Re: [Freeswitch-users] FS with SSL/TLS issues!
Moreover, i am unable to import "cafile.pem" in my 3CX phone. Could it be the reason if these errors? 1.Do i need client certificate to establish SSL connection with every softphone? 2.I have read that we can use FScomm phone to test but where does it get the certificate file ? 3.How can i check that FS has really seen the agent.pem file and OK for SSL/TLS SIP? Meanwhile i'll map domain name on FS server and will try to generate certificates using domain name. ...and ill be waiting for anyone to put his thoughts in , where i am! Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
Just go get a regular signed cert like you would for mod_ssl for apache, It will work. cat the key and the cert into agent.pem and the chain cert into cafile.pem and fire it up... I did this it works great. Startssl.com is a good alternative -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 ISN: 410*543 On Feb 22, 2013, at 1:59 AM, mehroz <mehroz.ashraf85@gmail.com> wrote:
Moreover, i am unable to import "cafile.pem" in my 3CX phone. Could it be the reason if these errors?
1.Do i need client certificate to establish SSL connection with every softphone? 2.I have read that we can use FScomm phone to test but where does it get the certificate file ? 3.How can i check that FS has really seen the agent.pem file and OK for SSL/TLS SIP?
Meanwhile i'll map domain name on FS server and will try to generate certificates using domain name. ...and ill be waiting for anyone to put his thoughts in , where i am! Thanks
Alright, Thanks for inputs! I can successfully Register with FS on TLS now. The issue was not the CA root certificate nor domains requirement or Signed certificate, but it was because client certificate were not properly installed. (For those , trying on 3CX , you have to rename "cafile.pem" to "root_cert_3CXPHONE.pem"). Now, the problem is internal calling! and i guess it is same as Limitations of the Freeswitch TLS/SSLv23 Implementation (FS-3877) <http://wiki.freeswitch.org/wiki/SIP_TLS#Limitations_of_the_Freeswitch_TLS.2FSSLv23_Implementation_.28FS-3877.29> . I have been looking for the solution but coudnt get that yet! I can make call from A but the called party B does not rang, but then i can call B to A and it rings (strange, as both are in same network , same NAT). Can one can guide me step wise what to do now ! ? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
TLS-NAT issue arises due to the fact that it tries to contact on other port (TCP session) as far as my understanding. Changing contact header can sort this issue and i have done it setting up <variable name="sip-force-contact" value="NDLB-tls-connectile-dysfunction"/> for each directory (user). This helped and worked :) I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" in SIP profile. Cheeers! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
Hi, I've done all you said but I'm in trouble indeed: TLS/SRTP enabled on two extension; If I try to dial between the two extensions I have: /mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION/ If I call outbound, it's ok, and the call is encrypted. Can you help me? Vic On 27/02/13 15:47, mehroz wrote:
TLS-NAT issue arises due to the fact that it tries to contact on other port (TCP session) as far as my understanding.
Changing contact header can sort this issue and i have done it setting up
<variable name="sip-force-contact" value="NDLB-tls-connectile-dysfunction"/> for each directory (user).
This helped and worked :)
I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" in SIP profile. Cheeers!
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
This will work better in 1.4 but the code in 1.2 is frozen so we can't improve it there. In the meantime you can just do this: make an acl that is default allow called tls add it to apply-nat-acl param in sofia profile where you have tls Now it will record the rport for every client in the contact path and it should resolve the existing connection. On Wed, Feb 27, 2013 at 3:32 PM, Vittorio Guglielmo < vittorio.guglielmo@gmail.com> wrote:
Hi,
I've done all you said but I'm in trouble indeed:
TLS/SRTP enabled on two extension;
If I try to dial between the two extensions I have:
*mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION*
If I call outbound, it's ok, and the call is encrypted.
Can you help me?
Vic
On 27/02/13 15:47, mehroz wrote:
TLS-NAT issue arises due to the fact that it tries to contact on other port (TCP session) as far as my understanding.
Changing contact header can sort this issue and i have done it setting up
<variable name="sip-force-contact" value="NDLB-tls-connectile-dysfunction"/> for each directory (user).
This helped and worked :)
I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" in SIP profile. Cheeers!
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
I did it, now I have: /mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION/ The only way I succeded was to add /bypass_media=true/ in the dialplan for each extension, but I have a huge latency in the voice (like 2/3 seconds). Vic On 27/02/13 22:54, Anthony Minessale wrote:
This will work better in 1.4 but the code in 1.2 is frozen so we can't improve it there. In the meantime you can just do this:
make an acl that is default allow called tls add it to apply-nat-acl param in sofia profile where you have tls
Now it will record the rport for every client in the contact path and it should resolve the existing connection.
On Wed, Feb 27, 2013 at 3:32 PM, Vittorio Guglielmo <vittorio.guglielmo@gmail.com <mailto:vittorio.guglielmo@gmail.com>> wrote:
Hi,
I've done all you said but I'm in trouble indeed:
TLS/SRTP enabled on two extension;
If I try to dial between the two extensions I have:
/mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION/
If I call outbound, it's ok, and the call is encrypted.
Can you help me?
Vic
On 27/02/13 15:47, mehroz wrote:
TLS-NAT issue arises due to the fact that it tries to contact on other port (TCP session) as far as my understanding.
Changing contact header can sort this issue and i have done it setting up
<variable name="sip-force-contact" value="NDLB-tls-connectile-dysfunction"/> for each directory (user).
This helped and worked :)
I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" in SIP profile. Cheeers!
