No NOTIFY MWI when registering via proxy.
Hello, I'm testing FS support for the header Path (FS is behind opensips). It pretty much works: I tested calling from one user to the other and calls work perfectly. However, I've noticed that when I register my terminal directly with FS without going thru the proxy, I receive an unsolicited NOTIFY containing Message-Waiting information. But when I register via proxy, FS doesn't send this NOTIFY. What could be causing this difference of behavior? (enabling debug (F8) doesn't show anything for registration handling). I have just updated to trunk Revision 14729 and this behavior persists. Also, is it possible to disable this unsolicited MWI notification? (I want to send it only to subscribed terminals). regards, takeshi
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
Hello, I'm testing FS support for the header Path (FS is behind opensips). It pretty much works: I tested calling from one user to the other and calls work perfectly. However, I've noticed that when I register my terminal directly with FS without going thru the proxy, I receive an unsolicited NOTIFY containing Message-Waiting information. But when I register via proxy, FS doesn't send this NOTIFY. What could be causing this difference of behavior? (enabling debug (F8) doesn't show anything for registration handling). I have just updated to trunk Revision 14729 and this behavior persists.
Also, is it possible to disable this unsolicited MWI notification? (I want to send it only to subscribed terminals).
OK. I got it this is a bad idea. Now I understand this unsolicited MWI notification is due to an implicit subscription and so, it is a good thing because we will reduce load by not having to deal with separate SUBSCRIBE requests. However I have yet to figure out why this NOTIFY is not sent when the REGISTER is relayed by a proxy.
Its because many phones cheat and just expect mwi without asking for it so we send one on register. There is an opt to disable it I think but I can't recall what it is atm On Sep 2, 2009 9:20 PM, "mayamatakeshi" <mayamatakeshi@gmail.com> wrote: On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi <mayamatakeshi@gmail.com> wrote: > > Hello, > I'm test... OK. I got it this is a bad idea. Now I understand this unsolicited MWI notification is due to an implicit subscription and so, it is a good thing because we will reduce load by not having to deal with separate SUBSCRIBE requests. However I have yet to figure out why this NOTIFY is not sent when the REGISTER is relayed by a proxy. _______________________________________________ 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
On Thu, Sep 3, 2009 at 12:05 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Its because many phones cheat and just expect mwi without asking for it so we send one on register. There is an opt to disable it I think but I can't recall what it is atm
Thanks. I've located the option in the mod sofia source: send-message-query-on-register It seems it is not in the wiki. I will update the mod sofia page.
On Sep 2, 2009 9:20 PM, "mayamatakeshi" <mayamatakeshi@gmail.com> wrote:
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi <mayamatakeshi@gmail.com> wrote: > > Hello, > I'm test...
OK. I got it this is a bad idea. Now I understand this unsolicited MWI notification is due to an implicit subscription and so, it is a good thing because we will reduce load by not having to deal with separate SUBSCRIBE requests.
However I have yet to figure out why this NOTIFY is not sent when the REGISTER is relayed by a proxy.
_______________________________________________ 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
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
Hello, I'm testing FS support for the header Path (FS is behind opensips). It pretty much works: I tested calling from one user to the other and calls work perfectly. However, I've noticed that when I register my terminal directly with FS without going thru the proxy, I receive an unsolicited NOTIFY containing Message-Waiting information. But when I register via proxy, FS doesn't send this NOTIFY. What could be causing this difference of behavior? (enabling debug (F8) doesn't show anything for registration handling).
I have enabled Sofia debug and I can see NTA is complaining about invalid URI when building the NOTIFY: nua: nua_notify: entering nua(0x9b3c1e8): sent signal r_notify nua(0x9b3c1e8): recv signal r_notify nua: nua_stack_set_params: entering nua(0x9b3c1e8): adding notify usage with event message-summary nta_leg_tcreate(0x9b74c68) nta outgoing create: invalid URI nta: outgoing_free(0x9b74928) nua(0x9b3c1e8): event r_notify 900 Internal error at nua_client.c:711 nua(0x9b3c1e8): removing notify usage with event message-summary My REGISTER relayed by opensips is this: REGISTER sip:test.com SIP/2.0 Record-Route: <sip:192.168.2.100;lr=on;ftag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5> Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKe62c.49e9f6c1.0 Via: SIP/2.0/UDP 192.168.2.121:5060 ;received=192.168.2.121;rport=5060;branch=z9hG4bKPj4uAYgDuRbilYy4lCWcjlDKIDAtf-9RdS Max-Forwards: 69 From: <sip:user1@test.com <sip%3Auser1@test.com>
;tag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 To: <sip:user1@test.com <sip%3Auser1@test.com>> Call-ID: JvQ.apMLiJtfHa7z4ShIfgBPi5jIbtBv CSeq: 14872 REGISTER Contact: <sip:user1@192.168.2.121:5060;nat=yes> Expires: 60 Authorization: Digest username="user1", realm="test.com", nonce="7d911eef-2c16-4deb-99f6-afcff9968a19", uri="sip:192.168.2.100", response="df29caeb78790b4527f1176622cbf192", algorithm=MD5, cnonce="5.EXCbM3RZTx6iOh1cvUzUvEZTs2eheG", qop=auth, nc=00000001 Content-Length: 0 Path: <sip:opensips@192.168.2.100 <sip%3Aopensips@192.168.2.100> ;lr;received=sip:192.168.2.121:5060>
I hope someone can point out a problem. I'm looking at NTA with gdb but I'm slow on this.
On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
Hello, I'm testing FS support for the header Path (FS is behind opensips). It pretty much works: I tested calling from one user to the other and calls work perfectly. However, I've noticed that when I register my terminal directly with FS without going thru the proxy, I receive an unsolicited NOTIFY containing Message-Waiting information. But when I register via proxy, FS doesn't send this NOTIFY. What could be causing this difference of behavior? (enabling debug (F8) doesn't show anything for registration handling).
