- FreeSWITCH-users - FreeSWITCH Mailing Lists

Bug Tracker and Source Viewer
by Brian West 04 Feb '07

04 Feb '07

A new website should be rolled out by monday but till then lets go over a few things. http://jira.freeswitch.org for bug tracking http://fisheye.freeswitch.org for source viewer Use these for bugs and source viewing. On another note... We are trying to work on a hardware abstraction layer for TDM hardware for both digital and analog lines. Can anyone wishing to be involved in this please join #freeswitch on IRC and we can outline an API/IOCTL list that we would like to see to interface with hardware from our software layer. Thanks, Brian West

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Threaded IVR/Voicemail?
by michael papet 03 Feb '07

03 Feb '07

Hi, I got things working with openser by having freeswitch register as an sip user with openser. Freeswitch is an sip client that handles voicemail. The problem with this is multiple users simultaneously using voicemail is not possible. The Freeswitch user is busy with a single caller. If I attempt to just forward the call to freeswitch's ip address:5060 (without freeswitch registering to openser) freeswitch does not respond at all. I assume I need to do something in sofia.conf.xml, but I'm not sure. Any advice, even if I'm missing the obvious, would be very helpful. Thanks. MP

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Re: [Freeswitch-users] Freeswitch Newbie Questions
by Anthony Minessale 03 Feb '07

03 Feb '07

You have a parse error in your file. Try installing the 100% clean default configs by removing all the files in /usr/local/freeswitch/conf and re-issuing make installall Also note there is a freeswitch bug tracker http://jira.freeswitch.org Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER: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: michael papet <mpapet(a)gmail.com> To: freeswitch-users <freeswitch-users(a)lists.freeswitch.org> Sent: Friday, February 2, 2007 3:04:41 PM Subject: [Freeswitch-users] Freeswitch Newbie Questions Okay, so I've figured out one or two things but have stumbled across a minor issue. I've got the following as my test plan:  <context name="default"> <extension name="playsound1" continue="true"> <condition field="source" expression="mod_sofia"/> <condition field="caller_id_name" expression="mpapet"> <action application="answer"/> <action application="playback" data="/var/sounds/congrats.wav"/> </condition> </extension> 2007-02-02 13:00:36 [NOTICE] switch_core.c:2802 switch_core_standard_on_execute() Execute playback(/var/sounds/congrats.wav) 2007-02-02 13:00:36 [DEBUG] mod_sndfile.c:138 sndfile_file_open() Opening File [/var/sounds/congrats.wav] 8000hz 2007-02-02 13:00:36 [DEBUG] switch_ivr.c:1125 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2007-02-02 13:00:36 [DEBUG] mod_sofia.c:2698 sip_i_state() Channel sofia/zed/mpapet(a)voip.katigo.mooo.com entering state [completed] It answers and starts playing the file according to the console but I hear no sound. Do I need some kind of other statement in there? Or, is it a NAT issue. Or? What does this error message mean? [DEBUG] mod_sofia.c:4585 event_callback() event [nua_r_register] status [401][Unauthorized] session: n/a FYI: If I remove freeswitch_combined.xml, the file default_context.xml does not get read/included from freeswitch.xml. Something like "cannot find default context" kept appearing in the console despite the presences of default_context.xml. My quick and dirty solution was to put the default context inside freeswitch.xml. I verified this in the freeswitch console with the error message. After the change, call works great. As an FYI for future users, I'm running openser and freeswitch on the same machine using virtual ip addresses. You need to modify sofia.conf.xml <param name="sip-ip" value="your.freeswitch.virtual-ip.address"/> to get it to register freeswitch to your sip server. _______________________________________________ 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 ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited

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Freeswitch Newbie Questions
by michael papet 02 Feb '07

02 Feb '07

Okay, so I've figured out one or two things but have stumbled across a minor issue. I've got the following as my test plan:  <context name="default"> <extension name="playsound1" continue="true"> <condition field="source" expression="mod_sofia"/> <condition field="caller_id_name" expression="mpapet"> <action application="answer"/> <action application="playback" data="/var/sounds/congrats.wav"/> </condition> </extension> 2007-02-02 13:00:36 [NOTICE] switch_core.c:2802 switch_core_standard_on_execute() Execute playback(/var/sounds/congrats.wav) 2007-02-02 13:00:36 [DEBUG] mod_sndfile.c:138 sndfile_file_open() Opening File [/var/sounds/congrats.wav] 8000hz 2007-02-02 13:00:36 [DEBUG] switch_ivr.c:1125 switch_ivr_play_file() Codec Activated L16@8000hz 1 channels 20ms 2007-02-02 13:00:36 [DEBUG] mod_sofia.c:2698 sip_i_state() Channel sofia/zed/mpapet(a)voip.katigo.mooo.com entering state [completed] It answers and starts playing the file according to the console but I hear no sound. Do I need some kind of other statement in there? Or, is it a NAT issue. Or? What does this error message mean? [DEBUG] mod_sofia.c:4585 event_callback() event [nua_r_register] status [401][Unauthorized] session: n/a FYI: If I remove freeswitch_combined.xml, the file default_context.xml does not get read/included from freeswitch.xml. Something like "cannot find default context" kept appearing in the console despite the presences of default_context.xml. My quick and dirty solution was to put the default context inside freeswitch.xml. I verified this in the freeswitch console with the error message. After the change, call works great. As an FYI for future users, I'm running openser and freeswitch on the same machine using virtual ip addresses. You need to modify sofia.conf.xml<param name="sip-ip" value=" your.freeswitch.virtual-ip.address"/> to get it to register freeswitch to your sip server.

