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wrong playback speed on shout streams and mp3 files
by Pekka Kurki 21 Mar '08

21 Mar '08

with the latest svn version the mp3 streams and files are played with higher speed. this used to work ok with an older version. my extension definition: <extension name="hki"> <condition field="destination_number" expression="^hki$">  <action application="answer"/> <action application="set" data="stream_prebuffer=131072"/>  <action application="set" data="stream_speed=1"/> <action application="playback" data="shout://217.30.180.252:8000"/> </condition> </extension> any ideas? br, pekka //(Mr.) Pekka K. Kurki Intellectics GesmbH http://www.intellectics.com Helenenstrasse 44, A-2500 Baden, Austria Tel. +43 780 815811 (NEW AREA CODE!!!) Mobile +43 681 10358833 Fax +43 780 815819 (NEW AREA CODE!!!) (Tel. US: 360-968-1576 (NEW!!) UK: 0870 4787064 FIN: 02-48084133) sip: pekka.kurki(a)sip.intellectics.com enum: +432256815811, ISN: 1*743 skype: pkkurki, pkkurki-sip (sip gateway) gtalk: pekkis50(a)gmail.com (sip gateway - experimental) gizmo: pekkis50 email: Pekka.Kurki(a)intellectics.com www: http://pekka.kurki.at push-to-call: http://www.intellectics.com/call.htm//

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RC1 Has Arrived!
by Anthony Minessale 20 Mar '08

20 Mar '08

FreeSWITCH Community, After nearly 2 years of non-stop development, FreeSWITCH has evolved from an idea and an empty directory into a fully operational soft-switch capable of withstanding the loads 8-10 times as large as anything else they were able to deploy in the past with other software according to the reports from our early adopters. I am happy to announce the immediate availability of FreeSWITCH 1.0 “Phoenix“ (Release Candidate 1). The Phoenix symbolizes the rebirth of a new era and I hope this new release will be the first step in a new age of voice-over-IP development we can all benefit from. We anticipate the official FreeSWITCH 1.0 “Phoenix” release date to be Monday, May 26th 2008. We will have a formal release presentation at this year’s ClueCon 2008 August 5th-7th 2008. http://www.cluecon.com I would like to thank our community for the testing, bug reporting, code contributions, feedback and financial donations that helped us get this far, especially those who were involved from day 1. Please take this opportunity to test drive FreeSWITCH as we will be making it available as binary packages for the first time to practice for the official 1.0 release. The code and arrangement of the configuration will not change apart from bug fixes and tweaks from this point forward. As always, the code will still be available in svn at the usual place. For download instructions please stay tuned to: http://wiki.freeswitch.org/wiki/Installation_Guide The packages will be online soon. Thanks Everyone! 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 ____________________________________________________________________________________ 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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mod_limit fall through
by bsnipes 20 Mar '08

20 Mar '08

I am current jumping to a different extension when a limit is reached and the limit_exceeded condition is hit by changing the 'dialed_ext' and calling: <action application="bridge" data="${sofia_contact($${domain}/${dialed_ext}@$${domain})}"/> I hit a problem when it hits the 'second line' for the user. I would like it to check via mod_limit if that extension is on the phone and respond appropriately at that point but it doesn't seem to hit the mod_limit for that second extension when it is called that way. Is there another way to call an extension and have it go through the limit process again? Thanks, Brian -- View this message in context: http://www.nabble.com/mod_limit-fall-through-tp16128842p16128842.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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session.recordFile and only silence
by Jonas Gauffin 20 Mar '08

20 Mar '08

Hello How will freeswitch handle if a caller comes to the voicemail and is only silent and then hangs up? (hangs up before the silence_secs value is exceeded) //Jonas

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jira seems to be having some problems
by Kurt Marasco 19 Mar '08

19 Mar '08

I'm trying to update a bug report at http://jira.freeswitch.org/ but the server is not properly responding. Thought that maybe someone here might be able to resolve the issue.

