July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Gafachi no passing caller number
by Klaus Teller 14 Jul '09

14 Jul '09

Hi, I tend to believe that we already had this working. Here is my origination string: {effective_caller_id_name=Paul Gascogne,effective_caller_id_number=16478343812}sofia/gateway/sip.gafachi.c… The caller number is not being passed to the destination. Is there something i'm missing? Thanks, Klaus. -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

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Re: [Freeswitch-users] Getting xml_request in LUA
by Eli Hayun 14 Jul '09

14 Jul '09

In the Perl example I found: > > > How to access request parameters and how to return data > > You have two hashes that are populated for you by freeswitch. Those > hashes are: > > * %XML_REQUEST > * %XML_DATA > > I want to use LUA to set the directory and dialplan xml. How do I get > the XML_REQUEST/XML_DATA from LUA? Is this the information you are looking for? http://wiki.freeswitch.org/wiki/Mod_lua#For_serving_configuration Hi Thanks fro your answer No, this is not the information i am looking for. What I need is the "section" value that FS pass when using HTTP to get the XML from. This information is not available (at least, I don't know how to get it) Thanks Eli

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Problem in Adding another user in default directory
by velusamy velu 14 Jul '09

14 Jul '09

Dear All, How to create another user agent like 1000 to 1919 in internal profile. Please provide some steps to do it.. Thanks in Advance, Velusamy

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mod_dingaling no audio
by Jingwei Yang 14 Jul '09

14 Jul '09

Hi Guys, Here's my situation: The freeswitch server and my machine are behind the same LAN. If I commented out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by *originate dingaling/gmail.com/userAAA@gmail.com &echo*). However, external calls have no sound at all no matter whether this param is commented out or not. Please kindly let me know what other params to set to resolve this issue. Thanks, -Jingwei

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Getting xml_request in LUA
by Eli Hayun 14 Jul '09

14 Jul '09

In the Perl example I found: How to access request parameters and how to return data You have two hashes that are populated for you by freeswitch. Those hashes are: * %XML_REQUEST * %XML_DATA I want to use LUA to set the directory and dialplan xml. How do I get the XML_REQUEST/XML_DATA from LUA? Thanks Eli Hayun

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Dialogic cards
by Tim Uckun 13 Jul '09

13 Jul '09

We have some older dialogic cards (D300 series E1 cards) and I am wondering if freeswitch can support these cards. Thanks.

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port restricted NAT, SRTP problem
by Роберт Тверитнер 13 Jul '09

