February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Hardware SIP device register problem (Planet VIP-161W)
by Oleg Belykh 22 Feb '12

22 Feb '12

The device can't get registered on freeswitch system with "481 Call Does Not Exist" connection log: recv 417 bytes from tcp/[172.16.20.20]:2055 at 16:38:37.513380: ------------------------------------------------------------------------ REGISTER sip:kustogroup.com SIP/2.0 Via: SIP/2.0/TCP 172.16.20.20:5060;rport;branch=z9hG4bK57a5b7e300000 From: <sip:1105@kustogroup.com>;tag=2f157a14 To: <sip:1105@kustogroup.com> Call-ID: 19361e92(a)172.16.20.20 CSeq: 1550332939 REGISTER Max-Forwards: 70 Contact: <sip:1105@172.16.20.20:5060;transport=tcp>;expires=900;events="message-summary" User-Agent: VOIP_Agent_001 Supported: path Content-Length: 0 ------------------------------------------------------------------------ send 623 bytes to tcp/[172.16.20.20]:2055 at 16:38:37.521226: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 172.16.20.20:5060;rport=2055;branch=z9hG4bK57a5b7e300000 From: <sip:1105@kustogroup.com>;tag=2f157a14 To: <sip:1105@kustogroup.com>;tag=vy75Dp57NUm8m Call-ID: 19361e92(a)172.16.20.20 CSeq: 1550332939 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20120218T213714Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="kustogroup.com", nonce="20781b2c-185b-e111-952d-002590016844", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 657 bytes from tcp/[172.16.20.20]:2055 at 16:38:37.538670: ------------------------------------------------------------------------ REGISTER sip:kustogroup.com SIP/2.0 Via: SIP/2.0/TCP 172.16.20.20:5060;rport;branch=z9hG4bKc91f046100001 From: <sip:1105@kustogroup.com>;tag=2f157a14 To: <sip:1105@kustogroup.com>;tag=vy75Dp57NUm8m Call-ID: 19361e92(a)172.16.20.20 CSeq: 1550332940 REGISTER Max-Forwards: 70 Contact: <sip:1105@172.16.20.20:5060;transport=tcp>;expires=900;events="message-summary" User-Agent: VOIP_Agent_001 Supported: path Authorization: Digest username="1105", realm="kustogroup.com", nonce="20781b2c-185b-e111-952d-002590016844", qop=auth, cnonce="53506aaa", nc=00000001, uri="sip:kustogroup.com", response="26cb514fc66213aef7873b857a5a42a6" Content-Length: 0 ------------------------------------------------------------------------ send 508 bytes to tcp/[172.16.20.20]:2055 at 16:38:37.540118: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/TCP 172.16.20.20:5060;rport=2055;branch=z9hG4bKc91f046100001 From: <sip:1105@kustogroup.com>;tag=2f157a14 To: <sip:1105@kustogroup.com>;tag=vy75Dp57NUm8m Call-ID: 19361e92(a)172.16.20.20 CSeq: 1550332940 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20120218T213714Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ Please help, i cant understand the problem, another hardware device (DLink gateway) got registered without any problem. Connection logs are similar. ------------------------------------------------------------------------ recv 406 bytes from udp/[172.16.0.25]:5060 at 16:35:54.213419: ------------------------------------------------------------------------ REGISTER sip:kustogroup.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bKe8a46c80fd1882b6 From: <sip:1999@kustogroup.com>;tag=801a8ebf-898043 To: <sip:1999@kustogroup.com> Call-ID: BD21-112A-46684814FA6035979AAA-004@SipHost CSeq:4722 REGISTER Contact: <sip:1999@172.16.0.25:5060> Expires:600 Max-Forwards:70 User-Agent:dlink 12-38-43929367-0.9.5.1.1928-SA7O4 Content-Length: 0 ------------------------------------------------------------------------ send 635 bytes to udp/[172.16.0.25]:5060 at 16:35:54.220591: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bKe8a46c80fd1882b6 From: <sip:1999@kustogroup.com>;tag=801a8ebf-898043 To: <sip:1999@kustogroup.com>;tag=p8FH3K0j75erF Call-ID: BD21-112A-46684814FA6035979AAA-004@SipHost CSeq:4722 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20120218T213714Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="kustogroup.com", nonce="dcc9c5ca-175b-e111-952d-002590016844", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 639 bytes from udp/[172.16.0.25]:5060 at 16:35:54.311703: ------------------------------------------------------------------------ REGISTER sip:kustogroup.com:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bKe489a6a305c49f70 From: <sip:1999@kustogroup.com>;tag=801a8ebf-898043 To: <sip:1999@kustogroup.com> Call-ID: BD21-112A-46684814FA6035979AAA-004@SipHost CSeq:4723 REGISTER Contact: <sip:1999@172.16.0.25:5060> Expires:600 Max-Forwards:70 Authorization:Digest username="1999",realm="kustogroup.com",nonce="dcc9c5ca-175b-e111-952d-002590016844",uri="sip:kustogroup.com:5060",response="2ea94376953abc90a78efaa831e79af4",algorithm=MD5,cnonce="3870847b",qop=auth,nc=00001273 User-Agent:dlink 12-38-43929367-0.9.5.1.1928-SA7O4 Content-Length: 0 ------------------------------------------------------------------------ send 590 bytes to udp/[172.16.0.25]:5060 at 16:35:54.341527: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.0.25:5060;branch=z9hG4bKe489a6a305c49f70 From: <sip:1999@kustogroup.com>;tag=801a8ebf-898043 To: <sip:1999@kustogroup.com>;tag=QH994eHp4e5aB Call-ID: BD21-112A-46684814FA6035979AAA-004@SipHost CSeq:4723 REGISTER Contact: <sip:1999@172.16.0.25:5060>;expires=600 Date: Sun, 19 Feb 2012 16:35:54 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20120218T213714Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0

