December 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSwitch Video Calls Bria iPhone Client
by Mike Burlingame 27 Dec '12

27 Dec '12

I have been playing around trying to get my iPad and iPhone using Bria to work with video it *seems* that FreeSwitch is passing the signaling to both devices and FreeSwitch is holding the media due to the NAT issue of the devices - I just wanted to know before I spent too much time on this (not a high priority) if anyone else has gotten FreeSwitch to work with Bria for iPhone or Android? Thanks

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CLIR gsmopen
by dchs dchs 27 Dec '12

27 Dec '12

Hello, Is there any way to set up CLIR on outgoing calls placed over gsmopen?Do Huawei modems even support such an AT command? Any pointers towards any info on how to accomplish this would be much appreciated!

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xml_curl and ivr menus
by Tim Meade 27 Dec '12

27 Dec '12

I am continually getting 'unable to find menu' while trying to retrieve an ivr_menu using xml_curl Using xml_curl debug_on My DialPlan <document type="freeswitch/xml"> <section name="dialplan" description="example_curl_dialplan"> <context name="public"> <extension name="ws_origination"> <condition field="destination_number" expression="^\d+$" break="on-true"> <action inline="true" application="set" data="continue_on_fail=true"/> <action inline="true" application="set" data="hangup_after_bridge=true"/> <action application="answer"/> <action application="ivr" data="tivr"/> </condition> </extension> </context> </section> And the ivr configuration returned by my xml_curl <include>   <menu name="tivr" greet-long="phrase:demo_ivr_main_menu" greet-short="phrase:demo_ivr_main_menu_short" invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" exit-sound="voicemail/vm-goodbye.wav" confirm-macro="" confirm-key="" tts-engine="flite" tts-voice="rms" confirm-attempts="3" timeout="10000" inter-digit-timeout="2000" max-failures="3" max-timeouts="3" digit-len="4">   <entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888(a)conference.freeswitch.org"/> <entry action="menu-exec-app" digits="2" param="transfer 9196 XML default"/>  <entry action="menu-exec-app" digits="3" param="transfer 9664 XML default"/>  <entry action="menu-exec-app" digits="4" param="transfer 9191 XML default"/>  <entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/>   <entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/> <entry action="menu-top" digits="9"/>  </menu> </include> Any ideas on the 'unable to find menu' ???? Thanks Tim

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Avoid loopback channels
by Jose Fco. Irles Durá 27 Dec '12

27 Dec '12

Hello everybody, I want to avoid using loopback channels in freeswitch, but I don't know the correct way to build some behaviors. For examle, a click2call (origination) to a "virtual destination": I have a logic in my dialplan that when somebody dials XXX, it sends the call through a round robin set of gateways (saving de last gw used with the "hash/db" app and dialing sequentially from the least used to most used if one of it fails), what is the freeswitch way for this without loopback channels? At this moment, i do this originate: In the example, first ring in 256987455 (the call goes through dialplan and fs bridge with diferent gws if some fails) and when it answers, I transfer the call to the same dialplan with a lua script (the scripts uses "transfer" app). The originate data is: {origination_uuid=uuid,bridge_generate_comfort_noise=true,origination_caller_id_name='12547895',origination_caller_id_number=12547895,effective_caller_id_number=12547895,effective_caller_id_name='12547895'}loopback/256987455/default &lua('originate.lua uuid 12547895) I think that with a "calculated" dial-string with the gateways to dial and with variables "execute_on_XXX" it's a possibily, but it's the only way? Another case is dial to one of this "virtual destinations" in a ring all group or dial to a sip device with call forwarding if no answer set at the pbx (not at the phone). Thanks and sorry if i don't write properly. -- Jose Fco. Irles Durá

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repeated registration messages in logs
by Archana Venugopan 27 Dec '12

27 Dec '12

Hi, After registering the phones successfully I am getting the below message continuously every few seconds in my logs and this is filling up the logs and freeswitch goes down sometimes. Can someone please tell me why I keep getting this message every few seconds? I have another server and there I don't see this issue. 2012-12-27 12:37:44.714558 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [Abarajit(a)vectonecloud.com] from ip 192.168.2.205 2012-12-27 12:41:58.744557 [WARNING] sofia_reg.c:1484 SIP auth challenge (REGISTER) on sofia profile 'internal' for [sana(a)vectonecloud.com] from ip 192.168.2.48 Regards, Archana

