August 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SPA8000
by Erkan Ünlü 17 Aug '10

17 Aug '10

Hi all, I have a Linksys SPA8000 but I can't register a line. With softphones the user is work. But with SPA8000 don't register. The register to other provider is worked. So I can configure the SPA8000 correctly. The sample user can be registered with softphone and some hardphones but with SPA8000 can't register. Maybe I must change the XML in the directory folder. Everybody have idea? <include> <user id="1000" mailbox="1000"> <params> <param name="password" value="1234"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="accountcode" value="1000"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1000"/> <variable name="effective_caller_id_number" value="1000"/> </variables> </user> </include>

3 4

playing a message while bridging
by babak yakhchali 17 Aug '10

17 Aug '10

Hi is it possible to play a message while bridge is taking place, like <action application="set" data="ringback=/message.wav"/> <action application="bridge" data="sofia/internal/211%192.168.11.30"/> I've tried many combinations but sometimes I get server failure and in the best case the message file is played like beep beep . .. (but it's a voice message)!!! and it never works for now I'm using answer and playback and then I bridge but I wanted to see if it is possible not to answer the call thanx

6 8

outbound call with hunting number as the originating caller
by Neil Patel 17 Aug '10

17 Aug '10

Hi All, I have an app running on FS making outbound calls over a PRI line. The calls all caller-id at the endpoints with the PRI's pilot number. Say I wanted to change it so that the endpoint receives the call from one of the block of hunting numbers that my line comes with (instead of the primary/pilot number), how do I originate a call to do that? Here's the command on fs_cli that I'm trying, that doesn't seem to work: > originate {destination_number=XXXXXXXXXX}openzap/smg_prid/a/YYYYYYYYYY@g2&echo Not sure which channel var(s) I should be setting, or even if that's the way to make this work. Maybe it's up to my PRI service provider to provide this functionality? In case it matters, I am hosting two PRIs on Sangoma hardware, here is my openzap.conf: [span wanpipe smg_prid] name => smg_prid trunk_type =>e1 b-channel => 1:1-15 b-channel => 1:17-31 trunk_type =>e1 b-channel => 2:1-15 b-channel => 2:17-31 Thanks in advance, Neil

2 2

Did FreeSWITCH modules provides Vicidials features?
by Patrick Hsieh 16 Aug '10

16 Aug '10

Hello, I am a new user of FreeSWITCH. I have installed the Asterisk and Vicidial applications in my notebook. Q1: I would like to know are there any similar features which already be implemented by FreeSWITCH itself? Q2: Which will be a efficient way to have the features in FreeSWITCH? Rewrite all features in FreeSWITCH or create a plug-in modules to Vicidial directly. Q3: Did FreeSWITCH already support the ACD and Predictive Dialing features? Which modules releated to the features? Please advices! I am willing to contribute myself to extend the FreeSWITCH features. Best Regards, Patrick MAJOR VICIDIAL FEATURES: - Inbound, Outbound and Blended call handling - Outbound agent-controlled, broadcast and predictive dialing - Full USA FTC-compliance capability - Web-based agent and administrative interfaces - Ability to have agents operate remotely - Integrated call recording - Three-Way calling within the agent application - Scheduled Callbacks: Agent-Only and Anyone - Web-configurable IVRs and Voicemail boxes - Scalable to hundreds of seats - Ability to use standard Telco lines and VOIP trunks - Open-Source AGPLv2 licensed, with no software licensing cost

3 2

SF Telephony Meetup Tonight
by Darren Schreiber 16 Aug '10

16 Aug '10

Hi folks, Sorry this is so last minute. If you'd like to join, I'm hosting a FreeSWITCH install-fest as part of the SF Telephony Meetup Group run by Zhao Lu. It's tonight in downtown San Francisco. If you're in town, come on by and install some FreeSWITCH goodies :-) Details are here: http://www.meetup.com/sftelephony/calendar/14161219/ There's no charge or anything for this event - it's just a general phone-geek meeting session. Thanks, Darren Schreiber

1 0

mod_skypopen (skype endpoint) changes, please test (REPOST)
by Giovanni Maruzzelli 16 Aug '10

