Notify message: Event: talk not working?
Hello, I tried to remotely anwer physical phones, which are in RINGING state, via event-socket. But the phones are not answering (tried Yealink an Aastra, see below). I send via event-socket sendevent NOTIFY. profile: internal. event-string: talk. user: 200. host: my.domain.com. content-type: application/simple-message-summary. State: ACTIVE I also set the "State" as described in http://wiki.freeswitch.org/wiki/Mod_event_socket#sendevent The following SIP is exchanged then (here Yealink): NOTIFY sip:200@192.168.1.129:5062 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKy8arj9a0SXNZN. Max-Forwards: 70. From: <sip:200@my.domain.com>;tag=515t5XNrSKFQF. To: <sip:200@my.domain.com>. Call-ID: c74eacdc-cca4-1231-c19a-b3a15c58d7e1. CSeq: 52146637 NOTIFY. Contact: <sip:mod_sofia@192.168.1.220:5060>. User-Agent: Freeswitch. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: talk. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: application/simple-message-summary. Content-Length: 0. SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKy8arj9a0SXNZN. From: <sip:200@my.domain.com>;tag=515t5XNrSKFQF. To: <sip:200@my.domain.com>;tag=669161475. Call-ID: c74eacdc-cca4-1231-c19a-b3a15c58d7e1. CSeq: 52146637 NOTIFY. User-Agent: Yealink SIP-T38G 38.70.0.125. Content-Length: 0. The call was not answered however and the Rsyslog does not show an entry to this message. I then tried an Aastra 6739i Phone and got the following: NOTIFY sip:208@192.168.1.118:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKpeya6gXmHa3rr. Max-Forwards: 70. From: <sip:208@my.domain.com>;tag=SXXjXe2S7De0K. To: <sip:208@my.domain.com>. Call-ID: 835658b2-cca1-1231-c19a-b3a15c58d7e1. CSeq: 52145935 NOTIFY. Contact: <sip:mod_sofia@192.168.1.220:5060>. User-Agent: Freeswitch. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: talk. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: application/simple-message-summary. Content-Length: 0. . SIP/2.0 489 Bad Event. Via: SIP/2.0/UDP 192.168.1.220;rport=5060;branch=z9hG4bKpeya6gXmHa3rr;received=192.168.1.220. From: <sip:208@my.domain.com>;tag=SXXjXe2S7De0K. To: <sip:208@my.domain.com>;tag=891443321. Call-ID: 835658b2-cca1-1231-c19a-b3a15c58d7e1. CSeq: 52145935 NOTIFY. Server: Aastra 6739i/3.2.2.2044. Content-Length: 0. Aastra shows me in rsyslog: 00:53:57.550000 : (SIP) INFO: EvOnPacketReceived: CSipEngine::, Packet do not match an existing Dialog, method: 6 00:53:57.550000 : (SIP) FUNC: OnPacketReceived: CSipGenericRequest(0x821DBC60) 00:53:57.550000 : (SIP) FUNC: EvInvalidNotify: CSipSubscribeEventDispatcher(0x821DC770) Reason 8002c405: Enter So I think, the Subscription-State may not be right, so Subscription-State: active might be better, right? mod_sofia.c, around line 1429 (as of 23-Oct-2013) should set this to "active" as far as I can see, but this is not the case here. But how can I solve this? Maybe there is another way? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
I just upgraded to newest git to confirm that the problem still persists. I still have Subscription-State: terminated;reason=noresource. when sending a Notify talk event to the phone. Nobody has a clue about this? How do you remotely answer a physical phone? Best regrads Peter On 11/20/13 18:34, Peter Steinbach wrote:
Hello,
I tried to remotely anwer physical phones, which are in RINGING state, via event-socket. But the phones are not answering (tried Yealink an Aastra, see below).
I send via event-socket
sendevent NOTIFY. profile: internal. event-string: talk. user: 200. host: my.domain.com. content-type: application/simple-message-summary. State: ACTIVE
I also set the "State" as described in http://wiki.freeswitch.org/wiki/Mod_event_socket#sendevent
The following SIP is exchanged then (here Yealink):
NOTIFY sip:200@192.168.1.129:5062 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKy8arj9a0SXNZN.
Max-Forwards: 70.
From: <sip:200@my.domain.com>;tag=515t5XNrSKFQF.
To: <sip:200@my.domain.com>.
Call-ID: c74eacdc-cca4-1231-c19a-b3a15c58d7e1.
CSeq: 52146637 NOTIFY.
Contact: <sip:mod_sofia@192.168.1.220:5060>.
User-Agent: Freeswitch.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Event: talk.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Subscription-State: terminated;reason=noresource.
Content-Type: application/simple-message-summary.
Content-Length: 0.
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKy8arj9a0SXNZN.
From: <sip:200@my.domain.com>;tag=515t5XNrSKFQF.
To: <sip:200@my.domain.com>;tag=669161475.
Call-ID: c74eacdc-cca4-1231-c19a-b3a15c58d7e1.