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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@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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Now you are just stuck with ordinary problems with how you have things configured. One line is not enough to tell you exactly why but a good guess is you have codecs that don't match. This is unrelated to TLS. On Wed, Feb 27, 2013 at 4:13 PM, Vittorio Guglielmo < vittorio.guglielmo@gmail.com> wrote:
I did it, now I have:
*mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION*
The only way I succeded was to add *bypass_media=true* in the dialplan for each extension, but I have a huge latency in the voice (like 2/3 seconds).
Vic
On 27/02/13 22:54, Anthony Minessale wrote:
This will work better in 1.4 but the code in 1.2 is frozen so we can't improve it there. In the meantime you can just do this:
make an acl that is default allow called tls add it to apply-nat-acl param in sofia profile where you have tls
Now it will record the rport for every client in the contact path and it should resolve the existing connection.
On Wed, Feb 27, 2013 at 3:32 PM, Vittorio Guglielmo < vittorio.guglielmo@gmail.com> wrote:
Hi,
I've done all you said but I'm in trouble indeed:
TLS/SRTP enabled on two extension;
If I try to dial between the two extensions I have:
*mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION*
If I call outbound, it's ok, and the call is encrypted.
Can you help me?
Vic
On 27/02/13 15:47, mehroz wrote:
TLS-NAT issue arises due to the fact that it tries to contact on other port (TCP session) as far as my understanding.
Changing contact header can sort this issue and i have done it setting up
<variable name="sip-force-contact" value="NDLB-tls-connectile-dysfunction"/> for each directory (user).
This helped and worked :)
I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" in SIP profile. Cheeers!
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
I thought that codec could be the cause, but all the extensions have only G.711 (alaw and ulaw). These is the log when I try to call between the extensions: 2013-02-27 23:53:13.085555 [NOTICE] switch_channel.c:968 New Channel sofia/internal/101@voice.wicube.net [771747c6-8130-11e2-a4a8-efe939e44db4] 2013-02-27 23:53:13.085555 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.085555 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.085555 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101@voice.wicube.net) Running State Change CS_NEW 2013-02-27 23:53:13.085555 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/101@voice.wicube.net) State NEW 2013-02-27 23:53:13.105591 [DEBUG] sofia.c:7729 IP 82.85.60.13 Rejected by acl "domains". Falling back to Digest auth. 2013-02-27 23:53:13.105591 [WARNING] sofia_reg.c:1502 SIP auth challenge (INVITE) on sofia profile 'internal' for [103@voice.wicube.net] from ip 82.85.60.13 2013-02-27 23:53:13.105591 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.105591 [DEBUG] sofia.c:1750 detaching session 771747c6-8130-11e2-a4a8-efe939e44db4 2013-02-27 23:53:13.145610 [DEBUG] sofia.c:1842 Re-attaching to session 771747c6-8130-11e2-a4a8-efe939e44db4 2013-02-27 23:53:13.145610 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.145610 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.145610 [DEBUG] sofia.c:7729 IP 82.85.60.13 Rejected by acl "domains". Falling back to Digest auth. 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5599 Channel sofia/internal/101@voice.wicube.net entering state [received][100] 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5610 Remote SDP: v=0 o=- 3570994402 3570994402 IN IP4 10.10.5.101 s=cpc_med c=IN IP4 10.10.5.101 t=0 0 m=audio 53526 RTP/SAVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:G5TO9FRh3y0+WkCMn336K2K4X3iaYBFRdNsPF9f5 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:Fn6vvWfGAQH497IRM9HaO+MEoMw+tjpeyAdA4kId a=ice-ufrag:75f854d0 a=ice-pwd:02886032 a=candidate:Ha0a0565 1 UDP 2130706431 10.10.5.101 53526 typ host a=candidate:H5587298 1 UDP 2130706431 5.88.114.152 53526 typ host a=candidate:Ha0a0565 2 UDP 2130706430 10.10.5.101 64080 typ host a=candidate:H5587298 2 UDP 2130706430 5.88.114.152 64080 typ host 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 AES_CM_128_HMAC_SHA1_80 inline:G5TO9FRh3y0+WkCMn336K2K4X3iaYBFRdNsPF9f5] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:3155 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:kbdihjLnWO2Mnu0tsc+VpZKyIwsw7eKVp7ODyoxx] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:3093 Set Codec sofia/internal/101@voice.wicube.net PCMU/8000 20 ms 160 samples 64000 bits 2013-02-27 23:53:13.165556 [DEBUG] switch_core_codec.c:111 sofia/internal/101@voice.wicube.net Original read codec set to PCMU:0 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5827 (sofia/internal/101@voice.wicube.net) State Change CS_NEW -> CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101@voice.wicube.net) Running State Change CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/101@voice.wicube.net) State INIT 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:86 sofia/internal/101@voice.wicube.net SOFIA INIT 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:126 (sofia/internal/101@voice.wicube.net) State Change CS_INIT -> CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/101@voice.wicube.net) State INIT going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101@voice.wicube.net) Running State Change CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:2003 (sofia/internal/101@voice.wicube.net) Callstate Change DOWN -> RINGING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/101@voice.wicube.net) State ROUTING 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:149 sofia/internal/101@voice.wicube.net SOFIA ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:117 sofia/internal/101@voice.wicube.net Standard ROUTING 2013-02-27 23:53:13.165556 [INFO] mod_dialplan_xml.c:557 Processing 101 <101>->103 in context default Dialplan: sofia/internal/101@voice.wicube.net parsing [default->101] continue=false Dialplan: sofia/internal/101@voice.wicube.net Regex (FAIL) [101] destination_number(103) =~ /^101$/ break=on-false Dialplan: sofia/internal/101@voice.wicube.net parsing [default->102] continue=false Dialplan: sofia/internal/101@voice.wicube.net Regex (FAIL) [102] destination_number(103) =~ /^102$/ break=on-false Dialplan: sofia/internal/101@voice.wicube.net parsing [default->103] continue=false Dialplan: sofia/internal/101@voice.wicube.net Regex (PASS) [103] destination_number(103) =~ /^103$/ break=on-false Dialplan: sofia/internal/101@voice.wicube.net Action bridge(USER/103@82.85.60.237) 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/101@voice.wicube.net) State Change CS_ROUTING -> CS_EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/101@voice.wicube.net) State ROUTING going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101@voice.wicube.net) Running State Change CS_EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/101@voice.wicube.net) State EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:242 sofia/internal/101@voice.wicube.net SOFIA EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:209 sofia/internal/101@voice.wicube.net Standard EXECUTE EXECUTE sofia/internal/101@voice.wicube.net bridge(USER/103@82.85.60.237) 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:1089 sofia/internal/101@voice.wicube.net EXPORTING[export_vars] [domain_name]=[82.85.60.237] to event 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:1089 sofia/internal/101@voice.wicube.net EXPORTING[export_vars] [domain_name]=[82.85.60.237] to event 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-02-27 