November 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Rejecting a call from JavaScript
by Yehavi Bourvine 02 Nov '09

02 Nov '09

Hello, We would like to handle an incoming call to a busy phone according to user's prefference: Some want waiting call, some want to just reject the call, and others want to send the call to voicemail. We have a small JavaScript which tests the status of the destination and the user's will and tries to act accordingly. Our problem is how to send busy. I tried session.hangup("USER_BUSY") but it always sends "temporary unavailable" which causes the orignator to think that the destination is out of order. What is the correct way to do so? Thanks! __Yehavi:

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Get error "415 Unsupported Media Type" when receiving call from softswitch
by Lei Tang 02 Nov '09

02 Nov '09

Hi all, I get a "415 Unsupported Media Type" when FS receiving call from a softswitch. I captured some packets, It seems that the softswitch use SIP-I protocol, does FS can handle SIP-I message? ===here is the invite messagefrom softswitch INVITE sip:xxxxx@xxxx:5060;user=phone SIP/2.0 Contact: <sip:xxxx@xxxx:5061> MIME-version: 1.0 Content-Type: multipart/mixed;boundary=Alcatel-boundary To: <sip:xxxx@xxxx:5060;user=phone> From: xxxx<sip:xxxx@xxxxx:5061;user=phone>;tag=73D332463135364195291201 P-Asserted-Identity: <sip:xxxx@xxxxx:5061;user=phone> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE Supported: 100rel,timer,replaces,diversion Expires: 155 Session-Expires: 1800 Min-SE: 90 Call-ID: 01FD034872814000000230A1@sip-3 Max-Forwards: 70 CSeq: 1 INVITE Timestamp: 10645 Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E Content-Length: 542 --Alcatel-boundary Content-Type: application/sdp v=0 o=- 5 8 IN IP4 xxxxxx s=SDP Data c=IN IP4 xxxxx t=0 0 m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97 a=rtpmap:110 speex/8000/1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 TELEPHONE-EVENT/8000 a=fmtp:110 mode=3 a=rtpmap:111 speex/8000/1 a=rtpmap:3 GSM/8000 a=ptime:30 a=fmtp:111 mode=2 a=rtpmap:112 speex/8000/1 --Alcatel-boundary Content-Type: application/ISUP;version=N/A .. .... ...0.D... ...V..0......1...n..p... --Alcatel-boundary-- =====response message from FS SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E From: xxxxx <sip:xxxxx@xxxx:5061;user=phone>;tag=73D332463135364195291201 To: <sip:xxxxx@xxxxx:5060;user=phone>;tag=m6XS44BKF4pSS Call-ID: 01FD034872814000000230A1@sip-3 CSeq: 1 INVITE Timestamp: 10645 0.000000 User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Accept: application/sdp Accept-Encoding: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 -- Lei.Tang lei.tlfly(a)gmail.com

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Fwd: Many CS_REPORTING state Zombie session
by Seven Du 02 Nov '09

02 Nov '09

Just suspicious would be possible that happened on sqlite stage? I manually deleted the channels from sqlite and nothing bad happend. just FYI. ---------- Forwarded message ---------- From: Seven Du <dujinfang(a)gmail.com> Date: Sun, 1 Nov 2009 10:24:32 +0800 Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session To: Thank you Anthony. We are on r14696 and no non-standard mods loaded. we even unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both inbound and outbound. I must be very careful if I upgrade to trunk and turn on rwlock debug because it's on production and the the problem not happening that much so would be hard to trace. But I will find time to test and report back. FYI, I also noticed that in some zombile channels, couple of INVITEs sent but never got a response. However, it sent out an ACK at last.

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Many CS_REPORTING state Zombie session
by Dome Charoenyost 01 Nov '09

01 Nov '09

Dear All, I'm disable all CDR and use nibble_bill (modify) and odbc_query for create CDR and dedcut balance. but i found many zombie channel and can't kill Like a 51555b02-c5ba-11de-af77-774e15238dbe,outbound,2009-10-30 18:11:38,1256951498,sofia/external/66848087977@xxx.xxx.xx.191:7050,CS_REPORTING,FreeSWITCH,66850304266,xxx.xxx.xxx.191,7050,,,XML,public,,,,, I'm using FreeSWITCH Version 1.0.5pre3 (hacked) How to fix ? Dome C.

