[CRIT] sofia_glue.c:1083 No RTP ports available!
Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet: sofia_glue.c:1083 No RTP ports available! Restarting FS fixes the problem. Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn't happening until the last week or two. Current version we're running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z Here's a quick snip from the console when attempting to dial out to show the context of the error: 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102@192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102@192.168.1.6:5060 Original read codec set to PCMU:0 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available! 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102@192.168.1.6:5060 [BREAK] 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102@192.168.1.6:5060) Callstate Change RINGING -> HANGUP 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com
Repeat the call again and dump a SIP trace here while you're at it On Sun, Jul 22, 2012 at 10:16 AM, Phil Quesinberry <philq@qsystemsengineering.com> wrote:
Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet:
sofia_glue.c:1083 No RTP ports available!
Restarting FS fixes the problem.
Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn’t happening until the last week or two.
Current version we’re running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z
Here’s a quick snip from the console when attempting to dial out to show the context of the error:
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102@192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102@192.168.1.6:5060 Original read codec set to PCMU:0
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO
2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available!
2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102@192.168.1.6:5060 [BREAK]
2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102@192.168.1.6:5060) Callstate Change RINGING -> HANGUP
2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Regards,
Phil Quesinberry
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002
http://www.qsystemsengineering.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
Join Us At ClueCon - Aug 7-9, 2012
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
Jira Jira jira On Jul 22, 2012 12:41 PM, "Phil Quesinberry" <philq@qsystemsengineering.com> wrote:
**
Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet:
sofia_glue.c:1083 No RTP ports available!
Restarting FS fixes the problem.
Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn’t happening until the last week or two.
Current version we’re running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z
Here’s a quick snip from the console when attempting to dial out to show the context of the error:
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102@192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/ 102@192.168.1.6:5060 Original read codec set to PCMU:0
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO
2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available!
2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102@192.168.1.6:5060 [BREAK]
2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/ 102@192.168.1.6:5060) Callstate Change RINGING -> HANGUP
2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Regards,
*******Phil Quesinberry*
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002
*****http://www.qsystemsengineering.com*<http://www.qsystemsengineering.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
Join Us At ClueCon - Aug 7-9, 2012
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
Tony, What can cause this error to happen really? Does FS allow re-use of RTP ports at all given if the media isn't going to the same destination, or doesn't the SDP header define some kind of session tracking mechnaism so ports can be reused? On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Jira Jira jira
On Jul 22, 2012 12:41 PM, "Phil Quesinberry" <philq@qsystemsengineering.com> wrote:
Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet:
sofia_glue.c:1083 No RTP ports available!
Restarting FS fixes the problem.
Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn’t happening until the last week or two.
Current version we’re running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z
Here’s a quick snip from the console when attempting to dial out to show the context of the error:
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102@192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102@192.168.1.6:5060 Original read codec set to PCMU:0
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO
2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available!
2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102@192.168.1.6:5060 [BREAK]
2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102@192.168.1.6:5060) Callstate Change RINGING -> HANGUP
2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Regards,
Phil Quesinberry
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002
http://www.qsystemsengineering.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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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
Sorry to but in, How many RTP ports have you got configured? If you only have a range of 5 and you have 5 concurrent calls, then no more RTP ports will be available. Doing a `netstat -anp | grep -i udp` should show what ports are open (on linux) On 23/07/2012, at 4:12 PM, curriegrad2004 wrote:
Tony,
What can cause this error to happen really? Does FS allow re-use of RTP ports at all given if the media isn't going to the same destination, or doesn't the SDP header define some kind of session tracking mechnaism so ports can be reused?
On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Jira Jira jira
On Jul 22, 2012 12:41 PM, "Phil Quesinberry" <philq@qsystemsengineering.com> wrote:
Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet:
sofia_glue.c:1083 No RTP ports available!
Restarting FS fixes the problem.
Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn’t happening until the last week or two.
Current version we’re running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z
Here’s a quick snip from the console when attempting to dial out to show the context of the error:
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102@192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102@192.168.1.6:5060 Original read codec set to PCMU:0
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO
2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available!
2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102@192.168.1.6:5060 [BREAK]
2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102@192.168.1.6:5060) Callstate Change RINGING -> HANGUP
2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Regards,
Phil Quesinberry
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002
http://www.qsystemsengineering.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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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Remember each call is 2 rtp ports. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 23, 2012 2:28 AM, "Chris Mylonas" <chris@opencsta.org> wrote:
Sorry to but in,
How many RTP ports have you got configured?
