January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] sip_profiles - Aliases Question
by Anthony Minessale 17 Jan '10

17 Jan '10

We do offer a triple your money back guarentee! Many params can change without restarting and many can't ip/port fo example. FYI, You don't sound sincere when you use the term appriciate when you are arguing with these guys voulenteering to help explain it to you. You should make sure you make the most of their assistance as they could have just said rtfm On Jan 16, 2010 11:07 PM, "Alfredo Quiroga-Villamil" <lawwton(a)gmail.com> wrote: mm_202: Thanks for the reply. I don't agree with your statement: "In a production system, you *shouldnt* have to change the main profiles at all." I think it's always good to be able to reload things dynamically in a non-disruptive way. The alias is a bit unusual. To be honest I had never seen that before in any of the Telecommunication Systems I've worked on. Not that is not a good idea, just one that at first glance doesn't seem too clear for some reason. Specially when you see two profiles both representing the same object listed under "sofia status". Being forced to have the alias doesn't seem like an appealing option, should be optional I think. I would like if possible a more detailed explanation on what would break if not present. In any case, I am sure there is a reason for these things. I am trying to understand how they all work together. Things are not as apparent in FS at first glance when compared to other systems; but the platform seems to be built with the purpose of offering a lot of flexibility which is really good. Appreciate the feedback. Alfredo Q-V Appreciate the help, Alfredo On Sat, Jan 16, 2010 at 10:35 PM, mm_202 <mastermind202(a)gmail.com> wrote: > > On Sat, Jan 16, 201...

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Trouble with Originate/Bridge from ESL
by Max Bridgewater 17 Jan '10

17 Jan '10

Hi, The following command works great from the command line: originate {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52 &bridge(sofia/internal/1005%74.63.243.52) But this one isn't working from the ESL: api originate {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/internal/1004%74.63.243.52 &bridge(sofia/internal/1005%74.63.243.52) The observed behavior is following: The A leg is dialed, then B leg is also dialed but immediately followed by a hangup. That is, the B user doesn't even has time to answer. Any idea what I'm doing wrong? Thanks, Max.

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No hangup tone after zap channel closed, tones in general
by François Legal 17 Jan '10

17 Jan '10

Hello, How shall I do to get a hangup tone on an FXS port after the channel is closed ? Is there something specific to configure in openzap ? For instance, if I call the following extension from an FXS port : I get no tone after the hangup application is called. I also wonder if there is some documentation on the tones.conf file format, and about the variables uk-ring, us-ring, bong-ring, sit in vars.xml (when are they used, what is the syntax). I could not find any info on wiki. Thank you for helping François

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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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