- FreeSWITCH-users - FreeSWITCH Mailing Lists

Case sensitivity in dialplan
by Richard Mace 21 Jul '13

21 Jul '13

Hi, If I wanted to add a "group_dial_" extension in the dialplan, would the "group name" "have" to be lower case like the default entries or doesn't it matter? Thanks in advance Richard

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Sip clients evaluation using freeswitch
by Karl Schmidt 20 Jul '13

20 Jul '13

I need good softphone clients for both Windoze and linux desktops - They hope they can have video support. I'm about to evaluate the current releases of following clients: http://linphone.org/ http://sflphone.org/features http://qutecom.org/ https://jitsi.org/ http://www.ekiga.org/ https://wiki.gnome.org/Empathy http://www.icanblink.com/ Any comments before I get started? Or clients I may have missed? Tests I might run? -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 Big business = big government. Big business exists as the result of government regulations designed to produce an unofficial cartel for friends of the political class. kps --------------------------------------------------------------------------------

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call hanging up on transfer
by Ashish gautam 19 Jul '13

19 Jul '13

I am transferring the current session to mod_callcenter like this: $session->transfer("47673503","XML","public"); where 47673503 is the extension number for the callcenter. The call is getting hungup as soon as the program sees the above line. The console shows this: 2013-07-15 11:38:04.221516 [DEBUG] switch_ivr.c:1815 (sofia/internal/ 5000(a)10.1.30.236) State Change CS_EXECUTE -> CS_ROUTING 2013-07-15 11:38:04.221516 [DEBUG] switch_core_session.c:1340 Send signal sofia/internal/5000(a)10.1.30.236 [BREAK] 2013-07-15 11:38:04.221516 [DEBUG] sofia.c:5741 Channel sofia/internal/ 5000(a)10.1.30.236 entering state [completed][200] 2013-07-15 11:38:04.221516 [DEBUG] sofia.c:5741 Channel sofia/internal/ 5000(a)10.1.30.236 entering state [ready][200] 2013-07-15 11:38:04.221516 [DEBUG] switch_core_session.c:860 Send signal sofia/internal/5000(a)10.1.30.236 [BREAK] 2013-07-15 11:38:04.221516 [NOTICE] switch_ivr.c:1822 Transfer sofia/internal/5000(a)10.1.30.236 to XML[47673503@public] 2013-07-15 11:38:04.221516 [DEBUG] switch_cpp.cpp:844 transfer result: 0 Please help. -- Ashish Gautam

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log-auth-failures warning not in freeswitch.log
by Lloyd Aloysius 19 Jul '13

19 Jul '13

Hi All, I have the following parameter in the profile <param name="log-auth-failures" value="true"/> and log file configuration I have the following - debug,info,notice,warning,err,crit,alert Default values. I see the log auth failures in console. But not writing into the freeswitch.log file. How to fix this? Thanks Lloyd * *

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freeTDM config
by Karl Schmidt 19 Jul '13

19 Jul '13

I figured this out - but many others would have given up.. I needed to answer incoming calls on a FXO port. From the wiki page http://wiki.freeswitch.org/wiki/FreeTDM There are several places where there is a data="freetdm/1/1/${destination_number} bit, but I'm not understanding the syntax of /1/1/ bit means? I think it is supposed to mean freetdm/span#/port#/ What makes the span# what it is ( the order in freetdm.conf or the name of the span? ) The port number corresponds to the defined channel number? (if so why two names? ) At a different place in the wiki there is expression="FreeTDM/1/[34]" This time there is a different capitalization for the word freetdm (with meaning?)- and the square brackets might mean the port number applies only to this leg? I did get it working with this extension: <extension name="fxo3">  <condition field="chan_name" expression="FreeTDM/2:1/" >  <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="11 XML default" /> </condition> </extension> I'm guessing that the 2:1 means the second span in freetdm.conf and the first extension of that span? (Now that is rather cryptic - even for this aging assembly programmer ) There is nothing like that on the wiki page - What in the world am I missing? BTW I would think that the way this SHOULD work would be as in: data="freetdm/FXO/3/${destination_number} freetdm.conf - spans : ;Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express) [span zt FXS] trunk_type => FXS fxs-channel => 1 fxs-channel => 2 [span zt FXO] trunk_type => FXO fxo-channel => 3 fxo-channel => 4 Give me some clear answers and I can write this up for others. -------------------------------------------------------------------------------- Karl Schmidt EMail Karl(a)xtronics.com Transtronics, Inc. WEB http://secure.transtronics.com 3209 West 9th Street Ph (785) 841-3089 Lawrence, KS 66049 FAX (785) 841-0434 If one rejects laissez faire on account of mans fallibility and moral weakness, one must for the same reason also reject every kind of government action. Ludwig von Mises --------------------------------------------------------------------------------

