- FreeSWITCH-users - FreeSWITCH Mailing Lists

Number of calls
by Voip service 23 Dec '11

23 Dec '11

Hi, I am new in voip, how many calls can one freeswitch box handle with 30 % of trans-coded calls and system configuration as 8GB RAM X3430 Xeon Processor, 2.4GHz, 8M Cache, Turbo

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Problems with polycom registering
by lloyd thomas 23 Dec '11

23 Dec '11

I had what I thought was the perfect set up on my FS box. Until I changed internet provider (BT) All was fine for a week but all of a sudden some of my Polycom phones won't register. The polycom phones are behind a NAT and my FS box is on a static IP address. I am using a multi company setup One of my polycom phones seems to register OK Using 'sofia profile xxxxxxxx siptrace on' I can see that one of my polycom phones will not register because FS is trying to send replies to the private IP address (192.168.101.16) Any Ideas? Am I missing something in one of the profiles Good one ------------------------------------------------------------------------ send 702 bytes to udp/[87.194.242.110]:65004 at 17:49:32.226606: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 87.194.242.110:65004;branch=z9hG4bK15c264eE5B5988B From: "Suze" <sip:200@suze.tele.phisys.net>;tag=CA76AF43-ED2A5944 To: <sip:200@suze.tele.phisys.net>;tag=7Uy88X4pc8yHc Call-ID: 918d5430-ffb07455-eb2121fe(a)87.194.242.110 CSeq: 888 REGISTER Contact: <sip:200@87.194.242.110:65004>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER";expires=30 Date: Tue, 20 Dec 2011 17:49:32 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2c009dd 2011-03-15 14-29-04 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------- Bad one ------------------------------------------------------------------------ send 680 bytes to udp/[81.137.114.169]:5060 at 17:50:17.659167: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.101.16:5060 ;branch=z9hG4bK2b8f42dA63D9E42;received=81.137.154.169 From: "Lloyd" <sip:200@phisys.tele.phisys.net>;tag=621EA6CE-AF1D573 To: <sip:200@phisys.tele.phisys.net>;tag=aQaKeFQ13219e Call-ID: 52e98627-d24590dc-528d7361(a)192.168.101.16 CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2c009dd 2011-03-15 14-29-04 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="phisys.tele.phi.co.uk", nonce="1319a358-2b33-11e1-b5e4-dd1099e4b2d2", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------

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Re: [Freeswitch-users] Problem with make current
by Peter Olsson 23 Dec '11

23 Dec '11

Try "make spandsp-reconf" Or just do a ./bootstrap.sh and ./configure. /Peter ----- Reply message ----- Från: "lloyd thomas" <lloydie.t(a)gmail.com> Datum: ons, dec 21, 2011 03:05 Rubrik: [Freeswitch-users] Problem with make current Till: "freeswitch-users" <FreeSWITCH-users(a)lists.freeswitch.org> Since I am not getting anywhere with my sip registration problem I decided to update FS from git using make current. Unfortunately I am getting the following errors. What can I do to resolve? In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or â...â before âtz_tâ make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 !DSPAM:4ef13c9932768811635433!

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Ringback on PRI line
by Ryan V 23 Dec '11

23 Dec '11

Hi, We are running freeswitch with Sangoma PRI and Analog cards. There is no ring back for calls coming in on PRI line. Calls coming in on FXO lines are fine. Calling party gets ring back. I have added following to file conf/dialplan/public/00_inbound_did.xml <action application="set" data="instant_ringback=true"/> <action application="set" data="ringback=%(2000,4000,440.0,480.0)"/> Any suggestions to resolve this problem? Thanks, Venkatesh K

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Seasons greetings from Kinetix Tele.com
by Vlasis Hatzistavrou (KTI) 23 Dec '11

23 Dec '11

Seasons greetings from Kinetix Tele.com <http://www.kinetixtele.com>

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failover with mod_lcr. how to set failover reasons with | in bridge ?
by Yuriy Nasida 23 Dec '11