I have enabled Sofia debug and I can see NTA is complaining about invalid URI when building the NOTIFY:
nua: nua_notify: entering nua(0x9b3c1e8): sent signal r_notify nua(0x9b3c1e8): recv signal r_notify nua: nua_stack_set_params: entering nua(0x9b3c1e8): adding notify usage with event message-summary nta_leg_tcreate(0x9b74c68) nta outgoing create: invalid URI nta: outgoing_free(0x9b74928) nua(0x9b3c1e8): event r_notify 900 Internal error at nua_client.c:711 nua(0x9b3c1e8): removing notify usage with event message-summary
My REGISTER relayed by opensips is this:
REGISTER sip:test.com SIP/2.0 Record-Route: <sip:192.168.2.100;lr=on;ftag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5> Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKe62c.49e9f6c1.0 Via: SIP/2.0/UDP 192.168.2.121:5060 ;received=192.168.2.121;rport=5060;branch=z9hG4bKPj4uAYgDuRbilYy4lCWcjlDKIDAtf-9RdS Max-Forwards: 69 From: <sip:user1@test.com <sip%3Auser1@test.com>
;tag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 To: <sip:user1@test.com <sip%3Auser1@test.com>> Call-ID: JvQ.apMLiJtfHa7z4ShIfgBPi5jIbtBv CSeq: 14872 REGISTER Contact: <sip:user1@192.168.2.121:5060;nat=yes> Expires: 60 Authorization: Digest username="user1", realm="test.com", nonce="7d911eef-2c16-4deb-99f6-afcff9968a19", uri="sip:192.168.2.100", response="df29caeb78790b4527f1176622cbf192", algorithm=MD5, cnonce="5.EXCbM3RZTx6iOh1cvUzUvEZTs2eheG", qop=auth, nc=00000001 Content-Length: 0 Path: <sip:opensips@192.168.2.100 <sip%3Aopensips@192.168.2.100> ;lr;received=sip:192.168.2.121:5060>
I hope someone can point out a problem. I'm looking at NTA with gdb but I'm slow on this.
The invalid URI nta is complaining about is the route_uri extracted from the Contact stored upon registration. The difference of behavior between INVITE (works) and NOTIFY (doesn't work) via proxy, seems to be because for INVITE, mod_sofia code (function sofia_glue_do_invite in sofia_glue.c) calls sofia_overcome_sip_uri_weakness to adjust the route_uri. But for a NOTIFY, this function is not called (and it cannot be called, as there's no session which is required as a parameter). In my case I can see that basically what sofia_overcome_sip_uri_weakness does is to remove the "<" , ">" around the route_uri. I messed with the code in sofia_glue_send_notify to just remove "<" and ">" and after that I was able to receive the NOTIFY. So I believe there is some code lacking in FS to properly permit UAs registering via proxy to receive NOTIFY. I might be wrong: if there is anyone using this scenario successfully, please let me know. Otherwise, I'll open a ticket on JIRA. regards, takeshi
Following up, did a bug get created for this issue? Mike On Sep 5, 2009, at 2:40 AM, mayamatakeshi wrote:
On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi <mayamatakeshi@gmail.com> wrote:
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi <mayamatakeshi@gmail.com> wrote: Hello, I'm testing FS support for the header Path (FS is behind opensips). It pretty much works: I tested calling from one user to the other and calls work perfectly. However, I've noticed that when I register my terminal directly with FS without going thru the proxy, I receive an unsolicited NOTIFY containing Message-Waiting information. But when I register via proxy, FS doesn't send this NOTIFY. What could be causing this difference of behavior? (enabling debug (F8) doesn't show anything for registration handling).
I have enabled Sofia debug and I can see NTA is complaining about invalid URI when building the NOTIFY:
nua: nua_notify: entering nua(0x9b3c1e8): sent signal r_notify nua(0x9b3c1e8): recv signal r_notify nua: nua_stack_set_params: entering nua(0x9b3c1e8): adding notify usage with event message-summary nta_leg_tcreate(0x9b74c68) nta outgoing create: invalid URI nta: outgoing_free(0x9b74928) nua(0x9b3c1e8): event r_notify 900 Internal error at nua_client.c:711 nua(0x9b3c1e8): removing notify usage with event message-summary
My REGISTER relayed by opensips is this:
REGISTER sip:test.com SIP/2.0 Record-Route: <sip:192.168.2.100;lr=on;ftag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5
Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKe62c.49e9f6c1.0 Via: SIP/2.0/UDP 192.168.2.121 : 5060 ;received = 192.168.2.121 ;rport=5060;branch=z9hG4bKPj4uAYgDuRbilYy4lCWcjlDKIDAtf-9RdS Max-Forwards: 69 From: <sip:user1@test.com>;tag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 To: <sip:user1@test.com> Call-ID: JvQ.apMLiJtfHa7z4ShIfgBPi5jIbtBv CSeq: 14872 REGISTER Contact: <sip:user1@192.168.2.121:5060;nat=yes> Expires: 60 Authorization: Digest username="user1", realm="test.com", nonce="7d911eef-2c16-4deb-99f6-afcff9968a19", uri="sip: 192.168.2.100", response="df29caeb78790b4527f1176622cbf192", algorithm=MD5, cnonce="5.EXCbM3RZTx6iOh1cvUzUvEZTs2eheG", qop=auth, nc=00000001 Content-Length: 0 Path: <sip:opensips@192.168.2.100;lr;received=sip:192.168.2.121:5060>
I hope someone can point out a problem. I'm looking at NTA with gdb but I'm slow on this.
The invalid URI nta is complaining about is the route_uri extracted from the Contact stored upon registration. The difference of behavior between INVITE (works) and NOTIFY (doesn't work) via proxy, seems to be because for INVITE, mod_sofia code (function sofia_glue_do_invite in sofia_glue.c) calls sofia_overcome_sip_uri_weakness to adjust the route_uri. But for a NOTIFY, this function is not called (and it cannot be called, as there's no session which is required as a parameter).
In my case I can see that basically what sofia_overcome_sip_uri_weakness does is to remove the "<" , ">" around the route_uri. I messed with the code in sofia_glue_send_notify to just remove "<" and ">" and after that I was able to receive the NOTIFY. So I believe there is some code lacking in FS to properly permit UAs registering via proxy to receive NOTIFY. I might be wrong: if there is anyone using this scenario successfully, please let me know. Otherwise, I'll open a ticket on JIRA.
regards, takeshi
On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris <mike@jerris.com> wrote:
Following up, did a bug get created for this issue?
