- FreeSWITCH-dev - FreeSWITCH Mailing Lists

Call Hold ans swapping on FXS
by sanjeev mudholkar 25 Jan '08

25 Jan '08

Hi, I'am trying to use Freeswitch(with FXS) as a UA. Everything is fine for a outgoing and incoming call. 1)On doing a hold (flash) on the FXS the call is put on hold but there no SIP signaling (i.e re-invite is not sent to the peer). How can I force freeswitch to send a re-invite? 2) I'am not able to establish a consultation call as it is conferenced with the first one as soon as the second call is connected. This also prevents from swaping between calls. The call swapping feature seems to have been removed by introducing the transfer/conference feature. Is there away to achieve call swapping ? Appreciate your help. Thanks in advance -Sanjeev

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Re: [Freeswitch-dev] (no subject)
by Brian West 23 Jan '08

23 Jan '08

FreeSWITCHers, I'm a little disappointed... 12 digs and one single comment? I know we can do better than that. Help spread the word! Thanks, Brian West On Jan 21, 2008, at 2:49 PM, Brian West wrote: > Only 7 diggs? Come on guys... I know there are more out there that > can digg it! ;) > > /b > > On Jan 21, 2008, at 2:49 PM, Michael Collins wrote: > >> FYI, I updated the latest release date/version on the Wikipedia page >> so that people don’t think FS has been idle since beta1/August 21, >> 2007! >> >> BTW, Anthony, nice work on the HD telephony article. >> -MC >> >> From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org >> ] On Behalf Of Anthony Minessale >> Sent: Sunday, January 20, 2008 11:36 AM >> To: freeswitch-users(a)lists.freeswitch.org >> Subject: [Freeswitch-users] HD Telephony, Secure Audio >> >> Now that we have 16k and Secure RTP working abd a RC1, >> it's time to spread the word a little. >> >> I cross-posted my story at digg >> >> http://digg.com/tech_news/HD_Telephony_HIP_or_Hype >> >> Check it out and digg it! >> >> >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale@hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888@conference.freeswitch.org >> iax:guest@conference.freeswitch.org/888 >> googletalk:conf+888@conference.freeswitch.org >> pstn:213-799-1400 >> >> >> Be a better friend, newshound, and know-it-all with Yahoo! Mobile. >> Try it now. >> _______________________________________________ >> 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 > > > _______________________________________________ > 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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probs with RTP timestamps (again?)
by Sluschny, Thomas 23 Jan '08

23 Jan '08

hi, i just tested a new version from 21.01. (FreeSWITCH Version 1.0.pre3 (7309M)). If i call a phone from FreeSwitch i hear in phone robot like voice, direction phone -> FreeSwitch is ok (i use FS on WinXP). A look to wireshark shows that FreeSwitch send all 60ms 3 packets with the same timestamp, in versions before timestamp was always incremented by 160 to be synchron to RTP data bytes. I remembered the same error was already there: http://jira.freeswitch.org/browse/MODENDP-40 and than fixed. Is it now back? or another bug? In this version i got also some CRITs, may be this has the same cause: 2008-01-23 11:47:50 [CRIT] switch_time.c:387 softtimer_runtime() Forward Clock Skew Detected! 2008-01-23 11:48:03 [CRIT] switch_time.c:387 softtimer_runtime() Forward Clock Skew Detected! 2008-01-23 11:48:54 [CRIT] switch_time.c:387 softtimer_runtime() Forward Clock Skew Detected! 2008-01-23 11:50:56 [CRIT] switch_time.c:375 softtimer_runtime() Reverse Clock Skew Detected! 2008-01-23 11:55:36 [CRIT] switch_time.c:387 softtimer_runtime() Forward Clock Skew Detected! so i went little bit deeper and come to Sleep() method from windows. this sleep is very inaccurate, it counts in 15ms steps, so a sleep(1) is the same as sleep(15). my idea was to implement a better timer with WaitableTimer and Wait For SingleObject. now i have a timer with 1ms resolution, but the problem remains :( regards, Thomas

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Snom Contacts and FreeSWITCH Interop
by Brian West 23 Jan '08