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FreeSWITCH Conference Call Tomorrow (Wed Feb 22)
by Michael Collins 21 Feb '12

21 Feb '12

Hey all! Yes, I promise that I will come up for air and hang out with the community tomorrow. :) We have a special guest joining us tomorrow: Stefan Wintermeyer from Amooma. He will be talking about Gemeinschaft v4 and all the interesting things that lead to going with FreeSWITCH instead of Asterisk. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_22 Look forward to speaking with you all tomorrow! -Michael

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SIP bridge
by Anto 21 Feb '12

21 Feb '12

Hello I have a internal equipment and I want to use a gateway FreeSWITCH servers against some providers. Let me know if it is possible to transfer SIP messages (bridge), such as early media, etc.. I would like these messages to stay at the gateway and not reach the supplier, Is possible?. I've been looking for information but have not found anything about it :-(. Thank you very much. Best regards Anto

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Optimal way to monitor call stats
by Muhammad Naseer Bhatti 21 Feb '12

21 Feb '12

Hi, I was trying to figure out what should be the optimal way to generate live stats to monitor calls. For example, connect to event_socket, subscribe to CHANNEL_ANSWER, CHANNEL_PROGRESS, CHANNEL_DESTROY etc .. and parse them accordingly? Or if there is anything better than this to handle. Since the switch is generating higher number of calls per second, if we opt to this method won't this be too heavy for the server resources? Thanks, Goni

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LOSE_RACE, NORMAL_CLEARING and billsec==0
by Joe Flemmings 21 Feb '12

21 Feb '12

When I call a ring-group and answer the call, both the LOSE_RACE and NORMAL_CLEARING channels have a billsec==0 in the CDR How can I fix this to get the real time on the channel with NORMAL_CLEARING? I'm using cdr csv

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COLP using To: Header
by Markus Lindenberg 21 Feb '12

