- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] funny effect after minimizing xml files
by Jason White 15 Jun '09

15 Jun '09

Durk de Beer <durk.debeer(a)isp.solcon.nl> wrote: > > Hello I've minimized de xml files where possible to make a dialplan that is > as short as possible. Now do I've this funny effect to dial my extensions > who are running from 200 to 207. It seams that I'm able to dial an > extension in closed in a number. So for instants if I dial 120275 extension > 202 will ring even tried it whit two extensions in a number like 202205 . > This results in the first extension ringing so 202205, 202 will ring > 205202, 205 will ring. At this time I'm unable to pinpoint the cause of > this behaviour. Could someone point me to the cause of this effect I don't understand the problem, but my general advice is this: learn to read the FreeSWITCH logs carefully. Make sure that the log level is set to "debug", as it is in the default configuration, then carefully check the log files to see which dialplan extension matched and how the call was processed.

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funny effect after minimizing xml files
by Durk de Beer 15 Jun '09

15 Jun '09

Hello I've minimized de xml files where possible to make a dialplan that is as short as possible. Now do I've this funny effect to dial my extensions who are running from 200 to 207. It seams that I'm able to dial an extension in closed in a number. So for instants if I dial 120275 extension 202 will ring even tried it whit two extensions in a number like 202205 . This results in the first extension ringing so 202205, 202 will ring 205202, 205 will ring. At this time I'm unable to pinpoint the cause of this behaviour. Could someone point me to the cause of this effect /d

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When to use OpenSIPS/Kamailio in front of FS?
by Gavin Henry 15 Jun '09

15 Jun '09

Hi, I'm excited reading all the threads about how FS blows Asterisk away so that you don't need OpenSIPS/Kamailio in front of FS. Surely there must be a point when it would be advisable to do that though, as mod_sofia can't be as good as a dedicated SIP proxy? Thanks. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com

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How to connect Freeswitch Windows, Freeswitch Linux and also vice versa?
by Edmar Cruz 15 Jun '09

15 Jun '09

Ip Windows: 192.168.0.104 Ip Linux: 192.168.0.105 My windows: My sample on sip_profiles/external/dialus.xml <extension name="dialus"> <condition field="destination_number" expression="^014(\d+)$"> <action application="bridge" data="sofia/external/$1@192.168.0.105:5080"/> </condition> </extension> My linux: My sample on sip_profiles/external/dialus2.xml <extension name="dialus2"> <condition field="destination_number" expression="^014(\d+)$"> <action application="bridge" data="sofia/external/$1@192.168.0.104:5080"/> </condition> </extension> I have a number on windows 01497710001, on linux 01497710002 Trying to call each other on windows I dial 0149771002 But error on switch_ivr_originate: INVALID_NUMBER_FORMAT Please help me with this urgent issue... Or send me instructions or xml code that will me to solve this issue... Thanks, Edmar -- View this message in context: http://www.nabble.com/How-to-connect-Freeswitch-Windows%2C-Freeswitch-Linux… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Possible to initiate multiple skype interfaces with the same id?
by Jingwei Yang 15 Jun '09

15 Jun '09

Hi Team, As the subject indicates, is there a possible way to do that? I've tried setting up two different skype instances with the same id in /usr/src/freeswitch/src/mod/endpoints/mod_skypiax/configs/2startskype.sh /usr/bin/Xvfb :101 -auth /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & su root -c "/bin/echo '*userAAA* aaa'| DISPLAY=:101 /usr/bin/skype --pipelogin &" /usr/bin/Xvfb :102 -auth /usr/local/freeswitch/conf/autoload_configs/skypiax.X.conf & su root -c "/bin/echo '*userAAA *aaa'| DISPLAY=:102 /usr/bin/skype --pipelogin &" The script can be run without any error and when I executed the ps command, I noticed there's only one skype process up. This is strange since normally there will be two (with two different ids) 3782 pts/1 00:00:00 Xvfb 3793 pts/1 00:00:00 dbus-launch 3800 pts/1 00:00:04 Xvfb 3805 pts/1 00:00:03 *skype* 3811 pts/1 00:00:00 dbus-launch Then I started freeswitch and hit an error when loading mod_skypiax: *freeswitch(a)localhost.localdomain> load mod_skypiax 2009-06-15 10:00:52 [WARNING] mod_skypiax.c:950 load_config() rev 13600[(nil)|37 ][WARNINGA 950 ][skypiax1 ][-1, 0, 0] STARTING interface_id=1 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error code 3 from X Server 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending message failed with status 3 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1184 X11_errors_handler() rev 13600[(nil)|37 ][ERRORA 1184 ][none ][-1,-1,-1] Received error code 3 from X Server 2009-06-15 10:00:52 [ERR] skypiax_protocol.c:1251 skypiax_send_message() rev 13600[(nil)|37 ][ERRORA 1251 ][none ][-1,-1,-1] Sending message failed with status 3 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:976 load_config() rev 13600[(nil)|37 ][NOTICA 976 ][skypiax1 ][-1, 0, 0] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2009-06-15 10:00:53 [NOTICE] mod_skypiax.c:986 load_config() rev 13600[(nil)|37 ][NOTICA 986 ][skypiax1 ][-1, 0, 0] Found a running Skype client, connected to its SKYPE API for interface_id=1, waiting 60 seconds for CURRENTUSERHANDLE==userAAA API CALL [load(mod_skypiax)] output: -ERR [module load file routine returned an error] 2009-06-15 10:00:53 [ERR] mod_skypiax.c:1010 load_config() rev 13600[(nil)|37 ][ERRORA 1010 ][skypiax1 ][-1, 0, 0] The Skype client to which we are connected FAILED to gave us CURRENTUSERHANDLE=userAAA, interface_id=1 FAILED to start. No Skype client logged in as 'userAAA' has been found. Please (re)launch a Skype client logged in as 'userAAA'. Skypiax exiting now 2009-06-15 10:00:53 [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_skypiax.so **Module load routine returned an error** * Please enlighten whether it's possible to start multiple skype instances with the same skype id. If possible, what are the correct configs? Thanks, -Jingwei