23 Jan '08

Does anyone have a way to contact a real person at Snom to get some feedback on these bugs? I have four bugs one of which they have actually confirmed (Ticket#2007123010000023) and said will be fixed but I have three more that I have reported but haven't received any response on so I re- reported them today... they were previously in the respond on Ticket#2007123010000023 but its been a while since I received anything: Ticket#2008012310000047 RFC 2833 doesn't work when SRTP is enabled. (Works great on Polycom) Ticket#2008012310000038 SDP bug related to RTP/SAVP vs RTP/AVP in sdp and not accepting SAVP when you should for SRTP. (Polycom works great on this one too offers RTP/SAVP and RTP/AVP) Ticket#2008012310000029 Phone lies about its true port number in registration and subscription when using TLS. Will say one thing when it really means another. (Again polycom does this correctly) Example the phone registers to FreeSWITCH using TLS you get this in the contact <sip:1000@10.0.1.241:2048;transport=tls;line=uwohhel1> <839>:netstat -na | grep 10.0.1.241 tcp 0 0 10.0.1.250:5061 10.0.1.241:2051 ESTABLISHED Notice they aren't the same port? This will never work and requires many hacks to make this work correctly and we shouldn't have too, it should correctly report its port on register and subscribe. I would like to get these issues resolved in the next firmware release. If someone from Snom can call me.. I'm in Central Time (CST) my cellphone is listed at the bottom of this email. Thanks, Brian West FreeSWITCH.org 918-424-9278

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Re: [Freeswitch-dev] How to disable Call Waiting tone on FXS and extensions as well??
by Anthony Minessale 21 Jan '08

21 Jan '08

There is no way to disable it. I guess I was so excited that i managed to implement it that I never bothered to make it configurable. You can use mod_limit in FS if you want to limit calls to one at a time as a alt solution. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Irshad Ahamed <irshad.fswitch(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org; freeswitch-users(a)lists.freeswitch.org Sent: Monday, January 21, 2008 12:53:17 AM Subject: [Freeswitch-dev] How to disable Call Waiting tone on FXS and extensions as well?? Hi, Please let me know how we can disable/enable call waiting on FXS and extensions as well. My assumption is it is by default enabled. But I want to disable as well. Thanks in advance... BR, Irshad. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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How to disable Call Waiting tone on FXS and extensions as well??
by Irshad Ahamed 21 Jan '08

21 Jan '08

Hi, Please let me know how we can disable/enable call waiting on FXS and extensions as well. My assumption is it is by default enabled. But I want to disable as well. Thanks in advance... BR, Irshad.

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Do we have support for Caller-ID on FXS???
by Irshad Ahamed 18 Jan '08

18 Jan '08

Hi, Do we support Caller-id on FXS? If so, how to configure for it? Thanks in advance... BR, Irshad.

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How to park a call??
by Irshad Ahamed 16 Jan '08

16 Jan '08

Can anyone help me out how I should park a call?? I see a command "uuid_park uuid", but it's of no use. BR, Irshad.

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Re: [Freeswitch-dev] Regex Mismatch
by Anthony Minessale 14 Jan '08

