January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

problem with hard phone
by jwssr 31 Jan '10

31 Jan '10

Iam having probles with my hard phone...van access ip0020...which is not completing the 'B' leg on incoming calls to it and on outgoing calls....immedialtely issuing a 488... and fs log states ... switch_core_codec.c:537 Codec G722 Exists but not at the desired implementation. 16000hz 20ms 2010-01-30 11:49:07.736157 [ERR] sofia_glue.c:2126 Can't load codec? ... I have multiple soft phones (sflphones extensions 1000-1019), a dlink wifi dph-540 (extension 4001 ip=192.168.1.190), and this van access ip0020 hard phone (extension 2001 ip=192.168.1.180).....fs running on 192.168.1.102. all phones sans '2001" work perfectly..to/fro vitelity and local on lan. Im almost certain that the problem lies in the port number assigned to the uri. wireshark shows that call was cancel because port could not be reached and port number is missing on ua shown by fs cli. Call-ID: af4cfd49b52bf8b3b3ec8db3e8a309e5(a)192.168.1.190 User: 4001(a)192.168.1.102 Contact: 4001 <sip:4001@192.168.1.190:5060> ... ... Call-ID: 4c1ed9ae64405367(a)192.168.1.180 User: 2001(a)192.168.1.102 Contact: "user" <sip:2001@192.168.1.180> ... notice missing port no. here Call-ID: 9be86b04-720f-4f7c-af52-b4f67dccf76b User: 1000(a)192.168.1.102 Contact: "jon" <sip:1000@192.168.1.101:5060> .... wireshark..... 292 14.903431 192.168.1.102 192.168.1.180 ICMP Destination unreachable (Port unreachable) 295 14.927845 192.168.1.102 192.168.1.180 SIP Request: BYE sip:2001@192.168.1.180 I have output from siptrace...but did not want to clutter up forum with too much detail but can provide. I also have screenshots of html config of phone... I would appreciate some help please thanks -- View this message in context: http://old.nabble.com/problem-with-hard-phone-tp27386032p27386032.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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freeswitch with T.38
by Sharad 30 Jan '10

30 Jan '10

Is paid freeswitch available with T.38 support. ? regards Sharad -- View this message in context: http://n2.nabble.com/freeswitch-with-T-38-tp4485495p4485495.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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External Profile Problem
by juan camilo ospina quintero 30 Jan '10

30 Jan '10

Hi, im trying to establish a simple conference using freeswitch and sailfin, sailfin is and application server that works with SipSevlets. the all thing works as follow. two softphone register with freeswitch, extension 1000 and 1001 1000 sends and invite to 1001, this invite goes to sailfin, i use this <extension name="redirect"> <condition field="network_addr" expression="^192\.168\.2\.9$"/> <condition field="destination_number" expression="^1(\d+)$"> <action application="bridge" data="sofia/external/$0@127.0.0.1:5070"/> </condition> </extension> this is in /freeswitch/conf/dialplan/default.xml this far all goes well, the servlet receives the invite, and sends back the invite to freeswitch, i put this <extension name="receive"> <condition field="network_addr" expression="^192\.168\.2\.9$"/> <condition field="destination_number" expression="^1(\d+)$"> <action application="transfer" data="$0 XML default"/> </condition> </extension> in /freeswitch/conf/dialplan/public.xml, but freeswitch returns this ------------------------------------------------------------------------ 2010-01-26 20:14:29.512927 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000(a)192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1] 2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup sofia/external/1000(a)192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING] send 632 bytes to udp/[192.168.2.9]:5070 at 01:14:29.517927: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9 From: <sip:1000@192.168.2.9>;tag=g4xfbi12-3 To: <sip:1001@192.168.2.9:5080>;tag=4r91165pvcycB Call-Id: 192.168.153.1_3_3990383226484831353 Cseq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING" Content-Length: 0 ------------------------------------------------------------------------ 2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1086 Session 9 (sofia/external/1000(a)192.168.2.9) Ended 2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1000(a)192.168.2.9 [CS_DESTROY] i dont understand why i doesnt work if in public.xml, i tell to transfer the call if extension starts with 1 and the caller ip address is 192.168.2.9 please if someone can help me _________________________________________________________________ Windows Live: Keep your friends up to date with what you do online. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so…

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little hangup problem - prepaid application
by Christian Löschenkohl 30 Jan '10

30 Jan '10

hello i try to implement a prepaid application and mod_nibblebill isn't my first choice. the implementation is quite ok so far. now my problem: - the call setup is normal (a little scripting) - the calls get ended by sched_hangup (max duration of calls) - then i would like to execute a script with api_hangup_hook in this last script i would also like to use some channel vars (a-number, call duration after answer) but i can't do that i try to use an outbound event socket script but i couldn't get the variables/information i need to calculate the amount charge any hint here br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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mutiple playback files (unescape_char) Windows
by Diego Toro 30 Jan '10

30 Jan '10

Greetings I have a next section dial plan: <action application="set" data="sound_prefix=$${base_dir}\sounds\es\co\callie\" /> <action application="set" data="playback_delimiter=!"/> <action application="playback" data="misc\8000\serror.wav!misc\8000\provide_reference_number.wav!digits\8000\5.wav" /> .... I have a problem using multiple playback files on Windows, the path misc\8000\serror.wav is changed by misc\8000 serror.wav. I check C code on switch_utils.c, cleanup_separated_string function call to unescape_char function which change \s by ' '. This is correct, but on Windows '\' is the path separator, so is not possible to use '\s', '\n'... into path file. I think this is possible to fix it. Thanks Diego Toro http://lacarretade.blogspot.com/

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Custom NOTIFY message in FS
by Mark Campbell-Smith 30 Jan '10

30 Jan '10

Hi! Is there any way that a custom NOTIFY message can be built and sent in FS without cutting code? I have a bunch of Linksys SPA's and I want to implement the Resync_From_SIP option, which enables a resync to be triggered via a SIP NOTIFY message. (ie Event: resync in the NOTIFY message) Is this at all possible? Thanks!

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ClueCon MMX - Save the Date!
by Michael Collins 30 Jan '10

30 Jan '10

ClueCon MMX (2010) will be here before you know it! Please mark your calendars: August 3-5, 2010. Start talking up ClueCon with your peers, coworkers, business owners, CEOs, potential sponsors, and anyone else you can think of. It's coming fast, so start getting ready now. Lots more information will be coming soon. Looking forward to seeing everyone this August! -ClueCon Team

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Posting a Bounty
by Jerry Richards 29 Jan '10

29 Jan '10

How do I post a new Bounty request? Thanks, Jerry

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wiki password recovery - no mail is send
by Christian Löschenkohl 29 Jan '10

29 Jan '10

hello the password recovery for the fs wiki doesn't seem to work. no e-mail is send when entering the username and press "e-mail new password". may i assist here, we do maintain a few wikis for ourself. br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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Openzap w/ DAHDi Linux 2.2.1
by Joseph L. Casale 29 Jan '10

29 Jan '10

Anyone running this with their latest 2.2.1 release successfully using Digium analog TDM cards? Is it known to work fine, or possibly not been tested yet? Thanks, jlc

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