April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

spx file format (speex)
by Bernard Fluixa 18 Apr '12

18 Apr '12

Hello, I'd like to know if anyone managed to play speex files with extension ".spx" without converting format with sox. When I try to, I get the error message below: 2012-04-18 16:23:17.191277 [ERR] switch_core_file.c:122 Invalid file format [spx] for [test.spx]! If there any specific module to install (such as mod_shout for mp3 files)? Thank you. Bernard

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creating a conference for 2 separate parked UUIDs, please help
by Anton VG 18 Apr '12

18 Apr '12

Guys, I need to create a conference for 2 UUID's , which already exists and parked. How do I create a new, empty conference and then send those 2 UUIDs there?

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Has the behavior of loopback endpoint changed since 92e89cb4532913a3fd7a4888f8c94df379775cc3 commit
by Anton Kvashenkin 18 Apr '12

18 Apr '12

Today, as always, i just made git pull to the latests updates and found strange behaviour @ vanilla configuraton of the extension "Local_Extension". FreeSWITCH Version 1.1.beta1 (git-e9abdac 2012-04-17 23-05-48 -0400) <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${ru-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <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="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> The purpose of that extension is simple, just rings to another extension and, for example, if the other side is busy go to voicemail. But with the lattest git it doesn't work at all. Why does it proceed "voicemail" into default context? Here is the log: http://pastebin.freeswitch.com/18886 Actually it works when using 92e89cb4532913a3fd7a4888f8c94df379775cc3 revision.

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Why "Creating SIP UA for profile" all failed
by Fu Jiantao 18 Apr '12

18 Apr '12

Hi, I'm trying to run freeswitch, and seems 5060 is not listenning, I'm running it on Windows, I'm sure that 5060 is not used by other applications, and 130.62.240.10 is the only ipv4 address on the machine. The external ip is different one. I've tried using 127.0.0.1 to replace the {local_ip_v4} in vars.xml, it remains the same. I've seen previous solution on this, but seems there're only two possibilities, 1) the port 5060 is already used. 2) the ip is not valid. I'm sure they are both not in my env. So, what are else possibilities for such errors? Thanks in advance! 2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 ext-rtp-ip [auto-nat] 2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 ext-sip-ip [auto-nat] 2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 rtp-timeout-sec [300] 2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 rtp-hold-timeout-sec [1800] 2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 force-register-domain [130.62.240.10] 2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 force-subscription-domain [130.62.240.10] 2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 force-register-db-domain [130.62.240.10] 2012-04-19 00:33:16.796875 [DEBUG] sofia.c:3527 challenge-realm [auto_from] 2012-04-19 00:33:16.796875 [NOTICE] sofia.c:4372 Started Profile internal [sofia_reg_internal] 2012-04-19 00:33:16.796875 [DEBUG] mod_sofia.c:5193 Waiting for profiles to start 2012-04-19 00:33:16.796875 [DEBUG] sofia.c:1747 Creating agent for internal 2012-04-19 00:33:16.828125 [ERR] sofia.c:1813 Error Creating SIP UA for profile: external 2012-04-19 00:33:16.843750 [ERR] sofia.c:1813 Error Creating SIP UA for profile: internal 2012-04-19 00:33:16.843750 [ERR] sofia.c:1813 Error Creating SIP UA for profile: internal-ipv6

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No ringback/transfer_ringback freetdm
by NeCoSec - Ulrich Schinz 18 Apr '12