Hello, yes. http://jira.freeswitch.org/browse/MODSOFIA-26
On Sep 5, 2009, at 2:40 AM, mayamatakeshi wrote:
On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
Hello, I'm testing FS support for the header Path (FS is behind opensips). It pretty much works: I tested calling from one user to the other and calls work perfectly. However, I've noticed that when I register my terminal directly with FS without going thru the proxy, I receive an unsolicited NOTIFY containing Message-Waiting information. But when I register via proxy, FS doesn't send this NOTIFY. What could be causing this difference of behavior? (enabling debug (F8) doesn't show anything for registration handling).
I have enabled Sofia debug and I can see NTA is complaining about invalid URI when building the NOTIFY:
nua: nua_notify: entering nua(0x9b3c1e8): sent signal r_notify nua(0x9b3c1e8): recv signal r_notify nua: nua_stack_set_params: entering nua(0x9b3c1e8): adding notify usage with event message-summary nta_leg_tcreate(0x9b74c68) nta outgoing create: invalid URI nta: outgoing_free(0x9b74928) nua(0x9b3c1e8): event r_notify 900 Internal error at nua_client.c:711 nua(0x9b3c1e8): removing notify usage with event message-summary
My REGISTER relayed by opensips is this:
REGISTER sip:test.com SIP/2.0 Record-Route: < sip:192.168.2.100;lr=on;ftag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5> Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKe62c.49e9f6c1.0 Via: SIP/2.0/UDP 192.168.2.121:5060 ;received=192.168.2.121;rport=5060;branch=z9hG4bKPj4uAYgDuRbilYy4lCWcjlDKIDAtf-9RdS Max-Forwards: 69 From: <sip:user1@test.com <sip%3Auser1@test.com>
;tag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 To: <sip:user1@test.com <sip%3Auser1@test.com>> Call-ID: JvQ.apMLiJtfHa7z4ShIfgBPi5jIbtBv CSeq: 14872 REGISTER Contact: <sip:user1@192.168.2.121:5060;nat=yes> Expires: 60 Authorization: Digest username="user1", realm="test.com", nonce="7d911eef-2c16-4deb-99f6-afcff9968a19", uri="sip:192.168.2.100", response="df29caeb78790b4527f1176622cbf192", algorithm=MD5, cnonce="5.EXCbM3RZTx6iOh1cvUzUvEZTs2eheG", qop=auth, nc=00000001 Content-Length: 0 Path: <sip:opensips@192.168.2.100 <sip%3Aopensips@192.168.2.100> ;lr;received=sip:192.168.2.121:5060>
I hope someone can point out a problem. I'm looking at NTA with gdb but I'm slow on this.
The invalid URI nta is complaining about is the route_uri extracted from the Contact stored upon registration. The difference of behavior between INVITE (works) and NOTIFY (doesn't work) via proxy, seems to be because for INVITE, mod_sofia code (function sofia_glue_do_invite in sofia_glue.c) calls sofia_overcome_sip_uri_weakness to adjust the route_uri. But for a NOTIFY, this function is not called (and it cannot be called, as there's no session which is required as a parameter).
In my case I can see that basically what sofia_overcome_sip_uri_weakness does is to remove the "<" , ">" around the route_uri. I messed with the code in sofia_glue_send_notify to just remove "<" and ">" and after that I was able to receive the NOTIFY. So I believe there is some code lacking in FS to properly permit UAs registering via proxy to receive NOTIFY. I might be wrong: if there is anyone using this scenario successfully, please let me know. Otherwise, I'll open a ticket on JIRA.
regards, takeshi
_______________________________________________ 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
On 9/12/09, mayamatakeshi <mayamatakeshi@gmail.com> wrote:
On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris <mike@jerris.com> wrote:
Following up, did a bug get created for this issue?
Hello, yes. http://jira.freeswitch.org/browse/MODSOFIA-26
Just to simplify things in case someone searches the list: Issue was solved on rev 14851. Thank you all.
Hello, I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console. I have explicitely set <param name="manage-presence" value="true"/> in the internal profile. When a phone is set up I get the following Snom Phone REGISTER => OpenSIPS=> Freeswitch Freeswitch OK => OpenSIPS=>Snom Phone Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch Freeswitch 202 Accepted => OpenSIPS=>Snom Phone Snom Phone PUBLISH => OpenSIPS=> Freeswitch Freeswitch 200 OK => OpenSIPS=>Snom Phone So presence generally seems to work. But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there. Any idea how to force the NOTIFY messages? Best regards Peter Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it: freeswitch@sip11.mydomain.com> nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 401 Unauthorized for REGISTER (6) nua(0x905a80): recv signal r_destroy nta_leg_destroy((nil)) nta: received PUBLISH sip:100@sip1.mydomain.com SIP/2.0 (CSeq 3) nta: PUBLISH (3) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905f10): event i_publish 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905f10): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x905f10): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua(0x905f10): sent signal r_destroy nta: sent 200 OK for PUBLISH (3) nua(0x905f10): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:mod_sofia@192.168.178.200:5062 SIP/2.0 (CSeq 2) nta: canonizing sip:mod_sofia@192.168.178.200:5062 with contact nta: SUBSCRIBE (2) going to existing leg nua: nua_stack_process_request: entering nta: sent 200 OK for SUBSCRIBE (2) nua(0x905560): event i_subscribe 200 OK nua: nua_application_event: entering nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 8) nta: REGISTER (8) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5dc073ba0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5dc073ba0): sent signal r_respond nua(0x7fd5dc073ba0): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua: nua_handle_destroy: entering nua(0x7fd5dc073ba0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (8) nua(0x7fd5dc073ba0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x8fc3d0): sent signal r_respond nua(0x8fc3d0): recv signal r_respond 200 OK nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (7) nua(0x8fc3d0): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:100@sip1.mydomain.com;user=phone SIP/2.0 (CSeq 1) nta: SUBSCRIBE (1) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nta_leg_tcreate(0x7fd5dc03add0) nua(0x7fd5dc078b70): adding notify usage with event message-summary nua(0x7fd5dc078b70): event i_subscribe 100 Trying nua: nua_application_event: entering nua(): refresh notify after 3600 seconds (in [3600..3600]) nua: nua_respond: entering nua(0x7fd5dc078b70): sent signal r_respond nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted nua: nua_stack_set_params: entering nta: sent 202 Accepted for SUBSCRIBE (1) mayamatakeshi schrieb:
On 9/12/09, *mayamatakeshi* <mayamatakeshi@gmail.com <mailto:mayamatakeshi@gmail.com>> wrote:
On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote:
Following up, did a bug get created for this issue?