23:53:13.165556 [DEBUG] switch_event.c:1608 Parsing variable [sip_invite_domain]=[82.85.60.237] 2013-02-27 23:53:13.165556 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[103@82.85.60.237] 2013-02-27 23:53:13.165556 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:103@82.85.60.13:37595 [77239eea-8130-11e2-a4b2-efe939e44db4] 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:4970 (sofia/internal/sip:103@82.85.60.13:37595) State Change CS_NEW -> CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:103@82.85.60.13:37595) State INIT 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:86 sofia/internal/sip:103@82.85.60.13:37595 SOFIA INIT 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:2618 sip:103@82.85.60.13:37595;transport=TLS;ob Setting proxy route to sofia/internal/sip:103@82.85.60.13:37595 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1361974937 1361974938 IN IP4 82.85.60.237 s=FreeSWITCH c=IN IP4 82.85.60.237 t=0 0 m=audio 30656 RTP/AVP 0 9 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:126 (sofia/internal/sip:103@82.85.60.13:37595) State Change CS_INIT -> CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:103@82.85.60.13:37595) State INIT going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:2003 (sofia/internal/sip:103@82.85.60.13:37595) Callstate Change DOWN -> RINGING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:103@82.85.60.13:37595) State ROUTING 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:149 sofia/internal/sip:103@82.85.60.13:37595 SOFIA ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:103@82.85.60.13:37595) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:103@82.85.60.13:37595) State ROUTING going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_CONSUME_MEDIA 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:103@82.85.60.13:37595) State CONSUME_MEDIA 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:103@82.85.60.13:37595) State CONSUME_MEDIA going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5599 Channel sofia/internal/sip:103@82.85.60.13:37595 entering state [calling][0] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] sofia.c:5599 Channel sofia/internal/sip:103@82.85.60.13:37595 entering state [terminated][488] 2013-02-27 23:53:14.705571 [DEBUG] switch_channel.c:2994 (sofia/internal/sip:103@82.85.60.13:37595) Callstate Change RINGING -> HANGUP 2013-02-27 23:53:14.705571 [NOTICE] sofia.c:6383 Hangup sofia/internal/sip:103@82.85.60.13:37595 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.705571 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/sip:103@82.85.60.13:37595 [KILL] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_HANGUP 2013-02-27 23:53:14.705571 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:103@82.85.60.13:37595) State HANGUP 2013-02-27 23:53:14.705571 [DEBUG] mod_sofia.c:503 Channel sofia/internal/sip:103@82.85.60.13:37595 hanging up, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:48 sofia/internal/sip:103@82.85.60.13:37595 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:103@82.85.60.13:37595) State HANGUP going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/sip:103@82.85.60.13:37595) State Change CS_HANGUP -> CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:103@82.85.60.13:37595) State REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:92 sofia/internal/sip:103@82.85.60.13:37595 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:103@82.85.60.13:37595) State REPORTING going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:103@82.85.60.13:37595) State Change CS_REPORTING -> CS_DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1488 Session 10 (sofia/internal/sip:103@82.85.60.13:37595) Locked, Waiting on external entities 2013-02-27 23:53:14.725551 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [USER] cause: [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [INFO] mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_channel.c:2994 (sofia/internal/101@voice.wicube.net) Callstate Change RINGING -> HANGUP 2013-02-27 23:53:14.725551 [NOTICE] mod_dptools.c:3172 Hangup sofia/internal/101@voice.wicube.net [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/101@voice.wicube.net [KILL] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:2678 sofia/internal/101@voice.wicube.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/101@voice.wicube.net) State EXECUTE going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101@voice.wicube.net) Running State Change CS_HANGUP 2013-02-27 23:53:14.725551 [NOTICE] switch_core_session.c:1506 Session 10 (sofia/internal/sip:103@82.85.60.13:37595) Ended 2013-02-27 23:53:14.725551 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/sip:103@82.85.60.13:37595 [CS_DESTROY] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/sip:103@82.85.60.13:37595) Callstate Change HANGUP -> DOWN 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:103@82.85.60.13:37595) State DESTROY 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:396 sofia/internal/sip:103@82.85.60.13:37595 SOFIA DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:99 sofia/internal/sip:103@82.85.60.13:37595 Standard DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:103@82.85.60.13:37595) State DESTROY going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/101@voice.wicube.net) State HANGUP 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:497 sofia/internal/101@voice.wicube.net Overriding SIP cause 488 with 488 from the other leg 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:503 Channel sofia/internal/101@voice.wicube.net hanging up, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 488 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:48 sofia/internal/101@voice.wicube.net Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/101@voice.wicube.net) State HANGUP going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/101@voice.wicube.net) State Change CS_HANGUP -> CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/101@voice.wicube.net) Running State Change CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/101@voice.wicube.net) State REPORTING 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:92 sofia/internal/101@voice.wicube.net Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/101@voice.wicube.net) State REPORTING going to sleep 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/101@voice.wicube.net) State Change CS_REPORTING -> CS_DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:14.765550 [DEBUG] switch_core_session.c:1488 Session 9 (sofia/internal/101@voice.wicube.net) Locked, Waiting on external entities 2013-02-27 23:53:14.765550 [NOTICE] switch_core_session.c:1506 Session 9 (sofia/internal/101@voice.wicube.net) Ended 2013-02-27 23:53:14.765550 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/101@voice.wicube.net [CS_DESTROY] 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/101@voice.wicube.net) Callstate Change HANGUP -> DOWN 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/101@voice.wicube.net) Running State Change CS_DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/101@voice.wicube.net) State DESTROY 2013-02-27 23:53:14.765550 [DEBUG] mod_sofia.c:396 sofia/internal/101@voice.wicube.net SOFIA DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:99 sofia/internal/101@voice.wicube.net Standard DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/101@voice.wicube.net) State DESTROY going to sleep Vic On 27/02/13 23:22, Anthony Minessale wrote:
Now you are just stuck with ordinary problems with how you have things configured. One line is not enough to tell you exactly why but a good guess is you have codecs that don't match. This is unrelated to TLS.