21 Feb '12

Hi everybody, it seems some phones (Snom at least) support COLP (connected line update) in ringing phase by having the To:-header changed in the 180 ringing to a-pty. Snom have a nice detailed illustration on their wiki: http://wiki.snom.com/Category:Connected_Line_Identification_Presentation_(C… So if Fritz (111) calls Hans (222): INVITE sip:222@myshop From: "Fritz" <sip:111@myshop>... To: <sip:222@myshop>... The response sent to 111 should look something like this: SIP/2.0 180 Ringing From: "Fritz" <sip:111@myshop>... To: "Hans" <sip:222@myshop>... So basically i suppose i want freeswitch to stuff a effective_caller_id_name (if set) into the To: header in the a-leg, given somthing like the following dialplan snippet: <action application="export" data="effective_callee_id_number=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_number)}"/> <action application="export" data="effective_callee_id_name=${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> Is this possible? Thanks, Markus

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FreeSWITCH Security Hardening Project
by Brian West 21 Feb '12

21 Feb '12

While sitting here in the hospital I have been talking to Philip Zimermann (ZRTP) via phone... And we have talked about a FreeSWITCH hardening project (NO JOKES PLEASE). This is taking FreeSWITCH and security to the MAX. I would like to start this project to make FreeSWITCH more secure in the long run. (not saying that its not secure now). Who would be interested in such a project? Thanks, -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian(a)freeswitch.org http://www.freeswitch.org

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Latest GIT build error for mod_dptools
by afshin afzali 21 Feb '12

21 Feb '12

Hi, In latest freeswitch in mod_dptools.c function audio_bridge_function there is unused "channel" variable which causes build error! BEST, -- afshin

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Virtualized Freeswitch?
by Tim Nelson 21 Feb '12

21 Feb '12

Greetings all- I'm quite experienced with Asterisk, and specifically, usage in a virtualized environment. It works great with a bit of tuning. I'm starting to work more with Freeswitch, and as such, I'm looking to implement it in a virtualized environment. My initial installation was onto a CentOS 6 OpenVZ container. However, the logs are showing this (many times): 2012-02-20 18:43:51.846882 [WARNING] switch_scheduler.c:114 Task was executed late by 280 seconds 1 heartbeat (core) 2012-02-20 18:43:51.846882 [WARNING] switch_scheduler.c:114 Task was executed late by 260 seconds 2 check_ip (core) 2012-02-20 18:46:21.496538 [CRIT] switch_time.c:798 Virtual Migration Detected! Syncing Clock 2012-02-20 18:46:21.496538 [CRIT] switch_time.c:810 If you see this message many times try setting the param enable-clock-nanosleep to true in switch.conf.xml or consider a nicer machine to run me on. I AM *FREE* afterall. I completely understand the issue with realtime applications such as VoIP in a virtualized environment with limited timing facilities. However, OpenVZ makes use of a shared kernel which in my prior experience, does not have the issues associated with fully virtualized technologies like VMWare, KVM, Xen, Hyper-V, etc. Am I incorrect or is this issue specific to Freeswitch (but apparently not so in use with Asterisk) ? There are a couple of wiki entries [1][2] which seem to indicate others are using Freeswitch with OpenVZ with success. Am I mistaken? Looking for some pointers if you can lend some insight into this issue. Maybe the errors are harmless? I have not actually pushed any traffic through this installation yet, due to the errors. Thanks! --Tim [1] http://wiki.freeswitch.org/wiki/FreeSWITCH_in_OpenVZ [2] http://wiki.freeswitch.org/index.php?title=Virtualization&redirect=no

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FS protocol interoperability
by Josh 21 Feb '12

21 Feb '12

Very simple query: If I configure my (external) gateway to use SRTP/ZRTP/SKYPE and then configure all my internal clients to register to FS using "regular" (understand unsecured) sip, would it be possible to route (i.e make and receive) calls between the gateway and the internally registered clients? If so, what are the likely problems I could face? Many thanks!

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