Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
On Wed, Apr 7, 2010 at 2:06 AM, Anca Vamanu a...@opensips.org wrote: Hi James, What you see happens because of a improvement that I made in pua_dialoginfo module. Now the presentity_uri for the callee ( the uri that will be used as RURI in Publish message) is taken from RURI of Invite in the moment you call dialoginfo_set(before this To header was used). From the trace it seems that you call this function after you do lookup(location). Move this call upper, before lookup() changes the RURI to the contact of the phone. Also you can check out the possibility to set custom uris to be used as presentity uri by setting some pseudovariables (http://www.opensips.org/html/docs/modules/devel/pua_dialoginfo.html#id227160 ). Hi Anca, Moving the dialoginfo_set() above location() seems to fix the problem! I will let you know how things go. I also added another patch to get the green light upon startup, but I believe this causes the light to go solid amber during the first call, but then all subsequent calls seem to work ok. I will also try and see if I can duplicate the stuck 'red' state. Thanks for all your help. Regards, -- Anca Vamanu www.voice-system.ro -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
Hi James, What you see happens because of a improvement that I made in pua_dialoginfo module. Now the presentity_uri for the callee ( the uri that will be used as RURI in Publish message) is taken from RURI of Invite in the moment you call dialoginfo_set(before this To header was used). From the trace it seems that you call this function after you do lookup(location). Move this call upper, before lookup() changes the RURI to the contact of the phone. Also you can check out the possibility to set custom uris to be used as presentity uri by setting some pseudovariables (http://www.opensips.org/html/docs/modules/devel/pua_dialoginfo.html#id227160 ). Regards, -- Anca Vamanu www.voice-system.ro James Lamanna wrote: The phones should never receive the Publish message. Please catch a trace containing this Publish and send it to me. What do you mean by before? Before updating from svn with my patch? Before I updated from 1.6.2 to SVN I think - I'll try and double-check. I have a revised patch that covers my case that I'll submit this afternoon with the trace. Any luck with the stuck BLF trace I submitted as well? I also think that my configuration might need some help wrt when and where to call presence functions - I will post that once I get in the office. I noticed that with svn I'm not getting any entries in pua or presentity, only in active_watchers. Ok here's what I saw, running the 1.6 branch SVN HEAD - it is attached in bad_publish.txt. Also, I've attached my opensips.cfg to see if I'm doing anything bad in there as well. Thanks. -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
James Lamanna wrote: On Mon, Apr 5, 2010 at 12:52 AM, Angel Marin an...@anmar.eu.org wrote: On 03/04/10 01:40, James Lamanna wrote: On Fri, Apr 2, 2010 at 4:52 AM, Anca Vamanu a...@opensips.org wrote: Hi James, I think that the problem is with those Notifies without a body sent by OpenSIPS when the phone was started. This is normal behavior, correct and conform RFC - when the presence server does not have any record for that presentity - it includes no body. But since you say that Linksys does not like this and since it was not that difficult to change, I just committed a patch that sends a Notify with an empty dialoginfo tuple as body when no published record is found. Please upgrade from svn and test this case again. Hi Anca, I saw your patch and upgraded to the SVN 1.6 branch. However, I do not believe I ever hit that code path. In my debug logs I see No record exists in hash table (notify.c:853). I'm not sure if fallback2db is true, but I don't think it is, so I never make it to that codepath. Give the attached patch a shot, it should do what you're looking for. Though I'm not sure generating an empty dialog without a presence entry is the correct approach, I mean, the monitored extension is not there, so why not hint it as a blinking light? In a cold start scenario, it'll go green once your re-subscribe time comes, so set it to a couple minutes or lower in the phone and you'll be good to go. Actually, it never changes from blinking to green in that scenario, even after multiple subscribes and registers. Also, I just noticed that I'm receiving a peculiar response, a 501 Not Implemented from the phones in response to a PUBLISH (I swear it wasn't doing this before!)... -- James Hi James, The phones should never receive the Publish message. Please catch a trace containing this Publish and send it to me. What do you mean by before? Before updating from svn with my patch? Regards, -- Anca Vamanu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
