Re: [asterisk-users] How to give users the capability to set CDR userfield for some calls
2013/1/18 Danny Nicholas da...@debsinc.com
Since Gosub is technically an application, you should be able to modify
this snippet in features.conf
testfeature = #9,peer,Playback,tt-monkeys ;Allow both the caller and
callee to play
;;tt-monkeys to the opposite
channel
To this
testfeature = #9,peer,Gosub,play-monkeys,s,1 ;Allow both the caller and
callee to play
;;tt-monkeys to the opposite
channel
And in extensions.conf add
[play-monkeys]
Exten = s,1,playback(tt-monkeys)
Exten = s,n,return()
Unfortunately, I couldn't make the above work (with Asterisk 1.8/FreePBX
2.10) nore I could pass arguments to a custom Macro.
Can you confirm it worked on your setup ? Which asterisk version did you
use ?
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
*Sent:* Friday, January 18, 2013 3:35 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] How to give users the capability to set
CDR userfield for some calls
** **
** **
2013/1/17 Kevin Larsen kevin.lar...@pioneerballoon.com
Possibly switch to using subroutines instead of Macros. Macros are being
deprecated in place of subroutines.
Interesting thing to try.
The trouble is I can't find any usable example of calling Gosub routines
from features.conf's application map.
I've found old references explaining that this is not supported but I
don't if it's still valid or not.
Any ex
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From:Olivier oza_4...@yahoo.fr
To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date:01/17/2013 10:29 AM
Subject:[asterisk-users] How to give users the capability to set
CDRuserfield for some calls
Sent by:asterisk-users-boun...@lists.digium.com
--
Hello,
To my surprise, with asterisk 1.8 (I've not tried with other versions), it
seems you cannot set CDR's userfield from within a dialplan macro called by
dynamic features.
See :
testfeature = *321,self/callee,Macro,toto
[macro-toto]
exten = s,1,Verbose(0,Into macro-toto with CDR(src) set to ${CDR(src)})
exten = s,n,Set(CDR(userfield)=foobar)
I'm planning to use this feature to let users mark in CDR an ongoing call
as malicious or important or whatever.
Any hint ?
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
--
_
-- 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
--
_
-- 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] Queues and distributed device state over WAN
Oliver wrote:
snip
Before diving into this, I've got the following question :
- let say we have two Asterisk servers A and B,
- both are interconnected through PSTN (no SIP trunk)
- agent Alice's phone is connected (ie registered) to server A
- Alice's phone can be reached from server A phones dialing
Local/6789
- Alice's phone can also be reached from server B phones dialing
Local/00123456789
1. How do you configure both servers so that Alice's phone becomes a
Queue Member from a server B given queue ?
Simply calling AddQueueMember on server B, passing Local/00123456789
as interface value (ie AddQueueMember second argument) ?
2. Then, how should server A publish Alice's phone status ?
How should server B consume Alice's phone status and associate it
with the Queue member activity ? Using AddQueueMember stateinterface
argument ?
Before trying to get distributed device state going:
Device names across all Asterisk proxy's participating in
'distributed device state' need to be unique.
IE. You can't have 'SIP/cisco1' exist on server A for ALICE, and
SIP/cisco1 on server B for BOB.
You need to get XMPP distributed device state working.
I followed
https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMP
P+PubSub
You need a reasonbly reliable WAN links to the jabber server support the
XMPP updates between servers.
Asterisk segfaults if it can't contact the jabber server!!! See
https://issues.asterisk.org/jira/browse/ASTERISK-18078
Then:
With Alice reqistered on Server A as SIP/cisco1
With Server B hosting the queue named 'queue1'.
;(on Server B)
queues.conf:
[queue1]
;what makes this work with distributed states is the 'State Interface'
parameter
...
member = Local/00123456789,0,ALICE,SIP/cisco1
Alice will need a number to ring to login/logout of queue1 hosted on Server
B;
Dialplan Example: on server B:
...
exten = s,n,Set(queuename=queue1)
exten = s,n,Set(interface=Local/00123456789)
exten = s,n,Set(penalty=0)
exten = s,n,Set(stateinterface=SIP/cisco1)
exten =
s,n(queue-add),AddQueueMember(${queuename},${interface},${penalty},options,,
${stateinterface})
And to remove the member;
...