16 Aug '10

Just in case you missed the weekend mailz: ======= Hi FreeSWITCHers, I've made some long due modifications to mod_skypopen, that maybe introduced bugs. Please test with the latest git and report any *new* problem (ok, old problems too), here in the mailing list, or - much better - in the Jira. ========= commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090 Author: Giovanni Maruzzelli <gmaruzz(a)gmail.com> Date: Fri Aug 13 16:19:20 2010 -0500 skypopen: now answer a call only when directed to do it (before was trying to answer any incoming call). Lot of changes to a messy part, so maybe some problem will come out... Signed-off-by: Giovanni Maruzzelli <gmaruzz(a)gmail.com> =========== Thank you to all, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

1 0

mod_xml_cdr flags b-leg with "a_"
by Nyamul Hassan 16 Aug '10

16 Aug '10

Hi, When a call hangs up normally, like either of the user kept the phone down, then there are no issues. However, when the call is hung up through a transfer, like mod_nibblebill does when "low_balance" or "no_balance" is reached, then the b-leg cdr.xml file gets prefixed with "a_". Is this a desired feature? Or a bug? Regards HASSAN

2 2

Proxy Media and Codec Prefer
by Dome Charoenyost 16 Aug '10

16 Aug '10

Dear All, I'm not sure it's bug or not ? I try to test codec negotate. with 3 FS box FS1 send PCMA to FS2 and FS2 send G729 to FS3 FS1 -----> [PCMU] ----> FS2 ------> [G729] -------> FS3 (10.20.2.49) My dialplan <action application="bridge" data="{absolute_codec_string='G729'}sofia/internal/2222@10.20.2.49:5080"/> It's work fine When i try <action application="set" data="proxy_media=true"/> <action application="bridge" data="{absolute_codec_string='G729'}sofia/internal/2222@10.20.2.49:5080"/> Call fail. because FS2 ignore absolute_codec_string='G729' and send only PCMU (I think by pass from FS1) It's bug or not ? Dome C.

9 26

Enterprise Originate and group_confirm
by Phillip Jones 16 Aug '10

16 Aug '10

Hi there, I have the following dialplan which uses enterprise originate and group_confirm: The plan is that two numbers are called simutanuously, each number has several gateway to use, incase a gateway fails. <action application="ring_ready" data=""/> <action application="set" data="group_confirm_cancel_timeout=1"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail= NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION,INCOMPATIBLE_DESTINATION,UNALLOCATED_NUMBER "/> <action application="bridge" data="{ignore_early_media=true,origination_caller_id_number=2155556240,group_confirm_file=prompts\press-1-to-accept-call-from.wav,group_confirm_key=1} [leg_timeout=25]sofia/gateway/quest/6095553828|[leg_timeout=25]sofia/gateway/broadvox1/6095553828 :_:{ignore_early_media=true,origination_caller_id_number=2155556240,group_confirm_file=prompts\press-1-to-accept-call-from.wav,group_confirm_key=1} [leg_timeout=25]sofia/gateway/quest/2155554374|[leg_timeout=25]sofia/gateway/broadvox1/2155554374"/> <action application="managed" data="PPNSystem.Telephony.LeaveVoicemailHandler"/> <action application="hangup" data="NORMAL_CLEARING"/> Every thing works great in terms of mutiple legs ringing, bridging, hanging up etc. I have one issue though: When either of the numbers, 6095553828 for example, answers the call but does not accept it (i.e. does not press 1) and hangs up, the dialplan tries the next gateway for the same number and therefore 6095553828 ends up getting another call even though he has rejected it. I was expecting that once the call is answered and rejected (not bridged) then that part of the dialplan would stop. Any help appreciated. Thanks Pj

