April 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] [ERR] sofia_reg.c:2110 "some gateway" Registration Failed with status Unauthorized [401]. failure #37 (Snabel Kaabiya)
by Snabel Kabiya 06 Apr '14

06 Apr '14

Hi, please see my answers inline: Thanks, Snabel > > ---------- Forwarded message ---------- > From: Steven Ayre <steveayre(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sun, 6 Apr 2014 09:59:43 -0400 > Subject: Re: [Freeswitch-users] [ERR] sofia_reg.c:2110 "some gateway" > Registration Failed with status Unauthorized [401]. failure #37 (Snabel > Kaabiya) > According to > https://wiki.freeswitch.org/wiki/Provider_Configuration:_DIDLogic nothing > particularly special is required. What you have should work. > > Does this gateway ever register successfully - does it register fine then > start getting the 401 later, or does it never register at all. If the > latter perhaps the username/password is incorrect? If it works at first > then starts to fail then it might be an error at DIDLogic's end and I > suggest you contact them to find out what is causing the 401. > [SNABEL] the gateway registers fine then start getting the 401 later, it works at first then starts to fail. i'm not sure it's an error at DIDLogic's end. because we have another server with freeswitch verion: FreeSWITCH Version 1.2.12+git~20130806T202138Z~22ab34c507 (git 22ab34c 2013-08-06 20:21:38Z) which works fine, we don't get this error there... > Also any modifications to the gateway (eg changing password) take more > than reloadxml to take effect. You need to run 'sofia profile <profilename> > killgw <gateway>', wait for the gateway to be removed (gives a log message > or check via 'sofia status gateway') then 'sofia profile <name> rescan' to > add the gateway with the new settings. Or restart FreeSWITCH. > [SNABEL] i did restarted freeswitch. > > By the way sip_cid_type is a variable not a parameter. See > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Variables - but > unknown parameters are ignored so it should make no difference to > registration; ping, ping-max and caller-id-in-from should also have no > effect on registration. > [SNABEL] i see, so i could just remove them... > -Steve > > > On 6 April 2014 03:37, Snabel Kabiya <snabel(a)lexifone.com> wrote: > >> Hi All, >> >> I'm still getting this error: >> >> [ERR] sofia_reg.c:2110 DIDLogic Registration Failed with status >> Unauthorized [401]. failure #72 >> >> I've looked at SIP trace and freeswitch logs but i can't figure out >> what's causing the registration failure. >> >> here is more info on the issue, i would really appreciate your help guys. >> >> *Server: * >> Linux Ubuntu-1204-precise-64-minimal 3.5.0-34-generic #55~precise1-Ubuntu >> SMP Fri Jun 7 16:25:50 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux >> >> *freeswitch version:* >> FreeSWITCH Version 1.2.23+git~20140317T213443Z~769a03a629~64bit (git >> 769a03a 2014-03-17 21:34:43Z 64bit) >> >> *Gateway configuration:* >> <gateway name="DIDLogic"> >> <param name="username" value="username"/> >> <param name="realm" value="sip.de.didlogic.net"/> >> <param name="password" value="password"/> >> <param name="register" value="true"/> >> <param name="context" value="public"/> >> <param name="caller-id-in-from" value="true"/> >> <param name="sip_cid_type" value="none"/> >> <param name="ping" value="30"/> >> <param name="ping-max" value="3"/> >> </gateway> >> >> >> >> *SIP trace:* >> >> 2014-04-06 09:32:05.472850 [NOTICE] sofia_reg.c:423 Registering DIDLogic >> send 894 bytes to udp/[46.165.252.40]:5060 at 06:32:05.492557: >> >> ------------------------------------------------------------------------ >> REGISTER sip:sip.de.didlogic.net;transport=udp SIP/2.0 >> Via: SIP/2.0/UDP 144.76.98.11:5080;rport;branch=z9hG4bKUZ0p2Fjget1KN >> Max-Forwards: 70 >> From: <sip:95299@sip.de.didlogic.net>;tag=XK0KSD17jemDN >> To: <sip:95299@sip.de.didlogic.net> >> Call-ID: de332712-ba83-11e3-b33f-0dc96adc5bb0 >> CSeq: 57892398 REGISTER >> Contact: <sip:gw+DIDLogic@144.76.98.11:5080;transport=udp;gw=DIDLogic> >> Expires: 3600 >> User-Agent: >> FreeSWITCH-mod_sofia/1.2.23+git~20140317T213443Z~769a03a629~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY >> Supported: path, replaces >> Authorization: Digest username="username", realm="sip.de.didlogic.net", >> nonce="U0DtvVNA7JFDwqn/iKN2y0i7jjJF/xDpK+45ZYA=", >> cnonce="tZUwATUmEjKDKHBU0sVf8A", algorithm=MD5, uri="sip: >> sip.de.didlogic.net;transport=udp", >> response="8aaadef8769d2234def69a699c53f95c", qop=auth, nc=0000018e >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> recv 446 bytes from udp/[46.165.252.40]:5060 at 06:32:05.520506: >> >> ------------------------------------------------------------------------ >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/UDP 144.76.98.11:5080 >> ;rport=5080;branch=z9hG4bKUZ0p2Fjget1KN >> From: <sip:95299@sip.de.didlogic.net>;tag=XK0KSD17jemDN >> To: <sip:95299@sip.de.didlogic.net >> >;tag=b27e1a1d33761e85846fc98f5f3a7e58.437e >> Call-ID: de332712-ba83-11e3-b33f-0dc96adc5bb0 >> CSeq: 57892398 REGISTER >> WWW-Authenticate: Digest realm="sip.de.didlogic.net", >> nonce="U0D2EVNA9OVIqS6bUrch5649kHnuK6PXK/Fii4A=", qop="auth" >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 894 bytes to udp/[46.165.252.40]:5060 at 06:32:05.520845: >> >> ------------------------------------------------------------------------ >> REGISTER sip:sip.de.didlogic.net;transport=udp SIP/2.0 >> Via: SIP/2.0/UDP 144.76.98.11:5080;rport;branch=z9hG4bKv8SF4a3KB3Q6g >> Max-Forwards: 70 >> From: <sip:95299@sip.de.didlogic.net>;tag=XK0KSD17jemDN >> To: <sip:95299@sip.de.didlogic.net> >> Call-ID: de332712-ba83-11e3-b33f-0dc96adc5bb0 >> CSeq: 57892399 REGISTER >> Contact: <sip:gw+DIDLogic@144.76.98.11:5080;transport=udp;gw=DIDLogic> >> Expires: 3600 >> User-Agent: >> FreeSWITCH-mod_sofia/1.2.23+git~20140317T213443Z~769a03a629~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY >> Supported: path, replaces >> Authorization: Digest username="username", realm="sip.de.didlogic.net", >> nonce="U0D2EVNA9OVIqS6bUrch5649kHnuK6PXK/Fii4A=", >> cnonce="tZUwATUmEjKDKHBU0sVf8A", algorithm=MD5, uri="sip: >> sip.de.didlogic.net;transport=udp", >> response="3dc99375cf925ad9af145899dcf83b1a", qop=auth, nc=0000018f >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> recv 446 bytes from udp/[46.165.252.40]:5060 at 06:32:05.536657: >> >> ------------------------------------------------------------------------ >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/UDP 144.76.98.11:5080 >> ;rport=5080;branch=z9hG4bKv8SF4a3KB3Q6g >> From: <sip:95299@sip.de.didlogic.net>;tag=XK0KSD17jemDN >> To: <sip:95299@sip.de.didlogic.net >> >;tag=b27e1a1d33761e85846fc98f5f3a7e58.b648 >> Call-ID: de332712-ba83-11e3-b33f-0dc96adc5bb0 >> CSeq: 57892399 REGISTER >> WWW-Authenticate: Digest realm="sip.de.didlogic.net", >> nonce="U0D2EVNA9OUTYGscaPkV4MvAB1IDDtpQK/FjjIA=", qop="auth" >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2014-04-06 09:32:05.532842 [ERR] sofia_reg.c:2110 DIDLogic Registration >> Failed with status Unauthorized [401]. failure #72 >> 2014-04-06 09:32:06.472844 [WARNING] sofia_reg.c:478 DIDLogic Failed >> Registration [904], setting retry to 2160 seconds. >> >> >>