CSeq: 52146637 NOTIFY.
User-Agent: Yealink SIP-T38G 38.70.0.125.
Content-Length: 0.
The call was not answered however and the Rsyslog does not show an entry to this message.
I then tried an Aastra 6739i Phone and got the following:
NOTIFY sip:208@192.168.1.118:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKpeya6gXmHa3rr.
Max-Forwards: 70.
From: <sip:208@my.domain.com>;tag=SXXjXe2S7De0K.
To: <sip:208@my.domain.com>.
Call-ID: 835658b2-cca1-1231-c19a-b3a15c58d7e1.
CSeq: 52145935 NOTIFY.
Contact: <sip:mod_sofia@192.168.1.220:5060>.
User-Agent: Freeswitch.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Event: talk.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Subscription-State: terminated;reason=noresource.
Content-Type: application/simple-message-summary.
Content-Length: 0.
.
SIP/2.0 489 Bad Event.
Via: SIP/2.0/UDP 192.168.1.220;rport=5060;branch=z9hG4bKpeya6gXmHa3rr;received=192.168.1.220.
From: <sip:208@my.domain.com>;tag=SXXjXe2S7De0K.
To: <sip:208@my.domain.com>;tag=891443321.
Call-ID: 835658b2-cca1-1231-c19a-b3a15c58d7e1.
CSeq: 52145935 NOTIFY.
Server: Aastra 6739i/3.2.2.2044.
Content-Length: 0.
Aastra shows me in rsyslog:
00:53:57.550000 : (SIP) INFO: EvOnPacketReceived: CSipEngine::, Packet do not match an existing Dialog, method: 6
00:53:57.550000 : (SIP) FUNC: OnPacketReceived: CSipGenericRequest(0x821DBC60)
00:53:57.550000 : (SIP) FUNC: EvInvalidNotify: CSipSubscribeEventDispatcher(0x821DC770) Reason 8002c405: Enter
So I think, the Subscription-State may not be right, so
Subscription-State: active
might be better, right?
mod_sofia.c, around line 1429 (as of 23-Oct-2013) should set this to "active" as far as I can see, but this is not the case here.
But how can I solve this? Maybe there is another way?
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
Wonder is someone can help. Using snom phones to record calls works sort of, but it only records one side of the call, what am I missing. I am using the record button on the phone. It can be pressed at any time during the call to start the recording and after stoping it. Darcy Primrose
Pressing the record button by default sends a sip info record on and record off header. You can log in to the phones web interface and change what it does. http://wiki.snom.com/FAQ/What_does_the_'Record'_button_on_the_phone_do Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of fs Sent: Sunday, November 24, 2013 11:26 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] snom call recording Wonder is someone can help. Using snom phones to record calls works sort of, but it only records one side of the call, what am I missing. I am using the record button on the phone. It can be pressed at any time during the call to start the recording and after stoping it. Darcy Primrose _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I am aware of how the snom phone works with recording, my problem, is when I do this it only records one side of the call. Darcy ----- Original Message ----- From: "Moishe Grunstein" <max@nysolutions.com> To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Sent: Sunday, November 24, 2013 11:40 PM Subject: Re: [Freeswitch-users] snom call recording
Pressing the record button by default sends a sip info record on and record off header. You can log in to the phones web interface and change what it does. http://wiki.snom.com/FAQ/What_does_the_'Record'_button_on_the_phone_do
Thanks,
Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of fs Sent: Sunday, November 24, 2013 11:26 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] snom call recording
Wonder is someone can help. Using snom phones to record calls works sort of, but it only records one side of the call, what am I missing. I am using the record button on the phone. It can be pressed at any time during the call to start the recording and after stoping it.
Darcy Primrose
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
----- No virus found in this message. Checked by AVG - www.avg.com Version: 2014.0.4158 / Virus Database: 3629/6839 - Release Date: 11/15/13
I need to update this problem, I have found that all my call recordings are only recording one side of the call, the side that originated the call. Any help would be appreciated. The largest problem however is the snom phone only recording on side of the call when the record button is depressed. Darcy ----- Original Message ----- From: "fs" <fs@voice2net.ca> To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Sent: Sunday, November 24, 2013 11:25 PM Subject: [Freeswitch-users] snom call recording
Wonder is someone can help. Using snom phones to record calls works sort of, but it only records one side of the call, what am I missing. I am using the record button on the phone. It can be pressed at any time during the call to start the recording and after stoping it.
Darcy Primrose
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
----- No virus found in this message. Checked by AVG - www.avg.com Version: 2014.0.4158 / Virus Database: 3629/6839 - Release Date: 11/15/13
Are you sure the phones do support this event ? As far as i know only Phoner Lite supports it (after enabling it in the config) Snom or Bria for example are ignoring this completely... (We have only Snoms here and i had the same problem. Thats why i wrote a patch for mod_snom so ist possible to simulate a keypress via a http request) regards, gw -----Ursprüngliche Nachricht----- Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Peter Steinbach Gesendet: Sonntag, 24. November 2013 21:17 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Notify message: Event: talk not working? I just upgraded to newest git to confirm that the problem still persists. I still have Subscription-State: terminated;reason=noresource. when sending a Notify talk event to the phone. Nobody has a clue about this? How do you remotely answer a physical phone? Best regrads Peter On 11/20/13 18:34, Peter Steinbach wrote:
Hello,
I tried to remotely anwer physical phones, which are in RINGING state, via event-socket. But the phones are not answering (tried Yealink an Aastra, see below).