If you only have a range of 5 and you have 5 concurrent calls, then no more RTP ports will be available. Doing a `netstat -anp | grep -i udp` should show what ports are open (on linux)
On 23/07/2012, at 4:12 PM, curriegrad2004 wrote:
Tony,
What can cause this error to happen really? Does FS allow re-use of RTP ports at all given if the media isn't going to the same destination, or doesn't the SDP header define some kind of session tracking mechnaism so ports can be reused?
On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Jira Jira jira
On Jul 22, 2012 12:41 PM, "Phil Quesinberry" < philq@qsystemsengineering.com> wrote:
Recently through several of the latest git pulls, FS stops routing
calls
with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet:
sofia_glue.c:1083 No RTP ports available!
Restarting FS fixes the problem.
Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn’t happening until the last week or two.
Current version we’re running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z
Here’s a quick snip from the console when attempting to dial out to show the context of the error:
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102@192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102@192.168.1.6:5060 Original read codec set to PCMU:0
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO
2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available!
2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102@192.168.1.6:5060 [BREAK]
2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102@192.168.1.6:5060) Callstate Change RINGING -> HANGUP
2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Regards,
Phil Quesinberry
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002
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
Join Us At ClueCon - Aug 7-9, 2012
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
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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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I have recently experienced this exact behavior. There doesn't need to be any active calls. It does seem related to the fact that an outbound call was previously made through google voice. A restart also resolves the issue for me as well. If there is already a JIRA ticket for this, can someone post the link? I can try and help by providing more testing details. On 23/07/2012 9:17 AM, Brian Foster wrote:
Remember each call is 2 rtp ports.
Brian Foster Endigo Computer LLC
Sent from a mobile device.
On Jul 23, 2012 2:28 AM, "Chris Mylonas" <chris@opencsta.org <mailto:chris@opencsta.org>> wrote:
Sorry to but in,
How many RTP ports have you got configured?
If you only have a range of 5 and you have 5 concurrent calls, then no more RTP ports will be available. Doing a `netstat -anp | grep -i udp` should show what ports are open (on linux)
On 23/07/2012, at 4:12 PM, curriegrad2004 wrote:
> Tony, > > What can cause this error to happen really? Does FS allow re-use of > RTP ports at all given if the media isn't going to the same > destination, or doesn't the SDP header define some kind of session > tracking mechnaism so ports can be reused? > > On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale > <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote: >> Jira Jira jira >> >> On Jul 22, 2012 12:41 PM, "Phil Quesinberry" <philq@qsystemsengineering.com <mailto:philq@qsystemsengineering.com>> >> wrote: >>> >>> Recently through several of the latest git pulls, FS stops routing calls >>> with the following error, usually within a few minutes of making a call >>> through Google Voice, not sure if receiving a call through GV will cause the >>> problem yet: >>> >>> sofia_glue.c:1083 No RTP ports available! >>> >>> Restarting FS fixes the problem. >>> >>> Before I file the requisite Jira, is anyone aware of this problem or a >>> potential configuration issue that might not be compatible with the newer >>> builds of FS? This wasn't happening until the last week or two. >>> >>> Current version we're running is: FreeSWITCH Version >>> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z >>> >>> Here's a quick snip from the console when attempting to dial out to show >>> the context of the error: >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to >>> pass zrtp-hash between legs >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ >>> not set, so not propagating zrtp-hash >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare >>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec >>> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> PCMU/8000 20 ms 160 samples 64000 bits >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 >>> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> Original read codec set to PCMU:0 >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. >>> Disable 2833 dtmf and switch to INFO >>> >>> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports >>> available! >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal >>> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> [BREAK] >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 >>> (sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060>) Callstate Change RINGING -> HANGUP >>> >>> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup >>> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >>> >>> Regards, >>> >>> Phil Quesinberry >>> >>> Q Systems Engineering, Inc. >>> >>> Electronic Controls and Embedded Systems Development >>> >>> (410) 969-8002 <tel:%28410%29%20969-8002> >>> >>> http://www.qsystemsengineering.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 >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> 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 >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> 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 > > Join Us At ClueCon - Aug 7-9, 2012 > > 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
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commit 524468be7ba5ee0887d2e975e4f131171c601a54 Author: Anthony Minessale <anthm@freeswitch.org> Date: Mon Jul 23 11:36:19 2012 -0500 FS-4373 --resolve this is probably the same issue nobody would file a bug about on the mailing list even after I begged them to from last weekend please please stop filing bugs on the mailing list. I will still work on them and try to fix them but you are making me do way more work. Unless everyone wants to pay for support I cannot keep this up. On Mon, Jul 23, 2012 at 8:17 AM, Adam Kelloway <adam.kelloway@newpace.ca>wrote:
I have recently experienced this exact behavior. There doesn't need to be any active calls. It does seem related to the fact that an outbound call was previously made through google voice. A restart also resolves the issue for me as well.
