August 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Hardware phones that do 48Khz Celt: Was Re: Is there any license G729?
by Craig Askings 29 Sep '11

29 Sep '11

Are there any hardware phones that support 48 Khz Celt and automated/mass deployment? Craig. 2009/6/30 Jason White <jason(a)jasonjgw.net>: > Brian West <brian(a)freeswitch.org> wrote: >> Everyone has this need for lower bandwidth calls... I tend to march the >> other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a >> single ulaw call) > > 48khz Celt (celt@48000 in your codec preferences) sounds wonderful with > FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. > This also works well in 48khz conferences. > > I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, > G.722, G.722.1 and (my current favourite) Celt all the way. > > > _______________________________________________ > 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 > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig(a)overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365

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xml_curl and gateways
by Peter P GMX 21 Feb '11

21 Feb '11

According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to set up gateways dynamically, but I do net get it to work: I always get : "Invalid profile" My assumptions for a right xml answer back to FS are as follows, but I think at least one of it is false: 1. ) I start with <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> 2.) I do not want to change settings, so I do NOT add a <settings> tag 3.) I want to enhance an existing profile, so I add a profile tag (<profile name='external'>) 4.) I assume that I can have an existing profile "external" in the xml conf files and can mix additional parts dynamically into the same profile 5.) Within the profile I want to add gateways, so I add a gateway tag (<gateways>) 6.) Then I add the gateways 7.) I add the closing tags Which one of the assumptions if wrong? The request I receive is as follows: {"key_name"=>"name", "action"=>"directory", "tag_name"=>"configuration", "key_value"=>"sofia.conf", "controller"=>"xml_curls", "hostname"=>"freeswitch", "section"=>"configuration"} I only answer once at startup with the follwing: <document type='freeswitch/xml'> <section name='configuration'> <configuration name='sofia.conf' description='RailsAdditionalConf'> <profile name='external'> <gateways> <gateway name="freeswitch2"> <param name="username" value="1007"/> <param name="realm" value="192.168.0.56:5061"/> <param name="password" value="xxxx"/> <param name="proxy" value="192.168.0.56"/> <param name="expire-seconds" value="30"/> <param name="register" value="true"/> <param name="register-transport" value="tls"/> <param name="retry-seconds" value="30"/> </gateway> </gateways> </profile> </configuration> </section> </document> Does anybody have a correct XML answer for a gateway and can post it? Best regards Peter

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Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

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Direct inward dialling
by Peter P GMX 07 Jun '10

07 Jun '10

Dear all, when I get an incoming call from my SIP provider I do receive an invite on the trunk number, e.g. in Germany 0xxxxx9. However I have an extension block of 0xxxxx90 to 0xxxxx999. In the dialplan I checked the condition field="destination_number" but this compares to the number of the trunk 0xxxxx9, so I do not get the direct inward dialling extensions. The SIP to-header contains the direct inward dialling number 0xxxxx90 to 0xxxxx999 but I have no success in accessing it in the dialplan. I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri, but no success. They are all empty. Any idea which variable to parse? Best regards Peter

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zrtp endpoints have different sas through fs 1.0.4
by Harondel J. Sibble 19 Jan '10