On Apr 6, 2010, at 3:24, Anca Vamanu a...@opensips.org wrote: James Lamanna wrote: On Mon, Apr 5, 2010 at 12:52 AM, Angel Marin an...@anmar.eu.org wrote: On 03/04/10 01:40, James Lamanna wrote: On Fri, Apr 2, 2010 at 4:52 AM, Anca Vamanu a...@opensips.org wrote: Hi James, I think that the problem is with those Notifies without a body sent by OpenSIPS when the phone was started. This is normal behavior, correct and conform RFC - when the presence server does not have any record for that presentity - it includes no body. But since you say that Linksys does not like this and since it was not that difficult to change, I just committed a patch that sends a Notify with an empty dialoginfo tuple as body when no published record is found. Please upgrade from svn and test this case again. Hi Anca, I saw your patch and upgraded to the SVN 1.6 branch. However, I do not believe I ever hit that code path. In my debug logs I see No record exists in hash table (notify.c: 853). I'm not sure if fallback2db is true, but I don't think it is, so I never make it to that codepath. Give the attached patch a shot, it should do what you're looking for. Though I'm not sure generating an empty dialog without a presence entry is the correct approach, I mean, the monitored extension is not there, so why not hint it as a blinking light? In a cold start scenario, it'll go green once your re-subscribe time comes, so set it to a couple minutes or lower in the phone and you'll be good to go. Actually, it never changes from blinking to green in that scenario, even after multiple subscribes and registers. Also, I just noticed that I'm receiving a peculiar response, a 501 Not Implemented from the phones in response to a PUBLISH (I swear it wasn't doing this before!)... -- James Hi James, The phones should never receive the Publish message. Please catch a trace containing this Publish and send it to me. What do you mean by before? Before updating from svn with my patch? Before I updated from 1.6.2 to SVN I think - I'll try and double-check. I have a revised patch that covers my case that I'll submit this afternoon with the trace. Any luck with the stuck BLF trace I submitted as well? I also think that my configuration might need some help wrt when and where to call presence functions - I will post that once I get in the office. I noticed that with svn I'm not getting any entries in pua or presentity, only in active_watchers. Thanks. -- James Regards, -- Anca Vamanu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
The phones should never receive the Publish message. Please catch a trace containing this Publish and send it to me. What do you mean by before? Before updating from svn with my patch? Before I updated from 1.6.2 to SVN I think - I'll try and double-check. I have a revised patch that covers my case that I'll submit this afternoon with the trace. Any luck with the stuck BLF trace I submitted as well? I also think that my configuration might need some help wrt when and where to call presence functions - I will post that once I get in the office. I noticed that with svn I'm not getting any entries in pua or presentity, only in active_watchers. Ok here's what I saw, running the 1.6 branch SVN HEAD - it is attached in bad_publish.txt. Also, I've attached my opensips.cfg to see if I'm doing anything bad in there as well. Thanks. -- James U opensips.ip:5060 - phone.nat.ip:8640 INVITE sip:000...@phone.nat.ip:8640 SIP/2.0..Record-Route: sip:opensips.ip;r2=on;lr=on;ftag=as0175fba8;did=9e5.a4630c74..Record-Route: sip:opensips.ip:5061;r2=on;lr=on;ftag=as0175fba8;did=9e5.a4630c74..Via: SIP/2.0/UDP opensips.ip;branch=z9hG4bK9b4b.3093b021.0..Via: SIP/2.0/UDP asterisk.server.ip:5060;received=asterisk.server.ip;branch=z9hG4bK58a5bbd7;rport=5060..From: 6266395478 sip:16266395...@asterisk.server.ip;tag=as0175fba8..To: sip:000...@opensips.ip:5061..Contact: sip:16266395...@asterisk.server.ip..Call-ID: 6355d0e82d3d60622ced249859e95...