23 Dec '11

Hello list.I have same issue and wondering if nobody still have not solution. In general I have possibility not to use the bridge with "|" but this issue looks strange. Anyone? I am still wondering if there is a documented/undocumented reason why continue_on_fail does not work with piped failover? From: Ivan Kovacevic [mailto:ivank at rogers.com] Sent: June-09-11 4:38 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: continue_on_fail failover with mod_lcr Hi Everyone, I would like to implement fail-over for my outbound gateway, but I would like to be able to pick certain cause codes which qualify to stop trying next gateway (specifically when I have bad number and I am getting sip:404 NO_ROUTE_DESTINATION, UNALLOCATED_NUMBER).I was using 1.0.6, but I moved to the newest git about month ago. After searching through lists and spending several hours playing with continue_to_fail, failure_causes and fail_on_single_reject I was able to make it work by specifying cause codes for which I want to fail-over and omitting ones that qualify to stop trying next gateway in variable continue_to_fail: So this setup is working for me and in the case I have bad number and x.x.x.x returns sip:404 (NO_ROUTE_DESTINATION or UNALLOCATED_NUMBER) it is not trying y.y.y.y gateway. <include> <extension name="myextension"> <condition field="destination_number" expression="^(01144\d+)$"> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,SERVICE_NOT_IMPLEMENTED,CALL_RE JECTED,NO_USER_RESPONSE,EXCHANGE_ROUTING_ERROR,INCOMPATIBLE_DESTINATION,NETWORK_OUT_OF_ORDER,RECOVERY_ON_TIMER_EXPIRE"/> <action application="bridge" data="sofia/myprofile/$1 at x.x.x.x"/> <action application="bridge" data="sofia/myprofile/$1 at y.y.y.y"/> <action application="bridge" data="sofia/myprofile/$1 at z.z.z.z"/> </condition> </extension> </include> However if I want to use "|" between my gateways - and the example below is not working. And no matter what x.x.x.x returns - it will try y.y.y.y and eventually z.z.z.z. <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,SERVICE_NOT_IMPLEMENTED,CALL_RE JECTED,NO_USER_RESPONSE,EXCHANGE_ROUTING_ERROR,INCOMPATIBLE_DESTINATION,NETWORK_OUT_OF_ORDER,RECOVERY_ON_TIMER_EXPIRE"/> <action application="bridge" data="sofia/myprofile/$1 at x.x.x.x|sofia/myprofile/$1 at y.y.y.y|sofia/myprofile/$1 at z.z.z.z"/> Unfortunately, we have to use pipe for fail-over since we are using mod_lcr to choose between outbound gateways. Any suggestions? Thanks, Ivan

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 66, Issue 148
by Rodney 23 Dec '11

23 Dec '11

MC- Here is the pastebin: http://pastebin.freeswitch.org/18064 I created small log and recreated my problem. 1. at main ivr, pressed 7, then at conf menu went to static 509 (ivr option 9), and then pressed # to get transfered to 759 which is my dynamic extension Rodney ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, December 23, 2011 2:20 AM Subject: FreeSWITCH-users Digest, Vol 66, Issue 148 ----- Forwarded Message ----- 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: transfer to dynamic con from static (Michael Collins) 2. Radio stream binding ? (Sherif Omran) 3. faxing and file permissions (Tim Titov) 4. int/ext dial tones not synchronous (georg(a)riseup.net) 5. Re: Radio stream binding ? (Sherif Omran) 6. Re: Feedback (jay binks) 7. Re: Ringback on PRI line (Ryan V) Pastebin the actual console output, including the stuff leading up to when you press #. -MC On Thu, Dec 22, 2011 at 3:38 PM, Rodney <notlikeme75(a)yahoo.com> wrote: everytime i use transfer from caller controls i get "destination out of order" on console. so i use the execute extension and get problems. is this correct? > > ><control action="execute_application" digits="#" data="transfer 759 xml default"/> > > >`rodney > > > hello guys, can any body help me to use bind_meta_app to connect a channel to a radio stream using gnuradio.org project? thanks in advance regards, Sherif When faxing with freeswitch the directory and containing file need to be readable by "other". Otherwise, mod_spandsp complains about the file being inaccessible. You can also see on mod_spandsp wiki people doing chmod o+r $TMPFAX. Why can't this be run under the freeswitch user only, so that you can do basic access control? Hello all, I made the first experiences with the dialplan, and after some while, I really got calls from the outside routed to one of my sip phones. However, my phone is ringing something like 1-3 seconds earlier than the dial tone the caller is hearing. I googled around, but found no solution. As a start, I just used this extension: <extension name="Local_Extension"> <condition field="destination_number" expression="^(4930XXXXXXXX)$"> <action application="bridge" data="user/12(a)192.168.1.93"/> </condition> </extension> Thanks, Georg Hello all, does mod_scout play asf streams? thanks --- On Fri, 12/23/11, Sherif Omran <sherifomran2000(a)yahoo.com> wrote: >From: Sherif Omran <sherifomran2000(a)yahoo.com> >Subject: [Freeswitch-users] Radio stream binding ? >To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> >Date: Friday, December 23, 2011, 3:00 AM > > >hello guys, > >can any body help me to use > >bind_meta_app > >to connect a channel to a radio stream using gnuradio.org project? > >thanks in advance > >regards, >Sherif > >-----Inline Attachment Follows----- > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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 > Agreed... put what you have currently on Git Hub so we can check it out... you sounded like you were already fairly advanced in your progress, so throwing something that mostly works on Git Hub should only take an hour. can wait to check it out... Jay On 22 December 2011 11:34, Jeff Lenk <jeff(a)jefflenk.com> wrote: Chad, > >Any update on this? Looking forward to seeing some of your contributions and >perhaps being able to merge them into the mainline code. > >Sincerely, >Jeff > >-- >View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Feedback-tp6941381p7117251.ht… > >Sent from the freeswitch-users mailing list archive at Nabble.com. > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > > >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 > -- Sincerely Jay Adding following entries into inbound dial plan solved the problem. <action application="ring_ready" /> <action application="set" data="ringback=data=ringback=${in-ring}"/> <action application="set" data="instant_ringback=true"/> Thanks, Venkatesh K On Fri, Dec 23, 2011 at 4:50 AM, Michael Collins <msc(a)freeswitch.org> wrote: This is probably a PRI config issue. Doublecheck the PRI protocol settings with your provider. Also, get a d-channel trace and drop it on pastebin. (see the freetdm wiki page for d-chan capture instructions.) > > >I'm sure Moy or one of the guys here will be happy to take a look. > > >-MC > > >On Thu, Dec 22, 2011 at 12:06 AM, Ryan V <ryan(a)kaevee.com> wrote: > >Hi, >> >>We are running freeswitch with Sangoma PRI and Analog cards. There is no ring back for calls coming in on PRI line. Calls coming in on FXO lines are fine. Calling party gets ring back. >> >>I have added following to file conf/dialplan/public/00_inbound_did.xml >> >> <action application="set" data="instant_ringback=true"/> >> <action application="set" data="ringback=%(2000,4000,440.0,480.0)"/> >> >>Any suggestions to resolve this problem? >> >>Thanks, >> >>Venkatesh K >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting(a)freeswitch.org >>http://www.freeswitchsolutions.com >> >>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>http://www.cudatel.com >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>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 >> >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >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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Paging and Intercom
by Rob Hutton 23 Dec '11