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Re: [Freeswitch-users] [ERR] sofia_reg.c:2110 "some gateway" Registration Failed with status Unauthorized [401]. failure #37 (Snabel Kaabiya)
by Snabel Kabiya 06 Apr '14

06 Apr '14

Hi All, I'm still getting this error: [ERR] sofia_reg.c:2110 DIDLogic Registration Failed with status Unauthorized [401]. failure #72 I've looked at SIP trace and freeswitch logs but i can't figure out what's causing the registration failure. here is more info on the issue, i would really appreciate your help guys. *Server: * Linux Ubuntu-1204-precise-64-minimal 3.5.0-34-generic #55~precise1-Ubuntu SMP Fri Jun 7 16:25:50 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux *freeswitch version:* FreeSWITCH Version 1.2.23+git~20140317T213443Z~769a03a629~64bit (git 769a03a 2014-03-17 21:34:43Z 64bit) *Gateway configuration:* <gateway name="DIDLogic"> <param name="username" value="username"/> <param name="realm" value="sip.de.didlogic.net"/> <param name="password" value="password"/> <param name="register" value="true"/> <param name="context" value="public"/> <param name="caller-id-in-from" value="true"/> <param name="sip_cid_type" value="none"/> <param name="ping" value="30"/> <param name="ping-max" value="3"/> </gateway> *SIP trace:* 2014-04-06 09:32:05.472850 [NOTICE] sofia_reg.c:423 Registering DIDLogic send 894 bytes to udp/[46.165.252.40]:5060 at 06:32:05.492557: ------------------------------------------------------------------------ REGISTER sip:sip.de.didlogic.net;transport=udp SIP/2.0 Via: SIP/2.0/UDP 144.76.98.11:5080;rport;branch=z9hG4bKUZ0p2Fjget1KN Max-Forwards: 70 From: <sip:95299@sip.de.didlogic.net>;tag=XK0KSD17jemDN To: <sip:95299@sip.de.didlogic.net> Call-ID: de332712-ba83-11e3-b33f-0dc96adc5bb0 CSeq: 57892398 REGISTER Contact: <sip:gw+DIDLogic@144.76.98.11:5080;transport=udp;gw=DIDLogic> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.2.23+git~20140317T213443Z~769a03a629~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: path, replaces Authorization: Digest username="username", realm="sip.de.didlogic.net", nonce="U0DtvVNA7JFDwqn/iKN2y0i7jjJF/xDpK+45ZYA=", cnonce="tZUwATUmEjKDKHBU0sVf8A", algorithm=MD5, uri="sip:sip.de.didlogic.net;transport=udp", response="8aaadef8769d2234def69a699c53f95c", qop=auth, nc=0000018e Content-Length: 0 ------------------------------------------------------------------------ recv 446 bytes from udp/[46.165.252.40]:5060 at 06:32:05.520506: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 144.76.98.11:5080;rport=5080;branch=z9hG4bKUZ0p2Fjget1KN From: <sip:95299@sip.de.didlogic.net>;tag=XK0KSD17jemDN To: <sip:95299@sip.de.didlogic.net >;tag=b27e1a1d33761e85846fc98f5f3a7e58.437e Call-ID: de332712-ba83-11e3-b33f-0dc96adc5bb0 CSeq: 57892398 REGISTER WWW-Authenticate: Digest realm="sip.de.didlogic.net", nonce="U0D2EVNA9OVIqS6bUrch5649kHnuK6PXK/Fii4A=", qop="auth" Content-Length: 0 ------------------------------------------------------------------------ send 894 bytes to udp/[46.165.252.40]:5060 at 06:32:05.520845: ------------------------------------------------------------------------ REGISTER sip:sip.de.didlogic.net;transport=udp SIP/2.0 Via: SIP/2.0/UDP 144.76.98.11:5080;rport;branch=z9hG4bKv8SF4a3KB3Q6g Max-Forwards: 70 From: <sip:95299@sip.de.didlogic.net>;tag=XK0KSD17jemDN To: <sip:95299@sip.de.didlogic.net> Call-ID: de332712-ba83-11e3-b33f-0dc96adc5bb0 CSeq: 57892399 REGISTER Contact: <sip:gw+DIDLogic@144.76.98.11:5080;transport=udp;gw=DIDLogic> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.2.23+git~20140317T213443Z~769a03a629~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: path, replaces Authorization: Digest username="username", realm="sip.de.didlogic.net", nonce="U0D2EVNA9OVIqS6bUrch5649kHnuK6PXK/Fii4A=", cnonce="tZUwATUmEjKDKHBU0sVf8A", algorithm=MD5, uri="sip:sip.de.didlogic.net;transport=udp", response="3dc99375cf925ad9af145899dcf83b1a", qop=auth, nc=0000018f Content-Length: 0 ------------------------------------------------------------------------ recv 446 bytes from udp/[46.165.252.40]:5060 at 06:32:05.536657: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 144.76.98.11:5080;rport=5080;branch=z9hG4bKv8SF4a3KB3Q6g