On Wed, Feb 27, 2013 at 4:13 PM, Vittorio Guglielmo <vittorio.guglielmo@gmail.com <mailto:vittorio.guglielmo@gmail.com>> wrote:
I did it, now I have:
/mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION/
The only way I succeded was to add /bypass_media=true/ in the dialplan for each extension, but I have a huge latency in the voice (like 2/3 seconds).
Vic
On 27/02/13 22:54, Anthony Minessale wrote:
This will work better in 1.4 but the code in 1.2 is frozen so we can't improve it there. In the meantime you can just do this:
make an acl that is default allow called tls add it to apply-nat-acl param in sofia profile where you have tls
Now it will record the rport for every client in the contact path and it should resolve the existing connection.
On Wed, Feb 27, 2013 at 3:32 PM, Vittorio Guglielmo <vittorio.guglielmo@gmail.com <mailto:vittorio.guglielmo@gmail.com>> wrote:
Hi,
I've done all you said but I'm in trouble indeed:
TLS/SRTP enabled on two extension;
If I try to dial between the two extensions I have:
/mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION/
If I call outbound, it's ok, and the call is encrypted.
Can you help me?
Vic
On 27/02/13 15:47, mehroz wrote:
TLS-NAT issue arises due to the fact that it tries to contact on other port (TCP session) as far as my understanding.
Changing contact header can sort this issue and i have done it setting up
<variable name="sip-force-contact" value="NDLB-tls-connectile-dysfunction"/> for each directory (user).
This helped and worked :)
I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" in SIP profile. Cheeers!
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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@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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 <tel:%2B19193869900>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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@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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You should turn on the sip trace too sofia global siptrace on Then you can see the 488 you are getting and try to figure out why. On Wed, Feb 27, 2013 at 4:56 PM, Vittorio Guglielmo < vittorio.guglielmo@gmail.com> wrote:
I thought that codec could be the cause, but all the extensions have only G.711 (alaw and ulaw).
These is the log when I try to call between the extensions:
2013-02-27 23:53:13.085555 [NOTICE] switch_channel.c:968 New Channel sofia/internal/101@voice.wicube.net [771747c6-8130-11e2-a4a8-efe939e44db4] 2013-02-27 23:53:13.085555 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.085555 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.085555 [DEBUG] switch_core_state_machine.c:415 ( sofia/internal/101@voice.wicube.net) Running State Change CS_NEW 2013-02-27 23:53:13.085555 [DEBUG] switch_core_state_machine.c:433 ( sofia/internal/101@voice.wicube.net) State NEW 2013-02-27 23:53:13.105591 [DEBUG] sofia.c:7729 IP 82.85.60.13 Rejected by acl "domains". Falling back to Digest auth. 2013-02-27 23:53:13.105591 [WARNING] sofia_reg.c:1502 SIP auth challenge (INVITE) on sofia profile 'internal' for [103@voice.wicube.net] from ip 82.85.60.13 2013-02-27 23:53:13.105591 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.105591 [DEBUG] sofia.c:1750 detaching session 771747c6-8130-11e2-a4a8-efe939e44db4 2013-02-27 23:53:13.145610 [DEBUG] sofia.c:1842 Re-attaching to session 771747c6-8130-11e2-a4a8-efe939e44db4 2013-02-27 23:53:13.145610 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.145610 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.145610 [DEBUG] sofia.c:7729 IP 82.85.60.13 Rejected by acl "domains". Falling back to Digest auth. 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5599 Channel sofia/internal/101@voice.wicube.net entering state [received][100] 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5610 Remote SDP: v=0 o=- 3570994402 3570994402 IN IP4 10.10.5.101 s=cpc_med c=IN IP4 10.10.5.101 t=0 0 m=audio 53526 RTP/SAVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:G5TO9FRh3y0+WkCMn336K2K4X3iaYBFRdNsPF9f5 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:Fn6vvWfGAQH497IRM9HaO+MEoMw+tjpeyAdA4kId a=ice-ufrag:75f854d0 a=ice-pwd:02886032 a=candidate:Ha0a0565 1 UDP 2130706431 10.10.5.101 53526 typ host a=candidate:H5587298 1 UDP 2130706431 5.88.114.152 53526 typ host a=candidate:Ha0a0565 2 UDP 2130706430 10.10.5.101 64080 typ host a=candidate:H5587298 2 UDP 2130706430 5.88.114.152 64080 typ host
2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 AES_CM_128_HMAC_SHA1_80 inline:G5TO9FRh3y0+WkCMn336K2K4X3iaYBFRdNsPF9f5] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:3155 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:kbdihjLnWO2Mnu0tsc+VpZKyIwsw7eKVp7ODyoxx] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5137 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:3093 Set Codec sofia/internal/101@voice.wicube.net PCMU/8000 20 ms 160 samples 64000 bits 2013-02-27 23:53:13.165556 [DEBUG] switch_core_codec.c:111 sofia/internal/101@voice.wicube.net Original read codec set to PCMU:0 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf send/recv payload to 101 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5827 ( sofia/internal/101@voice.wicube.net) State Change CS_NEW -> CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 ( sofia/internal/101@voice.wicube.net) Running State Change CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 ( sofia/internal/101@voice.wicube.net) State INIT 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:86 sofia/internal/101@voice.wicube.net SOFIA INIT 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:126 ( sofia/internal/101@voice.wicube.net) State Change CS_INIT -> CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 ( sofia/internal/101@voice.wicube.net) State INIT going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 ( sofia/internal/101@voice.wicube.net) Running State Change CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:2003 ( sofia/internal/101@voice.wicube.net) Callstate Change DOWN -> RINGING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 ( sofia/internal/101@voice.wicube.net) State ROUTING 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:149 sofia/internal/101@voice.wicube.net SOFIA ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:117 sofia/internal/101@voice.wicube.net Standard ROUTING 2013-02-27 23:53:13.165556 [INFO] mod_dialplan_xml.c:557 Processing 101 <101>->103 in context default Dialplan: sofia/internal/101@voice.wicube.net parsing [default->101] continue=false Dialplan: sofia/internal/101@voice.wicube.net Regex (FAIL) [101] destination_number(103) =~ /^101$/ break=on-false Dialplan: sofia/internal/101@voice.wicube.net parsing [default->102] continue=false Dialplan: sofia/internal/101@voice.wicube.net Regex (FAIL) [102] destination_number(103) =~ /^102$/ break=on-false Dialplan: sofia/internal/101@voice.wicube.net parsing [default->103] continue=false Dialplan: sofia/internal/101@voice.wicube.net Regex (PASS) [103] destination_number(103) =~ /^103$/ break=on-false Dialplan: sofia/internal/101@voice.wicube.net Action bridge( USER/103@82.85.60.237) 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:167 ( sofia/internal/101@voice.wicube.net) State Change CS_ROUTING -> CS_EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 ( sofia/internal/101@voice.wicube.net) State ROUTING going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 ( sofia/internal/101@voice.wicube.net) Running State Change CS_EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:477 ( sofia/internal/101@voice.wicube.net) State EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:242 sofia/internal/101@voice.wicube.net SOFIA EXECUTE 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:209 sofia/internal/101@voice.wicube.net Standard EXECUTE EXECUTE sofia/internal/101@voice.wicube.net bridge(USER/103@82.85.60.237) 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:1089 sofia/internal/101@voice.wicube.net EXPORTING[export_vars] [domain_name]=[82.85.60.237] to event 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:1089 sofia/internal/101@voice.wicube.net EXPORTING[export_vars] [domain_name]=[82.85.60.237] to event 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-02-27 23:53:13.165556 [DEBUG] switch_event.c:1608 Parsing variable [sip_invite_domain]=[82.85.60.237] 2013-02-27 23:53:13.165556 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[103@82.85.60.237] 2013-02-27 23:53:13.165556 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:103@82.85.60.13:37595[77239eea-8130-11e2-a4b2-efe939e44db4] 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:4970 (sofia/internal/ sip:103@82.85.60.13:37595) State Change CS_NEW -> CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_INIT 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:103@82.85.60.13:37595) State INIT 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:86 sofia/internal/ sip:103@82.85.60.13:37595 SOFIA INIT 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:2618 sip:103@82.85.60.13:37595;transport=TLS;ob Setting proxy route to sofia/internal/sip:103@82.85.60.13:37595 2013-02-27 23:53:13.165556 [DEBUG] sofia_glue.c:2647 Local SDP: v=0 o=FreeSWITCH 1361974937 1361974938 IN IP4 82.85.60.237 s=FreeSWITCH c=IN IP4 82.85.60.237 t=0 0 m=audio 30656 RTP/AVP 0 9 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:126 (sofia/internal/ sip:103@82.85.60.13:37595) State Change CS_INIT -> CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:103@82.85.60.13:37595) State INIT going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_channel.c:2003 (sofia/internal/ sip:103@82.85.60.13:37595) Callstate Change DOWN -> RINGING 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:103@82.85.60.13:37595) State ROUTING 2013-02-27 23:53:13.165556 [DEBUG] mod_sofia.c:149 sofia/internal/ sip:103@82.85.60.13:37595 SOFIA ROUTING 2013-02-27 23:53:13.165556 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:103@82.85.60.13:37595) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:103@82.85.60.13:37595) State ROUTING going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_CONSUME_MEDIA 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:103@82.85.60.13:37595) State CONSUME_MEDIA 2013-02-27 23:53:13.165556 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:103@82.85.60.13:37595) State CONSUME_MEDIA going to sleep 2013-02-27 23:53:13.165556 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:13.165556 [DEBUG] sofia.c:5599 Channel sofia/internal/ sip:103@82.85.60.13:37595 entering state [calling][0] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] sofia.c:5599 Channel sofia/internal/ sip:103@82.85.60.13:37595 entering state [terminated][488] 2013-02-27 23:53:14.705571 [DEBUG] switch_channel.c:2994 (sofia/internal/ sip:103@82.85.60.13:37595) Callstate Change RINGING -> HANGUP 2013-02-27 23:53:14.705571 [NOTICE] sofia.c:6383 Hangup sofia/internal/ sip:103@82.85.60.13:37595 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.705571 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/sip:103@82.85.60.13:37595 [KILL] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.705571 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_HANGUP 2013-02-27 23:53:14.705571 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:103@82.85.60.13:37595) State HANGUP 2013-02-27 23:53:14.705571 [DEBUG] mod_sofia.c:503 Channel sofia/internal/ sip:103@82.85.60.13:37595 hanging up, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:48 sofia/internal/sip:103@82.85.60.13:37595 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:103@82.85.60.13:37595) State HANGUP going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/sip:103@82.85.60.13:37595) State Change CS_HANGUP -> CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:103@82.85.60.13:37595) State REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:92 sofia/internal/sip:103@82.85.60.13:37595 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:103@82.85.60.13:37595) State REPORTING going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:103@82.85.60.13:37595) State Change CS_REPORTING -> CS_DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/sip:103@82.85.60.13:37595 [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1488 Session 10 (sofia/internal/sip:103@82.85.60.13:37595) Locked, Waiting on external entities 2013-02-27 23:53:14.725551 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [USER] cause: [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [INFO] mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_channel.c:2994 ( sofia/internal/101@voice.wicube.net) Callstate Change RINGING -> HANGUP 2013-02-27 23:53:14.725551 [NOTICE] mod_dptools.c:3172 Hangup sofia/internal/101@voice.wicube.net [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2013-02-27 23:53:14.725551 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/101@voice.wicube.net [KILL] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:2678 sofia/internal/101@voice.wicube.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:477 ( sofia/internal/101@voice.wicube.net) State EXECUTE going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 ( sofia/internal/101@voice.wicube.net) Running State Change CS_HANGUP 2013-02-27 23:53:14.725551 [NOTICE] switch_core_session.c:1506 Session 10 (sofia/internal/sip:103@82.85.60.13:37595) Ended 2013-02-27 23:53:14.725551 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/sip:103@82.85.60.13:37595 [CS_DESTROY] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/sip:103@82.85.60.13:37595) Callstate Change HANGUP -> DOWN 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/sip:103@82.85.60.13:37595) Running State Change CS_DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:103@82.85.60.13:37595) State DESTROY 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:396 sofia/internal/ sip:103@82.85.60.13:37595 SOFIA DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:99 sofia/internal/sip:103@82.85.60.13:37595 Standard DESTROY 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:103@82.85.60.13:37595) State DESTROY going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 ( sofia/internal/101@voice.wicube.net) State HANGUP 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:497 sofia/internal/101@voice.wicube.net Overriding SIP cause 488 with 488 from the other leg 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:503 Channel sofia/internal/101@voice.wicube.net hanging up, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 488 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:48 sofia/internal/101@voice.wicube.net Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:667 ( sofia/internal/101@voice.wicube.net) State HANGUP going to sleep 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:446 ( sofia/internal/101@voice.wicube.net) State Change CS_HANGUP -> CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:415 ( sofia/internal/101@voice.wicube.net) Running State Change CS_REPORTING 2013-02-27 23:53:14.725551 [DEBUG] switch_core_state_machine.c:749 ( sofia/internal/101@voice.wicube.net) State REPORTING 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:92 sofia/internal/101@voice.wicube.net Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:749 ( sofia/internal/101@voice.wicube.net) State REPORTING going to sleep 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:440 ( sofia/internal/101@voice.wicube.net) State Change CS_REPORTING -> CS_DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_session.c:1283 Send signal sofia/internal/101@voice.wicube.net [BREAK] 2013-02-27 23:53:14.765550 [DEBUG] switch_core_session.c:1488 Session 9 ( sofia/internal/101@voice.wicube.net) Locked, Waiting on external entities 2013-02-27 23:53:14.765550 [NOTICE] switch_core_session.c:1506 Session 9 ( sofia/internal/101@voice.wicube.net) Ended 2013-02-27 23:53:14.765550 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/101@voice.wicube.net [CS_DESTROY] 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:556 ( sofia/internal/101@voice.wicube.net) Callstate Change HANGUP -> DOWN 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:559 ( sofia/internal/101@voice.wicube.net) Running State Change CS_DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:569 ( sofia/internal/101@voice.wicube.net) State DESTROY 2013-02-27 23:53:14.765550 [DEBUG] mod_sofia.c:396 sofia/internal/101@voice.wicube.net SOFIA DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:99 sofia/internal/101@voice.wicube.net Standard DESTROY 2013-02-27 23:53:14.765550 [DEBUG] switch_core_state_machine.c:569 ( sofia/internal/101@voice.wicube.net) State DESTROY going to sleep
Vic
On 27/02/13 23:22, Anthony Minessale wrote:
Now you are just stuck with ordinary problems with how you have things configured. One line is not enough to tell you exactly why but a good guess is you have codecs that don't match. This is unrelated to TLS.
On Wed, Feb 27, 2013 at 4:13 PM, Vittorio Guglielmo < vittorio.guglielmo@gmail.com> wrote:
I did it, now I have:
*mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION*
The only way I succeded was to add *bypass_media=true* in the dialplan for each extension, but I have a huge latency in the voice (like 2/3 seconds).
Vic
On 27/02/13 22:54, Anthony Minessale wrote:
This will work better in 1.4 but the code in 1.2 is frozen so we can't improve it there. In the meantime you can just do this:
make an acl that is default allow called tls add it to apply-nat-acl param in sofia profile where you have tls
Now it will record the rport for every client in the contact path and it should resolve the existing connection.