@asterisk.server.ip..cseq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 69..Date: Tue, 06 Apr 2010 15:12:11 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Type: application/sdp..Content-Length: 280..P-hint: usrloc appliedv=0..o=root 6818 6818 IN IP4 asterisk.server.ip..s=session..c=IN IP4 opensips.ip..t=0 0..m=audio 35470 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..a=nortpproxy:yes.. # U phone.nat.ip:8640 - opensips.ip:5060 SIP/2.0 100 Trying..To: sip:000...@opensips.ip:5061..From: 6266395478 sip:16266395...@asterisk.server.ip;tag=as0175fba8..Call-ID: 6355d0e82d3d60622ced249859e95...@asterisk.server.ip..cseq: 102 INVITE..Via: SIP/2.0/UDP opensips.ip;branch=z9hG4bK9b4b.3093b021.0..Via: SIP/2.0/UDP asterisk.server.ip:5060;received=asterisk.server.ip;branch=z9hG4bK58a5bbd7;rport=5060..Record-Route: sip:opensips.ip;r2=on;lr=on;ftag=as0175fba8;did=9e5.a4630c74..Record-Route: sip:opensips.ip:5061;r2=on;lr=on;ftag=as0175fba8;did=9e5.a4630c74..Server: Linksys/SPA942-6.1.3(a)..Content-Length: 0 # U phone.nat.ip:8640 - opensips.ip:5060 SIP/2.0 180 Ringing..To: sip:000...@opensips.ip:5061;tag=c3f142fab548cb0i0..From: 6266395478 sip:16266395...@asterisk.server.ip;tag=as0175fba8..Call-ID: 6355d0e82d3d60622ced249859e95...@asterisk.server.ip..cseq: 102 INVITE..Via: SIP/2.0/UDP opensips.ip;branch=z9hG4bK9b4b.3093b021.0..Via: SIP/2.0/UDP asterisk.server.ip:5060;received=asterisk.server.ip;branch=z9hG4bK58a5bbd7;rport=5060..Record-Route: sip:opensips.ip;r2=on;lr=on;ftag=as0175fba8;did=9e5.a4630c74..Record-Route: sip:opensips.ip:5061;r2=on;lr=on;ftag=as0175fba8;did=9e5.a4630c74..Contact: 000-000-0002 sip:000...@192.168.1.151:8640..Server: Linksys/SPA942-6.1.3(a)..Content-Length: 0 # U opensips.ip:5060 - phone.nat.ip:8640 PUBLISH sip:000...@phone.nat.ip:8640 SIP/2.0..Record-Route: sip:opensips.ip;lr=on;ftag=501689153a5b5fe31491a47f27dc82f5-5f2e..Via: SIP/2.0/UDP opensips.ip;branch=z9hG4bKa371.56493fb4.0..Via: SIP/2.0/UDP opensips.ip;branch=z9hG4bKa371.46493fb4.0..To: sip:000...@phone.nat.ip:8640..From: sip:000...@phone.nat.ip:8640;tag=501689153a5b5fe31491a47f27dc82f5-5f2e..CSeq: 10 PUBLISH..Call-ID: 77b9fe2016a10a4c-30...@opensips.ip..content-length: 627..User-Agent: OpenSIPS (1.6.2-notls (x86_64/linux))..Max-Forwards: 69..Event: dialog..Expires: 43201..Content-Type: application/dialog-info+xml..P-hint: outbound?xml version=1.0?.dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=0 state=full entity=sip:000...@phone.nat.ip:8640dialog id=6355d0e82d3d60622ced249859e95...@asterisk.server.ip call-id=6355d0e82d3d60622ced249859e95...@asterisk.server.ip local-tag=c3f142fab548cb0i0 remote-tag=as0175fba8 direction=recipientstateearly/stateremoteidentity display=6266395478sip:16266395...@asterisk.server.ip/identitytarget uri=sip:16266395...@asterisk.server.ip//remotelocalidentitysip:000...@phone.nat.ip:8640/identitytarget uri=sip:000...@phone.nat.ip:8640//local/dialog/dialog-info. # U phone.nat.ip:8640 - opensips.ip:5060 SIP/2.0 501 Not Implemented..To: sip:000...@phone.nat.ip:8640;tag=1ded7a2d320fb7c4i0..From: sip:000...@phone.nat.ip:8640;tag=501689153a5b5fe31491a47f27dc82f5-5f2e..Call-ID: 77b9fe2016a10a4c-30...@opensips.ip..cseq: 10 PUBLISH..Via: SIP/2.0/UDP
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
On 03/04/10 01:40, James Lamanna wrote:
On Fri, Apr 2, 2010 at 4:52 AM, Anca Vamanu a...@opensips.org wrote:
Hi James,
I think that the problem is with those Notifies without a body sent by
OpenSIPS when the phone was started. This is normal behavior, correct
and conform RFC - when the presence server does not have any record for
that presentity - it includes no body. But since you say that Linksys
does not like this and since it was not that difficult to change, I just
committed a patch that sends a Notify with an empty dialoginfo tuple as
body when no published record is found. Please upgrade from svn and test
this case again.
Hi Anca,
I saw your patch and upgraded to the SVN 1.6 branch.
However, I do not believe I ever hit that code path.
In my debug logs I see No record exists in hash table (notify.c:853).
I'm not sure if fallback2db is true, but I don't think it is, so I
never make it to that codepath.
Give the attached patch a shot, it should do what you're looking for.
Though I'm not sure generating an empty dialog without a presence entry
is the correct approach, I mean, the monitored extension is not there,
so why not hint it as a blinking light? In a cold start scenario, it'll
go green once your re-subscribe time comes, so set it to a couple
minutes or lower in the phone and you'll be good to go.