From: <sip:95299@sip.de.didlogic.net>;tag=XK0KSD17jemDN To: <sip:95299@sip.de.didlogic.net >;tag=b27e1a1d33761e85846fc98f5f3a7e58.b648 Call-ID: de332712-ba83-11e3-b33f-0dc96adc5bb0 CSeq: 57892399 REGISTER WWW-Authenticate: Digest realm="sip.de.didlogic.net", nonce="U0D2EVNA9OUTYGscaPkV4MvAB1IDDtpQK/FjjIA=", qop="auth" Content-Length: 0 ------------------------------------------------------------------------ 2014-04-06 09:32:05.532842 [ERR] sofia_reg.c:2110 DIDLogic Registration Failed with status Unauthorized [401]. failure #72 2014-04-06 09:32:06.472844 [WARNING] sofia_reg.c:478 DIDLogic Failed Registration [904], setting retry to 2160 seconds. On Thu, Apr 3, 2014 at 5:08 PM, Snabel Kabiya <snabel(a)lexifone.com> wrote: > > Hi Brian, > > i know what the error means, but what i don't understand why i keep > getting this error after i upgraded freeswitch. > > i keep getting this error almost every day, I'm not sure if i need to > change some configuration or add new parameters to the gateway after i > upgraded freeswitch. > > what could cause it? please let me know if you need me to provide you with > more info regarding the environment I'm working on, or any config. files. > > i really appreciate your help. > > Thanks, > Snabel > > > On Thu, Apr 3, 2014 at 2:01 PM, <freeswitch-users-request(a)lists.freeswitch > .org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users(a)lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request(a)lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner(a)lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: [ERR] sofia_reg.c:2110 "some gateway" Registration Failed >> with status Unauthorized [401]. failure #37 (Brian West) >> 2. Audio degrades ON TRANSFER with specific MOH file (Sean Devoy) >> 3. gateway setup example ? (Jeremy Ardley) >> 4. No audio stun for a long time! (rockxsj) >> 5. any tutorial for using video call? (dwi yulianto) >> 6. Re: can i make all log from console have date stamp (dwi >> yulianto) >> >> >> ---------- Forwarded message ---------- >> From: Brian West <brian(a)freeswitch.org> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Wed, 2 Apr 2014 21:23:52 -0500 >> Subject: Re: [Freeswitch-users] [ERR] sofia_reg.c:2110 "some gateway" >> Registration Failed with status Unauthorized [401]. failure #37 >> You have a gateway called 'some gateway' thats failed to register. >> -- >> Brian West >> brian(a)freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Apr 2, 2014, at 8:56 AM, Snabel Kabiya <snabel(a)lexifone.com> wrote: >> >> > >> > [ERR] sofia_reg.c:2110 "some gateway" Registration Failed with status >> Unauthorized [401]. failure #37 >> >> >> >> ---------- Forwarded message ---------- >> From: Sean Devoy <sdevoy(a)bizfocused.com> >> To: "freeswitch-users(a)lists.freeswitch.org" < >> freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Thu, 3 Apr 2014 04:05:02 +0000 >> Subject: [Freeswitch-users] Audio degrades ON TRANSFER with specific MOH >> file >> >> Hi All, >> >> >> >> I have a user that has claimed that if she picks up incoming calls, the >> audio is excellent. However, if someone else picks up the call and >> transfers it to her, the audio is "terrible". >> >> >> >> While investigating I found that they are using their own custom MOH >> file. I suspect that the AUDIO configuration is being modified by the MOH >> feature and leaving her transferred call in a degraded state. But, I don't >> understand enough to actually see if that is the case. >> >> >> >> Please review the pastebin file: http://pastebin.freeswitch.org/22302 >> >> >> >> I think I have trimmed it down to the relevant data. Ext 302 has >> answered and is just initiating the transfer. >> >> >> >> If I am correct in my hypothesis, what is the correct solution? >> >> >> >> Thanks in advance, >> >> Sean >> >> >> ---------- Forwarded message ---------- >> From: Jeremy Ardley <jeremy.ardley(a)gmail.com> >> To: freeswitch-users(a)lists.freeswitch.org >> Cc: >> Date: Thu, 03 Apr 2014 10:54:47 +0800 >> Subject: [Freeswitch-users] gateway setup example ? >> Hi, >> >> I'm presently running a partial PBX using Kamailio and SEMS and I am >> interested in changing to freeswitch