February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Conference Call Tomorrow
by Michael Collins 07 Feb '12

07 Feb '12

Hello all! Just a reminder that we are having Travis Cross come back and talk to us about git again. I think we've all had some time to digest the information from his first presentation and hopefully we have some intelligent questions for him. I'm wondering if he can show us an example of git bisect. :) Here's the agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_08 FYI, I am trying to get caught up on the release notes so I've found a few new FreeSWITCH things that I'd like to have you all help me get documented. :) Thanks, Michael

1 0

Re: [Freeswitch-users] sip_invite_from_params
by George Sapak 07 Feb '12

07 Feb '12

i already have it setup, my calls go through to the right extensions and I have done some rewitting in public.xml file, but I have having some wierdness on the handsets with call ID, when the phone rings the ID show correctly but when its picked up it shows it with the ;national... sting included. Here is how I have done the routing and rewritte: <extension name="105"> <condition expression="public" field="context"/> <condition field="destination_number" expression="^403XXXXXXX(.+)$"> <action application="transfer" data="105 XML default"/> </condition> </extension> public.xml <extension name="set caller id" continue="true"> <condition field="caller_id_number" expression="^(\d+)"> <action application="set" data="caller_id_number=$1"/> <action application="set" data="effective_caller_id_number=$1"/> </condition> </extension> Thank You, George ----- Original Message ----- From: "Brian West" <brian(a)freeswitch.org> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Tuesday, February 7, 2012 8:43:54 AM Subject: Re: [Freeswitch-users] sip_invite_from_params Why? What is your goal? This should be a simple dialplan rewrite really. /b On Feb 7, 2012, at 9:27 AM, George Sapak wrote: can someone give me an example in use, I have looked at the wiki and its pretty unclear, I am trying to remove a string in an invite -> ;phone-context=national INVITE sip:403XXXXXXX;phone-context=national@10.185.16.170;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.185.16.169:5060;branch=z9hG4bKght6ecqg4eh7d5sluaj9cl2rr4 From: "Caller"< sip:403YYYYYYY;phone-context=national@10.185.16.169;user=phone;isup-oli=00 >;tag=SDj6q1a01-98964 To: < sip:403XXXXXXX;phone-context=national@10.185.16.170;user=phone > Call-ID: SDj6q1a01-2afa0c6426482b0cfa2885f32878c86c-o0t3g30 CSeq: 62218 INVITE Content-Type: application/sdp Contact: < sip:403YYYYYYY;phone-context=national@10.185.16.169:5060;transport=udp > User-Agent: Nortel SESM 14.0.6.0 Max-Forwards: 19 Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption,r eplaces,100rel,tdialog Allow: UPDATE,REFER x-nt-corr-id: 871f3514-29e7-1b21-a633-000e0cb7d3a0 x-nt-location: -1 Content-Length: 206 Route: < sip:403XXXXXXX;phone-context=national@10.185.16.170:5060;user=phone;lr > Thank You, George -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian(a)freeswitch.org 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

3 2

needs some advice to secure my system
by Bob Smith 07 Feb '12

07 Feb '12

Kristian, Huh ? I was the one who suggested the firewall filtering and OpenVPN in the first place, no need to teach granny how to suck eggs. ;-) Bob

2 1

Conference Auto Outcall to External Numbers
by Ryan Watkins 07 Feb '12

07 Feb '12

Morning/Afternoon/Evening all... I've been trying to setup a conference call that would use the auto outcall function, and it would call external numbers (like their cell). I've tried to setup extensions for the individuals, and set the call forwarding to their external number. Although the auto outcall function will successfully call the extensions when they are set to an internal SIP phone, when call forwarding is enabled the conference call will not call the external number. Any ideas on how to accomplish this? Thanks a lot! Ryan

3 9

Getting USER_NOT_REGISTERED for no reason ?
by Bob Smith 07 Feb '12

07 Feb '12

Hi Brian, Have tried your various suggestions without luck. Have also tried replacing <action application="transfer" data="1000 XML default"/> with <action application="bridge" data="sofia/sip/1000%openvpn_udp:_:sofia/sip/1000%internal"/> in the dial plan and that doesn't work either Everything resolves fine though...... freeswitch@internal> expand echo ${sofia_contact(openvpn_udp/1000)} sofia/openvpn_udp/sip:1000@10.82.1.6:3072;line=2sf2mz02 freeswitch@internal> expand echo ${sofia_contact(internal/1000)} sofia/internal/sip:1000@10.14.2.2:12592;rinstance=5e131f94a204160f And yes I have looked through the logs, but there's nothing as blatantly obvious in there as you seem to think there is ? Bob

2 1

Wrong system variables assignations on FreeSwitch system with two NICs
by rico-freeswitch＠ricozome.net 07 Feb '12