14 Jan '08

If you look closely at your trace, you dialed 1001 and it tried every extension in your dialplan and none of them match 1001. You need to setup something that will match it and route the call to the other phone. perhaps a regex of ^1001$ for an exact match. or ^1\d+$ for everything that starts with a 1 etc. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Aadilkhan Maniyar <amaniyar(a)velankani.com> To: Freeswitch-dev(a)lists.freeswitch.org Sent: Thursday, January 10, 2008 10:30:33 PM Subject: [Freeswitch-dev] Regex Mismatch  Hi, I have installed FreeSwitch on a OpenSuse box. As per the instructions given in the “Getting started” document, I am trying to make a call between tow Xlite softphones registered with the numbers 1000 and 1001 to FreeSwitch. The registration happens successfully but when I try to make a call between these two there is an error (FS responds with a 404). According to the logs there is a Regex Mismatch in one of the files. I am unable to understand where the problem is. I have attached the log with this email. I have not modified any of the files in the conf directory. Would be grateful if anyone could help me out. Thanks & Regards, Aadil -----Inline Attachment Follows----- freeswitch@d286> 2008-01-10 16:19:34 [DEBUG] sofia_reg.c:471 sofia_reg_handle_register() Register: From: [1001(a)10.0.16.29] Contact: ["1001" <sip:1001@10.0.16.29:5070>] Expires: [1800] nta_agent: received garbage from udp/10.0.16.29:5090/sip 2008-01-10 16:19:42 [DEBUG] sofia_reg.c:471 sofia_reg_handle_register() Register: From: [1000(a)10.0.16.29] Contact: ["1000" <sip:1000@10.0.16.32:5060>] Expires: [1800] nta_agent: received garbage from udp/10.0.16.32:5090/sip nta_agent: received garbage from udp/10.0.16.29:5090/sip 2008-01-10 16:19:53 [NOTICE] switch_channel.c:385 switch_channel_set_name() New Chan sofia/mydomain.com/1000@10.0.16.29:5060 [2f029610-85df-4049-bc8d-048df6d98465] 2008-01-10 16:19:53 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Channel sofia/mydomain.com/1000@10.0.16.29:5060 entering state [received] 2008-01-10 16:19:53 [DEBUG] sofia.c:1032 sofia_handle_sip_i_state() Remote SDP: v=0 o=1000 1665729037 1665729069 IN IP4 10.0.16.32 s=X-Lite c=IN IP4 10.0.16.32 t=0 0 m=audio 8000 RTP/AVP 0 8 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2008-01-10 16:19:53 [DEBUG] sofia_glue.c:1200 sofia_glue_negotiate_sdp() Audio Codec Compare [pcmu:0:8000]/[PCMU:0:8000] 2008-01-10 16:19:53 [DEBUG] sofia_glue.c:810 sofia_glue_tech_set_codec() Set Codec sofia/mydomain.com/1000@10.0.16.29:5060 pcmu/8000 20 ms 160 samples 2008-01-10 16:19:53 [DEBUG] sofia_glue.c:1164 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2008-01-10 16:19:53 [DEBUG] sofia.c:1132 sofia_handle_sip_i_state() sofia/mydomain.com/1000@10.0.16.29:5060 State Change CS_NEW -> CS_INIT 2008-01-10 16:19:53 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/mydomain.com/1000@10.0.16.29:5060 [BREAK] 2008-01-10 16:19:53 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/mydomain.com/1000@10.0.16.29:5060) State INIT 2008-01-10 16:19:53 [DEBUG] mod_sofia.c:92 sofia_on_init() SOFIA INIT 2008-01-10 16:19:53 [DEBUG] mod_sofia.c:105 sofia_on_init() sofia/mydomain.com/1000@10.0.16.29:5060 State Change CS_INIT -> CS_RING 2008-01-10 16:19:53 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/mydomain.com/1000@10.0.16.29:5060 [BREAK] 2008-01-10 16:19:53 [DEBUG] switch_core_state_machine.c:418 switch_core_session_run() (sofia/mydomain.com/1000@10.0.16.29:5060) State RING 2008-01-10 16:19:53 [DEBUG] mod_sofia.c:120 sofia_on_ring() SOFIA RING 2008-01-10 16:19:53 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_ring() Standard RING sofia/mydomain.com/1000@10.0.16.29:5060 2008-01-10 16:19:53 [INFO] mod_dialplan_xml.c:316 dialplan_hunt() Processing 1000->1001! 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(1001) =~ /^556$/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(1001) =~ /^9193$/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(1001) =~ /^9192$/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(1001) =~ /^9191$/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(1001) =~ /^(18(0{2}|8{2}|7{2}|6{2})\d{7})$/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions source(mod_sofia) =~ /mod_portaudio/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(1001) =~ /^1234$/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(1001) =~ /^1000$/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(1001) =~ /^2000$/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions source(mod_sofia) =~ /mod_sofia/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(1001) =~ /^4(\d+)/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions source(mod_sofia) =~ /mod_sofia/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(1001) =~ /^4(\d+)/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions rdnis() =~ /8881231234/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions source(mod_sofia) =~ /mod_iax/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:116 parse_exten() test conditions destination_number(1001) =~ /0911/ 2008-01-10 16:19:53 [DEBUG] mod_dialplan_xml.c:118 parse_exten() Regex mismatch 2008-01-10 16:19:53 [INFO] switch_core_state_machine.c:111 switch_core_standard_on_ring() No Route, Aborting 2008-01-10 16:19:53 [NOTICE] switch_core_state_machine.c:112 switch_core_standard_on_ring() Hangup sofia/mydomain.com/1000@10.0.16.29:5060 [CS_RING] [NO_ROUTE_DESTINATION] 2008-01-10 16:19:53 [DEBUG] switch_channel.c:1076 switch_channel_perform_hangup() Kill sofia/mydomain.com/1000@10.0.16.29:5060 [KILL] 2008-01-10 16:19:53 [DEBUG] switch_core_session.c:647 switch_core_session_signal_state_change() Kill sofia/mydomain.com/1000@10.0.16.29:5060 [BREAK] 2008-01-10 16:19:53 [DEBUG] switch_core_state_machine.c:347 switch_core_session_run() (sofia/mydomain.com/1000@10.0.16.29:5060) State HANGUP 2008-01-10 16:19:53 [DEBUG] mod_sofia.c:217 sofia_on_hangup() Channel sofia/mydomain.com/1000@10.0.16.29:5060 hanging up, cause: NO_ROUTE_DESTINATION 2008-01-10 16:19:53 [DEBUG] mod_sofia.c:239 sofia_on_hangup() Responding to INVITE with: 404 2008-01-10 16:19:53 [DEBUG] sofia.c:71 sofia_event_callback() event [nua_i_state] status [404][Call state] session: n/a 2008-01-10 16:19:53 [DEBUG] switch_core_state_machine.c:45 switch_core_standard_on_hangup() Standard HANGUP sofia/mydomain.com/1000@10.0.16.29:5060, cause: NO_ROUTE_DESTINATION 2008-01-10 16:19:53 [DEBUG] switch_core_session.c:706 switch_core_session_thread() Session 1 (sofia/mydomain.com/1000@10.0.16.29:5060) Locked, Waiting on external entities 2008-01-10 16:19:53 [INFO] switch_core_session.c:712 switch_core_session_thread() Session 1 (sofia/mydomain.com/1000@10.0.16.29:5060) Ended 2008-01-10 16:19:53 [NOTICE] switch_core_session.c:714 switch_core_session_thread() Close Channel sofia/mydomain.com/1000@10.0.16.29:5060 [CS_HANGUP] nta_agent: received garbage from udp/10.0.16.32:5090/sip 2008-01-10 16:20:01 [CONSOLE] switch_core.c:654 switch_core_destroy() End existing sessions 2008-01-10 16:20:01 [CONSOLE] switch_core.c:656 switch_core_destroy() Clean up modules. 2008-01-10 16:20:01 [NOTICE] switch_loadable_module.c:410 switch_loadable_module_unprocess() Deleting Endpoint 'sofia' 2008-01-10 16:20:01 [NOTICE] switch_loadable_module.c:503 switch_loadable_module_unprocess() Deleting API Function 'sofia' 2008-01-10 16:20:01 [NOTICE] switch_loadable_module.c:589 switch_loadable_module_unprocess() Deleting Chat interface 'sip' 2008-01-10 16:20:01 [NOTICE] switch_loadable_module.c:621 switch_loadable_module_unprocess() Deleting Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1] 2008-01-10 16:20:01 [CONSOLE] switch_loadable_module.c:1052 do_shutdown() Stopping: mod_sofia nta_agent: received garbage from udp/10.0.16.29:5090/sip ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Re: [Freeswitch-dev] Query on no_media variable
by Anthony Minessale 14 Jan '08