but I'm a bit overwhelmed by the many >> configuration files. >> >> My present setup is an upstream gateway providing DID services and >> outgoing services based on my IP only - no authentication. My internal >> phone network is based on three digit extensions. >> I need freeswitch to accept incoming DID calls and distributed them to >> mapped internal extensions - and take internal extension outgoing calls and >> send them as full national number calls via the upstream gateway. >> >> Is there a reference setup or tutorial somewhere that covers this >> scenario? >> >> Thanks, >> >> -- >> Jeremy Ardley >> >> >> ---------- Forwarded message ---------- >> From: rockxsj <rockxsj(a)gmail.com> >> To: freeswitch-users(a)lists.freeswitch.org >> Cc: >> Date: Thu, 03 Apr 2014 14:11:16 +0800 >> Subject: [Freeswitch-users] No audio stun for a long time! >> I can see that sometimes my server product some logs like: >> >> >> deb9157a-b9b1-11e3-8ee1-7b004e0e7c32 2014-04-02 00:03:44.083032 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 8c9f55cc-b9b8-11e3-9cca-7b004e0e7c32 2014-04-02 00:21:02.441130 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:32:04.021509 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:32:35.021902 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:33:05.061286 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:33:36.061678 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:34:06.101062 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:34:37.101453 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:35:07.121681 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:35:37.161065 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> a7801e9c-b9ba-11e3-9f47-7b004e0e7c32 2014-04-02 00:36:08.161457 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 74c1f71e-b9fb-11e3-bbbd-7b004e0e7c32 2014-04-02 08:14:35.502605 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> c323e07a-b9fb-11e3-bc46-7b004e0e7c32 2014-04-02 08:15:02.921673 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> fa17e09a-b9fb-11e3-bd18-7b004e0e7c32 2014-04-02 08:16:35.121138 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> fa17e09a-b9fb-11e3-bd18-7b004e0e7c32 2014-04-02 08:17:06.121531 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 43945544-ba03-11e3-8df9-7b004e0e7c32 2014-04-02 09:09:08.621875 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 05e67bae-ba09-11e3-a24c-7b004e0e7c32 2014-04-02 09:50:28.321208 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> b151ab62-ba09-11e3-a4da-7b004e0e7c32 2014-04-02 09:58:45.491465 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 552384de-ba0c-11e3-afc5-7b004e0e7c32 2014-04-02 10:14:03.691235 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 04bc9112-ba0b-11e3-aabc-7b004e0e7c32 2014-04-02 10:16:39.392036 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 04bc9112-ba0b-11e3-aabc-7b004e0e7c32 2014-04-02 10:17:37.411787 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 04bc9112-ba0b-11e3-aabc-7b004e0e7c32 2014-04-02 10:18:07.431169 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 2e316a40-ba10-11e3-8076-7b004e0e7c32 2014-04-02 10:41:28.251996 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 7396150e-ba10-11e3-8186-7b004e0e7c32 2014-04-02 10:43:24.291472 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 388ba6ac-ba12-11e3-8801-7b004e0e7c32 2014-04-02 10:56:48.411568 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:10:06.051664 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:10:36.091046 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:11:06.111419 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:11:37.111820 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:12:07.151202 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:12:38.151594 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:13:08.190977 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:13:39.191370 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 5c7c26d8-ba10-11e3-8125-7b004e0e7c32 2014-04-02 11:14:10.191752 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> abb38326-ba16-11e3-9cb0-7b004e0e7c32 2014-04-02 11:27:40.391902 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> 75cee1e2-ba1b-11e3-b14f-7b004e0e7c32 2014-04-02 12:01:57.731915 [ERR] >> switch_rtp.c:749 No audio stun for a long time! >> >> My server is not behind firewall or router, I want know why the >> freeswitch still product these logs? Is something wong with my config? >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: dwi yulianto <dwiyulianto.anto(a)gmail.