Hello, yes. http://jira.freeswitch.org/browse/MODSOFIA-26
Just to simplify things in case someone searches the list: Issue was solved on rev 14851. Thank you all. ------------------------------------------------------------------------
_______________________________________________ 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
connect to FS with fs_cli Issue the command: /events MESSAGE_QUERY MESSAGE_WAITING then leave some voice mails probably you have a mis-configuration where the user/domain/profile cannot be resolved to the correct sofia profile to send the notify The event starts out as a freeswitch event and is translated into the notify by mod_sofia but only if it can match the event to a real sip user On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX <Prometheus001@gmx.net> wrote:
Hello,
I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console.
I have explicitely set <param name="manage-presence" value="true"/> in the internal profile.
When a phone is set up I get the following Snom Phone REGISTER => OpenSIPS=> Freeswitch Freeswitch OK => OpenSIPS=>Snom Phone
Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch Freeswitch 202 Accepted => OpenSIPS=>Snom Phone
Snom Phone PUBLISH => OpenSIPS=> Freeswitch Freeswitch 200 OK => OpenSIPS=>Snom Phone So presence generally seems to work.
But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there.
Any idea how to force the NOTIFY messages?
Best regards Peter
Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it:
freeswitch@sip11.mydomain.com> nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 401 Unauthorized for REGISTER (6) nua(0x905a80): recv signal r_destroy nta_leg_destroy((nil)) nta: received PUBLISH sip:100@sip1.mydomain.com<sip%3A100@sip1.mydomain.com>SIP/2.0 (CSeq 3) nta: PUBLISH (3) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905f10): event i_publish 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905f10): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x905f10): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua(0x905f10): sent signal r_destroy nta: sent 200 OK for PUBLISH (3) nua(0x905f10): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:mod_sofia@192.168.178.200:5062 SIP/2.0 (CSeq 2) nta: canonizing sip:mod_sofia@192.168.178.200:5062 with contact nta: SUBSCRIBE (2) going to existing leg nua: nua_stack_process_request: entering nta: sent 200 OK for SUBSCRIBE (2) nua(0x905560): event i_subscribe 200 OK nua: nua_application_event: entering nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 8) nta: REGISTER (8) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5dc073ba0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5dc073ba0): sent signal r_respond nua(0x7fd5dc073ba0): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua: nua_handle_destroy: entering nua(0x7fd5dc073ba0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (8) nua(0x7fd5dc073ba0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x8fc3d0): sent signal r_respond nua(0x8fc3d0): recv signal r_respond 200 OK nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (7) nua(0x8fc3d0): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:100@sip1.mydomain.com<sip%3A100@sip1.mydomain.com>;user=phone SIP/2.0 (CSeq 1) nta: SUBSCRIBE (1) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nta_leg_tcreate(0x7fd5dc03add0) nua(0x7fd5dc078b70): adding notify usage with event message-summary nua(0x7fd5dc078b70): event i_subscribe 100 Trying nua: nua_application_event: entering nua(): refresh notify after 3600 seconds (in [3600..3600]) nua: nua_respond: entering nua(0x7fd5dc078b70): sent signal r_respond nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted nua: nua_stack_set_params: entering nta: sent 202 Accepted for SUBSCRIBE (1)
mayamatakeshi schrieb:
On 9/12/09, *mayamatakeshi* <mayamatakeshi@gmail.com <mailto:mayamatakeshi@gmail.com>> wrote:
On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote:
Following up, did a bug get created for this issue?
Hello, yes. http://jira.freeswitch.org/browse/MODSOFIA-26
Just to simplify things in case someone searches the list: Issue was solved on rev 14851. Thank you all. ------------------------------------------------------------------------
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Anthony, thanks for the hint, I receive events like the following RECV EVENT Event-Name: MESSAGE_WAITING Core-UUID: e71632c8-d948-11de-942b-0138c6269e37 FreeSWITCH-Hostname: sip11.mydomain.com FreeSWITCH-IPv4: 192.168.178.200 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-24 23:33:13 Event-Date-GMT: Tue, 24 Nov 2009 22:33:13 GMT Event-Date-Timestamp: 1259101993918617 Event-Calling-File: mod_voicemail.c Event-Calling-Function: update_mwi Event-Calling-Line-Number: 1738 MWI-Messages-Waiting: yes MWI-Message-Account: 200@sip1.mydomain.com MWI-Voice-Message: 4/1 (0/0) I think the problem may be the Freeswitch cluster we are working with. All phones register with realm (e.g. 200@sip1.mydomain.com). The FS hostname is sip11.mydomain.com resp. sip12.mydomain.com on the other host. With xml_curl we ensure that for both domain names a directory entry is passed back. That way it works nicely with registering phones, receiving voicemails, recording voicemails etc. but not for MWI. For recording and querying voicemails we use the realm instead of the domain name and that way it works. When a voicemail has finished recording - and at the time the above message occurs - I see 2 directory xml_curl requests with Event-Calling-File=mod_voicemail.c&Event-Calling-Function=resolve_id One I expect is for retrieving the MWI data and the other one for sending the VM email (which is sucessfully sent). Any hint how we can workaround this? Or is there a parameter to tell mod_voicemail that is should use the realm instead of the local hostname for sending MWI? Best regards Peter Anthony Minessale schrieb:
connect to FS with fs_cli
Issue the command:
/events MESSAGE_QUERY MESSAGE_WAITING
then leave some voice mails
probably you have a mis-configuration where the user/domain/profile cannot be resolved to the correct sofia profile to send the notify
The event starts out as a freeswitch event and is translated into the notify by mod_sofia but only if it can match the event to a real sip user
On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>> wrote:
Hello,
I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console.
I have explicitely set <param name="manage-presence" value="true"/> in the internal profile.