If there is already a JIRA ticket for this, can someone post the link? I can try and help by providing more testing details.
On 23/07/2012 9:17 AM, Brian Foster wrote:
Remember each call is 2 rtp ports.
Brian Foster Endigo Computer LLC
Sent from a mobile device. On Jul 23, 2012 2:28 AM, "Chris Mylonas" <chris@opencsta.org> wrote:
Sorry to but in,
How many RTP ports have you got configured?
If you only have a range of 5 and you have 5 concurrent calls, then no more RTP ports will be available. Doing a `netstat -anp | grep -i udp` should show what ports are open (on linux)
On 23/07/2012, at 4:12 PM, curriegrad2004 wrote:
Tony,
What can cause this error to happen really? Does FS allow re-use of RTP ports at all given if the media isn't going to the same destination, or doesn't the SDP header define some kind of session tracking mechnaism so ports can be reused?
On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Jira Jira jira
On Jul 22, 2012 12:41 PM, "Phil Quesinberry" < philq@qsystemsengineering.com> wrote:
Recently through several of the latest git pulls, FS stops routing
calls
with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet:
sofia_glue.c:1083 No RTP ports available!
Restarting FS fixes the problem.
Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn’t happening until the last week or two.
Current version we’re running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z
Here’s a quick snip from the console when attempting to dial out to show the context of the error:
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102@192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102@192.168.1.6:5060 Original read codec set to PCMU:0
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO
2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available!
2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102@192.168.1.6:5060 [BREAK]
2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102@192.168.1.6:5060) Callstate Change RINGING -> HANGUP
2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Regards,
Phil Quesinberry
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002 <%28410%29%20969-8002>
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
Join Us At ClueCon - Aug 7-9, 2012
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
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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________
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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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
-- Adam -- [image: NewPace Logo]
Adam Kelloway Software Engineer, NewPace phone +1 (902) 406–8375 x1031 email Adam.Kelloway@NewPace.com <Adam.Kelloway@newpace.com> aim/msn Adam.Kelloway@NewPace.ca
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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
Hey Tony, Thanks for fixing this... However, Everyone else, Open the freaking Jira, it doesn¹t take that long... Or take the 2 minutes to search jira to see if theres an existing ticket for your problem, and if you have more information add it... Why? It saves Developers like Tony time in tracking down the issue and getting you a fix... As much as I hate to say it, there could come a time where No Jira ticket == Its not really a problem... You think tony hasn¹t spent a crazy amount of time on FreeSWITCH? Lets do a little math... 50 weeks a year (he¹s supposed to get 2 weeks of holiday a year right?) and I know he works at least 40 hours a week on FreeSWITCH so that¹s 2000 hours a Year, then factor in that he has been working on it since atleast 2006... So that¹s 12000 he has spent working on FreeSWITCH. And that¹s at the low end of the estimate... Ohloh.net estimates something like $40million dollars worth of developer time has gone into FreeSWITCH ( http://www.ohloh.net/p/freeswitch/estimated_cost ) most of which is the developers doing their part for the love of the project (ie: not getting paid a dime)... So come on people... Just a little help... That¹s all we are asking for here.... On 7/23/12 11:37 AM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
commit 524468be7ba5ee0887d2e975e4f131171c601a54 Author: Anthony Minessale <anthm@freeswitch.org> Date: Mon Jul 23 11:36:19 2012 -0500
FS-4373 --resolve this is probably the same issue nobody would file a bug about on the mailing list even after I begged them to from last weekend
please please stop filing bugs on the mailing list. I will still work on them and try to fix them but you are making me do way more work. Unless everyone wants to pay for support I cannot keep this up.
On Mon, Jul 23, 2012 at 8:17 AM, Adam Kelloway <adam.kelloway@newpace.ca> wrote:
I have recently experienced this exact behavior. There doesn't need to be any active calls. It does seem related to the fact that an outbound call was previously made through google voice. A restart also resolves the issue for me as well.
If there is already a JIRA ticket for this, can someone post the link? I can try and help by providing more testing details.
On 23/07/2012 9:17 AM, Brian Foster wrote:
Remember each call is 2 rtp ports.
