June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Default Install Extensions NEWBIE Question
by Mehma Sarja 11 Jun '11

11 Jun '11

The default install on ubuntu renders a working system servicing 19 extensions. Can I use this as-is in a production environment which only requires within-LAN, inter-departmental voice connectivity? Mehma

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fetch password
by Sam 11 Jun '11

11 Jun '11

hello, If i want to store the vm pasword in database , how should i go ahead with the dialplan to fetch the pasword. <domain name="mydomain.com" <user id="101"> <params> <param name="vm-password" value="12345"/> </params> </user> </domain> Regards San

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Free ZRTP implementation?
by Janne Snabb 11 Jun '11

11 Jun '11

Hi, I wanted to compile FreeSWITCH with ZRTP support, but it looks like Zimmermann's commercial libzrtp has not been available for "evaluation" download since the beginning of this year. I found a couple of older messages asking about it in the mailing list archives, but no real solution. Does anyone have patches or is anyone working on a module which would use a free ZRTP implementation such as ZORG or GNU ZRTP? If not, I might be able to have a look at it at some point... no promises though. -- Janne Snabb / EPIPE Communications snabb(a)epipe.com - http://epipe.com/

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FIFO suddenly stopped working properly
by Joao Leme 10 Jun '11

10 Jun '11

Hi there, No change nor updates have been made in months. Everything was working great but now FIFO is not ringing the extensions while the first call is not HUNG UP...meaning, first call to go to FIFO rings the agents, and one person answer...when the second call comes in, it goes to fifo queue (caller listens the music) but no one is aware of the call, nor even the agents that are available. Only after the first call hangs up the 2nd goes in. I tried troubleshooting but no help. What I also noticed is that before, to display the caller ID on my sip client I had to use: <action application="set" data="origination_caller_id_number=${caller_id_number}"/> <action application="set" data="origination_caller_id_name=${caller_id_name}"/> but now, while testing, the caller id was showing when calling fifo without setting those parameters, but before it would just show the fifo queue name. All I can say is that I can't figure out why and how to fix these behavior if I have done no change to FreeSWITCH nor the config files. I also tried downloading the latest GIT and got the same response. THANKS, John

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Strange and annoying problem.
by Carmelo Fischetti - Digi Unit 10 Jun '11

10 Jun '11

Hello to everybody, I'm new to this list but I've been experimenting with FreeSWITCH for 3 months. My environment consists of a FreeSWITCH server and some (4-5) SIP softphones. We are noting a strange and annoying problem. When in call, after exactly 15 minutes, FreeSWITCH CLI logs the following lines: 2011-06-10 20:10:51.689289 [DEBUG] sofia.c:4769 Channel sofia/internal/1010(a)fscch00.dyndns-server.com entering state [calling][0] 2011-06-10 20:10:51.789283 [INFO] sofia.c:748 sofia/internal/1010(a)fscch00.dyndns-server.com Update Callee ID to "Outbound Call" <1010> 2011-06-10 20:10:51.789283 [DEBUG] sofia.c:4769 Channel sofia/internal/1010(a)fscch00.dyndns-server.com entering state [ready][200] 2011-06-10 20:10:51.789283 [DEBUG] sofia.c:4780 Remote SDP: v=0 o=amsip 0 0 IN IP4 192.168.1.100 s=talk c=IN IP4 192.168.1.100 t=0 0 m=audio 12100 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=setup:both a=ptime:20 2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:4658 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] 2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:2724 Already using GSM 2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:4772 Set 2833 dtmf send/recv payload to 101 2011-06-10 20:10:51.789283 [DEBUG] sofia.c:5238 Processing updated SDP 2011-06-10 20:10:51.789283 [DEBUG] sofia_glue.c:3009 Audio params are unchanged for sofia/internal/1010(a)fscch00.dyndns-server.com. My first question is: what does this mean? Is there some configuration parameter (where?) that triggers some event after 15 minutes? Actually, for unencrypted calls, nothing happens but when using SRTP, the log changes to: 2011-06-10 20:32:32.825843 [DEBUG] sofia.c:4769 Channel sofia/internal/1010(a)fscch00.dyndns-server.com entering state [calling][0] 2011-06-10 20:32:32.965833 [INFO] sofia.c:748 sofia/internal/1010(a)fscch00.dyndns-server.com Update Callee ID to "Outbound Call" <1010> 2011-06-10 20:32:32.965833 [DEBUG] sofia.c:4769 Channel sofia/internal/1010(a)fscch00.dyndns-server.com entering state [ready][200] 2011-06-10 20:32:32.965833 [DEBUG] sofia.c:4780 Remote SDP: v=0 o=amsip 0 0 IN IP4 192.168.1.100 s=talk c=IN IP4 192.168.1.100 t=0 0 m=audio 12665 RTP/SAVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=setup:both a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:YjUwMzQyNWU4YmY2ZjhiOGYyYTQ1MmZiOGNhMWEy a=encryption:optional a=ptime:20 2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:4463 Existing key is still valid. 2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:4658 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] 2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:2724 Already using GSM 2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:4772 Set 2833 dtmf send/recv payload to 101 2011-06-10 20:32:32.965833 [DEBUG] sofia.c:5238 Processing updated SDP 2011-06-10 20:32:32.965833 [DEBUG] sofia_glue.c:3009 Audio params are unchanged for sofia/internal/1010(a)fscch00.dyndns-server.com. 2011-06-10 20:32:33.485800 [ERR] switch_rtp.c:2560 Error: SRTP unprotect failed with code 7 (auth check failed) (the last line repeated indefinitely)... and one leg of the call is dropped (one of the phones cannot hear the other). Since this happens with various softphones, it's a server issue, I suppose. Any help would be greatly appreciated. Thanks in advance. Carmelo