exten = s,n(queue-remove),RemoveQueueMember(${queuename},${interface})
Alec Davis
--
_
-- 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] Realtime vs Static Files
Upgrading to the latest version didn't help. After about 30 minutes, Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as Registered on Asterisk1. It is something really amazing... Can you run sip show peers on each one of the servers and post the response? You said the second asterisk is completely opaque to your peers. Can you run a tcpdump on secondary server to see if for some obscure reason the phones try to contact the secondary asterisk? I'll monitor one peer using tcpdump over a few hours and then review the packets. However, SIP DEBUG isn't showing any REGISTER packets. Here's the sip peers output. Values and names have been hidden. Some appear as Unreachable the secondary server and some appear as OK. I think some are listed as OK because the endpoint routers are performing some type of SIP ALG and routing packets based on port number and not source ip address. However, from the SIP DEBUG output, it seems clear that the secondary server in this example is sending out Keepalives based on the information that the primary server has entered into the realtime DB. Show peers Output from a primary server Name/username HostDyn Forcerport ACL Port Status Realtime a201/A201 217.x.x.48D N 65229OK (88 ms) Cached RT a202 (Unspecified)D N 0UNREACHABLE Cached RT b201/44845287 78.x.x.101 D N 5060 OK (26 ms) Cached RT c201/s 193.x.x.174 D N 5060 OK (52 ms) Cached RT d201/d201 94.x.x.228 D N 5060 OK (33 ms) Cached RT e201/e20194.x.x.44 D N 55018 OK (40 ms) Cached RT e202/e20294.x.x.44 D N 55022 OK (46 ms) Cached RT e203/e20394.x.x.44 D N 55024 OK (40 ms) Cached RT e204/e20494.x.x.44 D N 55008 OK (40 ms) Cached RT e205/e20594.x.x.44 D N 55016 OK (41 ms) Cached RT e206/e20694.x.x.44 D N 55014 OK (40 ms) Cached RT e207/e20794.x.x.44 D N 55020 OK (41 ms) Cached RT e208/e20894.x.x.44 D N 5060 OK (41 ms) Cached RT e209/e20994.x.x.44 D N 55012 OK (40 ms) Cached RT e210/e21094.x.x.44 D N 55010 OK (41 ms) Cached RT e211/e21194.x.x.44 D N 55026 OK (38 ms) Cached RT e212/e21281.x.x.93D N 5060 OK (46 ms) Cached RT f201 (Unspecified)D N 0 UNREACHABLE Cached RT g201/g 78.x.x.207 D N 5060 OK (29 ms) Cached RT h201/h201 217.x.x.78 D N 38980OK (22 ms) Cached RT i201 (Unspecified)D N 0 UNREACHABLE Cached RT i203/ i203 109.x.x.103 D N 5060 OK (32 ms) Cached RT i204/ i204 109.x.x.103 D N 1025 OK (31 ms) Cached RT i205/ i205 81.x.x.144 D N 5060 OK (32 ms) Cached RT i206/ i206 109.x.x.103 D N 1035 OK (31 ms) Cached RT i207/ i207 109.x.x.103 D N 1032 OK (32 ms) Cached RT i208/ i208 109.x.x.103 D N 1024 OK (31 ms) Cached RT j201/s 94.x.x.62D N 57813 OK (35 ms) Cached RT o201/o201 92.x.x.86 D N 51824 OK (47 ms) Cached RT o202/o202 92.x.x.86 D N 58641 OK (48 ms) Cached RT o203/o203 92.x.x.86 D N 49172 OK (47 ms) Cached RT j204/j204 176.x.x.214 D N 34824 OK (49 ms) Cached RT k201/k201 2.x.x.169 D N 52757OK (53 ms) Cached RT k202/k202 (Unspecified)D N 0 UNKNOWNCached RT l201/l201(Unspecified)D N 0 UNKNOWNCached RT m201/s 92.x.x.95 D N 54020 OK (32 ms) Cached RT n201 (Unspecified)
Re: [asterisk-users] question on SIP trunk and AMI to place call
The Dial events are created by app_dial. So long as you are using app_dial to create your outbound channel, you should have that event. Channel technology shouldn't matter. I am using the same AMI method to start both calls. Action: Originate Channel: DAHDI/18/XX or Action: Originate Channel: SIP/machine/XX Jerry -- _ -- 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] Integration with Social Media, Email and Web call center
I was thinking of something similar, maybe using the URL field of the queue() app as to point to an internal broker that will then link to the message being used. In theory one could do this for all kinds of traffic, including e-mails. The part I don't really like is keeping an audio call open for the duration of the job, but it plays very well with existing queues. In the end, I guess Matt is always right :) l. 2013/1/24 Matt Riddell li...@venturevoip.com In the past I've sent calls to an agent in the queue with music on hold that contained a beep every 20 seconds (to remind them they're on a call) and then used the same code I do for screen popping to send them alternative records. I.E. web page, email, fax etc. It's stored in the database that that's what they were working on and then when they finished working on it they just hang up or press * to disconnect the call. That way you can use the standard Asterisk queues and they don't get bothered by anything else while they're working on it. Facebook might be a little harder as you wouldn't necessarily know when an incoming request came. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- 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] Is there a need to secure RTP ports?