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Thu, 3 Apr 2014 17:56:43 +0700 >> Subject: [Freeswitch-users] any tutorial for using video call? >> i try use freeswitch with video call, >> i already added video codec in vars.xml, but i still cant use video call >> with linphone as softphone, i am already checked "always start video" in >> linphone setting >> >> is there any other configuration for enable video feature in freeswitch? >> >> thanks. >> >> >> ---------- Forwarded message ---------- >> From: dwi yulianto <dwiyulianto.anto(a)gmail.com> >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> Cc: >> Date: Thu, 3 Apr 2014 18:01:12 +0700 >> Subject: Re: [Freeswitch-users] can i make all log from console have date >> stamp >> thanks all for reply, >> >> i will try to ask that to developer. >> >> >> On Thu, Apr 3, 2014 at 12:07 AM, William King < >> william.king(a)quentustech.com> wrote: >> >>> If you file a feature request for a freeswitch event to be fired when a >>> new TLS connection is made, one of the developers might be able to look >>> into adding that functionality. Then you would be able to hook in >>> mod_fail2ban and log the events you are looking to have watched by >>> fail2ban. >>> >>> William King >>> Senior Engineer >>> Quentus Technologies, INC >>> 1037 NE 65th St Suite 273 >>> Seattle, WA 98115 >>> Main: (877) 211-9337 >>> Office: (206) 388-4772 >>> Cell: (253) 686-5518 >>> william.king(a)quentustech.com >>> >>> On 04/02/2014 03:29 AM, dwi yulianto wrote: >>> > sorry, i mean ssl dos. >>> > >>> > if i use that, that give cpu activity increase to 100%. >>> > >>> > i want to make that log so i can use fail2ban to ban source IP attacker >>> > >>> > >>> > On Wed, Apr 2, 2014 at 5:02 PM, dwi yulianto < >>> dwiyulianto.anto(a)gmail.com >>> > <mailto:dwiyulianto.anto@gmail.com>> wrote: >>> > >>> > i try ssh dos and that give log in tport. but log about tport tls >>> > didnt have any data/time stamp. >>> > >>> > maybe if i can make that log into date/time stamp i can make regex >>> > for fail2ban to find log when attacker give new connection in tls >>> > eventually. >>> > >>> > am i need to change in tport_type_tls.c in line about SU_DEBUG_5 >>> > that have part with "new connection from " and change that with >>> > switch_log_printf. but actually i dont really understand about C. >>> so >>> > how i can change that with switch_log_printf? >>> > >>> > Thanks >>> > >>> > >>> > >>> > >>> > >>> > >>> > On Wed, Apr 2, 2014 at 3:35 PM, Steven Ayre <steveayre(a)gmail.com >>> > <mailto:steveayre@gmail.com>> wrote: >>> > >>> > Sofia stack logging is extremely verbose, far more so than you >>> > want to run in production. Doing so will likely hurt the >>> > performance of your system due to too much disk I/o >>> > >>> > What're you looking to capture? >>> > >>> > FS already has log messages for auth failures for fail2ban to >>> match >>> > >>> > >>> > >>> > >>> > On Tuesday, April 1, 2014, dwi yulianto >>> > <dwiyulianto.anto(a)gmail.com <mailto:dwiyulianto.anto@gmail.com >>> >> >>> > wrote: >>> > >>> > i wanna make log from tport.c in console or in log file >>> with >>> > date stamp, so i can use it for fail2ban. >>> > >>> > thanks. >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting(a)freeswitch.org <mailto:consulting@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 >>> > <mailto:FreeSWITCH-users@lists.freeswitch.org> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > 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 >>> > >>> >>> _________________________________________________________________________ >>> 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 >>> >> >> >> _______________________________________________ >> 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 >> >> >