14 Jan '08

I added it to the wiki here: http://wiki.freeswitch.org/wiki/Channel_Variables Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: sanjeev mudholkar <sanjeev.mudholkar(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Monday, January 14, 2008 4:07:52 AM Subject: Re: [Freeswitch-dev] Query on no_media variable Thanks for the info Anthony. That serves my purpose. By the way is this info documented anywhere as the documents available still talk about "no_media" variable. I have been looking into freeswitch wiki and http://www.0xdecafbad.com/files/fsdoc.pdf. Is there a updated document available for any such changes? Thanks -Sanjeev On Jan 11, 2008 4:48 PM, Anthony Minessale <anthmct(a)yahoo.com> wrote: the var name was changed to bypass_media Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: sanjeev mudholkar < sanjeev.mudholkar(a)gmail.com> To: freeswitch-dev(a)lists.freeswitch.org Sent: Friday, January 11, 2008 5:26:39 AM Subject: [Freeswitch-dev] Query on no_media variable Hi, I have been trying to use the variable "no_media" to enable the media flow directly between the registered endpoints. Irrespective of the value of "no_media" variable the media always flows through the freeswitch. Here is my setup Two SIP end points(1001 and 1002) registered to Freeswitch. Dial plan: <extension name="test"> <condition field="destination_number" expression="^(1002)$"> <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/$${domain}/1002%freeswitch_ip"/> </condition> </extension> My requirement is to set the media path(direct or through freeswitch) based on the destination and not for all SIP calls. (Note: I had a quick look at audio_bridge_function in mod_dptools.c. I do not see the no_media variable used. Please ignore this note if this is not the right function) Thanks in advance -Sanjeev Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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