July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how to configure pin for use in gsmopen ?
by Luis Azedo 04 Jul '13

04 Jul '13

Hi, how do i enter the sim card pin for use with Huawei K3715 ? thank you -- Luis Azedo

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SIP Trunking
by Ashwin Jain 04 Jul '13

04 Jul '13

Is there anyway in which we can authenticate SIP phones based on MAC addresses? -- Thanks and Regards, Ashwin Jain

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Upon re-INVITE, FS responds call does not exist. Help!
by Avi Marcus 04 Jul '13

04 Jul '13

PCAP of the SIP: http://ge.tt/923EDxk/v/0 I've got a lot of calls to a CLEC which the SIP says MetaSwitch. A about 930 seconds from the start, a timer kicks in with a re-INVITE but FS responds 481 Call Does Not Exist. Then the other side tears down the call, even though FS keeps sending media for another 10-246 seconds. pcapsipdump puts all the packets together, so why doesn't FS? The call ID seems to be the same. I'm on FreeSWITCH (Version 1.2.8 git 03f9daf 2013-04-24 20:52:10Z) I know it used to work with this carrier, and while they are investigating -- and claiming that it's not happening with anyone else -- I hoped someone could see something funny in the PCAP that would help. Thanks! -Avi

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attended transfer - fails
by Lloyd Aloysius 04 Jul '13

04 Jul '13

Hi All, I could not figure it out why the attended transfer is failing. UA1 - Call - UA2 UA2 Answer UA2 Transfer - Dial UA3 While UA3 is Ring UA2 Press the Transfer After transfer complete. *Issues* 1. Now UA1 hear the MOH only not the UA3 Ring back 2. After timeout the call never reach the UA3 voice mail ( This is the timeout option ) and the call is just hangup UA Bridge Data - {sip_invite_domain=alcan.mydomain.ca ,ignore_early_media=true,force_transfer_context=alcan.mydomain.ca ,originate_timeout=20}[leg_timeout=20]${sofia_contact( nelson(a)alcan.mydomain.ca)} Any help is appreciated. Thanks Lloyd

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mod_callcenter makes me BREAK_OUT
by Sayyed Mohammad Emami Razavi 04 Jul '13

04 Jul '13

After configuring agents and queues, when i call and join to the queue, call center makes me BREAK_OUT Any solution?! FS logs paste bin http://pastebin.freeswitch.org/21144

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ptime transcoding does not pass rfc2833
by Chris Martineau 04 Jul '13

04 Jul '13

Hi, Does anyone know if this is a bug or a setting? If I use an incoming call ptime of say 40 on gsm to save client bandwidth and output to a carrier at 20 the rtp is transcoded to the smaller ptime fine but any rfc2833 dtmf signalling is not passed through. I have tried the pass_rfc2833 setting both on and off but it has no effect. I have done some web searches but can find nothing refering to this issue. Any help you could offer would be greatly appreciated. FreeSWITCH version: 1.0.head (git-58c3c3a 2011-11-22 18-22-57 -0600) Regards Chris

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effective_caller_id_name &effective_caller_id_number - not working
by Avi Marcus 04 Jul '13