2 1

(no subject)
by Yudhistira Muhammad Iqbal 06 Apr '14

06 Apr '14

hy my name is yudhistira i'm trying to connecting my freeswuitch box to my analog pabx by fxo using voip card openvox a400e but i have some trouble when i do that i can call out to fxo but incoming call to my freeswitch box not runing well i hope there is somebody that can help me to fix my problem this my configuration ----------zt.conf---------------------------------- [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 rxgain => 0.0 txgain => 0.0 -----------finish----------------------------------- -----------freetdm.conf------------------------ [span zt FXS] fxs-channel => 1 [span zt FXO] name => freetdm trunk_type => fxo number => 112 fxo-channel => 2 ------------finish------------------------------------ ------------freetdm.conf.xml-------------------- <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="0"/> <param name="hold-music" value="$${moh_uri}"/> </settings>  <analog_spans> <span name="FXO"> <param name="dialplan" value="XML"/> <param name="context" value="default"/>  <param name="tonegroup" value="us"/> <param name="enable_callerid" value="true"/> <param name="enable-analog-option" value="call-swap"/> </span> </analog_spans> </configuration> -----------------------finish-------------------------------------------------------------- -----------------------default.xml------------------------------------------------------ <extension name="outgoing-FXO"> <condition field="destination_number" expression="^(\d{3})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="freetdm/2/1/${destination_number}"/> <action application="hangup"/> </condition> </extension> -------------------------------finish------------------------------------------------------- ------------------------------00_inbound_did.xml------------------------------------ <include> <extension name="public_did"> <condition field="destination_number" expression="^(\d{3})$"> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="freetdm/1/1/${destination_number}"/> <action application="transfer" data="200209 XML public"/> </condition> </extension> </include> -----------------------finish--------------------------------------------