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Mod_pjsip
by Meftah Tayeb 01 Nov '09

01 Nov '09

Hi my FreeSWITCH friends Today we talked about FreeSWITCH libraries and we found pjsip, which is a lot better than sofia, it can do everything sofia can do and more, it's thread-safe and performs well, I would love to see a mod_pjsip. The library is GPL but they also offer alternative licensing, don't get me wrong, Sofia is great but I think FreeSWITCH will benefit from a thread-safe SIP library. don't say that i need some stuf for free, but we need to make fs a very also we love freeswitch to run in a Embedded PC like the blackfin CPU/boards, Pjsip is very good for Embedded SIP Devicesawesome Open Source project With much love, DelphiWorld____ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __________ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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Small bug in switch_ivr_record_file (in trunk)
by Jonas Gauffin 01 Nov '09

01 Nov '09

switch_ivr_play_say.c, line 486. file = switch_core_session_sprintf(session, "%s%s%s%s", switch_str_nil(tfile), tfile ? "]" : "", prefix, SWITCH_PATH_SEPARATOR, file); There should be five %s, not four.

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SIP Proxy with direct media path
by Dmitry Gromov 01 Nov '09

01 Nov '09

Hi! I was a long time user of MySipSwitch service (www.mysipswitch.com) but it looks like they are planning to close their service. I can't use the replacement they have and would actually prefer some locally installed solution. Question - is it possible to use FreeSwitch exactly like MySipSwitch - just SIP Proxy with very flexible dial plan? With MySipSwitch it is possible to: - register service with multiple SIP providers as client - register your ATA with MySipSwitch - create smart dial plan (in Ruby) to route calls based on prefixes and/or call source and some other logic - accept and route calls from all registered providers MySipSwitch does not have any media capabilities - RTP is always established directly between call parties (no re-Invites either). If this is possible - is there any beginner guide for this? Also - I know that FreeSwitch could run in Windows as well as in Linux - would it be possible to use the above functionality in both OSes? Thanks, Dmitry -- DG NJ

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multihomed help
by Joseph L. Casale 01 Nov '09

01 Nov '09

I am setting up fs in pfsense. Following the multihomed tutorial (I also have a dedicated wan/lan int) if I set directory/default.xml domain=10.0.0.1 (my lan int ip) it breaks everything, but if I set <X-PRE-PROCESS cmd="set" data="domain=10.0.0.1"/> in conf/vars.xml I now get audio working correctly, *9999 etc plays MOH. Is there still something I have done wrong? conf/sip_profiles/internal.xml has the int lan ip's entered, and external.xml does not have any edits. Thanks for any info! jlc

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 40, Issue 179
by Ahmed Munir 01 Nov '09

