[Freeswitch-users] Bypass Media True Disables MOH
When I uncomment the following tag, internally held calls no longer hear MOH. param name=inbound-bypass-media value=true/ Is there a way to have the above uncommented and still provide MOH to held calls? Best Regards, Jerry ___ 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
Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail
I upgraded to wanpipe-3.5.8.7.tgz and Freeswitch version 1.0.5pre9 and the bug is still present. Would libpri possibly help? I'm currently using the native wanpipe PRI stack and default openzap configs in Freeswitch. Best Regards, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Monday, December 28, 2009 3:31 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed toVoice Mail you have to update the sangoma driver and probably FreeSWITCH for good measure. Its a known bug in the sangoma driver that has been fixed it the latest release. On Mon, Dec 28, 2009 at 5:19 PM, Jerry Richards jerry.richa...@teotech.com wrote: Hello All, I posted a FS log into the Pastebin at http://pastebin.freeswitch.org/11644. I am still having the problem where a PSTN-to-Internal call via a Sangoma A101D card stops ringing the internal phone after about 10 seconds. It should be ringing for 30 seconds and then go to Voice Mail (as an Internal-to-Internal call does). Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Tuesday, December 22, 2009 8:02 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: PSTN-to-Internal Call Does Not Get Routed to Voice Mail I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry ___ 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:+19193869900 ___ 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
Re: [Freeswitch-users] Adding H263 Video to Existing CallFailsFirst Time
Okay. I uncommented the following lines and the video start works as correctly: param name=media-option value=bypass-media-after-att-xfer/ param name=inbound-bypass-media value=true/ Thanks, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Tuesday, December 22, 2009 8:33 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing CallFailsFirst Time No. The following lines is commented out (internal.xml): !--param name=media-option value=bypass-media-after-att-xfer/-- !--param name=inbound-bypass-media value=true/-- Thanks, Jerry -Original Message- From: Peter P GMX [mailto:prometheus...@gmx.net] Sent: Tuesday, December 22, 2009 3:21 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing Call FailsFirst Time Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: After establishing an audio call between two Bria softphones, and then starting video at the caller phone, FS replies to the re-INVITE with a 200 OK with only the PCMU codec. This looks incorrect. The audio call previously negotiated to the speex/16000 codec, and the re-INVITE from the caller added the H263-1998 codec. If I re-attempt to start video at the caller, then it is successful. I put a Freeswitch log 11596 into the pastebin that contains the complete scenario: establishing audio call, first failed start video attempt, and second successful start video attempt. Best Regards, Jerry ___ 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-use rs http://www.freeswitch.org ___ 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
[Freeswitch-users] Presence Change Distribution
Is there a setting to control how fast FS distributes presence changes to subscribers? Currently, it appears to take several minutes before I see presence changes. I would like to see them almost instantaneously, if possible. Thanks and Best Regards, Jerry ___ 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
[Freeswitch-users] PSTN-to-Internal Call Does Not Get Routed to Voice Mail
I have a Freeswitch PBX server with an installed Sangoma A101D card connected to a PRI. Most everything works okay, however when I get an inbound call from the PSTN, if the call is not answered within about 12 seconds, the call ends (so it doesn't go to voice mail). If I make a call from one internal phone to another, then it will go to voice mail after 30 seconds. How can I get the external call to route to voice mail after 30 seconds? I put a new 11595 log into the pastebin. Do you know any Freeswitch setting that might cause this? If this issue has been addressed before, what string should I use to search for it, because I can't find it. Thanks, Jerry ___ 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
Re: [Freeswitch-users] Adding H263 Video to Existing Call FailsFirst Time
No. The following lines is commented out (internal.xml): !--param name=media-option value=bypass-media-after-att-xfer/-- !--param name=inbound-bypass-media value=true/-- Thanks, Jerry -Original Message- From: Peter P GMX [mailto:prometheus...@gmx.net] Sent: Tuesday, December 22, 2009 3:21 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding H263 Video to Existing Call FailsFirst Time Just a question, do you use Freeswitch in bypass-media-mode in this scenario? Then media negociation should be handled outside Freeswitch. Best regards Peter Jerry Richards schrieb: After establishing an audio call between two Bria softphones, and then starting video at the caller phone, FS replies to the re-INVITE with a 200 OK with only the PCMU codec. This looks incorrect. The audio call previously negotiated to the speex/16000 codec, and the re-INVITE from the caller added the H263-1998 codec. If I re-attempt to start video at the caller, then it is successful. I put a Freeswitch log 11596 into the pastebin that contains the complete scenario: establishing audio call, first failed start video attempt, and second successful start video attempt. Best Regards, Jerry ___ 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-use rs http://www.freeswitch.org ___ 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
[Freeswitch-users] Allow/Deny REGISTER Request Based on User-Agent Header
Is it possible to allow/deny REGISTER requests based on the User-Agent header? I need to know/manage what devices are registering. Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9
I found the issue with this. I did an svn checkout from the trunk, and then
I did a local svn export to another local folder. For some reason, the svn
export did not include the libs/openzap folder (which was not the case when
I got 1.0.5pre8). Must I do a separate svn export from the libs/openzap
folder?
Best Regards,
Jerry
-Original Message-
From: Brian West [mailto:br...@freeswitch.org]
Sent: Wednesday, December 16, 2009 2:28 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9
Need siptrace with this type sofia profile siptrace on replace
with your profile.
/b
On Dec 16, 2009, at 4:23 PM, Jerry Richards wrote:
I upgraded to the latest 1.0.5pre9 and now if I try to call from an
internal phone to an external number on my Sangoma PRI, I get a 502 Bad
Gateway
reply. Below is the console loglevel 7 output. It says the
destination is out-of-order. I'm not sure what this means. Any help is
appreciated.
2009-12-16 14:10:46.410656 [DEBUG] sofia.c:5285 0 acls to check for
proxy
2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5303 network ip is a proxy
[0]
2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5331 IP 192.168.72.32
Rejected by acl domains. Falling back to Digest auth.
2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5285 0 acls to check for
proxy
2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5303 network ip is a proxy
[0]
2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5331 IP 192.168.72.32
Rejected by acl domains. Falling back to Digest auth.