On Wed, Feb 27, 2013 at 3:32 PM, Vittorio Guglielmo < vittorio.guglielmo@gmail.com> wrote:
Hi,
I've done all you said but I'm in trouble indeed:
TLS/SRTP enabled on two extension;
If I try to dial between the two extensions I have:
*mod_dptools.c:3052 Originate Failed. Cause: INCOMPATIBLE_DESTINATION*
If I call outbound, it's ok, and the call is encrypted.
Can you help me?
Vic
On 27/02/13 15:47, mehroz wrote:
TLS-NAT issue arises due to the fact that it tries to contact on other port (TCP session) as far as my understanding.
Changing contact header can sort this issue and i have done it setting up
<variable name="sip-force-contact" value="NDLB-tls-connectile-dysfunction"/> for each directory (user).
This helped and worked :)
I suppose you should also have "ext-rtp-ip and "ext-sip-ip" = "nat-auto" in SIP profile. Cheeers!
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Anthony, sofia global siptrace on and in attachment are the two siptrace: 1. *101* ( Bria iOS 2.3.4 on Iphone 3GS) *===> 102* (Bria 3 release 3.5.0b on Mac OSX 10.7.5) 2. *102* *===> 101* Both of them have forced only ulaw (first) and alaw (second), and without TSL/SRTP they call each other without problems. The behavior is different in the two above cases: When I call from 101 to 102 I get a *Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]* due to, I believe a: *SIP/2.0 415 Unsupported Media Type* When I instead call from 102 to 101 I get a *Originate Failed. Cause: INCOMPATIBLE_DESTINATION* with a : *SIP/2.0 488 Not Acceptable Here* Vic On 28/02/13 00:23, Anthony Minessale wrote:
sofia global siptrace on
They probably do not accept calls over TLS that are not SAVP On Thu, Feb 28, 2013 at 6:36 AM, Vittorio Guglielmo < vittorio.guglielmo@gmail.com> wrote:
Anthony,
sofia global siptrace on
and in attachment are the two siptrace:
1. *101* ( Bria iOS 2.3.4 on Iphone 3GS) *===> 102* (Bria 3 release 3.5.0b on Mac OSX 10.7.5) 2. *102* *===> 101*
Both of them have forced only ulaw (first) and alaw (second), and without TSL/SRTP they call each other without problems.
The behavior is different in the two above cases:
When I call from 101 to 102 I get a *Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]* due to, I believe a: *SIP/2.0 415 Unsupported Media Type*
When I instead call from 102 to 101 I get a *Originate Failed. Cause: INCOMPATIBLE_DESTINATION* with a : * SIP/2.0 488 Not Acceptable Here*
Vic
On 28/02/13 00:23, Anthony Minessale wrote:
sofia global siptrace on
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Sorry for the repost. I had to change my email to be able to respond to the list. I did post in pastebin a snippet of the dial plan. Pretty much, every thing is set to the default installation. To clarify my issue, I need to be able to execute a dial plan when FreeSWITCH sends an INVITE in response to a REFER. So far, I am not able to locate a setting that would allow me to do that. The switch_ivr_originate.c file transitions the channel from CS_ROUTING to CS_CONSUME_MEDIA which I believe is bypassing the CS_EXECUTE to trigger executing the dialplan. thanks ------------------------------------- Michael Collins msc at freeswitch.org Tue Feb 26 22:17:12 MSK 2013 Previous message: [Freeswitch-users] dialplan action when sending a REFER to FreeSWITCH Next message: [Freeswitch-users] nibblebill, spidermonkey_odbc issue Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Would you mind putting a snippet of your dialplan into our pastebin? ( pastebin.freeswitch.org) I'm sure the folks here can help you figure it out. With FreeSWITCH there's almost always a way to do it and it's just a matter of figuring out what works for your scenario. -MC On Tue, Feb 26, 2013 at 10:44 AM, Hussein Said < Hussein.Said at cassidiancommunications.com> wrote:
Hi,****
I have an application that routes a call to an ad-hoc conference. That application then hunts an extension to answer the call by sending a REFER to FreeSWITCH which will send an INVITE from the conference to join. ****
I need a way to make the “hunted” extension the moderator so that it can control the conference. ****
I did set a moderator controls profile and can see it works if I change the dial plan(public.xml) to set the “member-flag=moderator”.****
Also, “mod-wait|rfc4579” are set in the conference.conf.xml.****
My challenge is that when I hunt for an extension and find one, I send a REFER to FREESwitch so that it sends an INVITE to the extension that needs to answer the call, and I could not find a variable that worked in the dialplan that I can use as a condition to set the “member-flag=moderator” for the channel that is about to join the conference. Is what I am attempting feasible?****
** **
Thanks****
** **
** **
** **
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
Can you please explain to me what does it mean ? And, in case, how can I demonstrate it, cause I need to buy a lot of clients to build a secured voip network. Thank you very much for your time and your effort on a so huge piece of software ( I bought all of your books ). Vic On 28/02/13 17:42, Anthony Minessale wrote:
They probably do not accept calls over TLS that are not SAVP
On Thu, Feb 28, 2013 at 6:36 AM, Vittorio Guglielmo <vittorio.guglielmo@gmail.com <mailto:vittorio.guglielmo@gmail.com>> wrote:
Anthony,
sofia global siptrace on
and in attachment are the two siptrace:
1. *101* ( Bria iOS 2.3.4 on Iphone 3GS) *===> 102* (Bria 3 release 3.5.0b on Mac OSX 10.7.5) 2. *102* *===> 101*
Both of them have forced only ulaw (first) and alaw (second), and without TSL/SRTP they call each other without problems.
