswitch. It has SRTP* and TLS support. It is quite old (2006) and I'm wondering if this softphone creates sip sessions (with tls enabled) properly. Here is a sip trace (INVITE message should be sufficient) : ------------------------------------------------------------------------ recv 1092 bytes from tls/[192.168.1.4]:1145 at 17:09:19.945705: ------------------------------------------------------------------------ INVITE sip:1002@192.168.1.2;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.4:1145;branch=z9hG4bK-k1yp4ytettb1;rport From: "1001" <sip:1001@192.168.1.2>;tag=9udth2g3o3 To: <sip:1002@192.168.1.2;user=phone> Call-ID: 50b43148ee7f-696c59fq7vjl@snomSoft-000413FFFFFF CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:1001@192.168.1.4:1145;transport=tls;line=ojn9itpa>;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snomSoft/5.3 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Content-Type: application/sdp Content-Length: 362 v=0 o=root 28625 28625 IN IP4 192.168.1.4 s=call c=IN IP4 192.168.1.4 t=0 0 m=audio 57428 RTP/AVP 0 8 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:IZpDi51+JeVKPOO3Mox0q3jZYmJorsThpl6b2jw1 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv I have read in rfc3261 that : "The use of "transport=tls" has consequently been deprecated" So does snom 360 create sessions properly or not? Freeswitch works with it quite good but I don't know if it is right according to rfc. I tested few free/opensource softphones such as Minisip, Lynxphone, PhonerLite and neither worked properly with freeswitch with srtp/tls enabled. PhonerLite crashed when the call was to be setup although it registered properly , Minisip created tls sip sessions in weird way (transport=tcp) that the second leg of the call wasn't created (I wrote about this here http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003179.html) and Lynxphone didn't even register. So snom will be first FREE softphone that works. *I know that this softphone has bad SRTP/SDES implementation (RTP/AVP instead of RTP/SAVP) but I think that it isn't such a big problem when media streams by-pass Freeswitch server (for example when calls are setup in local LAN and Inbound-no-media is set to true). Am I right? TIA for reply Chris PS Sorry for my english.

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Need help with Openzap and Sangoma A200 analog card
by George Howitt 19 May '08

19 May '08

Hello all, I have a Sangoma A200 card with 2 FXS and 2 FXO interfaces. Zaptel 1.4.10 .1 is installed, along with wanpipe-3.2.5. FreeSwitch is compiled and installed with OpenZap. However, I have no idea how to configure OpenZap for my card. I can't find any examples. Specifically, how do I configure /etc/openzap/openzap.conf ???

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TLS-Support
by Helmut Kuper 19 May '08

19 May '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I try to get SIPS via TLS/TCP up. Unfortunately I get this error in console as a non root user: tport_bind_server(0x80febb8) to */85.16.246.5:5061/sips nta: bind(85.16.246.5:5061;transport=*): Protocol not supported nua: initializing SIP stack failed tport_destroy(0x80febb8) 2008-05-19 14:27:06 [ERR] sofia.c:551 sofia_profile_thread_run() Error Creating SIP UA for profile: internal in sip_profiles/internal.xml I configured TLS support as follows: ~  ~ <param name="tls" value="true"/> ~  ~ <param name="tls-bind-params" value="transport=tls"/> ~  ~ <param name="tls-sip-port" value="5061"/> ~  ~ <param name="tls-cert-dir" value="$${base_dir}/conf.tls/ssl"/> ~  ~ <param name="tls-version" value="tlsv1"/> I did a "bin/gentls_cert setup" to generate a CA. Maybe I habe to compile FS with TLS support? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkgxeBoACgkQ4tZeNddg3dyCpwCgj7pfQvJKMWKevJVyeIU6dQmy xcwAnidYgk6rA6p0YziRHkbJ1biQYuBA =DIYB -----END PGP SIGNATURE-----

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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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