--
Angel Marin
http://anmar.eu.org/
Index: notify_body.c
===
--- notify_body.c (revision 6758)
+++ notify_body.c (working copy)
@@ -340,7 +340,6 @@
xmlNodePtr state_node = NULL;
str *body= NULL;
- str *pres_uri= NULL;
char buf[MAX_URI_SIZE+1];
if ( (pres_user-len + pres_domain-len + 1) MAX_URI_SIZE) {
@@ -353,23 +352,6 @@
memcpy(buf + pres_user-len + 5, pres_domain-s, pres_domain-len);
buf[pres_user-len + 5 + pres_domain-len]= '\0';
- pres_uri = (str*)pkg_malloc(sizeof(str));
- if(pres_uri == NULL)
- {
- LM_ERR(while allocating memory\n);
- return NULL;
- }
- memset(pres_uri, 0, sizeof(str));
- pres_uri-s = buf;
- pres_uri-len = pres_user-len + 5 + pres_domain-len;
-
- LM_DBG([pres_uri] %.*s\n, pres_uri-len, pres_uri-s);
-
- if ( pres_contains_presence(pres_uri)0 ) {
- LM_DBG(No record exists in hash_table\n);
- goto error;
- }
-
/* create the Publish body */
doc = xmlNewDoc(BAD_CAST 1.0);
if(doc==0)
@@ -427,10 +409,6 @@
xmlCleanupParser();
return body;
error:
- if ( pres_uri )
- {
- pkg_free(pres_uri);
- }
if(body)
{
if(body-s)
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
On Mon, Apr 5, 2010 at 12:52 AM, Angel Marin an...@anmar.eu.org wrote: On 03/04/10 01:40, James Lamanna wrote: On Fri, Apr 2, 2010 at 4:52 AM, Anca Vamanu a...@opensips.org wrote: Hi James, I think that the problem is with those Notifies without a body sent by OpenSIPS when the phone was started. This is normal behavior, correct and conform RFC - when the presence server does not have any record for that presentity - it includes no body. But since you say that Linksys does not like this and since it was not that difficult to change, I just committed a patch that sends a Notify with an empty dialoginfo tuple as body when no published record is found. Please upgrade from svn and test this case again. Hi Anca, I saw your patch and upgraded to the SVN 1.6 branch. However, I do not believe I ever hit that code path. In my debug logs I see No record exists in hash table (notify.c:853). I'm not sure if fallback2db is true, but I don't think it is, so I never make it to that codepath. Give the attached patch a shot, it should do what you're looking for. Though I'm not sure generating an empty dialog without a presence entry is the correct approach, I mean, the monitored extension is not there, so why not hint it as a blinking light? In a cold start scenario, it'll go green once your re-subscribe time comes, so set it to a couple minutes or lower in the phone and you'll be good to go. Actually, it never changes from blinking to green in that scenario, even after multiple subscribes and registers. Also, I just noticed that I'm receiving a peculiar response, a 501 Not Implemented from the phones in response to a PUBLISH (I swear it wasn't doing this before!)... -- James -- Angel Marin http://anmar.eu.org/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
Hi James, I think that the problem is with those Notifies without a body sent by OpenSIPS when the phone was started. This is normal behavior, correct and conform RFC - when the presence server does not have any record for that presentity - it includes no body. But since you say that Linksys does not like this and since it was not that difficult to change, I just committed a patch that sends a Notify with an empty dialoginfo tuple as body when no published record is found. Please upgrade from svn and test this case again. Regards, -- Anca Vamanu www.voice-system.ro James Lamanna wrote: On Thu, Apr 1, 2010 at 12:26 AM, Anca Vamanu a...@opensips.org wrote: [snip] Ok I think I got this somewhat working. I was missing a dialoginfo_set() in another INVITE path. However, does anyone know how, if you add a new phone, to make the presence initialize to idle? The BLF light blinks amber until I call the phone that is being monitored, then it will blink red, and go back to green when the call is terminated. Hi James, Can you please run a network trace and catch the Notify that goes first to the phone and makes it blink amber as you said? Send that to me to see how we can fix that. Hi Anca, Attached is a log of a startup of 2 phones. Each phone is monitoring the other phone. This is a clean startup, so all presence tables are empty. I think this may be a lack of any NOTIFYs being sent at startup to the phone that a phone is online. Thanks. -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
Hi James, For this case, indeed I see that when BYE is received the server generates a Notify with state confirmed.. which is really strange. Can you please run opensips in debug mode and catch the logs when BYE is processed. Send them to me in an e-mail. Thanks, -- Anca Vamanu www.voice-system.ro James Lamanna wrote: Also, I've found a case where the BLF light stays red, even when a call is hung up. This seems to happen in the intercom case, where the SIP URI is sip:u...@ip;intercom=true. It doesn't happen on every intercom call, but once it does happen, it is impossible to clear without clearing the presence tables and rebooting the phone.. -- James Can you please catch the traces for this case also? All the presence traffic that you see when you hang up the phone - the Publish generated by OpenSIPS and the Notify sent to the phone. Thanks, Hi Anca, Here's the full trace. I call one phone (01) from my cell phone, and 02 is monitoring it. When 01 hangs up, the light on 02 stays red. Thanks for all your help! -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
