Re: [asterisk-users] sip-user status
On 12/13/2012 11:39 PM, Hans Witvliet wrote: Hi all, I'm caught up in a struggle between people how can not make up their mind... Half way implementing a asterisk farm and they come up with another feature they've seen in kamaillo. What he showed me was this: three registered sip users, a) one changes his presence status on his softphone, and all see the status change. b) one calls another, and the third person see the status of the other two change to busy. I've seen code/dialplan snippets where you could change your status by dialling a specific extension, on which asterisk will react (and change some variables accordingly), but that is not what i'm looking for. It seems that kamaillo has build-in features to react on sip-simple changes. Can i perform the same trick with asterisk? if so, how? Hans. In * this is done via hints. The devices register with * that they want to be notified when the status of what they want to monitor changes. We, when * knows that it is doing something with the device, * changes the hint status of said device and then sends the notification of status change to the awaiting devices. -- Jim Lucas -- _ -- 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] Doubt regarding jabber
I have Asterisk server 1.8.19 with jabber enabled. On the other side i have openfire server with asterisk-im enabled. I have a doubt, whether my sip client connected with asterisk can send message to other sip client, which is connected to same asterisk server. I have jitsi as a sip client. If its possible. Than please suggest any documentation regarding this. any help?? THanks a lot Regards Asteriskhelp -- _ -- 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] Possible bug - queue doesn't play hold music
Hi Can someone else please check the following: We have installed asterisk 1.8.18.0 onto our development and test servers. They were previously on 1.8.7.0 When an inbound call executes a queue, I can see in the logs that the hold music is supposed to start playing but there is no sound. If the call is answered and the callee puts the caller on hold, I can see the same log message of hold music starting but this time the hold music can be heard. This is happening on both installations of 1.8.18.0. If other people are experiencing the same thing we can raise a bug on it. Log excerpts below with my comments after a # symbol -- Executing [s@ethn-xx-work:4] Queue(SIP/x.x.x.x-0061, test-ish,Tn,,,600) -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: no music heard == Using SIP RTP CoS mark 5 -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 answered SIP/x.x.x.x-0061 -- Stopped music on hold on SIP/x.x.x.x-0061 [2012-12-14 14:44:04] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: music can be heard [2012-12-14 14:44:12] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Stopped music on hold on SIP/x.x.x.x-0061 == Spawn extension (ethn-xx-work, s, 4) exited non-zero on 'SIP/x.x.x.x-0061' -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] It's possible a redudant Queue?
Hi all, I have a doubt. I have to create a queue with 3 phones, these phones can be reached via two redudant Asterisk server. I can pass a variable (the sip trunks) to the queue or should I do two queues with the different trunks? Danilo -- -- _ -- 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] It's possible a redudant Queue?
In my experience, you should set up two identical queues and configurations. With a little work, you should be able to let server 1 know the phone is in use by server 2 and vice versa. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi Sent: Friday, December 14, 2012 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] It's possible a redudant Queue? Hi all, I have a doubt. I have to create a queue with 3 phones, these phones can be reached via two redudant Asterisk server. I can pass a variable (the sip trunks) to the queue or should I do two queues with the different trunks? Danilo -- -- _ -- 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 -- _ -- 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] It's possible a redudant Queue?
Ok. I solved with this configuration:
into /etc/asterisk/extensions.conf
[queue_from_central]
exten=_ZXXX,1,NoOp( ** Into queue_from_central context ** )
same=
n,Set(peer_up=${IF($[SIPPEER(serverA,status)=OK]?trunk_serverA:trunk_serverB)})
same= n,Queue(call-center-${peer_up},c,,,60)
same=Hangup()
and into /etc/asterisk/queues.conf
[call-center-trunk_serverA]
music=default
strategy=rrmemory
retry=5
wrapuptime=10
announce-frequency=30
announce-position=yes
;; SIP USER
member = SIP/22001@trunk_serverA
member = SIP/22002@trunk_serverA
member = SIP/22003@trunk_serverA
[call-center-trunk_serverB]
music=default
strategy=rrmemory
retry=5
wrapuptime=10
announce-frequency=30
announce-position=yes
;; SIP USER
member = SIP/22001@trunk_serverB
member = SIP/22002@trunk_serverB
member = SIP/22003@trunk_serverB
Thanks Danny ;)
Bye :)
Il 14/12/12 16:59, Danny Nicholas ha scritto:
In my experience, you should set up two identical queues and configurations.
With a little work, you should be able to let server 1 know the phone is in
use by server 2 and vice versa.
--
--
_
-- 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] BRI D-channel goes up and down
Hi, I have a B410P card with span ports set up as span=3,1,0,CCS,AMI span=4,2,0,CCS,AMI span=5,3,0,CCS,AMI signalling = bri_cpe switchtype = euroisdn layer1_presence = ignore However, I keep getting these messages over and over again: [Dec 14 18:53:14] WARNING[22476]: sig_pri.c:1150 pri_find_dchan: Span 3: D-channel is down! == Primary D-Channel on span 3 up == Primary D-Channel on span 4 up == Primary D-Channel on span 5 down [Dec 14 18:53:25] WARNING[22478]: sig_pri.c:1150 pri_find_dchan: Span 5: D-channel is down! == Primary D-Channel on span 5 up == Primary D-Channel on span 4 down [Dec 14 18:53:30] WARNING[22477]: sig_pri.c:1150 pri_find_dchan: Span 4: D-channel is down! == Primary D-Channel on span 3 down [Dec 14 18:53:30] WARNING[22476]: sig_pri.c:1150 pri_find_dchan: Span 3: D-channel is down! == Primary D-Channel on span 4 up == Primary D-Channel on span 3 up It seems I can dial out and in but I'm afraid I may be losing some calls if they happen to dial in/out right when a span goes down. libpri-1.4.13 dahdi-2.6.1 asterisk-11.0.1 Any suggestions? Thanks, Vieri -- _ -- 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] BRI D-channel goes up and down
I have a B410P card with span ports set up as span=3,1,0,CCS,AMI span=4,2,0,CCS,AMI span=5,3,0,CCS,AMI signalling = bri_cpe switchtype = euroisdn layer1_presence = ignore However, I keep getting these messages over and over again: [Dec 14 18:53:14] WARNING[22476]: sig_pri.c:1150 pri_find_dchan: Span 3: D-channel is down! == Primary D-Channel on span 3 up == Primary D-Channel on span 4 up == Primary D-Channel on span 5 down [Dec 14 18:53:25] WARNING[22478]: sig_pri.c:1150 pri_find_dchan: Span 5: D-channel is down! == Primary D-Channel on span 5 up == Primary D-Channel on span 4 down [Dec 14 18:53:30] WARNING[22477]: sig_pri.c:1150 pri_find_dchan: Span 4: D-channel is down! == Primary D-Channel on span 3 down [Dec 14 18:53:30] WARNING[22476]: sig_pri.c:1150 pri_find_dchan: Span 3: D-channel is down! == Primary D-Channel on span 4 up == Primary D-Channel on span 3 up It seems I can dial out and in but I'm afraid I may be losing some calls if they happen to dial in/out right when a span goes down. libpri-1.4.13 dahdi-2.6.1 asterisk-11.0.1 You will not lose any calls. The BRI (layer 1), Q.921 (layer 2), and Q.931 (layer 3) specifications were designed for this behavior. The telco is bringing the protocol layers down to conserve power. Astersk/libpri currently does not initiate bringing down the protocol layers. If the telco has an incoming call, it will bring the layers back up before presenting you with the call. For outgoing calls, the layers are brought back up as well before the outgoing call is given to the telco. Richard -- _ -- 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