2013-01-23 18:20, Sebastian Arcus skrev: I have an Asterisk server with one SIP trunk to a SIP provider. As my server registers with the SIP provider, I don't have any SIP ports open at my end to the Internet. However, I have the RTP ports open (as SIP has some trouble with my NAT). You could try iptables with ip_conntrack_sip ip_nat_sip. If they are loaded and you accept calls from your sip provider on port 5060 iptables inspects the sip/sdp and traffic from the endpoints are considered RELATED. I've some research/testing to do myself on this topic (it's on my always growing todo-list of doom.. :-) Maybe you should check it out? -- Johan Wilfer -- _ -- 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] CEL / CELGenUserEvent via AGI / no error and no cel entry
Hi, I am using Asterisk 11.2.0. Channel Event Logging (CEL) ist activated and running. CEL entries are logged into an mysql database. So far so good. I want to do some extra cel logging and try the following via an AGI-Script: EXEC CELGenUserEvent test In the asterisk logfile I can see the following: -- AGI Script Executing Application: (CELGenUserEvent) Options: (test) (no errors or warnings) But there is no cel entry in my database. What is going wrong here, please? Best regards, -Thorsten- -- _ -- 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] Frames with invalid timing info
I'm now getting these errors: [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891164, src=RTP [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891174, src=RTP even *without* any transcoding. Suggestions? -- _ -- 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] question on SIP trunk and AMI to place call
I just put a break at dial_exec_full (app/app_dial.c for Asterisk 11.0.2) did my AMI call Action: Originate Async: yes Channel: SIP/testsystem/XXX (calls from my machine over SIP trunk to another 11.0.2 box that has a PRI card to make a call out to my cell) and did not get a break. Why is a SIP call not logging the Dial event as a DAHDI call does??? jerry -- _ -- 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] Realtime vs Static Files
2013/1/25 Dan Journo d...@keshercommunications.com Upgrading to the latest version didn't help. After about 30 minutes, Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as Registered on Asterisk1. It is something really amazing... Can you run sip show peers on each one of the servers and post the response? ** ** You said the second asterisk is completely opaque to your peers. Can you run a tcpdump on secondary server to see if for some obscure reason the phones try to contact the secondary asterisk? ** ** I'll monitor one peer using tcpdump over a few hours and then review the packets. However, SIP DEBUG isn't showing any REGISTER packets. ** ** Here's the sip peers output. Values and names have been hidden. Some appear as Unreachable the secondary server and some appear as OK. I think some are listed as OK because the endpoint routers are performing some type of SIP ALG and routing packets based on port number and not source ip address. However, from the SIP DEBUG output, it seems clear that the secondary server in this example is sending out Keepalives based on the information that the primary server has entered into the realtime DB. ** ** *Show peers Output from a primary server* Name/username HostDyn Forcerport ACL Port Status Realtime a201/A201 217.x.x.48D N 65229OK (88 ms) Cached RT a202 (Unspecified)D N 0UNREACHABLE Cached RT b201/44845287 78.x.x.101 D N 5060 OK (26 ms) Cached RT c201/s 193.x.x.174 D N 5060 OK (52 ms) Cached RT d201/d201 94.x.x.228 D N 5060 OK (33 ms) Cached RT e201/e20194.x.x.44 D N 55018OK (40 ms) Cached RT e202/e20294.x.x.44 D N 55022OK (46 ms) Cached RT e203/e20394.x.x.44 D N 55024OK (40 ms) Cached RT e204/e20494.x.x.44 D N 55008OK (40 ms) Cached RT e205/e20594.x.x.44 D N 55016OK (41 ms) Cached RT e206/e20694.x.x.44 D N 55014OK (40 ms) Cached RT e207/e20794.x.x.44 D N 55020OK (41 ms) Cached RT e208/e20894.x.x.44 D N 5060 OK (41 ms) Cached RT e209/e20994.x.x.44 D N 55012OK (40 ms) Cached RT e210/e21094.x.x.44 D N 55010OK (41 ms) Cached RT e211/e21194.x.x.44 D N 