When a phone is set up I get the following Snom Phone REGISTER => OpenSIPS=> Freeswitch Freeswitch OK => OpenSIPS=>Snom Phone
Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch Freeswitch 202 Accepted => OpenSIPS=>Snom Phone
Snom Phone PUBLISH => OpenSIPS=> Freeswitch Freeswitch 200 OK => OpenSIPS=>Snom Phone So presence generally seems to work.
But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there.
Any idea how to force the NOTIFY messages?
Best regards Peter
Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it:
freeswitch@sip11.mydomain.com <mailto:freeswitch@sip11.mydomain.com>> nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 401 Unauthorized for REGISTER (6) nua(0x905a80): recv signal r_destroy nta_leg_destroy((nil)) nta: received PUBLISH sip:100@sip1.mydomain.com <mailto:sip%3A100@sip1.mydomain.com> SIP/2.0 (CSeq 3) nta: PUBLISH (3) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905f10): event i_publish 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905f10): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x905f10): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua(0x905f10): sent signal r_destroy nta: sent 200 OK for PUBLISH (3) nua(0x905f10): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:mod_sofia@192.168.178.200:5062 <http://sip:mod_sofia@192.168.178.200:5062> SIP/2.0 (CSeq 2) nta: canonizing sip:mod_sofia@192.168.178.200:5062 <http://sip:mod_sofia@192.168.178.200:5062> with contact nta: SUBSCRIBE (2) going to existing leg nua: nua_stack_process_request: entering nta: sent 200 OK for SUBSCRIBE (2) nua(0x905560): event i_subscribe 200 OK nua: nua_application_event: entering nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 8) nta: REGISTER (8) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5dc073ba0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5dc073ba0): sent signal r_respond nua(0x7fd5dc073ba0): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua: nua_handle_destroy: entering nua(0x7fd5dc073ba0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (8) nua(0x7fd5dc073ba0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x8fc3d0): sent signal r_respond nua(0x8fc3d0): recv signal r_respond 200 OK nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (7) nua(0x8fc3d0): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:100@sip1.mydomain.com <mailto:sip%3A100@sip1.mydomain.com>;user=phone SIP/2.0 (CSeq 1) nta: SUBSCRIBE (1) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nta_leg_tcreate(0x7fd5dc03add0) nua(0x7fd5dc078b70): adding notify usage with event message-summary nua(0x7fd5dc078b70): event i_subscribe 100 Trying nua: nua_application_event: entering nua(): refresh notify after 3600 seconds (in [3600..3600]) nua: nua_respond: entering nua(0x7fd5dc078b70): sent signal r_respond nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted nua: nua_stack_set_params: entering nta: sent 202 Accepted for SUBSCRIBE (1)
mayamatakeshi schrieb: > > On 9/12/09, *mayamatakeshi* <mayamatakeshi@gmail.com <mailto:mayamatakeshi@gmail.com> > <mailto:mayamatakeshi@gmail.com <mailto:mayamatakeshi@gmail.com>>> wrote: > > > On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com> > <mailto:mike@jerris.com <mailto:mike@jerris.com>>> wrote: > > Following up, did a bug get created for this issue? > > > Hello, > yes. > http://jira.freeswitch.org/browse/MODSOFIA-26 > > > Just to simplify things in case someone searches the list: > Issue was solved on rev 14851. > Thank you all. > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@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
Peter, I had a similar problem, the way I found to make it work was setting the mailbox ID in the phone to match the FS domain/hostname. For instance using a Linksys SPA962, I set the "Voice Mail Server" in the extension tab to extension@domain using FS hostname as the domain. Regards, Juliano Peter P GMX escreveu:
Anthony, thanks for the hint,
I receive events like the following RECV EVENT Event-Name: MESSAGE_WAITING Core-UUID: e71632c8-d948-11de-942b-0138c6269e37 FreeSWITCH-Hostname: sip11.mydomain.com FreeSWITCH-IPv4: 192.168.178.200 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-24 23:33:13 Event-Date-GMT: Tue, 24 Nov 2009 22:33:13 GMT Event-Date-Timestamp: 1259101993918617 Event-Calling-File: mod_voicemail.c Event-Calling-Function: update_mwi Event-Calling-Line-Number: 1738 MWI-Messages-Waiting: yes MWI-Message-Account: 200@sip1.mydomain.com MWI-Voice-Message: 4/1 (0/0)
I think the problem may be the Freeswitch cluster we are working with. All phones register with realm (e.g. 200@sip1.mydomain.com). The FS hostname is sip11.mydomain.com resp. sip12.mydomain.com on the other host. With xml_curl we ensure that for both domain names a directory entry is passed back. That way it works nicely with registering phones, receiving voicemails, recording voicemails etc. but not for MWI. For recording and querying voicemails we use the realm instead of the domain name and that way it works.
When a voicemail has finished recording - and at the time the above message occurs - I see 2 directory xml_curl requests with Event-Calling-File=mod_voicemail.c&Event-Calling-Function=resolve_id One I expect is for retrieving the MWI data and the other one for sending the VM email (which is sucessfully sent).
Any hint how we can workaround this? Or is there a parameter to tell mod_voicemail that is should use the realm instead of the local hostname for sending MWI?
Best regards Peter
Anthony Minessale schrieb:
connect to FS with fs_cli
Issue the command:
/events MESSAGE_QUERY MESSAGE_WAITING
then leave some voice mails
probably you have a mis-configuration where the user/domain/profile cannot be resolved to the correct sofia profile to send the notify
The event starts out as a freeswitch event and is translated into the notify by mod_sofia but only if it can match the event to a real sip user
On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>> wrote:
Hello,
I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console.
I have explicitely set <param name="manage-presence" value="true"/> in the internal profile.
When a phone is set up I get the following Snom Phone REGISTER => OpenSIPS=> Freeswitch Freeswitch OK => OpenSIPS=>Snom Phone
Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch Freeswitch 202 Accepted => OpenSIPS=>Snom Phone
Snom Phone PUBLISH => OpenSIPS=> Freeswitch Freeswitch 200 OK => OpenSIPS=>Snom Phone So presence generally seems to work.
But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there.
Any idea how to force the NOTIFY messages?