2009-12-16 14:10:46.457607 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/5...@192.168.72.141:5060
[e58e763f-7688-4600-aa70-481bbc359f58]
2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3787 Channel
sofia/internal/5...@192.168.72.141:5060 entering state [received][100]
2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3798 Remote SDP:
v=0
o=TC 1100638826 1100638826 IN IP4 192.168.72.32 s=session c=IN IP4
192.168.72.32 t=0 0 m=audio 1760 RTP/AVP 0 18 4 101 a=rtpmap:0
PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000/1
a=ptime:20
a=ptime:20
2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3923
(sofia/internal/5...@192.168.72.141:5060) State Change CS_NEW -
CS_INIT
2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/5...@192.168.72.141:5060 [BREAK]
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/5...@192.168.72.141:5060) Running State Change CS_INIT
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/5...@192.168.72.141:5060) State INIT
2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:83
sofia/internal/5...@192.168.72.141:5060 SOFIA INIT
2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:111
(sofia/internal/5...@192.168.72.141:5060) State Change CS_INIT -
CS_ROUTING
2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send
signal sofia/internal/5...@192.168.72.141:5060 [BREAK]
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/5...@192.168.72.141:5060) State INIT going to sleep
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/5...@192.168.72.141:5060) Running State Change
CS_ROUTING
2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/5...@192.168.72.141:5060) State ROUTING
2009-12-16 14:10:46.458582 [DEBUG] mod_sofia.c:132
sofia/internal/5...@192.168.72.141:5060 SOFIA ROUTING
2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:78
sofia/internal/5...@192.168.72.141:5060 Standard ROUTING
2009-12-16 14:10:46.458582 [INFO] mod_dialplan_xml.c:408 Processing
Anonymous-93491028 in context default
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-unloop] continue=false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (PASS)
[unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (FAIL)
[unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-tod_example] continue=true
Dialplan: day of week[4] =~ 2-6 (PASS)
Dialplan: hour[14] =~ 9-18 (PASS)
Dialplan: sofia/internal/5...@192.168.72.141:5060 Date/Time Match
(PASS) [tod_example] break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Action
set(open=true)
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-holiday_example] continue=true
Dialplan: month[12] =~ 1 (FAIL)
Dialplan: sofia/internal/5...@192.168.72.141:5060 Date/Time Match
(FAIL) [holiday_example] break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-Mediant1000] continue=false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (FAIL)
[Mediant1000]
destination_number(93491028) =~ /^8(\d+)$/ break=on-false
Dialplan: sofia/internal/5
[Freeswitch-users] Getting 502 Bad Gateway with 1.0.5.pre9
I upgraded to the latest 1.0.5pre9 and now if I try to call from an internal
phone to an external number on my Sangoma PRI, I get a 502 Bad Gateway
reply. Below is the console loglevel 7 output. It says the destination is
out-of-order. I'm not sure what this means. Any help is appreciated.
2009-12-16 14:10:46.410656 [DEBUG] sofia.c:5285 0 acls to check for proxy
2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5303 network ip is a proxy [0]
2009-12-16 14:10:46.411629 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by
acl domains. Falling back to Digest auth.
2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5285 0 acls to check for proxy
2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5303 network ip is a proxy [0]
2009-12-16 14:10:46.452626 [DEBUG] sofia.c:5331 IP 192.168.72.32 Rejected by
acl domains. Falling back to Digest auth.
2009-12-16 14:10:46.457607 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/5...@192.168.72.141:5060
[e58e763f-7688-4600-aa70-481bbc359f58]
2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3787 Channel
sofia/internal/5...@192.168.72.141:5060 entering state [received][100]
2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3798 Remote SDP:
v=0
o=TC 1100638826 1100638826 IN IP4 192.168.72.32
s=session
c=IN IP4 192.168.72.32
t=0 0
m=audio 1760 RTP/AVP 0 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000/1
a=ptime:20
a=ptime:20
2009-12-16 14:10:46.457607 [DEBUG] sofia.c:3923
(sofia/internal/5...@192.168.72.141:5060) State Change CS_NEW - CS_INIT
2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal
sofia/internal/5...@192.168.72.141:5060 [BREAK]
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/5...@192.168.72.141:5060) Running State Change CS_INIT
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/5...@192.168.72.141:5060) State INIT
2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:83
sofia/internal/5...@192.168.72.141:5060 SOFIA INIT
2009-12-16 14:10:46.457607 [DEBUG] mod_sofia.c:111
(sofia/internal/5...@192.168.72.141:5060) State Change CS_INIT - CS_ROUTING
2009-12-16 14:10:46.457607 [DEBUG] switch_core_session.c:1018 Send signal
sofia/internal/5...@192.168.72.141:5060 [BREAK]
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/5...@192.168.72.141:5060) State INIT going to sleep
2009-12-16 14:10:46.457607 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/5...@192.168.72.141:5060) Running State Change CS_ROUTING
2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/5...@192.168.72.141:5060) State ROUTING
2009-12-16 14:10:46.458582 [DEBUG] mod_sofia.c:132
sofia/internal/5...@192.168.72.141:5060 SOFIA ROUTING
2009-12-16 14:10:46.458582 [DEBUG] switch_core_state_machine.c:78
sofia/internal/5...@192.168.72.141:5060 Standard ROUTING
2009-12-16 14:10:46.458582 [INFO] mod_dialplan_xml.c:408 Processing
Anonymous-93491028 in context default
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing [default-unloop]
continue=false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-tod_example] continue=true
Dialplan: day of week[4] =~ 2-6 (PASS)
Dialplan: hour[14] =~ 9-18 (PASS)
Dialplan: sofia/internal/5...@192.168.72.141:5060 Date/Time Match (PASS)
[tod_example] break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Action set(open=true)
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-holiday_example] continue=true
Dialplan: month[12] =~ 1 (FAIL)
Dialplan: sofia/internal/5...@192.168.72.141:5060 Date/Time Match (FAIL)
[holiday_example] break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-Mediant1000] continue=false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (FAIL) [Mediant1000]
destination_number(93491028) =~ /^8(\d+)$/ break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 parsing
[default-SangomaPRI] continue=false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Regex (PASS) [SangomaPRI]
destination_number(93491028) =~ /^9(\d+)$/ break=on-false
Dialplan: sofia/internal/5...@192.168.72.141:5060 Action
