November 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

TLS Audiocodes
by Szasz Szabolcs 26 Dec '13

26 Dec '13

Hi, Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to work TLS between freeswitch and a softphone (phonerlite), but I have problem with Audiocodes during the TLS authentication. I've loaded the certification but it still doesn't work. Can I debug the tls in freeswitch? Please help me. Thank you in advance. Szasz Szabolcs

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Client TLS certificate setup
by adahary 26 Dec '13

26 Dec '13

I have already installed the CA certificate in the FS conf/ssl (following FS/TLS wiki) and managed to get TLS registering with PhonerLite so I know that basically it should work with other sip/tls clients as well.I'm trying to setup other softphones with TLS/SIPS connection to Freeswitch and I'm failing to register with CSimpleSip/android and Jitsi/windows. I get this:tport_type_tls.c:607 tport_tls_accept() tport_tls_accept(0xb6e02268): new connection from tls/1.2.3.4:13886/sipstport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATINGtport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATINGtport_tls.c:962 tls_connect() tls_connect(0xb6e02268): TLS setup failed (error:00000001:lib(0):func(0):reason(1))tport.c:2092 tport_close() tport_close(0xb6e02268): tls/1.2.3.4:13886/sipsI will appriciate any help/instruction/tip on how to setup working TLS connection with CSimpleSip/android and Jitsi/windows clients.Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Client-TLS-certificate-setup-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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TLS problem
by Dror Lupu 26 Dec '13

26 Dec '13

Hi Guys, I've done the steps for TLS setup as shown on the Wiki (using the server's IP address in CN and ALT), but I can't get TLS to work (UDP/TCP works fine). All certificates seem to be in place and generated correctly. I get this error, when using Linphone (I installed the root CA on my machine). 1. tport.c:2745 tport_wakeup_pri() tport_wakeup_pri(0xb5062a60): events IN 2. tport.c:869 tport_alloc_secondary() tport_alloc_secondary(0xb5062a60): new secondary tport 0xb67c0410 3. tport_type_tls.c:607 tport_tls_accept() tport_tls_accept(0xb67c0410): new connection from tls/172.16.111.211:61942/sips 4. tport_tls.c:873 tls_connect() tls_connect(0xb67c0410): events NEGOTIATING 5. tport_tls.c:873 tls_connect() tls_connect(0xb67c0410): events NEGOTIATING 6. tport_tls.c:962 tls_connect() tls_connect(0xb67c0410): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) 7. tport.c:2092 tport_close() tport_close(0xb67c0410): tls/172.16.111.211:61942/sips 8. Any ideas? Thanks in advance, Dror.

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Conference timer synchronization with T1 FreeTDM - how?
by Bob Hartwig 20 Dec '13

20 Dec '13

Looking at the source for mod_conference.c, I see that the timer_name is hard-coded to "soft", i.e. the 1000 Hz. system clock. How can this work with a T1-based FreeTDM system, where the FreeTDM channels are presumably synchronized to the T1 clock source? I'm seeing some weirdness with my FreeTDM / Dahdi system, where only FreeTDM channels are hearing distortion on conference, and I'm almost certain that this is the culprit, but I have no idea how to proceed to fix it. I'm pretty new to FreeTDM, so I'm hoping that I'm missing something simple. Any thoughts? Thanks, Bob

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Gateway Registration Issues
by Aloysius Lloyd 15 Dec '13

15 Dec '13

Hi All, I am trying to connect IRISTEL ( CLEC ) in Canada and FreeSWITCH. I have the following gateway configuration <include> <gateway name="wwc_test"> <param name="auth-username" value="username"/> <param name="username" value="14161231234" /> <param name="password" value="password" /> <param name="proxy" value="sbc1.iristel.net" /> <param name="realm" value="iristel.net" /> <param name="expire-seconds" value="120"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="ping" value="25"/> </gateway> </include> The above configuration is not working. I have the following error in cli Error. *2010-05-13 08:34:15.987465 [ERR] sofia_reg.c:1501 14161231234 Registration Failed with status Conflict [409]. failure #1* ==== Also carrier saying there is conflict in the contact field.Here is how they receive now. *Contact: <sip:gw+test_iristel@10.20.30.1:5080 ;transport=udp;gw=test_iristel>.* But the contact field should be like below Contact: <14161231234@10.20.30.1:5080; Any suggestions? Thanks in advance. Lloyd

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Recordings out of sync again
by Kees Varekamp 11 Dec '13

11 Dec '13

2 years ago, I had this problem that recordings would get badly out of sync. I've never been able to pinpoint the problem but found that the voip client's "preserve bandwidth during silence" setting made the problem go away. But after moving to a new server, the problem has come back with a vengeance. And "preserve bandwidth during silence" does not seem to make any difference at all The new server runs on vmware - which got me thinking that running on a vm perhaps could cause such a problem? Has anyone had similar experiences? Thanks, Kees -- Kees Varekamp *MROffice Director* 44 Castleford street, 0604, Auckland, New Zealand Phone: +6492821656 Skype: kees.mroffice Web: mro <http://mro-global.com>ffice.org *MROffice - Meeting Market Research Needs*

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Recording G729
by Kees Varekamp 11 Dec '13