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MPL Confusion
by Muhammad Shahzad 15 Jun '09

15 Jun '09

Hi, I have some confusion about FreeSWITCH's Mozilla Public License 1.1. I do understand that me or any one can change provided code according to our customization needs and we are not bound to share our changes as long as we are not distributing it, right? Now, i have been doing R&D on MSN and Yahoo voice chat services, I have now completed by research and now would like to write up FS modules to communicate with these servers. But as you all know both MSN and Yahoo provide SIP based VOIP services, however they are not using standard SIP stack and have their own versions of customized SIP stack. So, in order to write an endpoint for these servers, instead of writing everything from scretch, i can using existing mod_sofia endpoint and customize it to make it compatible with MSN and Yahoo SIP stack. So here are my questions, 1. Is it possible under MPL, that i make a copy of mod_sofia as say mod_msn and develop it to work with MSN, similarly mod_yahoo for Yahoo voice chat service? 2. If yes, how can i mention my role in these modules development, i.e. as developer or as contributor? Also i wish to include my work, once completed, in FreeSWITCH, can you provide me the guidelines and / or eligibility criteria to do so, any link on FS site etc.? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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Re: [Freeswitch-users] mod_voicemail accounts on-the-fly
by Adam Wilt 14 Jun '09

14 Jun '09

Thanks. I would really like mod_voicemail to be database driven, instead of by XML and cURL. I noticed in the documentation that you can provide an ODBC handle in the voicemail.conf.xml file, and so I tried it with MySQL. It created the two tables, voicemail_msgs and voicemail_prefs properly, and voicemail_msgs works the way I expect; when a new voicemail is created it writes a new record in this table. But I don't understand the purpose of voicemail_prefs; when I add a record here with a username and password, mod_voicemail ignores it. So I still have to use the config file or xml_curl to set-up the users. It doesn't seem like mod_voicemail reads from or writes to this table.

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Error in Dialplan documentation?
by Lars Zeb 13 Jun '09

13 Jun '09

At http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall, near the top under From Dialplan, it says: Bridge the incoming call to extension 100 and 101. The '%' is used instead of the @ to indicate that the endpoints are registered locally. Separate multiple endpoints with a comma. The ${sip_profile} variable is defined in freeswitch.xml. <action application="bridge" data="sofia/sip/100%${sip_profile},sofia/sip/101%${sip_profile}"/> However, I cannot find ${sip_profile} in freeswitch.xml. Is the documentation correct? Thanks, Lars

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Re: [Freeswitch-users] mod_voicemail accounts on-the-fly
by Adam Wilt 13 Jun '09

13 Jun '09

I'm sorry, I said dialplan, but I was talking about the directory. So the directory file must be edited every time a new mailbox is created? > > Message: 7 > Date: Sat, 13 Jun 2009 12:58:57 -0400 > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] mod_voicemail accounts on-the-fly > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <F5F3AE5B-D6AD-440E-802A-360354822FC2(a)freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > Why would have to touch the dialplan? Just one regex to catch it > all.. and send it to voicemail. You still have to add users to the > directory for them to have a mailbox anyway. > > /b > > On Jun 13, 2009, at 12:18 PM, Adam Wilt wrote: > > > Hi, I have a question about mod_voicemail. I would like to use it > > independently of the dialplan and I was hoping to be able to add new > > voicemail accounts on-the-fly, without having to update the > > FreeSWITCH configuration files. But it seems to be forcing me to > > manually add each user to the dialplan. > > > > Does anybody know if mod_voicemail was made to work the way I'm > > trying to use it, and the best way to approach this? > > > > Thanks, > > Adam > > > > Brian West > brian(a)freeswitch.org > > -- Meet us at ClueCon! http://www.cluecon.com > > > > >

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mod_voicemail accounts on-the-fly
by Adam Wilt 13 Jun '09

13 Jun '09

Hi, I have a question about mod_voicemail. I would like to use it independently of the dialplan and I was hoping to be able to add new voicemail accounts on-the-fly, without having to update the FreeSWITCH configuration files. But it seems to be forcing me to manually add each user to the dialplan. Does anybody know if mod_voicemail was made to work the way I'm trying to use it, and the best way to approach this? Thanks, Adam

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