18 Apr '12

Hi there, first of all, my name is Uli. I'm reading this list for a while and today im gonna post my first question ;) I'm using a Sangoma A500 with FreeTDM and FreeSWITCH. My setup is running, better say _was_ running, perfectly. I'm not 100% sure, but maybe sunday or monday I updated freeswitch. Since there I have a strange thing. If someone is calling via ISDN to my FreeSWITCH-box there is no ringback tone on the callers phone (i.e. tooot...tooot...tooot...). My phone here is ringing, I can answer the call, after timeout voicemail answers. Here are some parts of my config: Extension, which is matched: <extension name="coming_from_isdn_bureau"> <condition field="destination_number" expression="^(<MYNUMBER>)$"> <action application="set" data="effective_caller_id_number=00${caller_id_number}"/> <action application="set" data="ringback=$${de-ring}"/> <action application="set" data="transfer_ringback=$${de-ring}"/> <action application="transfer" data="225 XML default"/> </condition> </extension> In my first running setup I did not set variable ringback and transfer_ringback. And there was a toot...tooot...toot... Observing that there is no ringback any more I decided to set these variables, but with no (or not the wanted) effect. Application info sais: variable_ringback: [%(1000,4000,425)] variable_transfer_ringback: [%(1000,4000,425)] Calling internal (sip->sip) is no problem. Ringtones are there. Calling from sip-phone to external number via ISDN (Sangoma A500) is also no problem. There I don't set ringback-variable... <extension name="ext_isdn_calls"> <condition field="caller_id_number" expression="^(222|223)$" break="never"> <action application="set" data="effective_caller_id_number=<MSN-A>"/> <anti-action application="set" data="effective_caller_id_number=<MSN-B>"/> </condition> <condition field="destination_number" expression="^(0)(\d+)$"> <action application="info"/> <action application="bridge" data="freetdm/wp1/a/$2"/> </condition> </extension> Maybe someone has an idea, where I can do further debugging or where there could be a misconfiguration... Best regards Uli P.s.: hope you can understand my german english ;)

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FreeSwitch Configuration files
by Rajesh Choudhary 18 Apr '12

18 Apr '12

Hello I am a newbie to freeswitch i have installed freeswitch on Ubuntu 10.04 LTS as per installation guide, but there are no conf files in /usr/local/freeswitch/conf folder.However when i run sofia profile external restart reloadxml in FS cli itruns successfully from where it is getting conf files kindly help regards Rajesh

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Freeswitch and network bonding
by Sharad Garg 18 Apr '12

18 Apr '12

Hi All I am trying to run the Freeswitch with network bonding. My scanerio is - eth0 - 192.168.4.25 eth1 - 192.168.4.30 bond0 - 192.168.4.65 This is the virtual IP which is to access the freeswitch. The issue is when I make a call to 192.168.4.65 which is a virtual ip of Freeswitch system, this call dies not reach at Freeswitch. When I do wireshark, wireshark shows `Destination unreachable'. The same freeswitch works well if I disable the network bonding. Here I would like to mention that I have not made any kind of settings in Freeswitch to run with network bonding. Can someone help me why call is not arriving at Freeswitch when network bonding is enabled. Thanks in advance. Best Regards Sharad

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SIP trace specific call or uuid
by Anton VG 18 Apr '12

18 Apr '12

HI! Is there a way to enable sip-trace on the specific call or uuid? I've been only able to find how to enable it per profile, but on a busy situation, the information amount is too high to be useful, so it's often necessary to trace a specific call or call to specific destination.

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Translating DTMF from RFC2833 to INFO
by Bernard Murphy 17 Apr '12

17 Apr '12

I appreciate this message refers to a very old post, but I have EXACTLY the same problem and would love to know if you got this fixed in the end and how ? Basically Cisco gateway not sending DTMF events to the ISDN when they are received by RFC2833 however do get sent when received by SIP INFO. [original post] http://lists.freeswitch.org/pipermail/freeswitch-users/2009-December/050780… Thanks Bernie

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 70, Issue 121
by Sharad Garg 17 Apr '12

17 Apr '12

Thanks for the suggestion... After defining the sip-ip & rtp-ip, call is arriving at Freeswitch & call is answered. But unfortnately, there is one way voice. FS to network side voice is OK but network to FS there is no voice. any suggestion for this issue also. Best Regards Sharad ----- Original Message ----- From: <freeswitch-users-request(a)lists.freeswitch.org> To: <freeswitch-users(a)lists.freeswitch.org> Sent: Tuesday, April 17, 2012 5:35 PM Subject: FreeSWITCH-users Digest, Vol 70, Issue 121 > 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: Freeswitch and network bonding (Karol) > 2. Re: Freeswitch and network bonding (Sharad Garg) > 3. Re: Freeswitch and network bonding (Anton Kvashenkin) > 4. Re: dynamic dialplan alternatives (huseyin kalyoncu) > -------------------------------------------------------------------------------- > _______________________________________________ > 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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