Brian Foster Endigo Computer LLC
Sent from a mobile device.
On Jul 23, 2012 2:28 AM, "Chris Mylonas" <chris@opencsta.org> wrote:
Sorry to but in,
How many RTP ports have you got configured?
If you only have a range of 5 and you have 5 concurrent calls, then no more RTP ports will be available. Doing a `netstat -anp | grep -i udp` should show what ports are open (on linux)
On 23/07/2012, at 4:12 PM, curriegrad2004 wrote:
Tony,
What can cause this error to happen really? Does FS allow re-use of RTP ports at all given if the media isn't going to the same destination, or doesn't the SDP header define some kind of session tracking mechnaism so ports can be reused?
On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote: >> Jira Jira jira >> >> On Jul 22, 2012 12:41 PM, "Phil Quesinberry" <philq@qsystemsengineering.com> >> wrote: > >>> > >>> Recently through several of the latest git pulls, FS stops routing > calls > >>> with the following error, usually within a few minutes of making a
call
> >>> through Google Voice, not sure if receiving a call through GV will > cause the > >>> problem yet: > >>> > >>> sofia_glue.c:1083 No RTP ports available! > >>> > >>> Restarting FS fixes the problem. > >>> > >>> Before I file the requisite Jira, is anyone aware of this problem or a > >>> potential configuration issue that might not be compatible with the > newer > >>> builds of FS? This wasn¹t happening until the last week or two. > >>> > >>> Current version we¹re running is: FreeSWITCH Version > >>> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z > >>> > >>> Here¹s a quick snip from the console when attempting to dial out to show > >>> the context of the error: > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for > zrtp-hash > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding > whether to > >>> pass zrtp-hash between legs > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 > CF_ZRTP_PASSTHRU_REQ > >>> not set, so not propagating zrtp-hash > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec > Compare > >>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec > >>> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> > PCMU/8000 20 ms 160 samples 64000 bits > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 > >>> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> > Original read codec set to PCMU:0 > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. > >>> Disable 2833 dtmf and switch to INFO > >>> > >>> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports > >>> available! > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send > signal > >>> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> > [BREAK] > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 > >>> (sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> > ) Callstate Change RINGING -> HANGUP > >>> > >>> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup > >>> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> > [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > >>> > >>> Regards, > >>> > >>> Phil Quesinberry > >>> > >>> Q Systems Engineering, Inc. > >>> > >>> Electronic Controls and Embedded Systems Development > >>> > >>> (410) 969-8002 <tel:%28410%29%20969-8002> > >>> > >>> http://www.qsystemsengineering.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 > >>> > >>> Join Us At ClueCon - Aug 7-9, 2012 > >>> > >>> 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 > >>> >> >> _________________________________________________________________________ >> 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 >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> 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 >>
_________________________________________________________________________
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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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
So do I, since the last rebuild from git a few days ago. In autoload_configs/switch.conf.xml, I have: <!-- RTP port range --> <param name="rtp-start-port" value="10000"/> <param name="rtp-end-port" value="20000"/> and iptables:-A INPUT -i eth0 -p udp -m udp --dport 10000:20000 -j ACCEPT The 'netstat' command returns about 15 open UDP ports. --- On Mon, 7/23/12, Adam Kelloway <adam.kelloway@newpace.ca> wrote: From: Adam Kelloway <adam.kelloway@newpace.ca> Subject: Re: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Date: Monday, July 23, 2012, 6:17 AM I have recently experienced this exact behavior. There doesn't need to be any active calls. It does seem related to the fact that an outbound call was previously made through google voice. A restart also resolves the issue for me as well. If there is already a JIRA ticket for this, can someone post the link? I can try and help by providing more testing details.
Atleast 2 RTP ports... Possibly more... Also keep in mind that each RTP port requires a handle... So if you have things set to tight you¹ll run into issues... Check the ulimits On 7/23/12 7:17 AM, "Brian Foster" <bdfoster@endigotech.com> wrote:
Remember each call is 2 rtp ports.
Brian Foster Endigo Computer LLC
Sent from a mobile device.
On Jul 23, 2012 2:28 AM, "Chris Mylonas" <chris@opencsta.org> wrote:
Sorry to but in,
How many RTP ports have you got configured?