1 0

Bypass_media
by Andre Demattia 05 Apr '14

05 Apr '14

Hi I need to have many bridges in one dial plan for fail over however each bridge needs a different media bypass. I've read media bypass has to be on leg a but bypass media after bridge can be set in the bridge area. Is it possible to get bypass media to work on bridge so I can set the media based on gateway? Thanks Andre

1 0

Configure aliases and auto-complete on start up.
by bruce＠sqls.net 05 Apr '14

05 Apr '14

Right now I'm building a lua based fax server for fs which includes a handy management script that can be used from the console. What I've done to make using my management tool easier is created aliases and auto-complete's for it from the fs_cli console. Anytime I restart freeswitch the aliases and auto-complete seem to get reset. Is there a configuration file that I can configure these in so they are loaded on start up? Thank you :)

7 18

RTP timestamps jumping backwards
by Richard Brady 05 Apr '14

05 Apr '14

Hi Anthony and team I have a situation where the timestamp in a RTP stream generated by FS jumps backwards. Version 1.2.22. It seems the cause of this is that media from the A leg starts late. So in the meantime we start generating media on the B leg with our own timestamp. Then media starts arriving on the A leg so we relay it and we start using that timestamp instead of our own. FS sets the marker bit when this happens but I'm not sure this makes it OK to turn back the timestamp. I think we should actually be using a new SSRC. Any thoughts? (happy to raise a Jira ticket and verify in master). Richard -- Richard Brady M: +44 (0)7771 623 348 T: +44 (0)20 8144 8160 E: rnbrady(a)gmail.com

3 7

ZRTP not being detected?
by Peter Villeneuve 05 Apr '14

05 Apr '14

Hi, I have latest master built from git on Debain 7. I have CSipSimple client with ZRTP enabled and why I try calling the ZRTP enrollment extension (9787), FS tells me that my client doesn't have ZRTP even though I can see in the logs that it does indeed see the ZRTP hash: [DEBUG] sofia.c:8293 sofia/internal/1000(a)my.domain.com receiving invite from 85.xx.xx.xx:58163 version: 1.5.12b git f50f04b 2014-03-28 18:33:47Z 64bit [DEBUG] switch_core_media.c:343 Found audio zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10 35928c6c5064434bac6ba255ac970c28387c31c5adb1997e2df248623f042123 Looking at the zrtp_agent.lua script I see session:getVariable("zrtp_secure_media_confirmed_audio"); However, the extension "is_zrtp_secure" in the features context mentions a slightly different variable <condition field="${zrtp_secure_media_confirmed}" expression="^true$"> Can this be the root of the problem, ie the mismatch between "zrtp_secure_media_confirmed_audio" in the lua script and "zrtp_secure_media_confirmed" in the fetaures context? What's the proper variable terminology to use here? Any idea why FS doesn't seem to detect my ZRTP hash? Cheers, Peter

1 1

Wiki update for NetBSD/OpenBSD
by Brian West 05 Apr '14

05 Apr '14

I would like to know if interested parties can try out these instructions and report back or update/expand on these install instructions. https://wiki.freeswitch.org/wiki/Installation_Guide#NetBSD_6 Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

1 0

cross compiling.
by Michael Jerris 05 Apr '14

05 Apr '14

If anyone is successfully cross compiling freeswitch, please drop me a line off list. I'm working on integrating more into the freeswitch build system to make this easier, and I need details of steps you need to run to make this work well so that i can integrate better in to the build system. Thanks Mike

2 1

Auto-record works in fs_cli but doesn't work in conference.xml
by Jenny Davis 05 Apr '14

05 Apr '14

I have version 1.2 stable installed with mod_shout enabled and running, when I test in the fs_cli recording to icecast works. However when I add param name auto-record shout:// to conference.xml nothing happens, no stream is showing in icecast and isn't being recorded. I have not seen a specific example on how to add auto-record for streaming to icecast, so A- where to add the param name would be helpful as well as B- exact context. Please and thank you!

3 11