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30 Second call drop.
by Paul 19 Jul '13

19 Jul '13

Hi guys, After hours of googling and trying every different config options I need some help! My scenario is as follows: Remote Network 1: 192.168.1.0/24 (10.8.0.X/32 via openVPN tun) Remote Network 2: 192.168.5.0/24 (10.9.0.X/32 via openVPN tun) Main Network (where Freeswitch resides): 10.0.0.0/24 My sip gateway/proxy: 10.0.0.40 (kamailio) Freeswitch: 10.0.0.34 I can fully use the VPN from both networks (ssh, web gui control, etc) phones can register, however here are my problems: When calling incoming DID none of the extensions ever get ring, it skips to voicemail... when leaving a message voicemail cuts of after 30-31 seconds exactly. I am wondering if freeswitch isn't detecting my client sending audio and interprets it as silence and therefore hangs up. The voicemail message itsel does contain everything said right up to the hang up. Sometimes I can call between extensions if they are on the same side of the VPN subnet (say 101 calling 101 and they both are coming from 192.168.1.0/24 via 10.8.0.0/24) If 100 is on 10.8.0/24 and 101 is on 10.9.0.0/24 freeswitch just says user is unavailable. I'm suspecting this is a NAT issue, all the reading I have done talks about "external sip and rtp ip" in my case my trunk hooks up from private ip to private ip, it's the kamailio that's supposed to do the NAT (which I have verified works fine, I can place outgoing calls through it from other PBXs no problem with two way audio and working as it should be). I'm not sure if there is a way to enable nat or not with freeswitch. One thing I have done (as I've found this worked for some people) is set all references to sip-ip ext-sip-ip, rtp-ip and ext-rtp-ip to 10.0.0.34 (my freeswitch eth0). There are no other adapters on the box. I am hoping that as this is my first time setting up freeswitch that one of you freeswitch experts can point me in the right direction :) Thanks in advance guys!

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Re: [Freeswitch-users] Sip trunk not ringing
by Jose Suero 19 Jul '13

19 Jul '13

This is my setup Sip provider ---- Nat(ec2)--- freeswitch --- opensips/rtpproxy --- Nat(ec2)--- sip phone Nikolay Rogoshchenkov <nickolayr(a)gmail.com> wrote: >_________________________________________________________________________ >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

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Sip trunk not ringing
by Jose Suero 19 Jul '13

19 Jul '13

I have a sip trunk configured on free switch as a gateway when some one calls the DID for that number call comes in and is bridge to an opensips server which sends it to the phone. Problem is that the person waiting online never hears the phone actually ringing. There's just silence until the other person picks up. Where do I configure a ring??

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Re: [Freeswitch-users] Any Interest in an extra Day @ Cluecon for New User Training?
by Jose Suero 19 Jul '13

19 Jul '13

This would have been good. I'll be on opensips training on Monday but would've love to see some freeswitch training Ken Rice <krice(a)freeswitch.org> wrote: >_________________________________________________________________________ >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

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Wrong Port Number?
by Brian Foster 19 Jul '13

19 Jul '13

Every so often (and I've been paying attention to it closer lately), "sofia status" in fs_cli reports something similar to this: http://pastebin.freeswitch.org/21171 If you look at the external profile's port number, it's the wrong one. It should be 5080. But this does not just happen on the external profile, it happens occasionally ON EVERY profile. Sometimes a few of the profiles have this affliction at the same time. This happens when mod_sofia or freeswitch itself is reloaded. If I'm not careful (i.e. I leave it this way), phones do not register, no one can answer important phone calls, the world crashes, etc. The bad thing about this is that I cannot for the life of me figure out how to change it manually. Everything is set correctly in vars.xml and the profiles as far as ports go (internal uses 5060, doublenat uses 5090, external uses 5080). Sometimes I restart freeswitch and get lucky, but most of the time no dice. Does anyone have any idea on this? freeswitch@internal> version FreeSWITCH Version 1.2.8+git~20130503T183844Z~65f0d16dee (git 65f0d16 2013-05-03 18:38:44Z) Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana

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