[asterisk-users] XMPP issues in Asterisk 11.6.0 for distributed device states...
I have been working with distributed device states in Asterisk using XMPP attached to an OpenFire server. I have it working well across two servers and want to roll it out across every server in my company. All servers are Asterisk 11.6.0. I am running into a problem that seems like it should be a bit easier to solve than it is seeming to be. On the third server I am rolling into this solution, I get plenty of the following: res_xmpp.c:1398 xmpp_pubsub_handle_error: Error performing operation on PubSub node device_state, 403. So, basically, servers 1 and 2 continue to hum along nicely updating their device state, but server 3 gets a 403 forbidden message when it tries to deal with device state. I believe this has to do with the permissions set up on the device state node. I have a small example that demonstrates the creation of a new node. In the Asterisk CLI, I ran 'xmpp create collection asterisk test' on server 3, which was successful and can be seen on servers 1 and 2 with 'xmpp list nodes asterisk' The debug output from server 3 for this is as follows: --- XMPP sent to 'asterisk' --- iq to='pubsub.xmpp' from='server3@xmpp/astvoip3' type='set' id='aaacy' pubsub xmlns='http://jabber.org/protocol/pubsub' create node='test'/ configure x xmlns='jabber:x:data' type='submit' field var='FORM_TYPE' type='hidden' valuehttp://jabber.org/protocol/pubsub#owner/value /field field var='pubsub#node_type' valuecollection/value /field field var='FORM_TYPE' type='hidden' valuehttp://jabber.org/protocol/pubsub#node_config/value /field field var='pubsub#deliver_payloads' value1/value /field field var='pubsub#persist_items' value1/value /field field var='pubsub#access_model' valuewhitelist/value /field /x /configure /pubsub /iq - --- XMPP sent to 'asterisk' --- iq to='pubsub.xmpp' from='server3@xmpp/astvoip3' type='set' id='aaacz' pubsub xmlns='http://jabber.org/protocol/pubsub#owner' affiliations node='test' affiliation jid='server1@xmpp' affiliation='owner'/ affiliation jid='server2@xmpp' affiliation='owner'/ affiliation jid='server1@xmpp/astvoip1' affiliation='owner'/ affiliation jid='server2@xmpp/astvoip2' affiliation='owner'/ /affiliations /pubsub /iq - As we can see, the first message creates the test node and sets the access model to whitelist, so only jids in the whitelist are allowed to modify it. The second message then sets the appropriate server 1 and server 2 jids to be owners, thus meeting the requirements of the whitelist. Since these nodes are persistent, it would appear that server 3 cannot properly access device_state because it was never whitelisted when the node was created originally. I am fairly certain that I can solve this by deleting all my nodes and letting them be recreated, but that seems extreme as I put more servers into the system. Any thoughts on a better way to handle xmpp and making sure new servers can access the proper nodes? Kevin Larsen - Systems Analyst - Pioneer Balloon Company-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] modify from field sip headers
Hello, Im trying to modify the 'From' field in my sip headers in order to include extra info (user=tel) as it follows: From : sip:005114824403@200.91.0.146;user =tel However asterisk is still doing this header: sip:111@1.1.1.1;tag=as167b4b82 Is there a way to accomplish this? Ive been also looking up AGI but without any success. Ideas are welcome! Kind regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] modify from field sip headers
Im trying to modify the 'From' field in my sip headers in order to
include extra info (user=tel) as it follows:
The default extensions.conf has this, it might help.
;---
; from-pstn-to-did
;
; The context is designed for providers who send the DID in the TO: SIP
header
; only. The format of this header is:
;
; To: sip:2125551212@172.31.74.25
;
; So the DID must be extracted between the sip: and the @, which this does
;
[from-pstn-toheader]
exten = _.,1,Goto(from-pstn,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
;---
--
_
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[asterisk-users] Which is more efficient for 1 to many broadcasting?
Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it just disregard those channels and only run mixes against unmuted channels? Now, if the answer is MOH is more efficient, can someone suggest a way for a channel to be the source of a MOH class? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which is more efficient for 1 to many broadcasting?
On Tue, Mar 18, 2014 at 1:02 PM, James Sharp ja...@fivecats.org wrote: Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it just disregard those channels and only run mixes against unmuted channels? Now, if the answer is MOH is more efficient, can someone suggest a way for a channel to be the source of a MOH class? What sort of channel count are you looking for? We did some load testing recently and found less people in a bridge is better then more. Audio source location didn't really matter much. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which is more efficient for 1 to many broadcasting?
On 3/18/2014 6:58 PM, Paul Belanger wrote: On Tue, Mar 18, 2014 at 1:02 PM, James Sharp ja...@fivecats.org wrote: Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it just disregard those channels and only run mixes against unmuted channels? Now, if the answer is MOH is more efficient, can someone suggest a way for a channel to be the source of a MOH class? What sort of channel count are you looking for? We did some load testing recently and found less people in a bridge is better then more. Audio source location didn't really matter much. A few hundred to start with, but as with everything, I'd like to scale up as far as I can. And, of course, it makes sense that less people in a bridge is better than more but that's not quite what I'm asking. Is it more efficient to have, for example, 701 people in a confbridge room (700 muted users + 1 person yapping) or to have 700 people dialed in and just running the MusicOnHold application with said person yapping away via some audio source. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