On Fri, Apr 2, 2010 at 4:52 AM, Anca Vamanu a...@opensips.org wrote: Hi James, I think that the problem is with those Notifies without a body sent by OpenSIPS when the phone was started. This is normal behavior, correct and conform RFC - when the presence server does not have any record for that presentity - it includes no body. But since you say that Linksys does not like this and since it was not that difficult to change, I just committed a patch that sends a Notify with an empty dialoginfo tuple as body when no published record is found. Please upgrade from svn and test this case again. Hi Anca, I saw your patch and upgraded to the SVN 1.6 branch. However, I do not believe I ever hit that code path. In my debug logs I see No record exists in hash table (notify.c:853). I'm not sure if fallback2db is true, but I don't think it is, so I never make it to that codepath. Apr 2 16:42:24 [12585] DBG:presence:send_notify_request: dialog info: Apr 2 16:42:24 [12585] DBG:presence:printf_subs: [pres_uri]= sip:000...@opensips.ip [to_user]= 02 [to_domain]= opensips.ip [w_user]= 01[w_domain]= opensips.ip [event]= dialog [status]= active [expires]= 60 [callid]= 9673ec38-b54b9...@192.168.1.165 [local_cseq]=0 [to_tag]= 89c56fdf6f5b6f30be24c8867d74b34a-8b2f [from_tag]= f9ba884d0d82aab [contact]= sip:000...@phone.nat.ip:6095 [record_route]= Apr 2 16:42:24 [12585] DBG:presence:search_phtable: pres_uri= sip:000...@opensips.ip Apr 2 16:42:24 [12585] DBG:presence:get_p_notify_body: No record exists in hash_table Apr 2 16:42:24 [12585] DBG:presence_dialoginfo:dlginfo_agg_nbody: [pres_user]=02 [pres_domain]= opensips.ip, [n]=0 Apr 2 16:42:24 [12585] DBG:presence_dialoginfo:build_dialoginfo: [pres_uri] sip:000...@opensips.ip Apr 2 16:42:24 [12585] DBG:presence:search_phtable: pres_uri= sip:000...@opensips.ip Apr 2 16:42:24 [12585] DBG:presence_dialoginfo:build_dialoginfo: No record exists in hash_table Apr 2 16:42:24 [12585] DBG:presence:send_notify_request: Could not get the notify_body Apr 2 16:42:24 [12585] DBG:presence:send_notify_request: headers: Max-Forwards: 70^M Event: dialog^M Contact: sip:s...@opensips.ip:5060^M Subscription-State: active;expires=50^M Apr 2 16:42:24 [12585] DBG:presence:build_dlg_t: CONTACT = sip:000...@phone.nat.ip:6095 Apr 2 16:42:24 [12585] DBG:tm:t_uac: next_hop=sip:000...@phone.nat.ip::6095 Apr 2 16:42:24 [12585] DBG:core:mk_proxy: doing DNS lookup... Apr 2 16:42:24 [12585] DBG:tm:dlg2hash: 63929 Apr 2 16:42:24 [12585] DBG:tm:print_request_uri: sip:000...@phone.nat.ip::6095 Apr 2 16:42:24 [12585] DBG:tm:set_timer: relative timeout is 50 Apr 2 16:42:24 [12585] DBG:tm:insert_timer_unsafe: [4]: 0x7f0f225b44d0 (580) Apr 2 16:42:24 [12585] DBG:tm:set_timer: relative timeout is 30 Apr 2 16:42:24 [12585] DBG:tm:insert_timer_unsafe: [0]: 0x7f0f225b4500 (35) Apr 2 16:42:24 [12585] INFO:presence:send_notify_request: NOTIFY sip:000...@opensips.ip via sip:000...@phone.nat.ip::6095 on Regards, -- Anca Vamanu www.voice-system.ro -- James James Lamanna wrote: On Thu, Apr 1, 2010 at 12:26 AM, Anca Vamanu a...@opensips.org wrote: [snip] Ok I think I got this somewhat working. I was missing a dialoginfo_set() in another INVITE path. However, does anyone know how, if you add a new phone, to make the presence initialize to idle? The BLF light blinks amber until I call the phone that is being monitored, then it will blink red, and go back to green when the call is terminated. Hi James, Can you please run a network trace and catch the Notify that goes first to the phone and makes it blink amber as you said? Send that to me to see how we can fix that. Hi Anca, Attached is a log of a startup of 2 phones. Each phone is monitoring the other phone. This is a clean startup, so all presence tables are empty. I think this may be a lack of any NOTIFYs being sent at startup to the phone that a phone is online. Thanks. -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
On Fri, Apr 2, 2010 at 4:40 PM, James Lamanna jlama...@gmail.com wrote: On Fri, Apr 2, 2010 at 4:52 AM, Anca Vamanu a...@opensips.org wrote: Hi James, I think that the problem is with those Notifies without a body sent by OpenSIPS when the phone was started. This is normal behavior, correct and conform RFC - when the presence server does not have any record for that presentity - it includes no body. But since you say that Linksys does not like this and since it was not that difficult to change, I just committed a patch that sends a Notify with an empty dialoginfo tuple as body when no published record is found. Please upgrade from svn and test this case again. Hi Anca, I saw your patch and upgraded to the SVN 1.6 branch. However, I do not believe I ever hit that code path. In my debug logs I see No record exists in hash table (notify.c:853). I'm not sure if fallback2db is true, but I don't think it is, so I never make it to that codepath. [snip] Setting fallback2db == 1 as a modparam doesn't hit it either because it aggregates the body. Also, that has another side effect of crashing opensips after about 1 minute as well. Last entry in the log in that case is termination due to SIGCHLD. Thanks. -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