01 Nov '09

Hi, Thanks for reply, it really helped me. One more thing to ask, how can we make decision against >,<, >=, <= in condition header? Like we use == for action and != for anti-action. Kindly highlight it. > > ---------- Forwarded message ---------- > From: Ahmed Munir <ahmedmunir007(a)gmail.com> > To: FreeSwitch <freeswitch-users(a)lists.freeswitch.org> > Date: Wed, 21 Oct 2009 15:37:15 +0500 > Subject: [Freeswitch-users] Call custom variable in condition > Hi, > > I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I > want to use it in condition, as I'm listing down the configuration below; > > <context name="SIP_incoming"> > <extension name="call-sip-extensions"> > <condition field="destination_number" expression="^(\d+)$"> > <action application="set" data="AUTHENTICATION_STATUS=0"/> > <action application="transfer" data="${AUTHENTICATION_STATUS} > XML Authen_Status"/> > </condition> > </extension> > </context> > > <context name="Authen_Status"> > <extension name="exten-auth-status"> > <condition field="AUTHENTICATION_STATUS" expression="^0$"> > <action application="answer"/> > <action application="playback" data="play.wav"/> > </condition> > </extension> > </context> > > > > > But unfortunately it is not working. Kindly advise me how to do implement > it(Note: I don't want to call script). And one more thing to ask how can I > transfer the values within the same context? > > -- > Regards, > > Ahmed Munir > > > > > ---------- Forwarded message ---------- > From: Ghulam Mustafa <mustafa.pk(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 21 Oct 2009 15:52:24 +0500 > Subject: Re: [Freeswitch-users] Call custom variable in condition > Ahmed, > > you can't use variables set by "set" application within a condition, though > it doesn't make sense. wondering if there is any logic behind this or it's > just a simple missing feature. anyone? > > -m > > Ahmed Munir wrote: > >> Hi, >> >> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I >> want to use it in condition, as I'm listing down the configuration below; >> >> <context name="SIP_incoming"> >> <extension name="call-sip-extensions"> >> <condition field="destination_number" expression="^(\d+)$"> >> <action application="set" data="AUTHENTICATION_STATUS=0"/> >> <action application="transfer" data="${AUTHENTICATION_STATUS} >> XML Authen_Status"/> >> </condition> >> </extension> >> </context> >> >> <context name="Authen_Status"> >> <extension name="exten-auth-status"> >> <condition field="AUTHENTICATION_STATUS" expression="^0$"> >> <action application="answer"/> >> <action application="playback" data="play.wav"/> >> </condition> >> </extension> >> </context> >> >> >> >> >> But unfortunately it is not working. Kindly advise me how to do implement >> it(Note: I don't want to call script). And one more thing to ask how can I >> transfer the values within the same context? >> >> -- >> Regards, >> >> Ahmed Munir >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > > > ---------- Forwarded message ---------- > From: Tihomir Culjaga <tculjaga(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 21 Oct 2009 13:13:13 +0200 > Subject: Re: [Freeswitch-users] Call custom variable in condition > consider this: > > > > <context name="SIP_incoming"> > <extension name="call-sip-extensions"> > <condition field="destination_number" expression="^(\d+)$"> > <action application="set" data="AUTHENTICATION_STATUS=0"/> > <action application="transfer" data="${AUTHENTICATION_STATUS} > XML Authen_Status"/> > </condition> > </extension> > </context> > > > > <context name="Authen_Status"> > <extension name="exten-auth-status"> > <condition field="${AUTHENTICATION_STATUS}" expression="^0$"> > <action application="answer"/> > <action application="playback" data="play.wav"/> > </condition> > </extension> > </context> > > > > > > here is one of my dialplan. I'm using execute_extension but it is quite the > same... > > > > <extension name="ServiceLookup"> > <condition field="destination_number" expression="(^300030)(.*)"> > <action application="lookup_service_destination" data="in > ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $ > 1, in ${network_addr}:5060, out red_contact, out authResult"/> > <action application="log" data="INFO ######################## > ServiceLookup ########################\n"/> > <action application="log" data="INFO ######################## > contact = '${red_contact}' ##############\n"/> > <action application="log" data="INFO ######################## > CallerNum = '${caller_id_number:6:16}' ##########\n"/> > <action application="log" data="INFO ######################## > RADIUS auth = '${authResult}' ##########\n"/> > > <action application="execute_extension" data="doRedirect XML > public"/> > </condition> > </extension> > > > <extension name="doRedirect"> > <condition field="destination_number" expression="^doRedirect$"/> > <condition field="${authResult}" expression="^0$|^60$"> > <action application="log" data="INFO ######################## > RADIUS auth OK!!!' ##########\n"/> > <action application="redirect" data="${red_contact}"/> > <anti-action application="log" data="INFO ######################## > RADIUS auth NOK!! ##########\n"/> > <anti-action application="respond" data="403 Forbidden"/> > </condition> > > </extension> > > > > > On Wed, Oct 21, 2009 at 12:37 PM, Ahmed Munir <ahmedmunir007(a)gmail.com>wrote: > >> Hi, >> >> I've declared a variable named AUTHENTICATION_STATUS in dialplan, and I >> want to use it in condition, as I'm listing down the configuration below; >> >> <context name="SIP_incoming"> >> <extension name="call-sip-extensions"> >> <condition field="destination_number" expression="^(\d+)$"> >> <action application="set" data="AUTHENTICATION_STATUS=0"/> >> <action application="transfer" data="${AUTHENTICATION_STATUS} >> XML Authen_Status"/> >> </condition> >> </extension> >> </context> >> >> <context name="Authen_Status"> >> <extension name="exten-auth-status"> >> <condition field="AUTHENTICATION_STATUS" expression="^0$"> >> <action application="answer"/> >> <action application="playback" data="play.wav"/> >> </condition> >> </extension> >> </context> >> >> >> >> >> But unfortunately it is not working. Kindly advise me how to do implement >> it(Note: I don't want to call script). And one more thing to ask how can I >> transfer the values within the same context? >> >> -- >> Regards, >> >> Ahmed Munir >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Ahmed Munir

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mod_t38gateway
by Tihomir Culjaga 01 Nov '09

01 Nov '09

does anybody know how does it work and how to use it in a dialplan? freeswitch@nemesis> freeswitch@nemesis> freeswitch@nemesis> load mod_t38gateway API CALL [load(mod_t38gateway)] output: +OK 2009-10-30 22:44:38.204268 [NOTICE] mod_t38gateway.c:147 T.38 gateway enabled 2009-10-30 22:44:38.204268 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [mod_t38gateway] 2009-10-30 22:44:38.204268 [NOTICE] switch_loadable_module.c:248 Adding Application 't38gateway' 2009-10-30 22:44:38.205374 [NOTICE] switch_loadable_module.c:270 Adding API Function 't38gateway' freeswitch@nemesis> freeswitch@nemesis> freeswitch@nemesis> freeswitch@nemesis> t38gateway API CALL [t38gateway()] output: -USAGE: <uuid> <command> freeswitch@nemesis> T.

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