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Error while making Attendent transfer
by veerabhadrarao` 27 Dec '12

27 Dec '12

hai, i am working on Freeswitch. I am getting error which i displayed below while i making attendent transfer using default and features xml files. And also tell me how to make successful attendent transfer in freeswitch 2012-12-27 18:29:16.662856 [CRIT] mod_local_stream.c:297 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1237] I am using twinkle and SJphone as soft phones. please help me thank you

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FS sends BYE after 32 seconds
by Алексей Дорофеев 27 Dec '12

27 Dec '12

Hello, all! I need some help. I have one problem with incoming calls - FS hung up after 32 seconds. I think, FS does not accepted ACK because the field RequestURI is different from the same field in INVITE... Call trace: ------------------------------------------------------------------------ recv 1054 bytes from udp/[10.10.132.203]:5060 at 20:56:09.448547: ------------------------------------------------------------------------ INVITE sip:gw+voip.mtt.ru@11.111.88.147:5080;transport=udp;gw=voip.mtt.ruSIP/2.0 Via: SIP/2.0/UDP 10.10.132.203:5060 ;branch=z9hG4bK-d8754z-4438ff39749bbf13-1---d8754z-;rport Via: SIP/2.0/UDP 10.10.132.203:5061 ;branch=z9hG4bK-giy2ccrl24zvsgqk;rport=5061 Max-Forwards: 69 Record-Route: <sip:10.10.132.203:5060;lr;transport=UDP> Contact: "Anonymous"<sip:10.10.132.203:5061> To: <sip:7812BBBBBBB@10.10.132.203> From: "7812AAAAAAA"<sip:+7812AAAAAAA@10.10.132.203>;tag=hmxjwhzuanhi5qq7.o Call-ID: 1356-548022-325095~o CSeq: 662 INVITE Expires: 300 Content-Disposition: session Content-Type: application/sdp User-Agent: Sippy h323-conf-id: 1868605416-458121599-1280693483-4049122446 cisco-GUID: 1868605416-458121599-1280693483-4049122446 Content-Length: 283 ------------------------------------------------------------------------ send 513 bytes to udp/[10.10.132.203]:5060 at 20:56:09.448869: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.132.203:5060 ;branch=z9hG4bK-d8754z-4438ff39749bbf13-1---d8754z-;rport=5060 Via: SIP/2.0/UDP 10.10.132.203:5061 ;branch=z9hG4bK-giy2ccrl24zvsgqk;rport=5061 Record-Route: <sip:10.10.132.203:5060;lr;transport=UDP> From: "7812AAAAAAA"<sip:+7812AAAAAAA@10.10.132.203>;tag=hmxjwhzuanhi5qq7.o To: <sip:7812BBBBBBB@10.10.132.203> Call-ID: 1356-548022-325095~o CSeq: 662 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.10b+git~20121209T055926Z~bfc3c17bcb Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.132.203:5060 ;branch=z9hG4bK-d8754z-4438ff39749bbf13-1---d8754z-;rport=5060 Via: SIP/2.0/UDP 10.10.132.203:5061 ;branch=z9hG4bK-giy2ccrl24zvsgqk;rport=5061 Record-Route: <sip:10.10.132.203:5060;lr;transport=UDP> From: "7812AAAAAAA"<sip:+7812AAAAAAA@10.10.132.203>;tag=hmxjwhzuanhi5qq7.o To: <sip:7812BBBBBBB@10.10.132.203>;tag=U98Z2BN38jemS Call-ID: 1356-548022-325095~o CSeq: 662 INVITE Contact: <sip:7812BBBBBBB@11.111.88.147:0;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.3.10b+git~20121209T055926Z~bfc3c17bcb Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "7812BBBBBBB" <sip:7812BBBBBBB@10.10.132.203 >;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ recv 466 bytes from udp/[10.10.132.203]:5060 at 20:56:09.594850: ------------------------------------------------------------------------ ACK sip:7812BBBBBBB@11.111.88.147;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.10.132.203:5060 ;branch=z9hG4bK-d8754z-a05ecb4295a54946-1---d8754z-;rport Via: SIP/2.0/UDP 10.10.132.203:5061 ;rport=5061;branch=z9hG4bK-z4gyohqxmtxf5hg6 Max-Forwards: 69 To: <sip:7812BBBBBBB@10.10.132.203>;tag=U98Z2BN38jemS From: "7812AAAAAAA"<sip:+7812AAAAAAA@10.10.132.203>;tag=hmxjwhzuanhi5qq7.o Call-ID: 1356-548022-325095~o CSeq: 662 ACK User-Agent: Sippy Content-Length: 0 nua_server.c:155 nua_stack_process_request() nua(0x1c7b6e0): strange ACK from <sip:+7812AAAAAAA@10.10.132.203> <<< *this exchange 200 OK - ACK repeats several times...* *>>> * ------------------------------------------------------------------------ send 691 bytes to udp/[10.10.132.203]:5060 at 20:56:41.566627: ------------------------------------------------------------------------ BYE sip:10.10.132.203:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5080;rport;branch=z9hG4bK0N7S27DDaFByF Route: <sip:10.10.132.203:5060;lr;transport=UDP> Max-Forwards: 70 From: <sip:7812BBBBBBB@10.10.132.203>;tag=U98Z2BN38jemS To: "7812AAAAAAA" <sip:+7812AAAAAAA@10.10.132.203>;tag=hmxjwhzuanhi5qq7.o Call-ID: 1356-548022-325095~o CSeq: 37941636 BYE Contact: <sip:7812BBBBBBB@11.11.88.147:0;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.3.10b+git~20121209T055926Z~bfc3c17bcb Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Reason: SIP;cause=408;text="ACK Timeout" Content-Length: 0 ------------------------------------------------------------------------ recv 333 bytes from udp/[10.10.132.203]:5060 at 20:56:41.586770: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:5080 ;received=11.11.88.147;rport=5080;branch=z9hG4bK0N7S27DDaFByF To: "7812AAAAAAA"<sip:+7812AAAAAAA@10.10.132.203>;tag=hmxjwhzuanhi5qq7.o From: <sip:7812BBBBBBB@10.10.132.203>;tag=U98Z2BN38jemS Call-ID: 1356-548022-325095~o CSeq: 37941636 BYE Server: Sippy Content-Length: 0 Thank you for you attention! Alex.