set(effective_caller_id_number=425740${caller_id_number})
Dialplan: sofia/internal/5...@192.168.72.141:5060 Action
bridge(openzap/smg_prid/a/3491...@g1)
2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:122
(sofia/internal/5...@192.168.72.141:5060) State Change CS_ROUTING -
CS_EXECUTE
2009-12-16 14:10:46.459538 [DEBUG] switch_core_session.c:1018 Send signal
sofia/internal/5...@192.168.72.141:5060 [BREAK]
2009-12-16 14:10:46.459538 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/5...@192.168.72.141:5060) State ROUTING going to sleep
2009-12-16
[Freeswitch-users] One-way Video
I am trying to bring up a video call, but not having much luck. We are only getting one-way video (i.e. the caller sees far-end video, but the callee does not). I added the H263/H264 tags to the pre-process global_codec_prefs and outbound_codec_prefs tags in vars.xml. Anyone have hints on making two-way video to work? Best Regards, Jerry ___ 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
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org http://pastebin.freeswitch.org/ On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ 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 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net/ #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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 ___ 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
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
Here is the Pastebin Link: http://pastebin.freeswitch.org/11432 Thanks, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Tuesday, December 08, 2009 12:35 PM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org http://pastebin.freeswitch.org/ On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ 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 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net/ #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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 ___ 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
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org http://pastebin.freeswitch.org/ On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ 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 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net/ #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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 ___ 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
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org http://pastebin.freeswitch.org/ On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ 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 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net/ #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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 ___ 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
[Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP
Hello, I just pasted a log in the Pastebin with Freeswitch logging enabled. Does anyone know a way to prevent FS from connecting the call prior to the callee answering? Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Thursday, November 05, 2009 3:50 PM To: 'freeswitch-users@lists.freeswitch.org' Subject: Want 183 w/SDP, but Get 200 w/SDP I am trying to make a call through a Gateway that sends the INVITE with no SDP and ONLY wants the 200 OK w/SDP when the callee answers. For some reason, Freeswitch answers the call with 200 OK w/SDP even before the callee answers the phone. Is this to provide ringback? Can I disable that action? Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Accessing Config Info From Database
MC, We would like the dialplan to route the call based on Presence, which is a database lookup. I should be able to do this in Lua, true? Jerry _ From: Michael Collins [mailto:m...@freeswitch.org] Sent: Monday, November 16, 2009 11:33 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Accessing Config Info From Database On Mon, Nov 16, 2009 at 9:36 AM, Jerry Richards jerry.richa...@teotech.com wrote: I have a bit of confusion about Lua scripting. When a script is invoked, should it always return an XML string that is used by FS? Or as in the case of dialplan examples, does it actually execute the dialplan (e.g. session:answer();)? Best Regards, Jerry Jerry, A Lua script that is explicitly called from the dialplan will indeed execute dialplan-ish stuff. For example, let's say you had this in conf/dialplan/default.xml: extension name=lua sample condition field=destination_number expression=9876 action application=lua data=/path/to/myluascript.lua/ /condition /extension Then myluascript.lua has something like: --Sample Lua script session:answer() session:sleep(1000) session:streamFile(/path/to/file.wav) session:hangup() Assuming an otherwise default install, the above Lua script would execute when a caller dialed 9876, or if a call was x-ferred to 9876. However, if you're wanting to use Lua to serve up a dialplan then it's totally different. Lua is not called from the dialplan; Lua provides the dialplan to FreeSWITCH. This latter case is the scenario discussed in the wiki section you referenced. (http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration) Are you trying to use Lua scripting for serving up a dynamic configuration of some sort? -MC ___ 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
Re: [Freeswitch-users] Accessing Config Info From Database
I have a bit of confusion about Lua scripting. When a script is invoked, should it always return an XML string that is used by FS? Or as in the case of dialplan examples, does it actually execute the dialplan (e.g. session:answer();)? Best Regards, Jerry -Original Message- From: Leon de Rooij [mailto:l...@scarlet-internet.nl] Sent: Friday, November 13, 2009 2:29 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Accessing Config Info From Database Hi, You can use mod_xml_curl (generate xml on a webserver): http://wiki.freeswitch.org/wiki/Mod_xml_curl or mod_xml_odbc (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Mod_xml_odbc or LUA together with luasql (generate xml in freeswitch): http://wiki.freeswitch.org/wiki/Lua#For_serving_configuration regards, Leon On Fri, 2009-11-13 at 13:59 -0800, Jerry Richards wrote: Is there a way to access configuration information from a database (e.g. SQL) rather than from the XML files? Best Regards, Jerry ___ 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-use rs http://www.freeswitch.org ___ 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
[Freeswitch-users] How To Disable MD5 Authentication?
How can I disable MD5 Authentication upon registration? Best Regards, Jerry ___ 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
[Freeswitch-users] Accessing Config Info From Database
Is there a way to access configuration information from a database (e.g. SQL) rather than from the XML files? Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Bug in Freeswitch/scripts/gentls_cert.in build file?