11 Dec '13

Hi all, I'm suffering from poor sound quality on my setup, using GSM on the a-leg and PCMU on the b-leg. I'm thinking it's mostly bandwidth/network related, so I'm considering switching to G729 on both legs, as that is the only compressed codec that the b-leg supports (it's a sip gateway). But a complicating factor is that the calls need to be recorded in mp3. So I have some questions that I hope someone here can answer: - Does transcoding (for example GSM<>PCMU) negatively affect the sound quality, even when the CPU seems to be ok with it? - If yes, does recording 2 legs using the same codec (say G729) to a different sound format (say mp3) have the same (negative) effect? - When you use G729 on both legs (using mod_com_g729), it is even possible to record to mp3? Thank you, Kees

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old calls hanging around
by Peter Hartmann 07 Dec '13

07 Dec '13

Hi, First, thanks for Freeswitch! I'm experiencing an issue where 'show calls' returns several calls that aren't actually happening both inbound and outbound. Has anyone seen this before? Rebooting the handset (Polycom IP 550) associated with that extension has no effect so it seems in FS. freeswitch@internal> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch d039dbb0-507e-4ede-be18-7bbae464167b,inbound,2013-10-05 15:55:49,1381002949,sofia/external/+1347xxxxxxx@flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,d039dbb0-507e-4ede-be18-7bbae464167b,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, d14331c4-98a2-48d1-9be1-9dbef822d094,inbound,2013-10-07 09:59:34,1381154374,sofia/external/+1212xxxxxxx@flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,d14331c4-98a2-48d1-9be1-9dbef822d094,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, a699e843-cdb8-4582-801f-7925dcebc15c,inbound,2013-10-07 10:44:28,1381157068,sofia/external/+1646xxxxxxx@flowroute.com,CS_EXECUTE,unknown ,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,a699e843-cdb8-4582-801f-7925dcebc15c,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 6023734b-a787-4460-98ab-dce3ea3cc19b,inbound,2013-10-07 10:49:05,1381157345,sofia/external/+1212xxxxxxx@flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,6023734b-a787-4460-98ab-dce3ea3cc19b,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 3d885c2d-20f6-4cb9-88ed-b4f838ef37e2,outbound,2013-10-07 11:49:54,1381160994,sofia/external/1347xxxxxxx,CS_EXCHANGE_MEDIA,Extension 1000,212xxxxxxx,10.10.10.100,1347xxxxxxx,,,ACTIVE,Outbound Call,1347xxxxxxx,SEND,85fbbc14-9219-48b9-a8fa-a02d59bc23b5,fs,Extension 1000,212xxxxxxx,,,,,,,,,,,,,,,,,,, bfeaa3a5-d5da-45cb-9c82-4293616630d4,inbound,2013-10-07 12:06:38,1381161998,sofia/external/+1212xxxxxxx@flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,bfeaa3a5-d5da-45cb-9c82-4293616630d4,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, f85e192b-455e-4208-a912-6ce84dae4c15,inbound,2013-10-07 13:37:17,1381167437,sofia/external/+1212xxxxxxx@flowroute.com,CS_EXECUTE,+1212xxxxxxx,+1212xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,f85e192b-455e-4208-a912-6ce84dae4c15,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 8c453960-a7f0-4ede-892b-c6fb1c1d41ea,inbound,2013-10-07 15:09:34,1381172974,sofia/external/+1347xxxxxxx@flowroute.com,CS_EXECUTE,+1347xxxxxxx,+1347xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,8c453960-a7f0-4ede-892b-c6fb1c1d41ea,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 77a21c7f-b871-48bc-8a21-a12d95b4a7d3,inbound,2013-10-07 15:41:20,1381174880,sofia/external/+1646xxxxxxx@flowroute.com,CS_EXECUTE,+1646xxxxxxx,+1646xxxxxxx,216.115.69.144,1000,,,ACTIVE,Outbound Call,1000,SEND,77a21c7f-b871-48bc-8a21-a12d95b4a7d3,fs,Outbound Call,1000,,,,,,,,,,,,,,,,,,, 9 total. Running: FreeSWITCH Version 1.2.13+git~20131002T213046Z~88be913119 (git 88be913 2013-10-02 21:30:46Z) Thanks much!

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Contact header in gateway invite
by Remco van Vugt 07 Dec '13

07 Dec '13

Hi, I have the following issue: a carrier wants the A number in the contact field i.e.: Contact: <sip:+31xxxxxxxxx@ip.ad.dr.ess;[add. params]> I know; the contact is not up to the called party but they want it anyway. I spent quite some time, and I found some earlier mailing list threads but no answer that solves my particular problem. I noticed that by setting the extension parameter in the gateway configuration, this results in the desired header format. However, this is not dynamic and therefore not the A number (freeswitch runs in a SBC configuration, passing traffic in and out of the network). Is there any way to either pass the extension= parameter from the dialplan or somehow change the contact header? Thanks, Remco

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JsSIP and wss
by Peter Steinbach 06 Dec '13

06 Dec '13

I am trying the Jssip demo with freeswitch and it works nicely with jssip 0.3.0 and the following configuration: ws_servers: 'ws://my_domain.com:8088' (My ws port is on 8088) However this only works with 'ws' but not with secure 'wss'. * In the log I can see that freeswitch sets wss-binding[:7443] * wss.pem is also present in the certs directory But with the config: ws_servers: 'wss://my_domain.com:7443' I do not even get a connection and I receive a 1006 error. Anybody has a working wss connection with Jssip and can share his config with me? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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