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pleas cancle the mail list for me ,thanks
by pengcz＠126.com 01 Feb '07

01 Feb '07

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Freeswitch Newbie Questions
by michael papet 31 Jan '07

31 Jan '07

Hi, I got freeswitch installed and I'm sorting out some basic issues with getting things going. The immediate goal is to have freeswitch recording/sending voicemail while openser does the basic register/userloc/nat detection/etc. A call from an openser client that does not answer should go to openswitch for voicemail processing using your voicemail script example. I'm assuming I can grab caller info from the "to" and use a sql lookup to finish the voicemail handling. 1. What is the string to use in freeswitch.xml to make speex the default codec? Sorry but it's not perfectly clear to me. 2. So far I've got freeswitch and openser running with freeswitch on a virtual IP and registering with openser. For testing, I'm dialing *86 from another openser client. It's routing right with the call ending at freeswitch and hanging up. How do I set a route to the voicemail script example? I'm assuming that's how I would handle it but I'm a newbie. Here's the debug: 2007-01-31 11:17:37 [DEBUG] mod_sofia.c:1270 sofia_on_hangup() Channel sofia/voicemail/mpapet(a)katigo.mooo.com hanging up, cause: NO_ROUTE_DESTINATION 2007-01-31 11:17:37 [DEBUG] mod_sofia.c:1290 sofia_on_hangup() Responding to INVITE with: 404 2007-01-31 11:17:37 [DEBUG] switch_core.c:2723 switch_core_standard_on_hangup() Standard HANGUP sofia/voicemail/mpapet(a)katigo.mooo.com, cause: NO_ROUTE_DESTINATION 2007-01-31 11:17:37 [DEBUG] switch_core.c:3468 switch_core_session_thread() Session 2 (sofia/voicemail/mpapet(a)katigo.mooo.com) Locked, Waiting on external entities 2007-01-31 11:17:37 [INFO] switch_core.c:3473 switch_core_session_thread() Session 2 (sofia/voicemail/mpapet(a)katigo.mooo.com) Ended 2007-01-31 11:17:37 [NOTICE] switch_core.c:3349 switch_core_session_destroy() Close Channel sofia/voicemail/mpapet(a)katigo.mooo.com 2007-01-31 11:17:37 [DEBUG] mod_sofia.c:4585 event_callback() event [nua_i_state] status [404][Call state] session: n/a Finally, I attempted to subscribe from both my yahoo and earthlink email accounts. Neither one sent me the usual automated response for subscribing.

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Re: [Freeswitch-users] problems with codecs when usinh jingle
by Anthony Minessale 31 Jan '07

31 Jan '07

Try speex@8000k or speex@16000k to make sure you define the specific speed you want. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER: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: Pekka K. Kurki <pekka.kurki(a)intellectics.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, January 19, 2007 2:02:20 AM Subject: [Freeswitch-users] problems with codecs when usinh jingle - when connecting to/from original google talk client the connection starts ok but if the preferred codecs is speex the speed is wrong - it sounds like playing an LP with 45 rpm..., pcmu is ok - when connectibg to/from jabber/psi/kopete the cocec speex 110 offered by jabber/psi/kopete is not accepted by freeswitch. uploading another speex codec doesn't work, perhaps not supported by the clients... ____________________________________________________________________________________ Any questions? Get answers on any topic at www.Answers.yahoo.com. Try it now.

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mfcr2
by Yeruult[ICN] 30 Jan '07

30 Jan '07

Hello all, Does Freeswitch supports mfcr2? I know mfcr2 is old protocol but that doesn't mean ppl don't use it. As I know there is still many guys from everywhere. As I understood asterisk v1.4 becoming complicated for Steve Underwood's unicall library. Is there any other open sources which supports mfcr2? -- Best regards, Yeruult[ICN] mailto:yeruult@icn-mn.net

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timeline
by Jonas Gauffin 28 Jan '07

28 Jan '07

Hello Can you give me a rough estimate on how long it would take to have a production quality / release candidate version of FreeSwitch? Are we talking about 6 months, 1 year, 2 years? thanks, Jonas

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pa switch
by Tamas Cseke 26 Jan '07

26 Jan '07

Hello i tested the new pa switch feature, it works on linux, but on windows pa switch <callid> doesn't work says "no such call", without argument i can switch between the last two calls, but if i have many calls for example 3 i can't switch to the first one, just to the 2nd, and third, beacause of the pa switch <callid> doesn't work Thanks, Tamas

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