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sip_profiles - Aliases Question
by Alfredo Quiroga-Villamil 17 Jan '10

17 Jan '10

All: Any ideas why there is an alias here? What does that exactly mean? Do I need to have that? How do I remove that? freeswitch@internal> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@192.168.1.148:5080 RUNNING (0) internal profile sip:mod_sofia@192.168.1.148:5060 RUNNING (0) 192.168.1.148 alias internal ALIASED ================================================================================================= 2 profiles 1 alias Lots of questions there all trying to figure out why it's showing up there. Thanks in advance, Alfredo Q-V

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Originate_timeout not working with latest SVN trunk
by Chris Chen 17 Jan '10

17 Jan '10

Hi there, if anybody has the similar issue like me regarding the originate_timeout variable. I have the following dialplan: <extension name="Support_Extension"> <condition field="destination_number" expression="^(support)$"> <action application="set" data="dialed_extension=$1"/> <action application="export" data="dialed_extension=$1"/> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features_nln"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftim$ <action application="bind_meta_app" data="3 b s execute_extension::cf XML features_nln"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="call_timeout=10"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="db" 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="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/1000@My_FS_Public_IP ,user/1001@My_FS_Public_IP"/> <action application="set" data="originate_timeout=15"/> <action application="bridge" data="sofia/gateway/unlimitel/6475552222"/> <action application="bridge" data="sofia/gateway/unlimitel/4165552222"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default_nln ${domain_name} 1001"/> </condition> </extension> in the past the call_timeout and originate_timeout all working as expected, but with latest SVN trunk since 16318, the originate_timeout not working, once the call forwarded to the cell phone by <action application="bridge" data="sofia/gateway/unlimitel/6475552222"/> <action application="bridge" data="sofia/gateway/unlimitel/4165552222"/> it never timed out so the voicemail never got executed. Any helps are greatly appreciated Thanks, Chris

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How do I carry dial-in number to extension?
by mailinglist 16 Jan '10

16 Jan '10

I have two DID's registered at my SIP. This works nicely, I've created groups for holding the local extensions, and one of my phones can subscribe to two SIP accounts, so I can distinguish between if it's one or the other DID (it will show which SIP account is called in the display). Problem is, another one of the phones can only subscribe to one SIP, and has no display (linksys sipura 901), any idea how to accomplish that I can distinnguish between the DID's on that one?

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reloadxml question
by Alfredo Quiroga-Villamil 16 Jan '10

16 Jan '10

All: I am starting to play a bit with FS to get more familiar with it. I am trying the following: 1- sofia status // command for example to see the list of profiles. 2- I then went ahead and renamed the example.com.xml under conf/directory/default to example.com.xml.orig 3- I ran then reloadxml using the fs_cli 4- sofia status still displays the same info as before for example.conf as a gateway. I then went ahead and stopped FW and re-started it. After I did this I re-ran sofia status and this time the example.conf gateway that was previously listed went away as it was supposed to. Am I missing something? Is reloadxml the proper way to reload configuration changes. If it is why is it not working for this case? Thanks in advance, Alfredo

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Re: [Freeswitch-users] Email2Pdf
by Mathieu Parent 16 Jan '10

16 Jan '10

On Fri, Jan 15, 2010 at 1:03 AM, Costa Zikalala <costa.zikalala(a)gmail.com> wrote: > Hi Mathieu > > Thank you for your very wonderful script. You are talking about http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/sathieu/email2pdf/ > I'm just trying understand it a bit and am not too good with Perl. > Kindly confirm the folowing for me: > - Only attachments and not the whole email is converted to PDF? No, the message body should also be converted (in the script, this is call $header ;). > - Is the PDF output saved in the variable $file_out? Yes, or to stdout, if $file_out is "-". more info with: ./email2pdf --man > - Will you be completing your script as Email2Fax for Freeswitch? The remaining thing will be added to FS wiki (don't have time now), as this is site-specific: - postfix config (see http://hylafax.sourceforge.net/howto/faxing.php#ss5.4) - script email2fax: + guess From: fax number from the email + run email2pdf + convert to tiff + call txfax within FS Cheers Mathieu Parent > Thanks again, > Costa > >

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RTCP information
by Jon Bruel 16 Jan '10

16 Jan '10

In a real setup with 5-20 VoIP calls a day, every now and then there are some problems with sound quality, and I need some tools to investigate the cause. The phones support RTCP, and I would like to hear if I can get the FS to relay those packets to some kind of logger, including the signalling information? /Jon

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ODBC Not Available!
by Woody Dickson 16 Jan '10

16 Jan '10

Hello, I am trying to get ODBC within the voicemail module to work, but I am getting the following error: 2010-01-16 14:32:32.584124 [CRIT] switch_core_sqldb.c:306 Failure! OBDC NOT AVAILABLE! 2010-01-16 14:32:32.584124 [ERR] mod_voicemail.c:214 Error Opening DB [root@e-d freeswitch]# isql my_odbc +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ In my voicemail.conf.xml, I have: <param name="odbc-dsn" value="my_odbc:root:123456"/> Does anyone know what may be wrong with my config? Thanks for your help. woody

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Eavesdrop in LUA
by rod 16 Jan '10

16 Jan '10

Hi all, I'm trying to do this in LUA: A call B and I'd like to setup a new call to C with eavesdrop of A conversation with B. I have no idea how to do this if someone can help. I switched to LUA cause I see no way to achieve this with dialplan (snippets are welcome). regards, rod

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FIFO Originate caller ID
by Steve Steffler 16 Jan '10

16 Jan '10

Hello, I found an archived conversation on this list regarding FIFO origination caller ID, and how to modify it. http://old.nabble.com/FIFO-Orgination_caller_id-td26487628.html There seems to be no easy way to customize the caller ID on originated calls from the FIFO to on-hook agents who were registered dynamically. Anthony states a method to do it using static entries in the fifo conf file, and the general rationale is that SCREEN POPS be the preferred method, with the added nudge that good SIP phones can change the caller ID when the bridge is complete as well, which is all well and good. The problem is for my application, all on-hook agents are using cellular phones, and they register dynamically. Also, none of my agents are in front of a computer, so a SIP display update on the phone or screen pop on the computer in front of them is not really an option, and the only way they can identify calls from my FIFO right now is because they are the ones with NO CALLER ID (in other words, their mobile phones do not display the name, and the number is not recognized because it is set by FreeSWITCH to be "fifo+fifoname" instead of being numeric. This is far from ideal. I am wondering if there is anyone on the list who knows how to configure the origination_caller_id_number/name variables for dynamically registered on-hook agents so that the caller ID from the FIFO customer's incoming call is displayed to them instead of the above mangled caller ID? I'm not disagreeing that it is an old-skewl way of thought, but in actuality it is just a way to interface with old-school telephony devices (i.e. non-Smartphone mobile phones) and I am not sure how to accomplish this. Any help/input would be appreciated. Regards, Steve

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Re: [Freeswitch-users] multiple leg and multiple rtp
by Anthony Minessale 16 Jan '10

16 Jan '10

Perhaps best not to help him anymore without an apology for the snap judgement and comparison to asterisk clearly designed to push our buttons. We'll be here when you realize we were trying to help you but I can't promise we will still have paitence...... On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike(a)jerris.com> wrote: You already are running on trunk. Mike On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH Version 1.0.5pre10 (16012M... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists...

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