Here is what is believed to be a bug found by Robert Hadley found in
Freeswitch1.0.4/scripts/gentls_cert.in build file:
Fix for gentls_cert remove to work:
[scripts]# diff gentls_cert.in gentls_cert.in~
129c129
if [ -d ${CONFDIR}/CA ]; then
---
if [ ! -d ${CONFDIR}/CA ]; then
Best Regards,
Jerry
___
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[Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP
I am trying to make a call through a Gateway that sends the INVITE with no SDP and ONLY wants the 200 OK w/SDP when the callee answers. For some reason, Freeswitch answers the call with 200 OK w/SDP even before the callee answers the phone. Is this to provide ringback? Can I disable that action? Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Dial Plan Question
Okay. Say we want 1000 internal user extensions and want them to be configured with individual dial plans that route the call based on the extension's callgroup, time-of-day, and presence. Would be okay to create a static XML dialplan file for each extension, so calls to/from each extension would be routed uniquely based upon these parameters? This approach sounds straightforward to us. Best Regards, Jerry -Original Message- From: Shelby Ramsey [mailto:sicfsl...@gmail.com] Sent: Tuesday, November 03, 2009 1:45 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Dial Plan Question I think the real question is what are you trying to do ... for some things it's very easy to just whip up a static XML file and be done with it. For others you probably want some sort of interaction with a DB. The options here are pretty endless: -- XML curl -- handing off the call to a script call from a static dial plan (use lua if there is going to be any load) -- event_socket -- mod_lcr But ultimately I think it's what you're trying to accomplish that matters. For a PBX install I'd say static files is probably about as easy as it is going to get. For delivering a service you'd probably want interaction with a DB. I've use XML curl a lot and have even starting using direct DB queries from static dialplans using mod_memcache and memcachedb (not memcache ... persistent storage). SDR Jerry Richards wrote: My understanding of DialPlan/CallRouting is that it can be accomplished via static XML tags, or alternatively, via a DialPlan Application that interfaces with the dptools module. Question: If my above assumption is true, how does one select one approach over the other? What is the criteria/considerations that would govern the decision? Best Regards, Jerry ___ 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-use rs http://www.freeswitch.org ___ 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
[Freeswitch-users] Error checking for PMP [general error]
When I start Freeswitch, I see an Error checking for PMP [general error] as shown below. Does anyone know what could cause this? [r...@teoproxy bin]# ./freeswitch Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch -waste. auto-adjusting stack size for optimal performance 2009-11-02 10:12:27.17579 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-02 10:12:27.18373 [DEBUG] switch_event.c:553 Create event dispatch thread 0 2009-11-02 10:12:27.428749 [INFO] switch_nat.c:392 Scanning for NAT 2009-11-02 10:12:27.428885 [DEBUG] switch_nat.c:152 Checking for PMP 1/5 2009-11-02 10:12:27.678480 [DEBUG] switch_nat.c:152 Checking for PMP 2/5 2009-11-02 10:12:27.679449 [DEBUG] switch_nat.c:152 Checking for PMP 3/5 2009-11-02 10:12:28.179388 [DEBUG] switch_nat.c:152 Checking for PMP 4/5 2009-11-02 10:12:29.179217 [DEBUG] switch_nat.c:152 Checking for PMP 5/5 2009-11-02 10:12:31.178879 [ERR] switch_nat.c:183 Error checking for PMP [general error] 2009-11-02 10:12:31.178902 [DEBUG] switch_nat.c:397 Checking for UPnP 2009-11-02 10:12:43.176881 [INFO] switch_nat.c:411 No PMP or UPnP NAT detected! 2009-11-02 10:12:43.210145 [INFO] switch_core_sqldb.c:538 Opening DB 2009-11-02 10:12:43.919804 [NOTICE] switch_scheduler.c:166 Starting task thread 2009-11-02 10:12:43.937881 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1257185563 2009-11-02 10:12:43.937980 [CONSOLE] switch_core.c:1449 Bringing up environment. 2009-11-02 10:12:43.937994 [CONSOLE] switch_core.c:1450 Loading Modules. 2009-11-02 10:12:43.938319 [INFO] switch_time.c:661 Timezone loaded 530 definitions 2009-11-02 10:12:43.938336 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_SOFTTIMER_MODULE] 2009-11-02 10:12:43.938351 [NOTICE] switch_loadable_module.c:228 Adding Timer 'soft' 2009-11-02 10:12:43.938413 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_PCM_MODULE] Best Regards, Jerry ___ 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
[Freeswitch-users] Dial Plan Question
My understanding of DialPlan/CallRouting is that it can be accomplished via static XML tags, or alternatively, via a DialPlan Application that interfaces with the dptools module. Question: If my above assumption is true, how does one select one approach over the other? What is the criteria/considerations that would govern the decision? Best Regards, Jerry ___ 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
[Freeswitch-users] WARNING On Inbound Call Question
I have my Freeswitch server with an installed Sangoma A101D card. Most
everything works okay, however, when I get an inbound call from the PSTN, I
see the following warning show up in the log. Additionally, the caller (on
the PSTN) does not hear ringback, and if the call is not answered within
about 12 seconds, the call ends (so it doesn't go to voice mail). If I make
a call from one internal phone to another, then it will go to voice mail
after 30 seconds.
Here are the two warnings:
[WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1]
Rc=0 CSid=0 Seq=11
[WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to
PROGRESS_MEDIA
Here is the log of the warning upon an inbound call:
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com 2009-11-02 09:06:01.664835
[WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0
Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176]
2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on
1:1 from DOWN to RING
2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig
[START]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound
channel OpenZAP/1:1/5384
2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel
OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384)
State Change CS_NEW - CS_INIT
2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal
OpenZAP/1:1/5384 [BREAK]
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398
(OpenZAP/1:1/5384) Running State Change CS_INIT
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481
(OpenZAP/1:1/5384) State INIT
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384)
State Change CS_INIT - CS_ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal
OpenZAP/1:1/5384 [BREAK]
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481
(OpenZAP/1:1/5384) State INIT going to sleep
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398
(OpenZAP/1:1/5384) Running State Change CS_ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484
(OpenZAP/1:1/5384) State ROUTING
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384
CHANNEL ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78
OpenZAP/1:1/5384 Standard ROUTING
2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing
4253813176-5384 in context default
Dialplan: OpenZAP/1:1/5384 parsing [default-unloop] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~
/^true$/ break=on-false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~
/^true$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-tod_example] continue=true
Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example]
Dialplan: OpenZAP/1:1/5384 Action set(open=true)
Dialplan: OpenZAP/1:1/5384 parsing [default-SangomaPRI] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI]
destination_number(5384) =~ /^9(\d+)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-global-intercept]
continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept]
destination_number(5384) =~ /^(5380)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-group-intercept] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept]
destination_number(5384) =~ /^\*8$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-intercept-ext] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext]
destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-redial] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384) =~
/^870$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-global] continue=true
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~
/^true$/ break=never
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: OpenZAP/1:1/5384 Absolute Condition [global]
Dialplan: OpenZAP/1:1/5384 Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: OpenZAP/1:1/5384 Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe
r})