If you only have a range of 5 and you have 5 concurrent calls, then no more RTP ports will be available. Doing a `netstat -anp | grep -i udp` should show what ports are open (on linux)
On 23/07/2012, at 4:12 PM, curriegrad2004 wrote:
Tony,
What can cause this error to happen really? Does FS allow re-use of RTP ports at all given if the media isn't going to the same destination, or doesn't the SDP header define some kind of session tracking mechnaism so ports can be reused?
On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Jira Jira jira
On Jul 22, 2012 12:41 PM, "Phil Quesinberry" <philq@qsystemsengineering.com> wrote: >> >> Recently through several of the latest git pulls, FS stops routing calls >> with the following error, usually within a few minutes of making a call >> through Google Voice, not sure if receiving a call through GV will cause the >> problem yet: >> >> sofia_glue.c:1083 No RTP ports available! >> >> Restarting FS fixes the problem. >> >> Before I file the requisite Jira, is anyone aware of this problem or a >> potential configuration issue that might not be compatible with the newer >> builds of FS? This wasn¹t happening until the last week or two. >> >> Current version we¹re running is: FreeSWITCH Version >> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z >> >> Here¹s a quick snip from the console when attempting to dial out to show >> the context of the error: >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to >> pass zrtp-hash between legs >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ >> not set, so not propagating zrtp-hash >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec >> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> PCMU/8000 20 ms 160 samples 64000 bits >> >> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 >> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> Original read codec set to PCMU:0 >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. >> Disable 2833 dtmf and switch to INFO >> >> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports >> available! >> >> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal >> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> [BREAK] >> >> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 >> (sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> ) Callstate Change RINGING -> HANGUP >> >> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup >> sofia/internal/102@192.168.1.6:5060 <http://102@192.168.1.6:5060> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >> >> Regards, >> >> Phil Quesinberry >> >> Q Systems Engineering, Inc. >> >> Electronic Controls and Embedded Systems Development >> >> (410) 969-8002 <tel:%28410%29%20969-8002> >> >> http://www.qsystemsengineering.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 >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> 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 >>
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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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It's just an internal allocation within FS, so ports can be reused as soon as it's not allocated anymore. Lets say you have 5 ports, first call will get port 1, then 2 etc. When the first call hangs up, port 1 will be returned, and may be reused again. /Peter 23 jul 2012 kl. 08:20 skrev "curriegrad2004" <curriegrad2004@gmail.com>:
Tony,
What can cause this error to happen really? Does FS allow re-use of RTP ports at all given if the media isn't going to the same destination, or doesn't the SDP header define some kind of session tracking mechnaism so ports can be reused?
On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Jira Jira jira
On Jul 22, 2012 12:41 PM, "Phil Quesinberry" <philq@qsystemsengineering.com> wrote:
Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet:
sofia_glue.c:1083 No RTP ports available!
Restarting FS fixes the problem.
Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn’t happening until the last week or two.
Current version we’re running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z
Here’s a quick snip from the console when attempting to dial out to show the context of the error:
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102@192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits
2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102@192.168.1.6:5060 Original read codec set to PCMU:0
2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO
2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available!
2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102@192.168.1.6:5060 [BREAK]
2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102@192.168.1.6:5060) Callstate Change RINGING -> HANGUP
2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
Regards,
Phil Quesinberry
Q Systems Engineering, Inc.
Electronic Controls and Embedded Systems Development
(410) 969-8002
http://www.qsystemsengineering.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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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
_________________________________________________________________________ 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
Join Us At ClueCon - Aug 7-9, 2012
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
!DSPAM:500ce9de32761973692718!
This means that the internal port allocator in FS doesn't have any RTP ports available. Either because you have not enough ports configured (less then you use), or if there is a bug somewhere not returning ports back to the system. You should report to Jira, and attach the needed logs in there. /Peter ________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Phil Quesinberry [philq@qsystemsengineering.com] Skickat: den 22 juli 2012 19:16 Till: freeswitch-users@lists.freeswitch.org Ämne: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet: sofia_glue.c:1083 No RTP ports available! Restarting FS fixes the problem. Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn’t happening until the last week or two. Current version we’re running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z Here’s a quick snip from the console when attempting to dial out to show the context of the error: 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102@192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102@192.168.1.6:5060 Original read codec set to PCMU:0 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available! 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102@192.168.1.6:5060 [BREAK] 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102@192.168.1.6:5060) Callstate Change RINGING -> HANGUP 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com !DSPAM:500c37ce32761523320785!
participants (9)
-
Adam Kelloway
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Anthony Minessale
-
Brian Foster
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Chris Mylonas
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curriegrad2004
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Ken Rice
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mtle
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Peter Olsson
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Phil Quesinberry