I send via event-socket
sendevent NOTIFY. profile: internal. event-string: talk. user: 200. host: my.domain.com. content-type: application/simple-message-summary. State: ACTIVE
I also set the "State" as described in http://wiki.freeswitch.org/wiki/Mod_event_socket#sendevent
The following SIP is exchanged then (here Yealink):
NOTIFY sip:200@192.168.1.129:5062 SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKy8arj9a0SXNZN.
Max-Forwards: 70.
From: <sip:200@my.domain.com>;tag=515t5XNrSKFQF.
To: <sip:200@my.domain.com>.
Call-ID: c74eacdc-cca4-1231-c19a-b3a15c58d7e1.
CSeq: 52146637 NOTIFY.
Contact: <sip:mod_sofia@192.168.1.220:5060>.
User-Agent: Freeswitch.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Event: talk.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Subscription-State: terminated;reason=noresource.
Content-Type: application/simple-message-summary.
Content-Length: 0.
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKy8arj9a0SXNZN.
From: <sip:200@my.domain.com>;tag=515t5XNrSKFQF.
To: <sip:200@my.domain.com>;tag=669161475.
Call-ID: c74eacdc-cca4-1231-c19a-b3a15c58d7e1.
CSeq: 52146637 NOTIFY.
User-Agent: Yealink SIP-T38G 38.70.0.125.
Content-Length: 0.
The call was not answered however and the Rsyslog does not show an entry to this message.
I then tried an Aastra 6739i Phone and got the following:
NOTIFY sip:208@192.168.1.118:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKpeya6gXmHa3rr.
Max-Forwards: 70.
From: <sip:208@my.domain.com>;tag=SXXjXe2S7De0K.
To: <sip:208@my.domain.com>.
Call-ID: 835658b2-cca1-1231-c19a-b3a15c58d7e1.
CSeq: 52145935 NOTIFY.
Contact: <sip:mod_sofia@192.168.1.220:5060>.
User-Agent: Freeswitch.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, precondition, path, replaces.
Event: talk.
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Subscription-State: terminated;reason=noresource.
Content-Type: application/simple-message-summary.
Content-Length: 0.
.
SIP/2.0 489 Bad Event.
Via: SIP/2.0/UDP 192.168.1.220;rport=5060;branch=z9hG4bKpeya6gXmHa3rr;received=192.168.1.220.
From: <sip:208@my.domain.com>;tag=SXXjXe2S7De0K.
To: <sip:208@my.domain.com>;tag=891443321.
Call-ID: 835658b2-cca1-1231-c19a-b3a15c58d7e1.
CSeq: 52145935 NOTIFY.
Server: Aastra 6739i/3.2.2.2044.
Content-Length: 0.
Aastra shows me in rsyslog:
00:53:57.550000 : (SIP) INFO: EvOnPacketReceived: CSipEngine::, Packet do not match an existing Dialog, method: 6
00:53:57.550000 : (SIP) FUNC: OnPacketReceived: CSipGenericRequest(0x821DBC60)
00:53:57.550000 : (SIP) FUNC: EvInvalidNotify: CSipSubscribeEventDispatcher(0x821DC770) Reason 8002c405: Enter
So I think, the Subscription-State may not be right, so
Subscription-State: active
might be better, right?
mod_sofia.c, around line 1429 (as of 23-Oct-2013) should set this to "active" as far as I can see, but this is not the case here.
But how can I solve this? Maybe there is another way?
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
That works fine with my testing on MASTER. -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Nov 24, 2013, at 2:16 PM, Peter Steinbach <lists@telefaks.de> wrote:
I just upgraded to newest git to confirm that the problem still persists.
I still have Subscription-State: terminated;reason=noresource. when sending a Notify talk event to the phone.
Nobody has a clue about this? How do you remotely answer a physical phone?
Best regrads Peter
Hello Brian, which phones are you using for that? Best regards Peter On 11/25/13 19:50, Brian West wrote:
That works fine with my testing on MASTER. -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Nov 24, 2013, at 2:16 PM, Peter Steinbach <lists@telefaks.de> wrote:
I just upgraded to newest git to confirm that the problem still persists.
I still have Subscription-State: terminated;reason=noresource. when sending a Notify talk event to the phone.
Nobody has a clue about this? How do you remotely answer a physical phone?
Best regrads Peter
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
participants (5)
-
Brian West
-
fs
-
Gerald Weber
-
Moishe Grunstein
-
Peter Steinbach