Dialplan: OpenZAP/1:1/5384 Action
hash(insert/${domain_name}-last_dial/global/${uuid})
[Freeswitch-users] IVR Intro Clipped
I notice that when I call IVR from the PSTN, the Welcome to Freeswitch... introduction is clipped at the beginning, so it sounds like come to Freeswitch. If I call 5000 internally, then I always hear the full introduction. What can I do to resolve this? My XML config looks like: extension name=ivr_demo condition field=destination_number expression=5000 action application=answer/ action application=start_dtmf/ action application=ivr data=demo_ivr/ /condition /extension Best Regards, Jerry ___ 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
Re: [Freeswitch-users] IVR Intro Clipped
I modified my dialplan as shown, but the clipping persists. Should the sleep be placed somewhere else? extension name=ivr_demo condition field=destination_number expression=5000 action application=sleep data=1000\ action application=answer/ action application=start_dtmf/ action application=ivr data=demo_ivr/ /condition /extension Best Regards, Jerry -Original Message- From: Brian West [mailto:br...@freeswitch.org] Sent: Wednesday, October 28, 2009 1:51 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] IVR Intro Clipped Sleep 1000 ms... we usually bring up media too fast before the other end is ready. /b On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote: I notice that when I call IVR from the PSTN, the Welcome to Freeswitch... introduction is clipped at the beginning, so it sounds like come to Freeswitch. If I call 5000 internally, then I always hear the full introduction. What can I do to resolve this? My XML config looks like: extension name=ivr_demo condition field=destination_number expression=5000 action application=answer/ action application=start_dtmf/ action application=ivr data=demo_ivr/ /condition /extension Best Regards, Jerry ___ 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 ___ 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
[Freeswitch-users] FS Training
Did the voting booth close? I was unable to vote. I'm not sure what link to click and I have had some strange issues with my FS account today. I would be interested in paid training. Do you have plans for offering a training session at your locale? Or would you travel onsite to provide training? Best Regards, Jerry ___ 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
[Freeswitch-users] Inbound DTMF Not Recognized By IVR
I installed FS on a machine with a Sangoma A101D (PRI) card and if I make an inbound call to the FS IVR, it does not recognize DTMF digits from the PSTN phone. If I call IVR from an internal phone, then it does recognize the DTMF digits. I have mostly default configurations for everything. Best Regards, JErry ___ 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
[Freeswitch-users] Stop/Restart of Freeswitch Causes Crash
Sometimes if I stop (using ... command) and then restart freeswitch (using ./freeswitch command), the program will crash and return to the Linux (CentOS 5.3) prompt. I am using version 1.0.4. I just pasted the freeswitch/terminal log into the Pastebin. Best Regards, Jerry ___ 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
[Freeswitch-users] 3rd Party Dial Plan Tool
Can anyone recommend a good 3rd party dialplan tool that will work with Freeswitch? Best Regards, Jerry ___ 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
[Freeswitch-users] Adding Leading Digits To CALLING Number of Outgoing Call
How do I use the dial plan to add leading digits to an outgoing call through a gateway? My internal phone number is 5380, but when FS sends the call to the gateway I want the CALLING party to be 4253495380. Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Adding Leading Digits To CALLING Number of Outgoing Call
Okay, I figured it out. I added the following line to the default.xml file,
just prior to the bridge action:
action application=set
data=effective_caller_id_number=425349${caller_id_number}/
Now, 425349 is prepended to the outgoing call's caller ID.
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Monday, October 19, 2009 9:14 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: Adding Leading Digits To CALLING Number of Outgoing Call
How do I use the dial plan to add leading digits to an outgoing call through
a gateway?
My internal phone number is 5380, but when FS sends the call to the gateway
I want the CALLING party to be 4253495380.
Best Regards,
Jerry
___
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
[Freeswitch-users] scripts/contrib/trixter/makemodconf.pl: No such file or directory
I am building Freeswitch on a Centos 5.3 machine and the last step below gets an error because there is no scripts/contrib folder. Anyone know why? ./configure make make all install sounds-install uhd-moh-install moh-install scripts/contrib/trixter/makemodconf.pl modules.conf /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml bash: scripts/contrib/trixter/makemodconf.pl: No such file or directory Best Regards, Jerry ___ 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
Re: [Freeswitch-users] scripts/contrib/trixter/makemodconf.pl: No such file or directory
Okay, I think the contrib folder moved up one level. So the Wiki installation documentation should probably be updated to reflect that. Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Friday, October 16, 2009 9:47 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: scripts/contrib/trixter/makemodconf.pl: No such file or directory I am building Freeswitch on a Centos 5.3 machine and the last step below gets an error because there is no scripts/contrib folder. Anyone know why? ./configure make make all install sounds-install uhd-moh-install moh-install scripts/contrib/trixter/makemodconf.pl modules.conf /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml bash: scripts/contrib/trixter/makemodconf.pl: No such file or directory Best Regards, Jerry ___ 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
Re: [Freeswitch-users] SLAs and BLAs
I gather from the mailing archive that BLAs are implemented using the draft-anil-sipping-bla-04.txt document. According to the draft, the Appearance Agent is supposed to initiate a SUBSCRIBE request, but I don't see FS doing this. What phone types/models are known to work with the FS BLA implementation? Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, October 05, 2009 3:24 PM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] SLAs and BLAs We are building our own in-house developed Teo phones. I also have CounterPath's Bria Professional phone. For test purposes, I have one snom phone and a couple Polycomm phones. Jerry -Original Message- From: Brian West [mailto:br...@freeswitch.org] Sent: Monday, October 05, 2009 11:02 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SLAs and BLAs First off what phones are you going to be using? /b On Oct 5, 2009, at 12:58 PM, Jerry Richards wrote: I can see how BLFs and Presence are managed, however I haven't found much documentation on SLAs and BLAs. What is the RFC(s) that Freeswitch used to implement SLAs and BLAs? Do they differ from BLFs? Best Regards, Jerry ___ 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
[Freeswitch-users] FW: FS Does Not Relay PresencePUBLISHToSubscribing Phones
I put the sqlite3 select query in the paste bin again, and prior to that, I entered the .dump command. The select command came back with the sqlite3 prompt, which I guess means it didn't find an entry. How do I go about isolating this problem? I'm using CounterPath's Bria Professional softphone. They are the same company that make the Eyebeam. Any ideas? Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Friday, October 02, 2009 11:28 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Does Not Relay PresencePUBLISHToSubscribing Phones I put the sqlite3 select query in the paste bin, and prior to that, I entered the .dump command. The select command came back with a ... prompt which I don't understand. I don't know enough about sqlite3 to know what that means? Best Regards, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Friday, October 02, 2009 10:52 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay PresencePUBLISHToSubscribing Phones connect to sqlite directly with sqlite3 app and try that sql stmt and see why it doesn't match anything. sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name, 'Away','away','192.168.72.38',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='presence') and sub_to_user='1001' and (sub_to_host='192.168.72.38' or presence_hosts like '%192.168.72.38%') and (sip_subscriptions.profile_name = 'external' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards jerry.richa...@teotech.com wrote: Okay, I put a log up on the pastebin that shows the PUBLISH event coming from a CounterPath Bria Professional phone. For some reason, FS is getting an error and not relaying the presence status to the subscriber. Best Regards, Jerry _ From: João Mesquita [mailto:jmesqu...@freeswitch.org] Sent: Thursday, October 01, 2009 8:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISHToSubscribing Phones Piece of advice, don't ask, just do it. ;) jmesquita On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards jerry.richa...@teotech.com wrote: If you have time to take a look, I could put a trace in the pastebin? Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Thursday, October 01, 2009 10:29 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, October 01, 2009 9:36 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the manage-presence parameter to true in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry
Re: [Freeswitch-users] FS Does Not RelayPresencePUBLISHToSubscribing Phones
Okay, I added the ; at the end of the sqlite3 select command and it just returned to the sqlite prompt. No error was returned. Do you see anything in my database (in the pastebin) that is incorrect? By the way, the select command I put in the pastebin refers to the external config, but the internal config does the same thing. Best Regards, Jerry -Original Message- From: Rupa Schomaker [mailto:r...@rupa.com] Sent: Friday, October 02, 2009 11:42 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not RelayPresencePUBLISHToSubscribing Phones You are missing the trailing ; On Fri, Oct 2, 2009 at 1:28 PM, Jerry Richards jerry.richa...@teotech.com wrote: I put the sqlite3 select query in the paste bin, and prior to that, I entered the .dump command. The select command came back with a ... prompt which I don't understand. I don't know enough about sqlite3 to know what that means? Best Regards, Jerry -- -Rupa ___ 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
[Freeswitch-users] SLAs and BLAs
I can see how BLFs and Presence are managed, however I haven't found much documentation on SLAs and BLAs. What is the RFC(s) that Freeswitch used to implement SLAs and BLAs? Do they differ from BLFs? Best Regards, Jerry ___ 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
Re: [Freeswitch-users] SLAs and BLAs
We are building our own in-house developed Teo phones. I also have CounterPath's Bria Professional phone. For test purposes, I have one snom phone and a couple Polycomm phones. Jerry -Original Message- From: Brian West [mailto:br...@freeswitch.org] Sent: Monday, October 05, 2009 11:02 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SLAs and BLAs First off what phones are you going to be using? /b On Oct 5, 2009, at 12:58 PM, Jerry Richards wrote: I can see how BLFs and Presence are managed, however I haven't found much documentation on SLAs and BLAs. What is the RFC(s) that Freeswitch used to implement SLAs and BLAs? Do they differ from BLFs? Best Regards, Jerry ___ 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
[Freeswitch-users] Call Forward All/Busy/No-Answer
How would I configure FS to Call Forward All or Call Forward when Busy or Call Forward when No-Answer? Can this be done at the server, rather than at the phone? Best Regards, Jerry ___ 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
Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISHToSubscribing Phones
Okay, I put a log up on the pastebin that shows the PUBLISH event coming from a CounterPath Bria Professional phone. For some reason, FS is getting an error and not relaying the presence status to the subscriber. Best Regards, Jerry _ From: João Mesquita [mailto:jmesqu...@freeswitch.org] Sent: Thursday, October 01, 2009 8:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISHToSubscribing Phones Piece of advice, don't ask, just do it. ;) jmesquita On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards jerry.richa...@teotech.com wrote: If you have time to take a look, I could put a trace in the pastebin? Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Thursday, October 01, 2009 10:29 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, October 01, 2009 9:36 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the manage-presence parameter to true in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry ___ 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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 ___ 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
Re: [Freeswitch-users] FS Does Not Relay PresencePUBLISHToSubscribing Phones
I put the sqlite3 select query in the paste bin, and prior to that, I entered the .dump command. The select command came back with a ... prompt which I don't understand. I don't know enough about sqlite3 to know what that means? Best Regards, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Friday, October 02, 2009 10:52 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay PresencePUBLISHToSubscribing Phones connect to sqlite directly with sqlite3 app and try that sql stmt and see why it doesn't match anything. sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name, 'Away','away','192.168.72.38',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='presence') and sub_to_user='1001' and (sub_to_host='192.168.72.38' or presence_hosts like '%192.168.72.38%') and (sip_subscriptions.profile_name = 'external' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards jerry.richa...@teotech.com wrote: Okay, I put a log up on the pastebin that shows the PUBLISH event coming from a CounterPath Bria Professional phone. For some reason, FS is getting an error and not relaying the presence status to the subscriber. Best Regards, Jerry _ From: João Mesquita [mailto:jmesqu...@freeswitch.org] Sent: Thursday, October 01, 2009 8:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISHToSubscribing Phones Piece of advice, don't ask, just do it. ;) jmesquita On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards jerry.richa...@teotech.com wrote: If you have time to take a look, I could put a trace in the pastebin? Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Thursday, October 01, 2009 10:29 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, October 01, 2009 9:36 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the manage-presence parameter to true in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry ___ 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch
Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH To Subscribing Phones
By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the manage-presence parameter to true in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry ___ 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
Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones
I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, October 01, 2009 9:36 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the manage-presence parameter to true in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry ___ 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones
If you have time to take a look, I could put a trace in the pastebin? Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Thursday, October 01, 2009 10:29 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones I am using two Bria Professional Version 2.5.4 Build 54835 softphones. Thanks, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Thursday, October 01, 2009 9:36 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH ToSubscribing Phones which phone is it, we tested it with eyebeam and it appears to work for us. On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com wrote: By the way, I see the following lines at the FS console, which might be a clue as to why this is happening. Could someone point me toward what might cause this? I set the manage-presence parameter to true in each XML file where I saw it defined. [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external) [ERR] sofia_presence.c:611 DUMP PRESENCE SQL ... [WARNING] sofia_presence.c:565 192.168.72.38 is an alias, skipping Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Wednesday, September 30, 2009 9:12 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry ___ 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
[Freeswitch-users] FS Does Not Relay Presence PUBLISH To Subscribing Phones
I have two phones configured to subscribe to each other's presence status. When I change the presence status in one phone, I see the SIP PUBLISH message going to FS, but I don't see FS relaying that presence status to the subscribing phone. Does anyone know why? Best Regards, Jerry ___ 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
[Freeswitch-users] Searching Mailing List Archives
Sorry for this mundane question, but how do I search mailing archives for keywords? The following link has no search option? http://lists.freeswitch.org/pipermail/freeswitch-users/ Thanks And Best Regards, Jerry ___ 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
[Freeswitch-users] Control of BLF Capabilty?