07 Feb '12

Hi guys, I'm facing a stupid issue on a fresh installed Debian Squeeze (amd64) server with two NICs, and FreeSwitch bottstraped from Git tree : - eth0 is connected to the local LAN (172.16.96.3/24) - eth1 is connected to Internet and have a public IP address. To be exact, eth0 is on a DMZ, but anyway, it's a RFC1918 address. FS is : FreeSWITCH Version 1.0.head (git-d2c9fb5 2012-02-06 14-12-22 -0600) The problem is when I start the server with its default config, mod_sofia automatically binds into the public IP address, and nothing listens on the local IP address ! thougth the CLI, I get this : freeswitch@internal> sofia status Name Type Data State ================================================================================================= 194.79.xxx.xxx alias internal ALIASED internal profile sip:mod_sofia@194.79.xxx.xxx:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) external profile sip:mod_sofia@194.79.xxx.xxx:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG ================================================================================================= 3 profiles 1 alias freeswitch@internal> eval ${local_ip_v4} 194.79.xxx.xxx I have to precise it is my first attemps to install FS, and I'm discovering the software, so I may forget something... But I didn't found a way to force FS to link "internal" profile with the local IP address. I also could modify sip_profiles/internal.xml and hardcode the correct IP address, but my feeling is it's not the right way, and I guess it would probably have some side effects on the setup ? Last, I tried both old 1.0.6 branch and lastest Git tree, both with compiled packages "ala" Debian way, and bootstraped, and I always got the same behavior... So my question is simple : is this normal, ? and how should I handle FS setup with mutiple NICs ? Thanks for your help, -- Eric Belhomme

2 1

FreeSwitch state machines for CHANNEL_STATE and CHANNEL_CALLSTATE
by David Artuñedo Guillén 07 Feb '12

07 Feb '12

Hi, I'm new to FreeSwitch world. I'm looking at using the mod_event_socket interface to write some simple call control app. The Events I'm interested at are CHANNEL_STATE and CHANNEL_CALLSTATE to recognize the status of the calls and how they progress. I have not found any docs describing the state machines for channels and Calls. I look up in the Source code as well and couldn't find the place where the state machine is implemented. Does anybody know where can I find this info? EG, when setting up some call through FS these are the events I am getting with the uuids associated: bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_INIT bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE RINGING bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_ROUTING bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_CONSUME_MEDIA 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_CALLSTATE ACTIVE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE EARLY bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE ACTIVE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_EXECUTE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_HIBERNATE 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_HIBERNATE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_RESET 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_RESET 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_SOFT_EXECUTE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_SOFT_EXECUTE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_CONSUME_MEDIA bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_EXCHANGE_MEDIA bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE HANGUP bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_HANGUP 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_PARK bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_REPORTING bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE DOWN bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_DESTROY Thanks in advance. David

3 3

Invalid UTF-8 character
by Vishal Kakkar 07 Feb '12

07 Feb '12

Hi All, Anybody have idea, what could cause this WARNING.. I am getting this one since i upgraded to latest git last week, dont know if its related to that though. *switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it* There are 100s of such lines getting printed on each call. So want to fix the same before i make it live. Thanks a lot.

2 1

needs some advice to secure my system
by Bob Smith 07 Feb '12

07 Feb '12

Hello Philippe, The idea I am currently working towards implementing is : - SIP Origination / Inbound SIP = IP range ACL with carrier - User Origination / Devlivery = OpenVPN + SNOM Handsets (they have a built-in OpenVPN client, quite cool !) You can lock down OpenVPN quite tight so it hardly reponds at all to unauthorised requests. I have only just started my testing, but other than some issues with inbound calls and multiple profiles that I'm trying to iron out at the moment, everything seems to be working ok. Bob

4 5

sip_invite_from_params
by George Sapak 07 Feb '12

07 Feb '12

can someone give me an example in use, I have looked at the wiki and its pretty unclear, I am trying to remove a string in an invite -> ;phone-context=national INVITE sip:403XXXXXXX;phone-context=national@10.185.16.170;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.185.16.169:5060;branch=z9hG4bKght6ecqg4eh7d5sluaj9cl2rr4 From: "Caller"<sip:403YYYYYYY;phone-context=national@10.185.16.169;user=phone;isup-oli=00>;tag=SDj6q1a01-98964 To: <sip:403XXXXXXX;phone-context=national@10.185.16.170;user=phone> Call-ID: SDj6q1a01-2afa0c6426482b0cfa2885f32878c86c-o0t3g30 CSeq: 62218 INVITE Content-Type: application/sdp Contact: <sip:403YYYYYYY;phone-context=national@10.185.16.169:5060;transport=udp> User-Agent: Nortel SESM 14.0.6.0 Max-Forwards: 19 Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption,r eplaces,100rel,tdialog Allow: UPDATE,REFER x-nt-corr-id: 871f3514-29e7-1b21-a633-000e0cb7d3a0 x-nt-location: -1 Content-Length: 206 Route: <sip:403XXXXXXX;phone-context=national@10.185.16.170:5060;user=phone;lr> Thank You, George

2 1