Best regards Peter
Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it:
freeswitch@sip11.mydomain.com <mailto:freeswitch@sip11.mydomain.com>> nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 401 Unauthorized for REGISTER (6) nua(0x905a80): recv signal r_destroy nta_leg_destroy((nil)) nta: received PUBLISH sip:100@sip1.mydomain.com <mailto:sip%3A100@sip1.mydomain.com> SIP/2.0 (CSeq 3) nta: PUBLISH (3) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905f10): event i_publish 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905f10): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x905f10): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua(0x905f10): sent signal r_destroy nta: sent 200 OK for PUBLISH (3) nua(0x905f10): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:mod_sofia@192.168.178.200:5062 <http://sip:mod_sofia@192.168.178.200:5062> SIP/2.0 (CSeq 2) nta: canonizing sip:mod_sofia@192.168.178.200:5062 <http://sip:mod_sofia@192.168.178.200:5062> with contact nta: SUBSCRIBE (2) going to existing leg nua: nua_stack_process_request: entering nta: sent 200 OK for SUBSCRIBE (2) nua(0x905560): event i_subscribe 200 OK nua: nua_application_event: entering nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 8) nta: REGISTER (8) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5dc073ba0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5dc073ba0): sent signal r_respond nua(0x7fd5dc073ba0): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua: nua_handle_destroy: entering nua(0x7fd5dc073ba0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (8) nua(0x7fd5dc073ba0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x8fc3d0): sent signal r_respond nua(0x8fc3d0): recv signal r_respond 200 OK nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (7) nua(0x8fc3d0): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:100@sip1.mydomain.com <mailto:sip%3A100@sip1.mydomain.com>;user=phone SIP/2.0 (CSeq 1) nta: SUBSCRIBE (1) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nta_leg_tcreate(0x7fd5dc03add0) nua(0x7fd5dc078b70): adding notify usage with event message-summary nua(0x7fd5dc078b70): event i_subscribe 100 Trying nua: nua_application_event: entering nua(): refresh notify after 3600 seconds (in [3600..3600]) nua: nua_respond: entering nua(0x7fd5dc078b70): sent signal r_respond nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted nua: nua_stack_set_params: entering nta: sent 202 Accepted for SUBSCRIBE (1)
mayamatakeshi schrieb: > > On 9/12/09, *mayamatakeshi* <mayamatakeshi@gmail.com <mailto:mayamatakeshi@gmail.com> > <mailto:mayamatakeshi@gmail.com <mailto:mayamatakeshi@gmail.com>>> wrote: > > > On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com> > <mailto:mike@jerris.com <mailto:mike@jerris.com>>> wrote: > > Following up, did a bug get created for this issue? > > > Hello, > yes. > http://jira.freeswitch.org/browse/MODSOFIA-26 > > > Just to simplify things in case someone searches the list: > Issue was solved on rev 14851. > Thank you all. > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@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
Try an alias on the sip profile. Mike On Nov 24, 2009, at 5:56 PM, Peter P GMX wrote:
Anthony, thanks for the hint,
I receive events like the following RECV EVENT Event-Name: MESSAGE_WAITING Core-UUID: e71632c8-d948-11de-942b-0138c6269e37 FreeSWITCH-Hostname: sip11.mydomain.com FreeSWITCH-IPv4: 192.168.178.200 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-24 23:33:13 Event-Date-GMT: Tue, 24 Nov 2009 22:33:13 GMT Event-Date-Timestamp: 1259101993918617 Event-Calling-File: mod_voicemail.c Event-Calling-Function: update_mwi Event-Calling-Line-Number: 1738 MWI-Messages-Waiting: yes MWI-Message-Account: 200@sip1.mydomain.com MWI-Voice-Message: 4/1 (0/0)
I think the problem may be the Freeswitch cluster we are working with. All phones register with realm (e.g. 200@sip1.mydomain.com). The FS hostname is sip11.mydomain.com resp. sip12.mydomain.com on the other host. With xml_curl we ensure that for both domain names a directory entry is passed back. That way it works nicely with registering phones, receiving voicemails, recording voicemails etc. but not for MWI. For recording and querying voicemails we use the realm instead of the domain name and that way it works.
When a voicemail has finished recording - and at the time the above message occurs - I see 2 directory xml_curl requests with Event-Calling-File=mod_voicemail.c&Event-Calling-Function=resolve_id One I expect is for retrieving the MWI data and the other one for sending the VM email (which is sucessfully sent).
Any hint how we can workaround this? Or is there a parameter to tell mod_voicemail that is should use the realm instead of the local hostname for sending MWI?
Best regards Peter
Anthony Minessale schrieb:
connect to FS with fs_cli
Issue the command:
/events MESSAGE_QUERY MESSAGE_WAITING
then leave some voice mails
probably you have a mis-configuration where the user/domain/profile cannot be resolved to the correct sofia profile to send the notify
The event starts out as a freeswitch event and is translated into the notify by mod_sofia but only if it can match the event to a real sip user
On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>> wrote:
Hello,
I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console.
I have explicitely set <param name="manage-presence" value="true"/> in the internal profile.
When a phone is set up I get the following Snom Phone REGISTER => OpenSIPS=> Freeswitch Freeswitch OK => OpenSIPS=>Snom Phone
Snom Phone SUBSCRIBE => OpenSIPS=> Freeswitch Freeswitch 202 Accepted => OpenSIPS=>Snom Phone
Snom Phone PUBLISH => OpenSIPS=> Freeswitch Freeswitch 200 OK => OpenSIPS=>Snom Phone So presence generally seems to work.
But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there.
Any idea how to force the NOTIFY messages?