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CISCO 2811 - FreeSwitch - IVR
by Schenk, Oliver 10 Jun '11

10 Jun '11

Hey all, I have successfully managed the following setup ... at least to a certain extent. Phone line -> CISCO VIC2-4FXO -> CISCO 2811 -> Sip -> Windows Server 2008 -> FreeSwitch -> Managed IVR (IAppPlugin) When I dial say the phone line's number (49794) it connects me to extension 1024 which has the managed C#.NET IVR sitting behind it. That works fine while everything is happy. However, when I hang up my phone. My IVR is still happily talking to no one until the relevant timeouts expire. Is this the normal way to implement it? Just wait for timeout? Or do I have some CISCO configuration issues? Shouldn't the CISCO tell FreeSwitch that "Hey ... the guy on the line has put down his phone receiver so you can end your session now."? Any comments? voice-port 0/3/0 input gain 10 output attenuation 10 no comfort-noise cptone AU connection plar opx 1024 impedance complex1 ! voice-port 0/3/1 ! voice-port 0/3/2 ! voice-port 0/3/3 ! ! ! ! ! dial-peer voice 2 voip destination-pattern .T session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711ulaw no vad ! ! sip-ua credentials username 1000 password * ******* realm default authentication username 1000 password * ******** retry invite 2 retry register 10 timers connect 100 registrar ipv4:192.168.255.**:5060 expires 3600 sip-server ipv4:192.168.255.**:5060 ! Regards, Oliver Schenk Control Systems Engineer WestNet Rail NOTICE - This e-mail and any files transmitted with it are confidential and are only for the use of the person to whom they are addressed. If you are not the intended recipient then you have received this e-mail in error; please advise us immediately if this is the case. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of WestNet Rail.

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Require help for 2833 dtmf problem( application: bridge, bypass_media=true)
by lrobot 10 Jun '11

10 Jun '11

Hi Freeswitch gay, I use freeswitch to bridge one incoming call to other target phone number. I use bypass_media mode. So the rtp is not go through the freeswitch, By I notice that When LegB receive 183 or 200 SDP message. Freeswitch change it 2833 telephone-event payload type. and forwarding the message to caller. That make the caller use wrong dtmf payload type to send dtmf What wrong with that? How can fix it? Any information or help is welcome. Thanks Lrobot

1 0

Logical OR in conditions
by Matthew Ralston 10 Jun '11

10 Jun '11

Hi, What is the correct condition->break value to use in order to create a logical OR with conditions? I have two extensions which I'd like to merge into one. I'd like the actions to be run if either of the conditions matches: <extension name="call_forward_enable"> <condition field="destination_number" expression="^\*21\*([0-9]+)#$"> <action application="javascript" data="callForwardEnable.js"/> <action application="hangup"/> </condition> </extension>  <extension name="call_forward_enable"> <condition field="destination_number" expression="^7217([0-9]+)9$"> <action application="javascript" data="callForwardEnable.js"/> <action application="hangup"/> </condition> </extension> So the result would be something like: <extension name="call_forward_enable"> <condition field="destination_number" expression="^\*21\*([0-9]+)#$" break="on-true" /> <condition field="destination_number" expression="^7217([0-9]+)9$"> <action application="javascript" data="callForwardEnable.js"/> <action application="hangup"/> </condition> </extension> I can't seem to get it to work and my brain is melting trying to wrap my head around the different break values. Cheers, Matt

2 1

Enumerate variables in a session
by Yungwei Chen 09 Jun '11

09 Jun '11

Hi, I'm wondering if there's way to enumerate the list of variable names in a session in javascript. Thanks.

4 4

continue_on_fail failover with mod_lcr
by Ivan Kovacevic 09 Jun '11

09 Jun '11

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(a)x.x.x.x"/> <action application="bridge" data="sofia/myprofile/$1(a)y.y.y.y"/> <action application="bridge" data="sofia/myprofile/$1(a)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@x.x.x.x|sofia/myprofile/$1@y.y.y.y|sofia/myprofile/$1@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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