James Lamanna wrote: On Wed, Mar 31, 2010 at 9:16 PM, James Lamanna jlama...@gmail.com wrote: On Wed, Mar 31, 2010 at 10:28 AM, Anca Vamanu a...@opensips.org wrote: James Lamanna wrote: Anca Vamanu Wrote: Andrew, this patch is already in 1.6.2 and trunk. James, the first thing that you need to check is that you receive Subscribes from the phones with event 'dialog'. And indeed as Andrew said, you need to load pua and pua_dialoginfo modules. Ok thanks. I'll upgrade to 1.6.2. Do I still need to explicitly call dialoginfo_set()? Yes, you have to call it. Hi Anca, I'm still having problems getting this to work at all. I've now upgraded to 1.6.2. Here is my entire config: [snip] Ok I think I got this somewhat working. I was missing a dialoginfo_set() in another INVITE path. However, does anyone know how, if you add a new phone, to make the presence initialize to idle? The BLF light blinks amber until I call the phone that is being monitored, then it will blink red, and go back to green when the call is terminated. Hi James, Can you please run a network trace and catch the Notify that goes first to the phone and makes it blink amber as you said? Send that to me to see how we can fix that. Regards, -- Anca Vamanu www.voice-system.ro Thanks. -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
James Lamanna wrote: On Wed, Mar 31, 2010 at 9:28 PM, James Lamanna jlama...@gmail.com wrote: On Wed, Mar 31, 2010 at 9:16 PM, James Lamanna jlama...@gmail.com wrote: On Wed, Mar 31, 2010 at 10:28 AM, Anca Vamanu a...@opensips.org wrote: James Lamanna wrote: Anca Vamanu Wrote: Andrew, this patch is already in 1.6.2 and trunk. James, the first thing that you need to check is that you receive Subscribes from the phones with event 'dialog'. And indeed as Andrew said, you need to load pua and pua_dialoginfo modules. Ok thanks. I'll upgrade to 1.6.2. Do I still need to explicitly call dialoginfo_set()? Yes, you have to call it. Hi Anca, I'm still having problems getting this to work at all. I've now upgraded to 1.6.2. Here is my entire config: [snip] Ok I think I got this somewhat working. I was missing a dialoginfo_set() in another INVITE path. However, does anyone know how, if you add a new phone, to make the presence initialize to idle? The BLF light blinks amber until I call the phone that is being monitored, then it will blink red, and go back to green when the call is terminated. Also, I've found a case where the BLF light stays red, even when a call is hung up. This seems to happen in the intercom case, where the SIP URI is sip:u...@ip;intercom=true. It doesn't happen on every intercom call, but once it does happen, it is impossible to clear without clearing the presence tables and rebooting the phone.. -- James Can you please catch the traces for this case also? All the presence traffic that you see when you hang up the phone - the Publish generated by OpenSIPS and the Notify sent to the phone. Thanks, -- Anca Vamanu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
On Thu, Apr 1, 2010 at 12:26 AM, Anca Vamanu a...@opensips.org wrote: [snip] Ok I think I got this somewhat working. I was missing a dialoginfo_set() in another INVITE path. However, does anyone know how, if you add a new phone, to make the presence initialize to idle? The BLF light blinks amber until I call the phone that is being monitored, then it will blink red, and go back to green when the call is terminated. Hi James, Can you please run a network trace and catch the Notify that goes first to the phone and makes it blink amber as you said? Send that to me to see how we can fix that. Hi Anca, Attached is a log of a startup of 2 phones. Each phone is monitoring the other phone. This is a clean startup, so all presence tables are empty. I think this may be a lack of any NOTIFYs being sent at startup to the phone that a phone is online. Thanks. -- James opensips_init.log Description: Binary data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
Also, I've found a case where the BLF light stays red, even when a call is hung up. This seems to happen in the intercom case, where the SIP URI is sip:u...@ip;intercom=true. It doesn't happen on every intercom call, but once it does happen, it is impossible to clear without clearing the presence tables and rebooting the phone.. -- James Can you please catch the traces for this case also? All the presence traffic that you see when you hang up the phone - the Publish generated by OpenSIPS and the Notify sent to the phone. Thanks, Hi Anca, Here's the full trace. I call one phone (01) from my cell phone, and 02 is monitoring it. When 01 hangs up, the light on 02 stays red. Thanks for all your help! -- James opensips_red.log Description: Binary data ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
Hi, Andrew, this patch is already in 1.6.2 and trunk. James, the first thing that you need to check is that you receive Subscribes from the phones with event 'dialog'. And indeed as Andrew said, you need to load pua and pua_dialoginfo modules. Regards, -- Anca Vamanu www.voice-system.ro Andrew Pogrebennyk wrote: James, Are you using pua_dialoginfo to get device state? If so are you telling the dialog module to monitor the interesting dialogs and calling dialoginfo_set()? Note that once you get this working you will likely need this fix: http://sourceforge.net/tracker/?func=detailatid=1086412aid=2847397group_id=232389 On 31.03.2010 08:31, James Lamanna wrote: Sorry, I realized I had a configuration error on my phone, but the presence still does not work. The phone now subscribes to the event: dialog. Here are relevant parts of my opensips config: modparam(presence, server_address, sip:s...@xxx.xxx.xxx.xxx:5060) modparam(presence, expires_offset, 10) modparam(presence_xml, force_active, 1) modparam(presence_dialoginfo, force_single_dialog, 1) I have also verified that handle_subscribe() is being called when a SUBSCRIBE message comes in. Calling the phone doesn't seem to produce any PUBLISH messages or anything pertaining to presence. Thanks. -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
On Wed, Mar 31, 2010 at 10:28 AM, Anca Vamanu a...@opensips.org wrote:
James Lamanna wrote:
Anca Vamanu Wrote:
Andrew, this patch is already in 1.6.2 and trunk.
