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Multiple registrations: Only calling the first element in the list in this mode
by Peter P GMX 18 Jan '10

18 Jan '10

Hello, I habe the following behaviour when I call a user which is registered twice with 2 phones via bridge user/100@domain both phones are ringing. This is correct as I allow multiple registrations in a profile However when I call multiple endpoints via bridge user/100@domain,user/101@domain,user/102@domain only one phone with number100 is ringing. Console log shows "Only calling the first element in the list in this mode.": 2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable string 0 = [presence_id=100@domain] 2010-01-12 19:52:18.236361 [DEBUG] switch_ivr_originate.c:1734 variable string 1 = [transfer_fallback_extension=100] 2010-01-12 19:52:18.236361 [WARNING] switch_ivr_originate.c:2048 Only calling the first element in the list in this mode. 2010-01-12 19:52:18.236361 [NOTICE] switch_channel.c:613 New Channel sofia/internal/sip:100@10.11.12.203:2048 [9b95fcdc-ffab-11de-9ba2-13b2daa7ce61] Is there any way to work around this? I need all phones to be ringing in this scenario. Best regards Peter

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eavesdrop problem?
by Nikolay Kondratyev 18 Jan '10

18 Jan '10

Hi all, I want to use eavesdrop application. Playing with it I found that when one tries to eavesdrop caller the feature works ok. But when trying to eavesdrop callee eavesdrop attempt failes. I just updated to the latest version from http://latest.freeswitch.org <http://latest.freeswitch.org/> [freeswitch@freeswitch log]$ fs_cli -x version FreeSWITCH Version 1.0.5-20100115-0400 (16318M) My setup is as following: I have 3 internal extensions: 2853, 2840, 2879 (all are xlite). 1. 2879 calls 2840. Then 2853 dials 882879 - eavesdrop worked as expected. 2. 2840 calls 2879. Then 2853 dials 882879 - eavesdrop failed. I attached logs for both cases. I don't believe it's intended behavior. Can anybody please advise if it is a configuration or a software problem? Thanks and regards, Nikolay.

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Questions about mod_fifo
by Jingwei Yang 18 Jan '10

18 Jan '10

Hi Guys, I'm implementing an ACD system using ESL and mod_fifo. Based on what Anthony suggested in this post: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg01652.… *You can make an event socket application that listens for FIFO events and keeps track of what FIFOs are currently busy and when there are people waiting you can have that script generate a call to a group of SIP phones so when the first one answers, it sends them in as an agent where they can field the calls. * 1. How should I handle the concurrent issues? If I bridge a user to two agents and both of them answers, how does FS take care of this situation? Will a slower agent get a busy tone automatically? 2. If the socket application is brought up after some users have called in, what command should I use to check the busy queues? fifo list? 3. Am I using fifo list and fifo count correctly? here's the testing dialplan: <extension name="in_fifo"> <condition field="destination_number" expression="^779$"> <action application="set" data="fifo_music=$${hold_music}"/> <action application="answer"/> <action application="fifo" data="*myq *in"/> </condition> </extension> when a call comes in and gets queued, these are the results of some commands I tried. freeswitch(a)localhost.localdomain> fifo list API CALL [fifo(list)] output: <fifo_report> <fifo name="cool_fifo(a)192.168.1.104" consumer_count="0" caller_count="0" waiting_count="0" importance="0"> <callers></callers> <consumers></consumers> </fifo> </fifo_report> freeswitch(a)localhost.localdomain> fifo list myq API CALL [fifo(list myq)] output: <fifo_report></fifo_report> freeswitch(a)localhost.localdomain> fifo count myq API CALL [fifo(count myq)] output: none It seems *myq* doesn't get created at all? Please enlighten. Thanks and best regards, -Jingwei

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IMPORTANT -- Call for bug updates
by Michael Jerris 18 Jan '10

18 Jan '10

To all FreeSWITCH users- If you have bugs open on http://jira.freeswitch.org please login today and post a status update on these bugs (even if they appear to be awaiting comment by the development team). If you have a patch, please update these patches to svn trunk so that they may be reviewed. I know many patches have been sitting for quite some time but I will make a strong effort to review and merge patches that are ready to go in. If you have a bug, please update to svn trunk and comment if the issue still exists or is now fixed in trunk. If you bug has a comment requesting more information, please provide it. If you don't have any open bugs, and are not currently using recent svn trunk, I would appreciate it if you could carve a little bit of time out of your days and test out trunk. Feel free to look through jira and find a bug you are interested in and test it as well. We are working towards having the most stable and feature rich release of FreeSWITCH yet and we need your support and assistance to do so. As always, if you have a new bug to file, please do so as soon as possible and try to get as much information as possible on the bug. There are some good guidelines for reporting at http://wiki.freeswitch.org/wiki/Reporting_Bugs As always, many thanks to the community for all your hard work and support. Mike

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Re: [Freeswitch-users] Trouble with Originate/Bridge from ESL [SOLVED]
by Max Bridgewater 18 Jan '10

18 Jan '10

Given that I'm using Java, I had to escape the ringback value's quote twice: originate {ringback=\\'%(2000,4000,440.0,480.0)\\'}sofia/internal/1004%74.63.243.52 &bridge(sofia/internal/1005%74.63.243.52) On Sun, Jan 17, 2010 at 3:51 PM, Max Bridgewater <max.bridgewater(a)gmail.com>wrote: > One more piece of information: the call is being terminated by Freeswitch > with the event: > > Event-Name: CHANNEL_HANGUP > Hangup-Cause: NO_ANSWER > > Which is strange because B leg doesn't even have the time to answer. > > > > On Sun, Jan 17, 2010 at 1:55 PM, Max Bridgewater < > max.bridgewater(a)gmail.com> wrote: > >> 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 media after Originate
by Mohamed Hassan 18 Jan '10

18 Jan '10

Hi Freeswitch users thank you in advance as i am new to freeswitch iam facing a problem while writing a Callback service i tried to bridge the two legs through one sip provider the call estaplished successfully but without media passing and after some searching i found that there is a bug with sofia loopback so i shoulkd handle the two legs with different sip gateways and after trying i found the same error using cli : originate {ignore_early_media=true}sofia/gateway/prov/00xxxxxxxxxxx 9999 XML callback and here the 9999 context: <context name="callback"> <extension name="9999"> <condition field="destination_number" expression="9999"> <action application="sleep" data="5000"/> <action application="answer"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="[ignore_early_media=false]sofia/gateway/prov/00xxxxxxxxxx"/> </condition> </extension> </context>

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Change Domain Freeswitch
by Edmar Cruz 18 Jan '10

18 Jan '10

Dear All, How can i change the domain of my freeswitch 52.236.125.12 to sip.grandminister.com to be able to detect the presence of the user whos online or not... Thanks Edmar -- View this message in context: http://old.nabble.com/Change-Domain-Freeswitch-tp27104680p27104680.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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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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