04 Jul '13

... but it's not in the siptrace at all... Can we get more lines of the fs_cli log? From the setting of CID to the actual log? -Avi On Tue, Jul 2, 2013 at 3:10 AM, Moishe Grunstein <max(a)nysolutions.com>wrote: > What type of caller ID format is your ITSP expecting? Who is the ITSP?**** > > http://wiki.freeswitch.org/wiki/Channel_Variables#Caller_ID_Related**** > > ** ** > > ** ** > > ** ** > > Thanks,**** > > ** ** > > Moishe Grunstein**** > > Tornado Computer Systems, Inc.**** > > 212.400.7650 888.IPPBX.US > *Service Request Email: support(a)nysolutions.com ***** > > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified**** > > [image: cid:image001.jpg@01C72F94.9EE45D60] <http://www.nysolutions.com/>* > *** > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS**** > > ** ** > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Lloyd > Aloysius > *Sent:* Monday, July 01, 2013 7:39 PM > *To:* FreeSWITCH Users Help > *Subject:* spam>spam>[Freeswitch-users] effective_caller_id_name > &effective_caller_id_number - not working**** > > ** ** > > Hi All,**** > > ** ** > > When sending calls out from gateway the caller id name and number not > working. Please find the freeswitch cli log and sip invite trace. Any help > is appreciated.**** > > ** ** > > *freeswitch cli***** > > ** ** > > EXECUTE sofia/sipinterface_1/barbara(a)alcan.mydomain.caset(effective_caller_id_name=Alcan Cable) > **** > > 2013-07-01 19:23:21.676990 [DEBUG] mod_dptools.c:1373 sofia/sipinterface_1/ > barbara(a)alcan.mydomain.ca SET [effective_caller_id_name]=[Alcan Cable]**** > > EXECUTE sofia/sipinterface_1/barbara(a)alcan.mydomain.caset(effective_caller_id_number= > 9052066900)**** > > 2013-07-01 19:23:21.676990 [DEBUG] mod_dptools.c:1373 sofia/sipinterface_1/ > barbara(a)alcan.mydomain.ca SET [effective_caller_id_number]=[9052066900]*** > * > > ** ** > > *SIP Trace***** > > ** ** > > ** ** > > INVITE sip:4164750401@alcan.mydomain.ca SIP/2.0**** > > ** ** > > Via: SIP/2.0/UDP 192.168.240.105:5060;branch=z9hG4bK17bfd0ad4**** > > ** ** > > Max-Forwards: 70**** > > ** ** > > Content-Length: 285**** > > ** ** > > To: 4164750401 <sip:4164750401@alcan.mydomain.ca>**** > > ** ** > > From: Barbara Ladner <sip:barbara@alcan.mydomain.ca>;tag=02f4abe9bef3a7f** > ** > > ** ** > > Call-ID: 43016333178f7b0882c3767aab886c1d(a)192.168.240.105**** > > ** ** > > CSeq: 1894816474 INVITE**** > > ** ** > > Supported: timer**** > > ** ** > > Allow-Events: talk,hold,conference**** > > ** ** > > Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO**** > > ** ** > > Content-Type: application/sdp**** > > ** ** > > Contact: Barbara Ladner <sip:barbara@192.168.240.105:5060;transport=udp>** > ** > > ** ** > > Supported: replaces**** > > ** ** > > User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45** > ** > > ** ** > > ** ** > > ** ** > > v=0**** > > ** ** > > o=MxSIP 0 575681134 IN IP4 192.168.240.105**** > > ** ** > > s=SIP Call**** > > ** ** > > c=IN IP4 192.168.240.105**** > > ** ** > > t=0 0**** > > ** ** > > m=audio 3000 RTP/AVP 0 8 18 101**** > > ** ** > > a=rtpmap:0 PCMU/8000**** > > ** ** > > a=rtpmap:8 PCMA/8000**** > > ** ** > > a=rtpmap:18 G729/8000**** > > ** ** > > a=rtpmap:101 telephone-event/8000**** > > ** ** > > a=fmtp:101 0-15**** > > ** ** > > a=ptime:20**** > > ** ** > > a=silenceSupp:off - - - -**** > > ** ** > > ** ** > > SIP/2.0 100 Trying**** > > ** ** > > Via: SIP/2.0/UDP 192.168.240.105:5060 > ;branch=z9hG4bK17bfd0ad4;received=99.247.64.220;rport=16511**** > > ** ** > > From: Barbara Ladner <sip:barbara@alcan.mydomain.ca>;tag=02f4abe9bef3a7f** > ** > > ** ** > > To: 4164750401 <sip:4164750401@alcan.mydomain.ca>**** > > ** ** > > Call-ID: 43016333178f7b0882c3767aab886c1d(a)192.168.240.105**** > > ** ** > > CSeq: 1894816474 INVITE**** > > ** ** > > User-Agent: FreeSWITCH-mod_sofia/1.5.3b+git~20130628T160527Z~a52a604fbb*** > * > > ** ** > > Content-Length: 0**** > > ** ** > > ** ** > > Thanks**** > > Lloyd**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > >

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[announce] mod_GsmOpen: IMEI and IMSI based device discovery on Linux
by Giovanni Maruzzelli 04 Jul '13

04 Jul '13

Hello FreeSWITCHers, mod_gsmopen allows for connecting to the cellular network for originating and terminating voice calls and SMSes using cheap dongles (eg: internet keys). Many peoples found annoying to have to discover by trial and error which are the correct ttyUSB devices name to write in configuration file. Also because on Linux they can (and do) change at reboot if you use multiple dongles. IMEI is the identifier-id built in in each dongle, unique worldwide for each individual device - like MAC address for Ethernet. IMSI is the worldwide unique-id built in each SIM card. With this new configuration parameters you will just specify the IMEI and/or the IMSI and mod_gsmopen will automatically discover the correct devices names (so, you can leave them out from the config file). So, being done at startup, is reboot safe, and you can also move around the dongles in different USB ports or hubs, etc (not while FreeSWITCH is running, heh ;) ). Please, update to the latest code, and have a look at the wiki page: http://wiki.freeswitch.org/wiki/GSMopen#IMEI_and_IMSI_automatic_device_disc… -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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crypto tag in sdp in webrtc INVITE message
by Iwan Budi Kusnanto 04 Jul '13

04 Jul '13

Hi All, Need some clarification here. It seems that by default Freeswitch will use DTLS in webrtc. According to this http://tools.ietf.org/html/rfc4568, crypto tag is used by SDES. And according to this https://bugzilla.mozilla.org/show_bug.cgi?id=825515, crypto tag shouldn't be in the SDP. Is it correct to have crypto tag in SDP when we use DTLS? FYI, I have made some modification to jssip and make it works with firefox 23 & Freeswitch https://groups.google.com/forum/#!topic/jssip/wXWrcBYHJ28 But it only works when the call initiated by firefox. It gives me no audio when the call initiated by SIP phone or using originate command. And i guess one of the causes is this crypto tag existence. Thanks. -- Iwan Budi Kusnanto

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IP Phone w/ Broken G722 codec
by Nandy Dagondon 04 Jul '13

04 Jul '13

Hello guys, I have IP phones that (incorrectly) presents SDP G722:9/16000. Searching the mailing list archives, I found G722_8. Is this the workaround to this problem? Is this still active in the master HEAD since I found out that mod_g722.c was deleted sometime ago. How do I activate it? Thanks. /Nandy

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