The behavior is different in the two above cases:
When I call from 101 to 102 I get a *Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]* due to, I believe a: *SIP/2.0 415 Unsupported Media Type*
When I instead call from 102 to 101 I get a *Originate Failed. Cause: INCOMPATIBLE_DESTINATION* with a : *SIP/2.0 488 Not Acceptable Here*
Vic
On 28/02/13 00:23, Anthony Minessale wrote:
sofia global siptrace on
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I mean, the phone may expect SRTP to be offered over the TLS socket. You were offering regular RTP profile "AVP" with no crypto in it and they rejected it as unsupported. That makes me guess they require you to enable srtp "SAVP" + crypto tags or they will drop the call. try adding {sip_secure_media=true} in the outbound call to the phone. On Thu, Feb 28, 2013 at 11:03 AM, Vittorio Guglielmo < vittorio.guglielmo@gmail.com> wrote:
Can you please explain to me what does it mean ?
And, in case, how can I demonstrate it, cause I need to buy a lot of clients to build a secured voip network.
Thank you very much for your time and your effort on a so huge piece of software ( I bought all of your books ).
Vic
On 28/02/13 17:42, Anthony Minessale wrote:
They probably do not accept calls over TLS that are not SAVP
On Thu, Feb 28, 2013 at 6:36 AM, Vittorio Guglielmo < vittorio.guglielmo@gmail.com> wrote:
Anthony,
sofia global siptrace on
and in attachment are the two siptrace:
1. *101* ( Bria iOS 2.3.4 on Iphone 3GS) *===> 102* (Bria 3 release 3.5.0b on Mac OSX 10.7.5) 2. *102* *===> 101*
Both of them have forced only ulaw (first) and alaw (second), and without TSL/SRTP they call each other without problems.
The behavior is different in the two above cases:
When I call from 101 to 102 I get a *Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]* due to, I believe a: *SIP/2.0 415 Unsupported Media Type*
When I instead call from 102 to 101 I get a *Originate Failed. Cause: INCOMPATIBLE_DESTINATION* with a : * SIP/2.0 488 Not Acceptable Here*
Vic
On 28/02/13 00:23, Anthony Minessale wrote:
sofia global siptrace on
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Thank you so much Anthony, that solved the problem !!! Vic On 28/02/13 18:29, Anthony Minessale wrote:
I mean, the phone may expect SRTP to be offered over the TLS socket. You were offering regular RTP profile "AVP" with no crypto in it and they rejected it as unsupported. That makes me guess they require you to enable srtp "SAVP" + crypto tags or they will drop the call.
try adding {sip_secure_media=true} in the outbound call to the phone.
On Thu, Feb 28, 2013 at 11:03 AM, Vittorio Guglielmo <vittorio.guglielmo@gmail.com <mailto:vittorio.guglielmo@gmail.com>> wrote:
Can you please explain to me what does it mean ?
And, in case, how can I demonstrate it, cause I need to buy a lot of clients to build a secured voip network.
Thank you very much for your time and your effort on a so huge piece of software ( I bought all of your books ).
Vic
On 28/02/13 17:42, Anthony Minessale wrote:
They probably do not accept calls over TLS that are not SAVP
On Thu, Feb 28, 2013 at 6:36 AM, Vittorio Guglielmo <vittorio.guglielmo@gmail.com <mailto:vittorio.guglielmo@gmail.com>> wrote:
Anthony,
sofia global siptrace on
and in attachment are the two siptrace:
1. *101* ( Bria iOS 2.3.4 on Iphone 3GS) *===> 102* (Bria 3 release 3.5.0b on Mac OSX 10.7.5) 2. *102* *===> 101*
Both of them have forced only ulaw (first) and alaw (second), and without TSL/SRTP they call each other without problems.
The behavior is different in the two above cases:
When I call from 101 to 102 I get a *Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]* due to, I believe a: *SIP/2.0 415 Unsupported Media Type*
When I instead call from 102 to 101 I get a *Originate Failed. Cause: INCOMPATIBLE_DESTINATION* with a : *SIP/2.0 488 Not Acceptable Here*
Vic
On 28/02/13 00:23, Anthony Minessale wrote:
sofia global siptrace on
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 <tel:%2B19193869900>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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@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@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@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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Guys!!!! As discussed earlier i enabled <variable name="sip-force-contact" value="NDLB-tls-connectile-dysfunction"/> forr each directory/user to handle TLS connection port issue by changing contact address/port by setting it. NOW.... I am experiencing another issue due to it and that is re-INVITE after one , resulting in Updating SDP and video session starts in the audio call (as i have to make audio and video call both ) I have tried all possible values like "session-timer" setting to 0 and large value but no luck, and INVITE is received exactly after one minute. ANy one have idea what exactly is happening and what is the solution? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
Attached is the SIP messages trace. Starting from the invite packets after 1 min. Communication is between user 9292 and 332244 sip_trace_1.txt <http://freeswitch-users.2379917.n2.nabble.com/file/n7588151/sip_trace_1.txt> -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
Try doing what i told the other guy, get rid of the NDLB flag and setup a nat-acl that is a global allow. On Fri, Mar 1, 2013 at 10:35 AM, mehroz <mehroz.ashraf85@gmail.com> wrote:
Attached is the SIP messages trace. Starting from the invite packets after 1 min. Communication is between user 9292 and 332244 sip_trace_1.txt < http://freeswitch-users.2379917.n2.nabble.com/file/n7588151/sip_trace_1.txt
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp75877... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
participants (5)
-
Anthony Minessale
-
Brian West
-
Hussein Said
-
mehroz
-
Vittorio Guglielmo