Is there a way in FS to selectively deny a BLF presence subscription request for the sake of privacy? So that groups could be defined that are allowed to be monitor or be monitored? And others that are not allowed to monitor or be monitored? Best Regards, Jerry ___ 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
[Freeswitch-users] Outbound INVITE rejected with 480 Temp Unavail, Reason MANDATORY_IE_MISSING
Hello All, I have an internal extension that needs to send an INVITE without SDP body (Content Length 0). Freeswitch is replying with 480 Temporarily Unavailable with reason MANDATORY_IE_MISSING. Would anyone know what I need to do to enable this? Best Regards, Jerry ___ 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
[Freeswitch-users] OCS Support?
Hi, Does Freeswitch support OCS? We are interested in having our desktop PC control our in-house desktop phones (e.g. initiate call, answer call, hold call, etc.) using the uaCSTA protocol. Best Regards, Jerry ___ 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
Re: [Freeswitch-users] FS Presence Implementation
I think you're referring to the SIP SIMPLE implementation as the default FS presence mechanism. This is fine and I can use that protocol. The question I still have regards the plain text content in the body of the SIP MESSAGE method. What is the format of this plain text for presence that is compatible with the FS implementation? Best Regards, Jerry _ From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Tuesday, September 15, 2009 11:53 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Presence Implementation the default config ships with presence enabled for SIP if you have a phone that supports it, all you have to do is enable it on the phone. On Tue, Sep 15, 2009 at 1:30 PM, Jerry Richards jerry.richa...@teotech.com wrote: Also, is presence conveyed as any string? Or is presence a predefined list of status? Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Tuesday, September 15, 2009 8:46 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Presence Implementation I would like to modify my SIP phone and my gateway to convey/exchange presence information. Could someone point me toward the FS presence documentation? I've seen bits and pieces. Also, I think presence can be communicated via more than one protocol. Thanks And Best Regards, Jerry ___ 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
[Freeswitch-users] FS Presence Implementation
I would like to modify my SIP phone and my gateway to convey/exchange presence information. Could someone point me toward the FS presence documentation? I've seen bits and pieces. Also, I think presence can be communicated via more than one protocol. Thanks And Best Regards, Jerry ___ 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
Re: [Freeswitch-users] FS Presence Implementation
Also, is presence conveyed as any string? Or is presence a predefined list of status? Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Tuesday, September 15, 2009 8:46 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: FS Presence Implementation I would like to modify my SIP phone and my gateway to convey/exchange presence information. Could someone point me toward the FS presence documentation? I've seen bits and pieces. Also, I think presence can be communicated via more than one protocol. Thanks And Best Regards, Jerry ___ 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
[Freeswitch-users] Pastebin Username/Password Not Accepted
What account do I need to create to post logs in the Pastebin? I tried my mailing list username/password, and also tried a jira.freeswitch.org username/password. Neither of these were accepted. Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Pastebin Username/Password Not Accepted
Aha... I have been notified that I failed the test. The username/password is given in the authentication pop-up itself. My bad... -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, September 14, 2009 8:13 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: Pastebin Username/Password Not Accepted What account do I need to create to post logs in the Pastebin? I tried my mailing list username/password, and also tried a jira.freeswitch.org username/password. Neither of these were accepted. Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Inbound Gateway Call Not Working
Okay. I got the Grandstream Gateway's 1-stage dialing working with
Freeswitch (Thank You, Michael Collins and Thank All You Developers for
creating this really slick Softswitch/PBX).
Here are the changes/additions I made to the XML files:
conf/sip_profiles/exernal/grandstreamGXW4104.xml (added file):
include
gateway name=192.168.72.186
param name=username value=1000/
param name=password value=1234/
param name=proxy value=192.168.72.186/
param name=register value=false/
param name=extension value=1000/
/gateway
/include
conf/dialplan/default.xml (added to existing file):
extension name=GrandstreamTest
condition field=destination_number expression=^(9{0,1}\d{10})$
action application=bridge
data=sofia/gateway/192.168.72.186/$...@192.168.72.186/
/condition
/extension
conf/dialplan/public.xml (added to existing file):
extension name=GrandstreamTest
condition field=destination_number expression=^(5000)$
action application=transfer data=$1 XML default/
/condition
/extension
conf/autoload_configs/acl.conf.xml (added to existing file):
list name=lan default=allow
node type=allow cidr=192.168.72.186/32/
...
/list
...
list name=domains default=deny
node type=allow cidr=192.168.72.186/32/
...
/list
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, September 11, 2009 1:27 PM
To: 'freeswitch-users@lists.freeswitch.org'; 'Michael Collins'
Subject: RE: Inbound Gateway Call Not Working
Thanks. I added the node type=allow cidr=x.x.x.x/32/ to both the
lan list and domain list in the acl.conf.xml file and it does not try to
authenticate anymore.
However, now it replies to the INVITE with a 480 TEMPORARILY UNAVAILABLE.
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, September 11, 2009 10:57 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: RE: Inbound Gateway Call Not Working
By the way, the FS DEBUG console is saying the following when an inbound
call is made:
Rejected by acl domains. Falling back to Digest auth.
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, September 11, 2009 10:25 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: Inbound Gateway Call Not Working
I am trying to configure a Grandstream gateway to work with FS. I can make
outbound calls without a problem. However, inbound calls are getting a 403
Forbidden from FS in response to the INVITE from the gateway.
Now, the INVITE's from address is the caller's number (e.g. 111222),
which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy
Authentication Required and the gateway uses username Anonymous and the
uri sip:4...@192.168.72.38 (4000 is the destination for all calls from the
gateway).
Is there an example configuration for this scenario?