James, the first thing that you need to check is that you receive
Subscribes from the phones with event 'dialog'. And indeed as Andrew
said, you need to load pua and pua_dialoginfo modules.
Ok thanks. I'll upgrade to 1.6.2.
Do I still need to explicitly call dialoginfo_set()?
Yes, you have to call it.
Hi Anca,
I'm still having problems getting this to work at all.
I've now upgraded to 1.6.2.
Here is my entire config:
debug=3 # debug level (cmd line: -dd)
fork=yes
log_stderror=no # (cmd line: -E)
log_facility=LOG_LOCAL0
tos=0x60
# Uncomment these lines to enter debugging mode
#fork=no
log_stderror=yes
debug=6
check_via=no# (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
listen=udp:my.ip.address:5060
listen=udp:my.ip.address:5061
# -- module loading --
#set module path
#mpath=/usr/local/lib/opensips/modules/
mpath=/usr/local/lib64/opensips/modules/
# Uncomment this if you want to use SQL database
loadmodule db_mysql.so
loadmodule sl.so
loadmodule maxfwd.so
loadmodule textops.so
loadmodule avpops.so
loadmodule tm.so
loadmodule rr.so
loadmodule dialog.so
loadmodule signaling.so
loadmodule options.so
loadmodule localcache.so
loadmodule usrloc.so
loadmodule presence.so
loadmodule presence_xml.so
loadmodule presence_dialoginfo.so
loadmodule pua.so
loadmodule pua_dialoginfo.so
#loadmodule pua_bla.so
loadmodule pua_usrloc.so
loadmodule registrar.so
loadmodule mi_fifo.so
loadmodule xlog.so
# Uncomment this if you want digest authentication
# db_mysql.so must be loaded !
loadmodule auth.so
loadmodule auth_db.so
# !! Nathelper
loadmodule nathelper.so
# - setting module-specific parameters ---
# -- mi_fifo params --
modparam(mi_fifo, fifo_name, /tmp/opensips_fifo)
modparam(usrloc, db_mode, 2)
modparam(usrloc|dialog|dispatcher|presence|presence_xml|pua|avpops,
db_url, mysql://opensips:myp...@localhost/opensips)
modparam(avpops,avp_table,usr_preferences)
#modparam(dispatcher, force_dst, 1)
# Only use username
#modparam(dispatcher, flags, 1)
# Store passwords for 1 hour in cache
modparam(auth,username_spec,$avp(i:54))
modparam(auth,password_spec,$avp(i:55))
modparam(auth,calculate_ha1,1)
modparam(auth_db, db_url,
mysql://opensipsro:mypas...@localhost/opensips)
modparam(auth_db, calculate_ha1, yes)
modparam(auth_db, password_column, password)
modparam(auth_db, load_credentials, $avp(i:55)=password)
modparam(rr, enable_full_lr, 1)
modparam(dialog, dlg_flag, 4)
modparam(dialog, profiles_with_value, caller)
modparam(usrloc,nat_bflag,6)
modparam(nathelper,sipping_bflag,8)
#modparam(nathelper, natping_interval, 30)
modparam(nathelper, ping_nated_only, 1) # Ping only clients behind NAT
#modparam(nathelper, natping_interval, 30)
modparam(nathelper, sipping_from, sip:pin...@my.ip.address)
modparam(nathelper, rtpproxy_sock, unix:/var/run/rtpproxy/rtpproxy.sock)
modparam(presence, server_address, sip:s...@my.ip.address:5060)
modparam(presence, expires_offset, 10)
modparam(presence_xml, force_active, 1)
modparam(presence_dialoginfo, force_single_dialog, 1)
modparam(pua_dialoginfo, presence_server, sip:s...@my.ip.address:5060)
modparam(pua_dialoginfo, include_callid, 1)
modparam(pua_dialoginfo, include_tags, 1)
modparam(pua_dialoginfo, caller_confirmed, 1)
modparam(pua_usrloc, default_domain, my.ip.address)
modparam(pua_usrloc, presence_server, sip:s...@my.ip.address:5060)
# - request routing logic ---
# main routing logic
route{
if (!is_method(NOTIFY))
xlog(L_INFO, New request - Request/failure/branch routes:
M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n);
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483,Too Many Hops);
exit;
};
if (msg:len = 2048 ) {
sl_send_reply(513, Message too big);
exit;
};
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence of received
if (nat_uac_test(3)) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (is_method(REGISTER) || !is_present_hf(Record-Route)) {
#xlog(L_INFO, LOG:Someone trying to register from
private IP,
rewriting\n);
#xlog(L_INFO, $rb\n);
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
On Wed, Mar 31, 2010 at 9:16 PM, James Lamanna jlama...@gmail.com wrote: On Wed, Mar 31, 2010 at 10:28 AM, Anca Vamanu a...@opensips.org wrote: James Lamanna wrote: Anca Vamanu Wrote: Andrew, this patch is already in 1.6.2 and trunk. James, the first thing that you need to check is that you receive Subscribes from the phones with event 'dialog'. And indeed as Andrew said, you need to load pua and pua_dialoginfo modules. Ok thanks. I'll upgrade to 1.6.2. Do I still need to explicitly call dialoginfo_set()? Yes, you have to call it. Hi Anca, I'm still having problems getting this to work at all. I've now upgraded to 1.6.2. Here is my entire config: [snip] Ok I think I got this somewhat working. I was missing a dialoginfo_set() in another INVITE path. However, does anyone know how, if you add a new phone, to make the presence initialize to idle? The BLF light blinks amber until I call the phone that is being monitored, then it will blink red, and go back to green when the call is terminated. Thanks. -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