2 5

TLS problem using SNOM phones
by Matthias Reinacher 16 Aug '10

16 Aug '10

Hello all, i have a problem using different Snom phones (300/320/820/821) with Freeswitch and TLS in a internet-wide setup (phones being registered via internet at a FS w/ public IP). Sometimes (25-50% of cases) the first try to call someone (internal) won't go through. Logs from phone, Freeswitch and ssldump show that the phone sends an INVITE, the FS asks for more credentials (digest auth b/c phone IP not in ACL), the phone answers with an INVITE w/ more auth credentials -- and this second INVITE package is not received by the FS. It does arrive at the server though, as verified by ssldump. Interestingly, if one presses "Cancel" on the phone, a new TCP/SSL connection is created, apparently the old one died -- see logs from phone, FS, and ssldump below. Also, Snom 820/821 show the registrar as not registered after such an action (presumably b/c original SSL connection is dead). Has anybody encountered this behaviour and can verify it? Are there any ideas what causes the problem here (FS, OpenSSL, phone?) and if there is any remedy for it? I was planning on using FS as a production phone system w/ encrypted signalling and audio using Snom phones. I can't do that right now b/c of the abovementioned problem, and i have not yet found a solution. Any help is greatly appreciated. TIA Matthias Logs: phone, FS, ssldump, please also see annotation enclosed in "--- ... ---" - phone: Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:225 (1280 bytes): INVITE sip:9999@XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone> Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:1001@192.168.0.114:3967;transport=tls;line=ze6y9ju8>;reg-id=1 X-Serialnumber: 0004134517B7 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 476 v=0 o=root 739742427 739742427 IN IP4 192.168.0.114 s=call c=IN IP4 192.168.0.114 t=0 0 m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Received from tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:363 (350 bytes): SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone> Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 Received from tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:524 (844 bytes): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone>;tag=vjyU4aSFye19F Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="XXXX.dyndns.org", nonce="a3346430-8ce9-11df-9347-29b67737619e", algorithm=MD5, qop="auth" Content-Length: 0 Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:528 (418 bytes): ACK sip:9999@XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone>;tag=vjyU4aSFye19F Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:1001@192.168.0.114:3967;transport=tls;line=ze6y9ju8>;reg-id=1 Content-Length: 0 --- the following packet is not received by FS --- Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:22:544 (1541 bytes): INVITE sip:9999@XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-7g007jateb7f;rport From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone> Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:1001@192.168.0.114:3967;transport=tls;line=ze6y9ju8>;reg-id=1 X-Serialnumber: 0004134517B7 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="1001",realm="XXXX.dyndns.org",nonce="a3346430-8ce9-11df-9347-29b67737619e",uri="sip:9999@XXXX.dyndns.org;user=phone",qop=auth,nc=00000001,cnonce="4fd48758",response="bb47aafad2fc3980d7540b0d2cbf1b03",algorithm=MD5 Content-Type: application/sdp Content-Length: 476 v=0 o=root 739742427 739742427 IN IP4 192.168.0.114 s=call c=IN IP4 192.168.0.114 t=0 0 m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- before this packet is sent, a new TCP/SSL connection is negotiated --- Sent to tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:29:936 (386 bytes): CANCEL sip:9999@XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone> Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Content-Length: 0 Received from tls:XXX.XXX.XXX.XXX:5061 at 11/7/2010 14:41:30:562 (345 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport=4655;received=78.53.13.206 From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone>;tag=XUQm659jUQQvB Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL Content-Length: 0 - Freeswitch: recv 1280 bytes from tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:33.774063: ------------------------------------------------------------------------ INVITE sip:9999@XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone> Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:1001@192.168.0.114:3967;transport=tls;line=ze6y9ju8>;reg-id=1 X-Serialnumber: 0004134517B7 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 476 v=0 o=root 739742427 739742427 IN IP4 192.168.0.114 s=call c=IN IP4 192.168.0.114 t=0 0 m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 350 bytes to tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:33.777861: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=XXX.XXX.XXX.XXX From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone> Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2010-07-11 14:41:33.778081 [DEBUG] sofia.c:5999 IP XXX.XXX.XXX.XXX Rejected by acl "domains". Falling back to Digest auth. send 844 bytes to tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:33.864787: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=XXX.XXX.XXX.XXX From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone>;tag=vjyU4aSFye19F Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="XXXX.dyndns.org", nonce="a3346430-8ce9-11df-9347-29b67737619e", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 418 bytes from tls/[XXX.XXX.XXX.XXX]:3967 at 12:41:34.082329: ------------------------------------------------------------------------ ACK sip:9999@XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone>;tag=vjyU4aSFye19F Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:1001@192.168.0.114:3967;transport=tls;line=ze6y9ju8>;reg-id=1 Content-Length: 0 --- here, the 2nd INVITE packet w/ credentials is missing, note time difference to next packet --- ------------------------------------------------------------------------ recv 386 bytes from tls/[XXX.XXX.XXX.XXX]:4655 at 12:41:41.905857: ------------------------------------------------------------------------ CANCEL sip:9999@XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone> Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Content-Length: 0 ------------------------------------------------------------------------ send 345 bytes to tls/[XXX.XXX.XXX.XXX]:4655 at 12:41:41.906402: ------------------------------------------------------------------------ SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport=4655;received=XXX.XXX.XXX.XXX From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone>;tag=XUQm659jUQQvB Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL Content-Length: 0 - ssldump: 1 254 2335.8000 (417.3058) C>SV3.1(1300) application_data --------------------------------------------------------------- INVITE sip:9999@XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone> Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:1001@192.168.0.114:3967;transport=tls;line=ze6y9ju8>;reg-id=1 X-Serialnumber: 0004134517B7 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 476 v=0 o=root 739742427 739742427 IN IP4 192.168.0.114 s=call c=IN IP4 192.168.0.114 t=0 0 m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --------------------------------------------------------------- 1 255 2335.8043 (0.0043) S>CV3.1(138) application_data --------------------------------------------------------------- SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 --------------------------------------------------------------- 1 256 2335.8044 (0.0000) S>CV3.1(183) application_data --------------------------------------------------------------- From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone> Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE --------------------------------------------------------------- 1 257 2335.8045 (0.0000) S>CV3.1(89) application_data --------------------------------------------------------------- User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 --------------------------------------------------------------- 1 258 2335.8935 (0.0890) S>CV3.1(161) application_data --------------------------------------------------------------- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport=3967;received=78.53.13.206 --------------------------------------------------------------- 1 259 2335.8935 (0.0000) S>CV3.1(84) application_data --------------------------------------------------------------- From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz --------------------------------------------------------------- 1 260 2335.8935 (0.0000) S>CV3.1(85) application_data --------------------------------------------------------------- To: <sip:9999@XXXX.dyndns.org;user=phone>;tag=vjyU4aSFye19F --------------------------------------------------------------- 1 261 2335.8935 (0.0000) S>CV3.1(72) application_data --------------------------------------------------------------- Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 INVITE --------------------------------------------------------------- 1 262 2335.8935 (0.0000) S>CV3.1(542) application_data --------------------------------------------------------------- User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="XXXX.dyndns.org", nonce="a3346430-8ce9-11df-9347-29b67737619e", algorithm=MD5, qop="auth" Content-Length: 0 --------------------------------------------------------------- 1 263 2336.1083 (0.2147) C>SV3.1(438) application_data --------------------------------------------------------------- ACK sip:9999@XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-re694j8n03bd;rport