Thanks and Best Regards,
Jerry
___
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[Freeswitch-users] Inbound Gateway Call Not Working
I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 111222), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username Anonymous and the uri sip:4...@192.168.72.38 (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Inbound Gateway Call Not Working
By the way, the FS DEBUG console is saying the following when an inbound call is made: Rejected by acl domains. Falling back to Digest auth. Best Regards, Jerry -Original Message- From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Friday, September 11, 2009 10:25 AM To: 'freeswitch-users@lists.freeswitch.org' Subject: Inbound Gateway Call Not Working I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 111222), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username Anonymous and the uri sip:4...@192.168.72.38 (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry ___ 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
[Freeswitch-users] [ERR] mod_sofia.c:2645 Invalid Gateway
I have phones registered internally and can call among them. However, when I dial 711 from an internal phone, freeswitch replies with 484 Address Incomplete with reason INVALID_NUMBER_FORMAT. At the server console, I see the following error: [ERR] mod_sofia.c:2645 Invalid Gateway Does anyone know why I get this error? Is there something more I must do to add the gateway below? I already added the following to the usr/local/freesitch/conf/dialplan/default.xml: extension name=Testing - Mediant 1000 condition field=destination_number expression=^(711)$ action application=bridge data=sofia/gateway/mediant1000/$1/ /condition /extension I already created a usr/local/freeswitch/conf/sip_profiles/external/mediant1000.xml file: include gateway name=192.168.72.253 param name=username value=TEOGateWay/ param name=password value=ti0w...@b/ param name=register value=false/ /gateway /include Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Minimum/Recommended Freeswitch SystemConfiguration
Mitul, Thank you for your reply. Freeswitch is new to me, so I am not yet able to take measurements of FS under a load of traffic. I was just asking for future planning purposes. After I do some more development with it perhaps I can record some of these measurements. Thanks and Regards, Jerry -Original Message- From: Mitul Limbani [mailto:mi...@enterux.com] Sent: Friday, September 04, 2009 2:21 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Minimum/Recommended Freeswitch SystemConfiguration Jerry, As far as I understand freeswitch, it using kernel to thread and this operation eats good amount of RAM, but since the internal strructure of fs is to store all these sip details in runtime sqlite db, which is compressed text data earlier written in XML but while fs loads this configs it gets it in sqlite and that's what it used instead of asterisks astdb. Although what you see as recommended config for 500 users is true but it also depends on which processor you are trying this on. Intel or AMD is still ok but if you trying it on arm I don't have any data as such, interestingly if you have some test hardware scenario you can actually test and let us all know about it, it's quite useful bit of info that can be positioned on the FS Wiki, in case you want to take this experiment offlist do write to me, im interested to document :) Look forward to hear from you, Thanks Regards, Mitul Limbani, Founder CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 05-Sep-2009, at 12:03 AM, Jerry Richards jerry.richa...@teotech.com wrote: Under the Minimum/Recommended System Requirements, what is meant by We recommend you plan for 50% duty cycle? What is this duty cycle? Also, I see that the system requirements indicate Freeswitch recommends 1GB RAM and 50MB disk space. I guess I'm wondering how the number of extensions and external interfaces drive size of RAM and disk space? For example, would these recommendations support 100 extensions and one external interface? 500 extensions and 10 external interfaces? Etc.? Best Regards, Jerry ___ 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 ___ 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
[Freeswitch-users] Minimum/Recommended Freeswitch System Configuration
Under the Minimum/Recommended System Requirements, what is meant by We recommend you plan for 50% duty cycle? What is this duty cycle? Also, I see that the system requirements indicate Freeswitch recommends 1GB RAM and 50MB disk space. I guess I'm wondering how the number of extensions and external interfaces drive size of RAM and disk space? For example, would these recommendations support 100 extensions and one external interface? 500 extensions and 10 external interfaces? Etc.? Best Regards, Jerry ___ 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
[Freeswitch-users] Duplicate Extension Registration
I submitted this to the dev-list, but maybe it should be in the user-list: Can I register two phones to the same Line-ID? That is, does Freeswitch support a configuration where multiple endpoints have the same extension number, auth-id and password? And if so, do I have control over whether an inbound call causes both to ring or not? Thanks and Best Regards, Jerry ___ 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
[Freeswitch-users] Presence Feature
Does Freeswitch support Presence via SIMPLE protocol? Can it maintain presence? I presume this would be a SUBSCRIBE/NOTIFY arrangement? Best Regards, Jerry ___ 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
[Freeswitch-users] Scalability
Hello All, Does anyone know what the capacity of a stand-alone Freeswitch, in terms of how many users? Also, when that number is exceeded, how can Freeswitch server be distributed to accommodate a larger installation? Best Regards, Jerry ___ 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
[Freeswitch-users] RTP Packet Routing
Hello All, I noticed Freeswitch becomes the middle-man, handling RTP traffic for an active call. How do I configure it so it allows the two SIP endpoints to send RTP packet to each other directly? Best Regards, Jerry ___ 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
[Freeswitch-users] Cannot create outgoing channel type [error] cause: [FACILITY_NOT_SUBSCRIBED]
Hello All, I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP machine for the first time using the Getting Started Guide. I can register three lines (1000, 1001, and 1002), but when I attempt to call one phone to the other I hear the operator say: The person at extension 1000 is not available... Also, the Freeswitch log shows: Cannot create outgoing channel type [error] cause: [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause: [FACILITY_NOT_SUBSCRIBED] Does anyone know why I get this error? Best Regards, Jerry ___ 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
Re: [Freeswitch-users] Cannot create outgoing channel type [error]cause: [FACILITY_NOT_SUBSCRIBED]
Yes. This a stand-alone Windows XP machine. Jerry -Original Message- From: Brian West [mailto:br...@freeswitch.org] Sent: Monday, August 24, 2009 12:33 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Cannot create outgoing channel type [error]cause: [FACILITY_NOT_SUBSCRIBED] Are you trying to test everything on the same machine? /b On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote: Hello All, I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP machine for the first time using the Getting Started Guide. I can register three lines (1000, 1001, and 1002), but when I attempt to call one phone to the other I hear the operator say: The person at extension 1000 is not available... Also, the Freeswitch log shows: Cannot create outgoing channel type [error] cause: [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause: [FACILITY_NOT_SUBSCRIBED] Does anyone know why I get this error? Best Regards, Jerry ___ 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
