September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

CudaTel Communications Server Version 1.0 Released
by Michael Collins 22 Sep '09

22 Sep '09

At ClueCon 2009 we had an exciting announcement: Barracuda Networks and the FreeSWITCH team have been working together to create a new PBX appliance. Dubbed the CudaTel Communications Server, this new communications platform is both feature-rich and easy-to-use. We are pleased to announce that version 1.0 of the CudaTel Communcations Server has been released! The feature list for this affordable system is impressive: Automatic phone provisioning Multi-party conferencing Group calling SIP phone and provider support Automated attendant Voicemail TMD hardware option High definition codec support (G.722, G.722.1, G.722.1c) Call recording Active Directory and LDAP integration Encrypted VoIP support Many more features are included, all of which are controlled by an intuitive Web-based interface. We invite you to visit the CudaTel <http://www.cudatel.com/> website or call 989-720-4000 for more information or to request evaluation units. -The FreeSWITCH Team

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ALLOTTED_TIMEOUT hangup cause?
by Nicolas Brenner 22 Sep '09

22 Sep '09

Hi, Today, while trying to bridge some calls I started to get a ALLOTTED_TIMEOUT hangup cause on the second leg. I looked for info on the Wiki and Google, but I couldn't find a detailed explanation. Does anybody know what does it mean exactly? Thanks! Nicolas

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Some Newbie questions about dialplan and local Sip registration
by Filip Lyncker 22 Sep '09

22 Sep '09

Hi List, for the first experiments with freeswitch I downloaded the Windows installation. Now Im trying to get my 2 Sipphones get connected to. Later I want connect the freeswitch to my asterisk gateway. I find the examples pretty complex therfore Im trying to build up a simple solution to understand the functions from the scratch .. my current problem is , that I cant route my local sips to each other ( registration seems to work now). the next is , that freeshwitch is not able to connect to asterisk. but I will describe this later. I installed in the Directory a xml file ( called 22.xml) with the following content : <include> <domain name="$${domain}"> <user id="22" mailbox="22"> <params> <param name="password" value="Xk21%"></param> <param name="vm-password" value="22"></param> <param name="sip-port" value="5060"></param> </params> <variables> <variable name="accountcode" value="22"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 22"></variable> <variable name="effective_caller_id_number" value="22"></variable> </variables> </user> <user id="24" mailbox="24"> <params> <param name="password" value="dudeldum"></param> <param name="vm-password" value="24"></param> <param name="sip-port" value="5060"></param> </params> <variables> <variable name="accountcode" value="24"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 24"></variable> <variable name="effective_caller_id_number" value="24"></variable> </variables> </user> </domain> </include> This seems to be ok now. Now I want to dial from 22 to 24 , wherefore I configured this dialplan : <include> <context name="any"> <condition field="destination_number" expression="^(2[0-9])$"> <action application="bridge" data="user/${dialed_extension}"/> </condition> </include> wich doesnt work , mybe b/c the user/${dialed_extension} I dont know... Freeswitch says: [INFO] switch_core_state_machine.c:136 No Route, Aborting [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/24(a)192.168.1.34 [CS_ROUTING] [NO_ROUTE_DESTINATION] [NOTICE] switch_core_session.c:1086 Session 17 (sofia/internal/24(a)192.168.1.34) Ended [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/24(a)192.168.1.34 [CS_DESTROY] Im sure , for you guys this cant be a big deal;) Next Point is my Asterisk registration , mybe you can help me out here to .. : In the sip-profiles/external I installed the my_asterisk.xml with that content : <include> <gateway name="asterisk"> <param name="username" value="28"></param> <param name="password" value="test"></param> <param name="realm" value="28"></param> <param name="proxy" value="192.168.1.119"></param> <param name="register" value="true"></param> </gateway> </include> Freeswitch allways complains a timeout error : [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request Timeout [408]. failure #17 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry to 540 seconds. it seems that It cant connect ( I also tried out to explicit set the port to 5060 b/c I read something about 5080 .. : <param name="sip-port" value="5060"></param> but this didnt help) In my Asterisk I set in the sip.conf the entry 28 with the pw test .... If someone could help me with my first steps I would be verrry thankful ;)) cheers Filip -- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis

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Mod_perl or ESL
by Nagalenoj 22 Sep '09

22 Sep '09

Dear friends, I want to know which is the better way to do route calls and control calls. I've did a experiment which can be done in both ways, Mod_perl and ESL. I don't know which one is better to take. When I see some earlier posts, It is given like Mod_perl has some limitations and I don't know what kind of limitations they are., Can someone say which is better to use and how it is better? -- View this message in context: http://www.nabble.com/Mod_perl-or-ESL-tp25520023p25520023.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Unable to set internal call to registered sip user
by Filip Lyncker 22 Sep '09

