September 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Tutorial about integration of Kamailio 3.3.x and FreeSwitch 1.2.x
by Daniel-Constantin Mierla 24 Sep '12

24 Sep '12

Hello, just to let everyone know that I updated my tutorial about using Kamailio SIP Server and FreeSwitch for building telephony systems. This time I used latest stable branches from the two projects. The tutorial is available online at: * http://asipto.com/u/k33fs It follows the same concept of using Kamailio for handling authentication, registration and routing logic, with FreeSwitch providing the media services (voicemail, conferencing, ivr, ...) and SBC functionality. One thing promised with the previous version was ability to generate user directory automatically from Kamailio's subscriber table. It is done now for voicemail using a Lua script that connects to Kamailio's database and builds the appropriate XML directory file. Hope is going to be useful for many out there! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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using spandsp fax tone detection from originate command
by Adam Johnston 24 Sep '12

24 Sep '12

Hi all, I've been trying to use the spandsp_start_fax_detect application (with txfax as the callback) from the originate command but I keep getting a hangup before it completes. So far I've tried calling the app directly: originate {ignore_early_media='true'}sofia/gateway/mygateway/18005551234 &spandsp_start_fax_detect('txfax /tmp/myfax.tif 3 ced') Inline dialplan: originate {ignore_early_media='true'}sofia/gateway/mygateway/18005551234 'spandsp_start_fax_detect:txfax /tmp/myfax.tif 3 ced' inline ..plus some other permutations but all end with the same result: [NOTICE] mod_spandsp.c:107 Enabling fax detection 'txfax' '/tmp/myfax.tif' [DEBUG] switch_core_media_bug.c:456 Attaching BUG to sofia/external/18005551234 [NOTICE] switch_core_state_machine.c:226 sofia/external/18005551234 has executed the last dialplan instruction, hanging up. Am I misunderstanding how spandsp_start_fax_detect works? Adam Johnston

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CUSTOM event messages missing FreeSWITCH-Hostname
by James Bravo 24 Sep '12

24 Sep '12

Hi We've recently upgraded from version 1.0.head to 1.2.0-rc2 and noticed that certain events sent from php code using the ESP library have certain core headers missing, in particular 'FreeSWITCH-Hostname'. For example, using '/events plain all' in fs_cli, version 1.0.head would show RECV EVENT Event-Subclass: rmCTI::TRANSFER_ANSWER Core-UUID: 47527f1f-b6c3-4f74-9f16-32f565ee4123 FreeSWITCH-Hostname: s01pbx01.test.com FreeSWITCH-Switchname: s01pbx01.test.com FreeSWITCH-IPv4: 211.141.141.131 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2012-08-03 10:43:06 Event-Date-GMT: Fri, 03 Aug 2012 09:43:06 GMT Event-Date-Timestamp: 1343986986238409 Event-Calling-File: mod_event_socket.c Event-Calling-Function: read_packet Event-Calling-Line-Number: 1188 Command: sendevent CUSTOM Event-Name: CUSTOM Channel-Name: sofia/internal1/1202(a)sip.test.com rmCTI-UUID: aebc98dc-22da-4ad1-8234-912487329123 rmCTI-Other-UUID: b24a8783-16b1-495e-9c4e-5ea52c362123 rmCTI-Info: Transfer to new b-leg answered where as version 1.2.0-rc2 it shows RECV EVENT Event-Subclass: rmCTI::TRANSFER_ANSWER Command: sendevent CUSTOM Event-Name: CUSTOM Channel-Name: sofia/internal1/1202(a)sip.test.com rmCTI-UUID: c3c7b331-0b8b-4ea5-834f-9d72fd446123 rmCTI-Other-UUID: db75c60b-a258-4b47-8d51-a1b7ac86f123 rmCTI-Info: Transfer to new b-leg answered Event-UUID: f47597b2-9e8c-4508-be57-3e1eed35c123 Is there any reason these core headers are missing? Is it something to do with CUSTOM events? Thanks in advance James Bravo

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Softphone with callee name display?
by Yehavi Bourvine 24 Sep '12

24 Sep '12

Hello, Polycom and Snom (at least) can display the name of the called person (sent by P-Remote-ID or P-Asserted-ID fields). Is there a softphone that does the same? I couldn't find suh one so far... Thanks! __Yehavi:

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openVZ and FS: core dump!?
by Dennis 24 Sep '12

24 Sep '12

hi all, at the moment we are playing arround with openVZ (container) and freeswitch. as far as we have read, openVZ and fs are working quite well together. but we are getting a lot of fs core dumps, when testing. we are using actual versions of fs, which are running very smooth on normal/real servers without openVZ. the only information we get out of the core dump is: "Program terminated with signal 6, Aborted." are there any settings or things, which have to be looked at, when running fs in an openVZ container? thank you for your help! dennis