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New to Freeswitch
by Vijai Ganapathy 26 Dec '12

26 Dec '12

Dear all, I am new to freeswitch and I have a question to ask. I am planning to use gsm_open to have small call center setup. >From the local market I could get Huawei E1732 model Dongle only and I am wondering whether freeswitch / Linux Kernel would support this model for voice ? Also would ATOM or Phenom would be enough to run 10 seat call center ? Want to check with experts before jumping on to actual setup. -- Vijai Ganapathy Mailto: Vijai.Ganapathy(a)gmail.com

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XML DialPlan Gateay dialing redundancy
by Sean Devoy 26 Dec '12

26 Dec '12

I need a little help with a complex XML dialplan. Also, I am on the Stable release so I do not believe the much needed nested conditions exists for me yet. My problem is faulty gateway providers who periodically fail to connect the call with [temp failure] because "our routing tables where not up to date." So what I am after is an xml dialplan to do this If (it is a 10 or 11 digit dial) Dial using gateway 1 If no answer Go to voice mail Else if gateway failure Dial using gateway 2 If no answer Go to voice mail Else if gateway failure Dial using gateway 3 If no answer Go to voice mail Else if gateway failure Play a message that the number is screwed! Perhaps I need to breakdown and use a scripting language like LUA or JavaScript for this, but I would rather keep everything in XML if possible. Any thoughts? Happy Holidays to all, Sean

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Check user status before bridge
by Lloyd Aloysius 26 Dec '12

26 Dec '12

Hello, Is there a way using lua to check the user status before bridge a call. I want to check if the user is already on another the call or session? Thanks Lloyd

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