23 Dec '11

How is everyone doing intercom and paging? We are having problems with Grandstream phones where they put an active call on hold to answer the second call if they see the "auto answer" headers. Do all brands behave this way or is this something unique to Grandstream? I was thinking earlyier about handling this from the switch, but since current phones allow you to log in to multiple accounts even across multiple switches, it would be impossible to do this from the switch level...

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Excluding busy extensions from intercom
by Rob Hutton 23 Dec '11

23 Dec '11

I am trying to get a basic "All Page" setup working and have used the wiki and example configs to put together the following dialplan: <extension name="main_number_10" continue="true"> <condition field="destination_number" expression="^\*40$"> <action application="set" data="api_hangup_hook=conference 40 kick all"/> <action application="answer"/> <action application="set" data="conference_auto_outcall_caller_id_name=${effective_caller_id_name}"/> <action application="set" data="conference_auto_outcall_caller_id_number=${effective_caller_id_number}"/> <action application="set" data="conference_auto_outcall_timeout=5"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> <action application="set" data="sip_exclude_contact=${presence_id}"/> <action application="conference_set_auto_outcall" data="user/115@$${location_1}"/> <action application="conference_set_auto_outcall" data="user/100@$${location_1}"/> <action application="conference_set_auto_outcall" data="user/warehouse@$${location_1}"/> <action application="conference" data="40@conference_2"/> <action application="conference" data="40 kick all"/> </condition> </extension> Everything works as expected, except that intercom forces any active calls to hold, and likewise forces the originating call on hold. Is there a way of excluding any active users? I have tried both ${network_addr} and ${presence_id} in the sip_exclude_contact variable, and neither causes the orginator to be excluded. Console log is here: http://pastebin.freeswitch.org/18022

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Feedback
by Chad Vogel 23 Dec '11

23 Dec '11

Hello, I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users. Here are some of the changes we have made: * We converted FS core to C++ and having it compiling to .NET 4.0 CLI * FS now runs inside its own namespace * We replace FS_CLI with a PowerShell shell app * Supports SQL Server support via the Native SQL Server Client API * Support for windows clustering (up to 32 node active/active cluster) * Ties more directly into the Win32 API and has less reliance on open source Libs * Replaces OpenSSL sockets with Windows encrypted sockets * Added windows performance monitoring * No longer need to use mod_managed for managed modules * Memory management relies on .Net garbage collection * Added support for Microsoft Speech * VoiceXML 3.0 and SCXML support * No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages) * Fixed RTP clock timing issues in virtualized environments * New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)

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