13 Jul '09

Hi guys! I'm a novice in VoIP world, and may be missing some important concepts, but recently I've faced a problem with client softphone residing behind a port-restricted NAT and a public FS server and can't find an explanation on why it is happening and how to escape it. Okey, the problem is as follows. I have a client residing behind port restricted NAT. It can register at our public server and can issue a looped call and hear itself perfectly well. But when I call to speek to this natted guy from computer exposed to web without any routers he able to hear me for merely a second and then I become muted and he hear nothing. On the other side I can hear this guy quite good, though with slight jittery sound. If I set bypass_media param in our server's external profile to true - everything works as it supposed to - we hear one another. But still there is a problem with call originating from that guy - it is being interrupted after some time (after about 30 sec). Both clients are capable to do STUN and ICE and have these options enabled. Calls are secured with TLS and SRTP enabled on our server. FreeSWITCH is installed on Windows Server 2008 box with open UDP traffic and TCP, UDP ports 5080,5081 opened in order to expose an external profile. As far as I understand, with bypass_media param disabled FreeSWITCH is acting as media proxy and it is unable to do ICE and that should be a reason why that guy can't hear me. In overmentioned peer2peer scenario switching to no media mode is acceptable, but still there is a question whether this call with media flow bypassing FreeSWITCH is secured? I guess not. Cause I don't have any certificates installed on clients. Also, we've plans to use our FreeSWITCH as a media conference server. And of course this guy failes to connect to the testing one. Below are some of configurations: vars.xml ... <X-PRE-PROCESS cmd="set" data="external_rtp_ip=74.208.167.44"/> ... external.xml ... <param name="inbound-bypass-media" value="true"/> ... public dialplan <extension name="secure_media" continue="true"> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"> <action application="set" data="sip_secure_media=true"/> <action application="export" data="sip_secure_media=true"/> <action application="playback" data="misc/8000/call_secured.wav"/> </condition> </extension> <extension name="guard_auth" continue="true"> <condition field="${sip_authorized}" expression="^true$" break="never"> <anti-action application="respond" data="407"/> </condition> </extension> <extension name="simple_conference"> <condition field="destination_number" expression="^(.*)\.conference\..{2}$"> <action application="conference" data="$1"/> </condition> </extension> <extension name="one2one"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/external/$1%${domain_name}"/> </condition> </extension> My SDP: Remote SDP: v=0 o=- 3456517465 3456517465 IN IP4 91.79.44.168 s=pjmedia c=IN IP4 91.79.44.168 t=0 0 a=X-nat:2 m=audio 1142 RTP/SAVP 103 102 104 117 3 0 8 9 101 a=rtpmap:103 speex/16000 ... a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:1143 IN IP4 91.79.44.168 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ee+Xv9etM5t5w3DH5B1hR+i9lrt7BHQhzJIwFv7d a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:ELeudfX0mgsL+0u7qzGOqdEfg891fMn281BJszkS a=ice-ufrag:70db7dd4 a=ice-pwd:708804fe a=candidate:H 1 UDP 39 91.79.44.168 1142 typ host ... And this is the other guy SDP: Remote SDP: v=0 o=- 3456517469 3456517470 IN IP4 85.140.191.254 s=pjmedia c=IN IP4 85.140.191.254 t=0 0 a=X-nat:8 m=audio 23374 RTP/SAVP 103 101 a=rtpmap:103 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:23375 IN IP4 85.140.191.254 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:2m4YBCRid0h+5AIaIGmXaqelQsSuK3HP1jtAMoiG a=ice-ufrag:182f5a61 a=ice-pwd:1d690863 a=candidate:S 1 UDP 31 85.140.191.254 23374 typ srflx raddr 192.168.2.2 rport 2796 ... PS sorry for a long post

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Help
by Ney Frota 13 Jul '09

13 Jul '09

Help

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Gateway Settings from-domain and caller-id-in-from
by Dale 13 Jul '09

13 Jul '09

Hello, I have been playing around with gateway settings today and noticed something that I wasn't sure if it was a bug or if its just the way it works. When I have from-domain set in my gateway config it correctly uses the configured from domain. If I then set caller-id-in-from to true the configured from domain is no longer used and it reverts back to the ip address configured for that sip_profile. Is that expected or is that a bug? Thanks, -Dale

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Re: [Freeswitch-users] Help Regarding memory leak with freeswitch
by Rajagopal, Sridhar (Sridhar) 13 Jul '09

13 Jul '09

Hi , Thanks very much for the help. Please let me know which module of the code has fix. Do I need to update freeswitch core library or sofia-sip library or is it necessary to update entire freeswitch code. Regards, Sridhar ________________________________________ From: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org [freeswitch-users-request(a)lists.freeswitch.org] Sent: Monday, July 13, 2009 9:17 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 37, Issue 67 Send Freeswitch-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: Dialogic cards (Anthony Minessale) 2. Re: Help Regarding memory leak with freeswitch (Anthony Minessale) 3. Re: Setting channel variables using event socket (Anthony Minessale) 4. Re: Dialogic cards (Steve Underwood) 5. Re: Dialogic cards (Steve Underwood) 6. Re: Dialogic cards (Anthony Minessale) ---------------------------------------------------------------------- Message: 1 Date: Mon, 13 Jul 2009 10:08:26 -0500 From: Anthony Minessale <anthony.minessale(a)gmail.com> Subject: Re: [Freeswitch-users] Dialogic cards To: freeswitch-users(a)lists.freeswitch.org Message-ID: <191c3a030907130808s37e52ddg3ba567d6a5a1034b(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Dialogic is coming to ClueCon this year (this aug 4th) and they are sponsoring the conference. I can discuss the possibility of supporting their cards at that time. On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun <timuckun(a)gmail.com> wrote: > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > > Thanks. > > _______________________________________________ > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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