On Wed, Mar 31, 2010 at 9:28 PM, James Lamanna jlama...@gmail.com wrote: On Wed, Mar 31, 2010 at 9:16 PM, James Lamanna jlama...@gmail.com wrote: On Wed, Mar 31, 2010 at 10:28 AM, Anca Vamanu a...@opensips.org wrote: James Lamanna wrote: Anca Vamanu Wrote: Andrew, this patch is already in 1.6.2 and trunk. James, the first thing that you need to check is that you receive Subscribes from the phones with event 'dialog'. And indeed as Andrew said, you need to load pua and pua_dialoginfo modules. Ok thanks. I'll upgrade to 1.6.2. Do I still need to explicitly call dialoginfo_set()? Yes, you have to call it. Hi Anca, I'm still having problems getting this to work at all. I've now upgraded to 1.6.2. Here is my entire config: [snip] Ok I think I got this somewhat working. I was missing a dialoginfo_set() in another INVITE path. However, does anyone know how, if you add a new phone, to make the presence initialize to idle? The BLF light blinks amber until I call the phone that is being monitored, then it will blink red, and go back to green when the call is terminated. Also, I've found a case where the BLF light stays red, even when a call is hung up. This seems to happen in the intercom case, where the SIP URI is sip:u...@ip;intercom=true. It doesn't happen on every intercom call, but once it does happen, it is impossible to clear without clearing the presence tables and rebooting the phone.. -- James Thanks. -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
Hi, I'm trying to get presence (BLF) working with some Linksys 942 phones. I've noticed that I get the error, handle_subscribe: Missing or unsupported event header field value I did a trace and the phone is trying to subscribe to the x-spa-cti event. Is there a way to support/fix this? Is there something that supports that event? Thank you. This is opensips 1.6.1. -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
Sorry, I realized I had a configuration error on my phone, but the presence still does not work. The phone now subscribes to the event: dialog. Here are relevant parts of my opensips config: modparam(presence, server_address, sip:s...@xxx.xxx.xxx.xxx:5060) modparam(presence, expires_offset, 10) modparam(presence_xml, force_active, 1) modparam(presence_dialoginfo, force_single_dialog, 1) I have also verified that handle_subscribe() is being called when a SUBSCRIBE message comes in. Calling the phone doesn't seem to produce any PUBLISH messages or anything pertaining to presence. Thanks. -- James On Tue, Mar 30, 2010 at 7:56 PM, James Lamanna jlama...@gmail.com wrote: Hi, I'm trying to get presence (BLF) working with some Linksys 942 phones. I've noticed that I get the error, handle_subscribe: Missing or unsupported event header field value I did a trace and the phone is trying to subscribe to the x-spa-cti event. Is there a way to support/fix this? Is there something that supports that event? Thank you. This is opensips 1.6.1. -- James ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Presence and Linksys phones - not working 1.6.1
James, Are you using pua_dialoginfo to get device state? If so are you telling the dialog module to monitor the interesting dialogs and calling dialoginfo_set()? Note that once you get this working you will likely need this fix: http://sourceforge.net/tracker/?func=detailatid=1086412aid=2847397group_id=232389 On 31.03.2010 08:31, James Lamanna wrote: Sorry, I realized I had a configuration error on my phone, but the presence still does not work. The phone now subscribes to the event: dialog. Here are relevant parts of my opensips config: modparam(presence, server_address, sip:s...@xxx.xxx.xxx.xxx:5060) modparam(presence, expires_offset, 10) modparam(presence_xml, force_active, 1) modparam(presence_dialoginfo, force_single_dialog, 1) I have also verified that handle_subscribe() is being called when a SUBSCRIBE message comes in. Calling the phone doesn't seem to produce any PUBLISH messages or anything pertaining to presence. Thanks. -- James -- Sincerely, Andrew Pogrebennyk ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