22 Sep '09

Dear List, I read the documentation, but Im still confused about how to dial a internal registered sip user. I configured the both sip phones in the directory in my local.xml file : <include> <domain name="$${domain}"> <user id="22" mailbox="22"> <params> <param name="password" value="Xk21%"></param> <param name="vm-password" value="22"></param> <param name="sip-port" value="5060"></param> </params> <variables> <variable name="accountcode" value="22"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 22"></variable> <variable name="effective_caller_id_number" value="22"></variable> </variables> </user> <user id="24" mailbox="24"> <params> <param name="password" value="dudeldum"></param> <param name="vm-password" value="24"></param> <param name="sip-port" value="5060"></param> </params> <variables> <variable name="accountcode" value="24"></variable> <variable name="user_context" value="default"></variable> <variable name="effective_caller_id_name" value="Extension 24"></variable> <variable name="effective_caller_id_number" value="24"></variable> </variables> </user> </domain> </include> It seems, that they can connect to the freeswitch. I configured the dialplan like following : <include> <context name="default"> <extension name="diallocal"> <condition field="destination_number" expression="^(2[0-9])$">  <action application="bridge" data="user/${dialed_extension}@${domain_name}"></action>   </condition> </extension> ... If I call from the sip user 24 to 22 , freeswitch logs the following and gives an busy tone immediately: freeswitch@Bigfish> 2009-09-22 13:50:29.367114 [NOTICE] switch_channel.c:602 New Channel sofia/internal/24(a)192.168.1.34 [decc119c-a973-6b4c-bf11-ec251c653cda] 2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing 24->22 in context default 2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user [@192.168.1.34] 2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed. Cause: SUBSCRIBER_ABSENT 2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup sofia/internal/24(a)192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session 13 (sofia/internal/24(a)192.168.1.34) Ended 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/24(a)192.168.1.34 [CS_DESTROY] thanks again for your help ... regards, Filip -- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis

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No ring tone while recording incoming call. Please help.
by Svetik VOIP 22 Sep '09

22 Sep '09

Hi, I have trouble recording incoming calls with FreeSwitch. I have followed the instruction from Misc. Dialplan Tools record session (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session) It works well for outgoing calls, but I have the problem with incoming calls. The person who is calling does not hear ring tone, he hears just the silence until I pick up the phone. Everything else is working, we can talk, conversation is recorded. Here is a copy of my dialplan for incoming calls /usr/local/freeswitch/conf/dialplan/public/voipms.xml <include> <extension name="voipms">  <condition field="destination_number" expression="XXXXXXXXXX">  <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 2009"/> <action application="set" data="RECORD_SOFTWARE=FreeSwitch"/> <action application="set" data="RECORD_ARTIST=FreeSwitch"/> <action application="set" data="RECORD_COMMENT=FreeSwitch"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="set" data="RECORD_ANSWER_REQ=true"/> <action application="set" data="ringback=${us-ring}"/> <action application="record_session" data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="bridge" data="user/user1@${domain_name}"/> </condition> </include> for outcoming calls I have a similar code added to the /usr/local/freeswitch/conf/dialplan/default/user1.xml and it works well. I have tried to move the line <action application="set" data="ringback=${us-ring}"/> between the lines <action application="record_session and <action application="bridge" but it did not solve my problem. Any ideas what am I doing wrong and how to fix it? Igor

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Displaying matched extension during a call
by Anil Kumar S. R. 22 Sep '09

22 Sep '09

Hi All, I am new to Freeswitch. So please bear with me if I ask any silly questions. * Can anyone of you please tell me how to display the extension name which has matched an incoming/outgoing call. * And can you please elaborate what does '<action application="info"/>' mean. * Suppose we have set a variable in the extension of the dialplan XML. Is there anyway we can display this variable on CLI for our debugging purposes. Regards, -- Anil Kumar S. R. http://sranil.googlepages.com/ "The best way to succeed in this world is to act on the advice you give to others."

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BLF docs/howto?
by Dmitry Bely 21 Sep '09

21 Sep '09

Have not found anything usable in the wiki/mail list archives. I'm trying to setup BLF (busy lamp field) for Grandstream GXP-2000 phone. It offers BLF/eventlist BLF modes. Does Freeswitch supports both including the latter (RFC4662)? How to setup BLF on Freeswitch side? Are there any examples? - Dmitry Bely

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Can I set "From:" field in originate command?
by Mikhail Krivushin 21 Sep '09

21 Sep '09

-- С уважением, Кривушин Михаил г. Томск сот. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME(a)jabber.ru skype: mkrivushin

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How to set the IP of REGISTER message??
by Brad Tuan 21 Sep '09

21 Sep '09

If my PC has two IP address, How to decide the IP which using to send REGISTER message?? For example, I have two IP , 172.30.30.XXX for External and 192.168.60.XXX for Internal, and now i want to add two gateway, one is 210.XXX.XXX.XXX and the other one is 192.168.60.30, How to change the IP that i want to use??

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