Best regards Peter
Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it:
freeswitch@sip11.mydomain.com <mailto:freeswitch@sip11.mydomain.com>> nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 401 Unauthorized for REGISTER (6) nua(0x905a80): recv signal r_destroy nta_leg_destroy((nil)) nta: received PUBLISH sip:100@sip1.mydomain.com <mailto:sip%3A100@sip1.mydomain.com> SIP/2.0 (CSeq 3) nta: PUBLISH (3) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905f10): event i_publish 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905f10): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x905f10): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua(0x905f10): sent signal r_destroy nta: sent 200 OK for PUBLISH (3) nua(0x905f10): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:mod_sofia@192.168.178.200:5062 <http://sip:mod_sofia@192.168.178.200:5062> SIP/2.0 (CSeq 2) nta: canonizing sip:mod_sofia@192.168.178.200:5062 <http://sip:mod_sofia@192.168.178.200:5062> with contact nta: SUBSCRIBE (2) going to existing leg nua: nua_stack_process_request: entering nta: sent 200 OK for SUBSCRIBE (2) nua(0x905560): event i_subscribe 200 OK nua: nua_application_event: entering nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 8) nta: REGISTER (8) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5dc073ba0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5dc073ba0): sent signal r_respond nua(0x7fd5dc073ba0): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua: nua_handle_destroy: entering nua(0x7fd5dc073ba0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (8) nua(0x7fd5dc073ba0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com <http://sip1.mydomain.com> SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x8fc3d0): sent signal r_respond nua(0x8fc3d0): recv signal r_respond 200 OK nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (7) nua(0x8fc3d0): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:100@sip1.mydomain.com <mailto:sip%3A100@sip1.mydomain.com>;user=phone SIP/2.0 (CSeq 1) nta: SUBSCRIBE (1) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nta_leg_tcreate(0x7fd5dc03add0) nua(0x7fd5dc078b70): adding notify usage with event message-summary nua(0x7fd5dc078b70): event i_subscribe 100 Trying nua: nua_application_event: entering nua(): refresh notify after 3600 seconds (in [3600..3600]) nua: nua_respond: entering nua(0x7fd5dc078b70): sent signal r_respond nua(0x7fd5dc078b70): recv signal r_respond 202 Accepted nua: nua_stack_set_params: entering nta: sent 202 Accepted for SUBSCRIBE (1)
mayamatakeshi schrieb:
On 9/12/09, *mayamatakeshi* <mayamatakeshi@gmail.com
<mailto:mayamatakeshi@gmail.com>
<mailto:mayamatakeshi@gmail.com <mailto:mayamatakeshi@gmail.com>>> wrote:
On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com> <mailto:mike@jerris.com <mailto:mike@jerris.com>>> wrote:
Following up, did a bug get created for this issue?
Hello, yes. http://jira.freeswitch.org/browse/MODSOFIA-26
Just to simplify things in case someone searches the list: Issue was solved on rev 14851. Thank you all.
------------------------------------------------------------------------
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org
<mailto:FreeSWITCH-users@lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@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
I tried now with phones directly attached to the freeswitch (without an OpenSIPS in between). I also added the alias. But the behaviour is as before: No notify message from freeswitch, neither after register nor after a voicemail is recorded. Best regards Peter Brian West schrieb:
Yes an alias will be required for every domain you run on the profile so it can find it.
/b
On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
Try an alias on the sip profile.
Mike
_______________________________________________ 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
Did you check the 2 replies that told you you need aliases in your sofia profile to translate the domain found in your message_waiting to the right profile? Both Brian and Mike answered you. On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX <Prometheus001@gmx.net> wrote:
I tried now with phones directly attached to the freeswitch (without an OpenSIPS in between). I also added the alias. But the behaviour is as before: No notify message from freeswitch, neither after register nor after a voicemail is recorded.
Best regards Peter Brian West schrieb:
Yes an alias will be required for every domain you run on the profile so it can find it.
/b
On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
Try an alias on the sip profile.
Mike
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
based on your example past sip1.mydomain.com is the domain in the packet and thus the profile should have an alias for this. Then the user must reside in your sip db with the user 200 and domain sip1.mydomain.com if you dont have this consider the force-register-domain and force-register-db-domain to normalize the host names. On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Did you check the 2 replies that told you you need aliases in your sofia profile to translate the domain found in your message_waiting to the right profile? Both Brian and Mike answered you.
On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX <Prometheus001@gmx.net>wrote:
I tried now with phones directly attached to the freeswitch (without an OpenSIPS in between). I also added the alias. But the behaviour is as before: No notify message from freeswitch, neither after register nor after a voicemail is recorded.
Best regards Peter Brian West schrieb:
Yes an alias will be required for every domain you run on the profile so it can find it.
/b
On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
Try an alias on the sip profile.
Mike
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Hello Anthony, I did some checks today Here is how the phones are registered: mysql> select sip_host, presence_hosts, server_user,server_host, hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1; +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ | sip_host | presence_hosts | server_user | server_host | hostname | sip_realm | mwi_user | mwi_host | +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ | sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 | sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com | +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ IPs are: 10.11.12.1 sip1.mydomain.com (common cluster IP) 10.11.12.2 sip11.mydomain.com 10.11.12.3 sip12.mydomain.com (not used at this point) XML-Curl for the directory is: <document type="freeswitch/xml"> <section name="directory"> <domain name="sip1.mydomain.com"> <user id="100"> <params> <param name="password" value="pass"/> <param name="vm-password" value="pass"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true"/> <param name="vm-mailto" value="email@domain.net"/> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> <param name="http-allowed-api" value="voicemail"/> </params> <variables> <variable name="accountcode" value="800"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 100"/> <variable name="effective_caller_id_number" value="100"/> </variables> </user> </domain> </section> </document> The internal profile has the following alias: <profile name="internal" domain="$${domain}"> <aliases> <alias name="$${domain}"/> <alias name="sip1.mydomain.com"/> <alias name="default"/> </aliases> With $${domain} being sip11.mydomain.com Phones are registering to sip1.mydomain.com, Voicemail works, but MWI does not. Any hint what I should change to make this work? Best regards Peter Anthony Minessale schrieb:
based on your example past
sip1.mydomain.com <http://sip1.mydomain.com> is the domain in the packet and thus the profile should have an alias for this. Then the user must reside in your sip db with the user 200 and domain sip1.mydomain.com <http://sip1.mydomain.com>
if you dont have this consider the force-register-domain and force-register-db-domain to normalize the host names.
On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote:
Did you check the 2 replies that told you you need aliases in your sofia profile to translate the domain found in your message_waiting to the right profile? Both Brian and Mike answered you.
On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>> wrote:
I tried now with phones directly attached to the freeswitch (without an OpenSIPS in between). I also added the alias. But the behaviour is as before: No notify message from freeswitch, neither after register nor after a voicemail is recorded.