From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone>;tag=vjyU4aSFye19F Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:1001@192.168.0.114:3967;transport=tls;line=ze6y9ju8>;reg-id=1 Content-Length: 0 --- here, the 2nd INVITE packet is received, but it doesn't arrive at FS --- --------------------------------------------------------------- 1 2336.1229 (0.0145) S>C TCP FIN 1 264 2336.2127 (0.0898) C>SV3.1(1561) application_data --------------------------------------------------------------- INVITE sip:9999@XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:3967;branch=z9hG4bK-7g007jateb7f;rport From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone> Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:1001@192.168.0.114:3967;transport=tls;line=ze6y9ju8>;reg-id=1 X-Serialnumber: 0004134517B7 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom821/8.4.4 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="1001",realm="XXXX.dyndns.org",nonce="a3346430-8ce9-11df-9347-29b67737619e",uri="sip:9999@XXXX.dyndns.org;user=phone",qop=auth,nc=00000001,cnonce="4fd48758",response="bb47aafad2fc3980d7540b0d2cbf1b03",algorithm=MD5 Content-Type: application/sdp Content-Length: 476 v=0 o=root 739742427 739742427 IN IP4 192.168.0.114 s=call c=IN IP4 192.168.0.114 t=0 0 m=audio 50506 RTP/SAVP 0 8 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:0AKzFxymFJ8eHsDOerrZiw8RZhgB9pmDU0q0+4lj a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --------------------------------------------------------------- --- a new TCP/SSL connection is initiated for the next packet --- 1 2336.2128 (0.0001) C>S TCP FIN New TCP connection #7: XXX.XXX.XXX (4655) <-> XXX.XXX.XXX (5061) 7 1 0.1214 (0.1214) C>SV3.1(63) Handshake ClientHello Version 3.1 random[32]= 3c 26 79 5c 23 37 ce ec 6b 40 cc 95 14 16 f8 e9 fd 5e 36 da dd 56 ab 2a 8e 0e 15 aa 4c d7 93 98 cipher suites TLS_RSA_WITH_RC4_128_MD5 TLS_RSA_WITH_RC4_128_SHA TLS_RSA_WITH_NULL_MD5 TLS_RSA_WITH_NULL_SHA TLS_DH_anon_WITH_3DES_EDE_CBC_SHA TLS_DH_anon_WITH_RC4_128_MD5 TLS_RSA_WITH_DES_CBC_SHA TLS_RSA_EXPORT1024_WITH_RC4_56_SHA TLS_RSA_EXPORT1024_WITH_DES_CBC_SHA TLS_DH_anon_WITH_DES_CBC_SHA compression methods NULL 7 2 0.1219 (0.0005) S>CV3.1(74) Handshake ServerHello Version 3.1 random[32]= 4c 39 bc 05 60 47 db 31 51 42 29 ae 9f 45 c5 2f 56 5b 02 dd 4b 3b c1 f7 f3 b7 80 9b 8b 16 7a a0 session_id[32]= f5 49 8b 05 df 1d 7a 30 6d b5 58 79 b6 a3 e4 1d c2 c1 98 fc de f5 64 8f 85 cc 7c 7f a9 08 9f d7 cipherSuite TLS_RSA_WITH_RC4_128_SHA compressionMethod NULL 7 3 0.1219 (0.0000) S>CV3.1(644) Handshake Certificate 7 4 0.1219 (0.0000) S>CV3.1(4) Handshake ServerHelloDone 7 5 0.4413 (0.3193) C>SV3.1(134) Handshake ClientKeyExchange EncryptedPreMasterSecret[128]= 38 71 d5 4f 2f 4f 66 08 37 92 57 ef 4a bd 13 60 6f 7c 0f 23 47 83 a5 59 95 25 91 58 10 c2 b9 ea 8d 12 88 d9 33 06 a8 8c aa e6 17 b8 91 35 0f 56 49 a7 9f 77 04 9c fe b1 17 70 c0 a3 5f bc 77 ff 25 41 a5 2f cd f3 24 d7 3c da 9f a7 86 30 a8 64 7d ca cd 06 6c d4 a0 fc 18 b7 5a 04 69 45 ae 28 4e 0f bf 3c 49 87 fc f6 91 3b 1d 00 6e 81 46 5f e3 7e cd a2 40 b2 37 3b c3 fc 0a 17 8b aa 76 88 7 6 0.4413 (0.0000) C>SV3.1(1) ChangeCipherSpec 7 7 0.4413 (0.0000) C>SV3.1(36) Handshake Finished verify_data[12]= 64 c2 1f cb 54 7c ec 60 0a 45 db e4 7 8 0.4504 (0.0091) S>CV3.1(1) ChangeCipherSpec 7 9 0.4504 (0.0000) S>CV3.1(36) Handshake Finished verify_data[12]= 17 f3 73 52 66 96 4a 87 ba 6d 42 f5 7 10 0.5550 (0.1046) C>SV3.1(406) application_data --------------------------------------------------------------- CANCEL sip:9999@XXXX.dyndns.org;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz To: <sip:9999@XXXX.dyndns.org;user=phone> Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL Max-Forwards: 70 Reason: SIP;cause=487;text="Request terminated by user" Content-Length: 0 --------------------------------------------------------------- 7 11 0.5559 (0.0008) S>CV3.1(163) application_data --------------------------------------------------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/TLS 192.168.0.114:4655;branch=z9hG4bK-7g007jateb7f;rport=4655;received=78.53.13.206 --------------------------------------------------------------- 7 12 0.5560 (0.0000) S>CV3.1(84) application_data --------------------------------------------------------------- From: "Matthias" <sip:1001@XXXX.dyndns.org>;tag=rllbt2d9pz --------------------------------------------------------------- 7 13 0.5560 (0.0000) S>CV3.1(85) application_data --------------------------------------------------------------- To: <sip:9999@XXXXXXXX.dyndns.org;user=phone>;tag=XUQm659jUQQvB --------------------------------------------------------------- 7 14 0.5560 (0.0000) S>CV3.1(72) application_data --------------------------------------------------------------- Call-ID: 5379263cf7de-1vkl60xxknr0 CSeq: 2 CANCEL

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