19 Jan '10

I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp capable endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone client, 2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i (symbian s60) and an O2 Xda Flame (windows mobile 5). All 3 endpoints are registered with FS using the default extensions of 1000- 1003 With global_setvar zrtp_secure_media=true the zrtp negotiation between end points happens but the SAS never matches,below is console output for a call between 2 of the endpoints 2009-08-23 14:10:17.643073 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1003(a)10.12.13.45! 2009-08-23 14:10:21.257568 [NOTICE] sofia.c:3794 Channel [sofia/internal/sip:1000@10.12.13.166:5062] has been answered 2009-08-23 14:10:21.275521 [NOTICE] switch_ivr_originate.c:2015 Channel [sofia/internal/1003(a)10.12.13.45] has been answered 2009-08-23 14:10:22.232053 [WARNING] mod_sofia.c:810 We were told to use ptime 20 but what they meant to say was 80 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2009-08-23 14:11:34.496118 [NOTICE] sofia.c:322 Hangup sofia/internal/sip:1000@10.12.13.166:5062 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-08-23 14:11:34.512100 [NOTICE] switch_ivr_bridge.c:1016 Hangup sofia/internal/1003(a)10.12.13.45 [CS_EXECUTE] [NORMAL_CLEARING] 2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1086 Session 16 (sofia/internal/sip:1000@10.12.13.166:5062) Ended 2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/sip:1000@10.12.13.166:5062 [CS_DESTROY] 2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1086 Session 15 (sofia/internal/1003(a)10.12.13.45) Ended 2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1003(a)10.12.13.45 [CS_DESTROY] Of note, with the endpoints registered through the Ekiga sip server, the sas DOES match on both ends. With global_setvar zrtp_secure_media=false, the endpoints can't detect a zrtp peer. Reading the list archives hasn't enlightened me. I see this comment from 2008 http://www.nabble.com/Freeswitch-and-Twinkle-and-ZRTP- td18518140.html#a18518343 On Jul 17, 2008, at 4:23 PM, Michael Jerris wrote: > it should in bypass_media or proxy_media modes. in the other modes we > are in the media path and would not know how to handle the encrypted > packets. > > Mike Is this still relevant? Or is there some other setting not covered here http://wiki.freeswitch.org/wiki/ZRTP to make this work properly? I ask firstly about this in the context of a peer 2 peer zrtp communication between the endpoints, then secondly in the case of FS acting as a trusted middleman as in section 2 here http://www.zfoneproject.com/docs/asterisk/man/html/u_guide.html#passthrough Lastly how does one implement the security enrollment as noted above with FS -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help(a)pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice)

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How can I join two freeswitch on two servers?
by Edmar Cruz 18 Dec '09

18 Dec '09

I like to connect two freeswitch, call each other, communicate and vice versa. Can you give me an example for that? Thanks -- View this message in context: http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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OSP Interop w/ Trans Nexus
by Gregory Boehnlein 02 Dec '09

02 Dec '09

Anyone know if Freeswitch can work w/ OSP and use something like Transnexus for CDR/Rating?

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Is there any license G729?
by Edmar Cruz 04 Nov '09

04 Nov '09

Hi, Is there any available license G729 for freeswitch? I need it to match G729 of Asterisks? If any please help me and instructions how to install this thing. Thanks -- View this message in context: http://www.nabble.com/Is-there-any-license-G729--tp24251951p24251951.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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EXCHANGE_ROUTING_ERROR -- how to diagnose?
by Matthew Fong 23 Oct '09

23 Oct '09

A few users of mine have been getting hung-up on after leg b of the bridge hangsups. I looked in the logs and they are being hungup with an EXCHANGE_ROUTING_ERROR hangup cause. The problem is only occurring intermittently and only when both leg a and leg b are both passed thru external gateways (the problem does *not* exist if leg a is a softphone connected directly to FS). I'm wondering if there are any known incompatibilities with equipment out there that would cause this error. Otherwise, I assume the best way to diagnosis this further is to ngrep, which I'll do, but thought I'd ask here first. Thanks. --matt hello hunter - hosted predictive dialer & voice broadcasting http://www.hellohunter.com

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Re: [Freeswitch-users] 2 voicemail questions
by Mark Campbell-Smith 20 Oct '09

20 Oct '09

Hi! >> 1. Can I email the voicemail message to multiple email addresses? A comma separated list does not work in the extensions.xml file (1000.xml), but it does work if I hard code the email addresses into the notify-voicemail.tpl file. Could this be added to the switch so that it can handle comma separated lists? >> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? >api Hangup hook? i want the 'voicemail' application to appear to call the extension to notify the user that there is a waiting message. This is an extract from my dialplan.xml: <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/>  <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> <condition field="${vm_boxcount(${destination_number}@${domain_name})}" expression="^(1)$"> <action application="log" data="MSGBOX ${vm_boxcount(${dialed_extension}@${domain_name})}"/> <action application="set" data="api_hangup_hook=originate sofia/internal_nat/${dialed_etension}%${domain_name} default default Message 4000 4000 3"/> This only works if the B leg (ie voicemail application) hangs up first. This would be an unusual situation and does not achieve what I want... is there any other way to achieve this? Thanks > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > <param name="vm-mailto" value="me(a)myemail.com"/> Try a comma sep. list. Not sure if it will work. > > 2. How can I make Freeswitch dial a number AFTER a voicemail is left? api Hangup hook? I g > From: Brian West <brian(a)freeswitch.org> > On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote: > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > <param name="vm-mailto" value="me(a)myemail.com"/> > Try a comma sep. list. Not sure if it will work.

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