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FreeSWITCH HA Database changes in 1.2
by Colin Mason 24 Sep '12

24 Sep '12

Hello, I noticed recently on FS 1.2 that the table "sip_recovery" wasn't being created automatically by FreeSWITCH. If I specify: <param name="core-recovery-db-dsn" value="maxpowersoft_odbc:freeswitch:blahblah"/> In switch.conf.xml, FreeSWITCH will create a table called "recovery". So my questions are: 1. Have you renamed the sip_recovery table to recovery? 2. Is "core-recovery-db-dsn" a required parameter for HA FreeSWITCH to function? Thanks, Colin

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Remove UPDATE from Allow list in the INVITE
by Stephen Wilde 24 Sep '12

24 Sep '12

It's possible to remove the UPDATE from the Allow list in the INVITE? I have an issue with a customer where the solution is not to send the UPDATE in the Allow list. I have tried with dialplan: <action application="export" data="ignore_display_updates=true" /> without succes. Any advise? Stephen

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Fifo api command register agents (add fifo_member)
by Peter van Raamsdonk 24 Sep '12

24 Sep '12

Hi all, I started to implement fifo in my application to handle calls from know phones to unknown extensions and specific dial string/ extension. I use the api commands with esl (java client), outbound socket (with sendMsg). http://wiki.freeswitch.org/wiki/Mod_event_socket I looked at a lot of examples, for me I thought the right sequence to would be to add (register) user agents first (determined by myself) and later add caller in which all user agents would ring and the call would be handled after accepting the call from free user agent. My problem is now the right notation of the add fifo_member. I think I should use this for registering agents; http://wiki.freeswitch.org/wiki/Mod_fifo#Configure_for_Agent_Callback http://wiki.freeswitch.org/wiki/Mod_fifo#Simple%20On-hook%20Agent%20Login/L… http://wiki.freeswitch.org/wiki/Variable_fifo_member_wait sendmsg <uuid> call-command: execute execute-app-name: <one of the applications> execute-app-arg: <application data> sendMsg call-command: execute execute-app-name: set execute-app-arg: "fifo_member(add TestQue sofia/internal/1001%123.123.123.123)" I don't know if the () can be used in sendMsg, my queue name needs quotes "\'" and the callerid needs "@" or "%" with domain at end. When I execute the command I don't get errors, response is ok and in log of freeswitch I also see no errors (only set command). However when I execute fifo list or (verbose_list) from FS CLI, I don't see any user agents registered. If I put the caller in my queue, I see only the caller registered (callers) http://wiki.freeswitch.org/wiki/Mod_fifo#Terminology http://wiki.freeswitch.org/wiki/Mod_fifo#list.7Clist_verbose Below works, only my created queue is missing lag, call timeout and simcount! Put a caller into a FIFO queue sendMsg call-command: execute execute-app-name: fifo execute-app-arg: TestQue in somesoundfile.wav someothersoundfilemusic.wav Kind regards, Peter Ps, I hope somebody can advise !!!!

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Re: [Freeswitch-users] Some questions from a relative newbie
by Stanislav Sinyagin 24 Sep '12

24 Sep '12

in this case, you would have two OpenSIPS boxes and two FreeSwitch boxes. The easiest scenario would be to run them in active/standby mode, so that only one box is handling the calls, and the other one starts acting only when it realizes that the master box is gone. Here is one of possible examples: http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS There are also some scenarios for active/active redundancy, but that requires much more engineering efforts. >________________________________ > From: D'Arcy Cain <darcy(a)Vex.Net> >To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >Cc: Stanislav Sinyagin <ssinyagin(a)yahoo.com> >Sent: Sunday, September 23, 2012 10:07 PM >Subject: Re: [Freeswitch-users] Some questions from a relative newbie > >On Sun, 23 Sep 2012 11:24:59 -0700 (PDT) >Stanislav Sinyagin <ssinyagin(a)yahoo.com> wrote: > >> by the way, do you consider running a fault-tolerant service with >> OpenSIPS/Kamailio/Repro as a border controller and two FreeSWITCH boxes >> behind? > >Thanks. I have added that to my reading list. However, I am not sure >how useful that scenario would be. Doesn't the OpenSIPS router just >become my new point of failure? If I add another box I can get my >provider to fail over to it. That makes them the point of failure but >they already are so I kind of depend on them not to go down anyway. > >-- >D'Arcy J.M. Cain >System Administrator, Vex.Net >http://www.Vex.Net/ IM:darcy@Vex.Net > > >

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Freeswitch+sangoma a101 can not hear ring back tone
by James zhu 24 Sep '12

24 Sep '12

hello:I install fw and sangoma A101 card with sangoma isdn. after few days, caller can not hear the ring back tone, but the voice worked. the hardware is ok, compare with normal status, the alert message lost. how do i set that to enforce to send alert message? Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com

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