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Re: [Freeswitch-users] Wikipbx status
by Traun Leyden 18 Mar '08

18 Mar '08

The biggest incompatibility right now is with mod_voicemail, which was released shortly after wikipbx was released, and I haven't had a chance to go back and make this work. It's a high priority item though. Wikipbx comes with its own voicemail IVR, but its rather simplistic and does not support things like MWI (message waiting indicator). The other thing to be aware of is that it is heavily SIP oriented, so if you are going to be using non-SIP protocols you will probably run into some issues. Hope this helps! > I didn't know about wikipbx until today and it looks like nice from the > screenshots. Is it current with freeswitch in it's features support? > > Brian

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mod_limit
by Brian Snipes 18 Mar '08

18 Mar '08

A little background: I would like our phones to have the ability to roll a call over to a second registration on the phone if the first registration is in use. An example: line1 is registered as 317, line2 as 417. If line1 is in use I would like the call to 317 to go to 417. If 417 is in use then have it rollover to 317 voicemail. Using suggestions on the irc channel I've added extensions for limit and limit_exceeded but when a second call comes into the 317 extension it gets a hangup with the following: 2008-03-17 12:46:33 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Brian Snipes->limit_exceeded@default 2008-03-17 12:46:33 [NOTICE] switch_ivr.c:982 switch_ivr_session_transfer() Transfer sofia/default/305@192.168.10.2:5060 to enum[limit_exceeded@default] 2008-03-17 12:46:34 [INFO] switch_core_state_machine.c:113 switch_core_standard_on_ring() No Route, Aborting Here is what I have cobbled together through reading and experimenting. I'm not sure if all the set/export statements are needed or not. I used the info application to try and figure out what condition to test for but I'm not doing something right. <extension name="limit" continue="true"> <condition> <action application="limit" data="inbound ${destination_number} 1"/> </condition> </extension> <extension name="limit_exceeded">      <condition field="sip_req_user" expression="^(317)$"> <action application="set" data="destination_number=417"/> <action application="export" data="destination_number=417"/> <action application="set" data="dialed_ext=417"/> <action application="export" data="dialed_ext=417"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="${sofia_contact($${domain}/${dialed_ext}@$${domain})}"/> </condition> </extension> Many thanks, Brian

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Re: [Freeswitch-users] IAX and CDR
by Anthony Minessale 18 Mar '08

18 Mar '08

our IAX2 support is limited. We are using our own fork of the iaxclient library from http://iaxclient.wiki.sourceforge.net/ This is the client lib used in most iax soft phones and is not suitable as-is for a server. To avoid causing commotion for the other software that uses this lib we just forked it and added our own mutexes and a few other things. (this is not a hostile fork as I have commit access on the real one. It's just because there are several experimental changes.) Even with our changes, the original lib has no trunking support and still barely qualifies to be used in a server so it's not worth it to add the support. The real solution is to write an entire IAX2 stack from scratch that actually can scale for use on a server. This is within my capability but not prudent because the IAX2 protocol is, in reality, an Asterisk Specific proprietary protocol and it wouldn't be fair for us to have a nicer IAX2 implementation than them. Not to mention, there is nearly no demand for this protocol among our user base. There was an openiax http://www.openiax.org that was underway. Maybe someday if that gets off the ground we can look at supporting it. But all signs point to the fact that we would need major motivation to expand our current iax2 support beyond what it currently is. 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: Michael Collins <mcollins(a)fcnetwork.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 17, 2008 6:37:37 PM Subject: Re: [Freeswitch-users] IAX and CDR  Not entirely intuitive, but there are two references to IAX2 on this page: http://wiki.freeswitch.org/wiki/Current_Status I believe I heard in passing that FS does not currently support IAX trunking, but I’d wait for confirmation from one of the devs to be sure. -MC From: freeswitch-users-bounces@lists.freeswitch.org[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Vijay Ramnarine Sent: Monday, March 17, 2008 4:17PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] IAXand CDR How much IAX does FS support? does it support IAX trunking? How about CDR ? If there is somewhere I can just go to in order to read up about these, pleasetell me link -----Inline Attachment Follows----- _______________________________________________ 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 ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Wikipbx status
by Brian Snipes 18 Mar '08

18 Mar '08

I didn't know about wikipbx until today and it looks like nice from the screenshots. Is it current with freeswitch in it's features support? Brian

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IAX and CDR
by Vijay Ramnarine 17 Mar '08

17 Mar '08

How much IAX does FS support? does it support IAX trunking? How about CDR ? If there is somewhere I can just go to in order to read up about these, please tell me link

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