Best regards Peter Brian West schrieb: > Yes an alias will be required for every domain you run on the profile > so it can find it. > > /b > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > > >> Try an alias on the sip profile. >> >> Mike >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@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
Hello, i now changed the $${domain} name of the server to the domain name the phones register with. Now messaging (MWI, notify) works. Best regards Peter Peter P GMX schrieb:
Hello Anthony,
I did some checks today Here is how the phones are registered:
mysql> select sip_host, presence_hosts, server_user,server_host, hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1; +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ | sip_host | presence_hosts | server_user | server_host | hostname | sip_realm | mwi_user | mwi_host | +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ | sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 | sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com | +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ IPs are: 10.11.12.1 sip1.mydomain.com (common cluster IP) 10.11.12.2 sip11.mydomain.com 10.11.12.3 sip12.mydomain.com (not used at this point)
XML-Curl for the directory is: <document type="freeswitch/xml"> <section name="directory"> <domain name="sip1.mydomain.com"> <user id="100"> <params> <param name="password" value="pass"/> <param name="vm-password" value="pass"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true"/> <param name="vm-mailto" value="email@domain.net"/> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> <param name="http-allowed-api" value="voicemail"/> </params> <variables> <variable name="accountcode" value="800"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 100"/> <variable name="effective_caller_id_number" value="100"/> </variables> </user> </domain> </section> </document>
The internal profile has the following alias: <profile name="internal" domain="$${domain}"> <aliases> <alias name="$${domain}"/> <alias name="sip1.mydomain.com"/> <alias name="default"/> </aliases> With $${domain} being sip11.mydomain.com
Phones are registering to sip1.mydomain.com, Voicemail works, but MWI does not. Any hint what I should change to make this work?
Best regards Peter
Anthony Minessale schrieb:
based on your example past
sip1.mydomain.com <http://sip1.mydomain.com> is the domain in the packet and thus the profile should have an alias for this. Then the user must reside in your sip db with the user 200 and domain sip1.mydomain.com <http://sip1.mydomain.com>
if you dont have this consider the force-register-domain and force-register-db-domain to normalize the host names.
On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote:
Did you check the 2 replies that told you you need aliases in your sofia profile to translate the domain found in your message_waiting to the right profile? Both Brian and Mike answered you.
On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>> wrote:
I tried now with phones directly attached to the freeswitch (without an OpenSIPS in between). I also added the alias. But the behaviour is as before: No notify message from freeswitch, neither after register nor after a voicemail is recorded.
Best regards Peter Brian West schrieb: > Yes an alias will be required for every domain you run on the profile > so it can find it. > > /b > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > > >> Try an alias on the sip profile. >> >> Mike >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@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
Hello, i now changed the $${domain} of the server to the domain name the phones register with. Now messaging (MWI, notify) works. Thanks to all for your support. Best regards Peter Peter P GMX schrieb:
Hello Anthony,
I did some checks today Here is how the phones are registered:
mysql> select sip_host, presence_hosts, server_user,server_host, hostname, sip_realm, mwi_user,mwi_host from sip_registrations limit 1; +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ | sip_host | presence_hosts | server_user | server_host | hostname | sip_realm | mwi_user | mwi_host | +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ | sip1.mydomain.com | sip1.mydomain.com | 136 | 10.11.12.2 | sip11.mydomain.com | sip1.mydomain.com | 136 | sip1.mydomain.com | +-------------------+-------------------+-------------+-------------+--------------------+-------------------+----------+-------------------+ IPs are: 10.11.12.1 sip1.mydomain.com (common cluster IP) 10.11.12.2 sip11.mydomain.com 10.11.12.3 sip12.mydomain.com (not used at this point)
XML-Curl for the directory is: <document type="freeswitch/xml"> <section name="directory"> <domain name="sip1.mydomain.com"> <user id="100"> <params> <param name="password" value="pass"/> <param name="vm-password" value="pass"/> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true"/> <param name="vm-mailto" value="email@domain.net"/> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> <param name="http-allowed-api" value="voicemail"/> </params> <variables> <variable name="accountcode" value="800"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 100"/> <variable name="effective_caller_id_number" value="100"/> </variables> </user> </domain> </section> </document>
The internal profile has the following alias: <profile name="internal" domain="$${domain}"> <aliases> <alias name="$${domain}"/> <alias name="sip1.mydomain.com"/> <alias name="default"/> </aliases> With $${domain} being sip11.mydomain.com
Phones are registering to sip1.mydomain.com, Voicemail works, but MWI does not. Any hint what I should change to make this work?
Best regards Peter
Anthony Minessale schrieb:
based on your example past
sip1.mydomain.com <http://sip1.mydomain.com> is the domain in the packet and thus the profile should have an alias for this. Then the user must reside in your sip db with the user 200 and domain sip1.mydomain.com <http://sip1.mydomain.com>
if you dont have this consider the force-register-domain and force-register-db-domain to normalize the host names.
On Fri, Nov 27, 2009 at 3:11 PM, Anthony Minessale <anthony.minessale@gmail.com <mailto:anthony.minessale@gmail.com>> wrote:
Did you check the 2 replies that told you you need aliases in your sofia profile to translate the domain found in your message_waiting to the right profile? Both Brian and Mike answered you.
On Thu, Nov 26, 2009 at 5:55 PM, Peter P GMX <Prometheus001@gmx.net <mailto:Prometheus001@gmx.net>> wrote:
I tried now with phones directly attached to the freeswitch (without an OpenSIPS in between). I also added the alias. But the behaviour is as before: No notify message from freeswitch, neither after register nor after a voicemail is recorded.
Best regards Peter Brian West schrieb: > Yes an alias will be required for every domain you run on the profile > so it can find it. > > /b > > On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: > > >> Try an alias on the sip profile. >> >> Mike >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@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
Does the alias you added match the one that you saw in the event? The alias is 100% for sure the fix for this issue, please check again. Mike On Nov 26, 2009, at 6:55 PM, Peter P GMX wrote:
I tried now with phones directly attached to the freeswitch (without an OpenSIPS in between). I also added the alias. But the behaviour is as before: No notify message from freeswitch, neither after register nor after a voicemail is recorded.
Best regards Peter Brian West schrieb:
Yes an alias will be required for every domain you run on the profile so it can find it.
/b
On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
Try an alias on the sip profile.
Mike
participants (6)
-
Anthony Minessale
-
Brian West
-
Juliano - Terra
-
mayamatakeshi
-
Michael Jerris
-
Peter P GMX