55026OK (38 ms) Cached RT e212/e21281.x.x.93D N 5060 OK (46 ms) Cached RT f201 (Unspecified)D N 0UNREACHABLE Cached RT g201/g 78.x.x.207 D N 5060 OK (29 ms) Cached RT h201/h201 217.x.x.78 D N 38980 OK (22 ms) Cached RT i201 (Unspecified)D N 0UNREACHABLE Cached RT i203/ i203 109.x.x.103 D N 5060 OK (32 ms) Cached RT i204/ i204 109.x.x.103 D N 1025 OK (31 ms) Cached RT i205/ i205 81.x.x.144 D N 5060 OK (32 ms) Cached RT i206/ i206 109.x.x.103 D N 1035 OK (31 ms) Cached RT i207/ i207 109.x.x.103 D N 1032 OK (32 ms) Cached RT i208/ i208 109.x.x.103 D N 1024 OK (31 ms) Cached RT j201/s 94.x.x.62D N 57813OK (35 ms) Cached RT o201/o201 92.x.x.86 D N 51824OK (47 ms) Cached RT o202/o202 92.x.x.86 D N 58641OK (48 ms) Cached RT o203/o203 92.x.x.86 D N 49172OK (47 ms) Cached RT j204/j204 176.x.x.214 D N 34824OK (49 ms) Cached RT k201/k201 2.x.x.169 D N 52757OK (53 ms) Cached RT k202/k202 (Unspecified)D N 0UNKNOWNCached RT l201/l201(Unspecified)D N 0UNKNOWNCached RT m201/s 92.x.x.95 D N 54020OK (32 ms) Cached RT n201 (Unspecified)
Re: [asterisk-users] Frames with invalid timing info
At a command prompt (not at the Asterisk CLI), if you run dahdi_tool and hit F1, what does it say? This is what I see: http://i.imgur.com/je7qRHa.png On Fri, Jan 25, 2013 at 8:20 AM, Richard Kenner ken...@gnat.com wrote: I'm now getting these errors: [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891164, src=RTP [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891174, src=RTP even *without* any transcoding. Suggestions? -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Quoting error with gotoiftime
I'm getting the following error, and none of us can figure out why:
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token', expecting
$end; Input:
=
^
Here is the code that generates it:
[scottsdale#queues-account]
exten = s,1,GotoIfTime(8:00-16:55,mon-fri,*,*?queue)
exten = s,n,Goto(scottsdale#queues-closed,s,1)
exten = s,n(queue),ExecIf($[${prefix} =
]?Queue(azenglish):Queue(azspanish))
exten = s,n(callback),Playback(scottsdale-q/${prefix}callbackmessage)
exten = s,n,Voicemail(@scottsdale,s)
exten = s,n,Hangup
Here is the rest of the call progress surrounding it, which seems to be
working anyway:
-- Executing [2@scottsdale#queues-aax:1]
Goto(SIP/televolve-1-1c7d, scottsdale#queues-account,s,1) in new
stack
-- Goto (scottsdale#queues-account,s,1)
-- Executing [s@scottsdale#queues-account:1]
GotoIfTime(SIP/televolve-1-1c7d, 8:00-16:55,mon-fri,*,*?queue) in
new stack
-- Goto (scottsdale#queues-account,s,3)
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token', expecting
$end; Input:
=
^
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [s@scottsdale#queues-account:3]
ExecIf(SIP/televolve-1-1c7d, ?Queue(azenglish):Queue(azspanish))
in new stack
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- 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] Quoting error with gotoiftime
Looks to me like ${prefix} contains nothing but two quotes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Friday, January 25, 2013 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Quoting error with gotoiftime
I'm getting the following error, and none of us can figure out why:
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror: ast_yyerror():
syntax error: syntax error, unexpected 'token', expecting $end; Input:
=
^
Here is the code that generates it:
[scottsdale#queues-account]
exten = s,1,GotoIfTime(8:00-16:55,mon-fri,*,*?queue)
exten = s,n,Goto(scottsdale#queues-closed,s,1)
exten = s,n(queue),ExecIf($[${prefix} =
]?Queue(azenglish):Queue(azspanish))
exten = s,n(callback),Playback(scottsdale-q/${prefix}callbackmessage)
exten = s,n,Voicemail(@scottsdale,s) exten = s,n,Hangup
Here is the rest of the call progress surrounding it, which seems to be working
anyway:
-- Executing [2@scottsdale#queues-aax:1] Goto(SIP/televolve-1-1c7d,
scottsdale#queues-account,s,1) in new stack
-- Goto (scottsdale#queues-account,s,1)
-- Executing [s@scottsdale#queues-account:1]
GotoIfTime(SIP/televolve-1-1c7d, 8:00-16:55,mon-fri,*,*?queue) in new
stack
-- Goto (scottsdale#queues-account,s,3) [Jan 25 09:07:19] WARNING[19258]:
ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax error: syntax error,
unexpected 'token', expecting $end; Input:
=
^
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [s@scottsdale#queues-account:3]
ExecIf(SIP/televolve-1-1c7d, ?Queue(azenglish):Queue(azspanish)) in
new stack
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- 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] Quoting error with gotoiftime
Error doesn't occur in 11.2.1
-- Executing [1260@default:1] Answer(SIP/sipuser-0001, ) in new
stack
-- Executing [1260@default:2] Goto(SIP/sipuser-0001,
scottsdale#queues-account,s,1) in new stack
-- Goto (scottsdale#queues-account,s,1)
-- Executing [s@scottsdale#queues-account:1]
GotoIfTime(SIP/sipuser-0001, 8:00-16:55,mon-fri,*,*?queue) in new
stack
-- Goto (scottsdale#queues-account,s,3)
-- Executing [s@scottsdale#queues-account:3]
ExecIf(SIP/sipuser-0001, 1?Queue(azenglish):Queue(azspanish)) in new
stack
== Spawn extension (scottsdale#queues-account, s, 3) exited non-zero on
'SIP/sipuser-0001'
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Friday, January 25, 2013 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Quoting error with gotoiftime
I'm getting the following error, and none of us can figure out why:
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token', expecting
$end; Input:
=
^
Here is the code that generates it:
[scottsdale#queues-account]
exten = s,1,GotoIfTime(8:00-16:55,mon-fri,*,*?queue)
exten = s,n,Goto(scottsdale#queues-closed,s,1)
exten = s,n(queue),ExecIf($[${prefix} =
]?Queue(azenglish):Queue(azspanish))
exten = s,n(callback),Playback(scottsdale-q/${prefix}callbackmessage)
exten = s,n,Voicemail(@scottsdale,s)
exten = s,n,Hangup
Here is the rest of the call progress surrounding it, which seems to be
working anyway:
-- Executing [2@scottsdale#queues-aax:1]
Goto(SIP/televolve-1-1c7d, scottsdale#queues-account,s,1) in new
stack
-- Goto (scottsdale#queues-account,s,1)
-- Executing [s@scottsdale#queues-account:1]
GotoIfTime(SIP/televolve-1-1c7d, 8:00-16:55,mon-fri,*,*?queue) in
new stack
-- Goto (scottsdale#queues-account,s,3)
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token', expecting
$end; Input:
=
^
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [s@scottsdale#queues-account:3]
ExecIf(SIP/televolve-1-1c7d, ?Queue(azenglish):Queue(azspanish))
in new stack
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- 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] How to implement priority queuing within a single queue ?
Hi, Let say that in a call center, callers are recognized and categorized in 4 priority levels (priority 1 for Very Very Important Personalities, 2 for VIP, and so on) before entering a Queue. How can you make sure a priority 2 caller is answered before priority 3 callers, for instance ? I can think of several solutions but none really pleases me : 1. Have 4 different queues, set penalty value and let each caller enter one queue depending on its own priority. I don't like this solution because I foresee editing stats for 4 queues instead of one is harder. 2. Iterate over each call waiting in the queue and insert new call with Queue's position argument accordingly valued. I don't like this one because I'm afraid coding this won't be so easy. What would you suggest ? 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] Quoting error with gotoiftime
What version does the error occur on? I suspect more recent versions of
Asterisk removes extraneous quotes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, January 25, 2013 11:20 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Quoting error with gotoiftime
Error doesn't occur in 11.2.1
-- Executing [1260@default:1] Answer(SIP/sipuser-0001, ) in new stack
-- Executing [1260@default:2] Goto(SIP/sipuser-0001,
scottsdale#queues-account,s,1) in new stack
-- Goto (scottsdale#queues-account,s,1)
-- Executing [s@scottsdale#queues-account:1]
GotoIfTime(SIP/sipuser-0001, 8:00-16:55,mon-fri,*,*?queue) in new stack
-- Goto (scottsdale#queues-account,s,3)
-- Executing [s@scottsdale#queues-account:3] ExecIf(SIP/sipuser-0001,
1?Queue(azenglish):Queue(azspanish)) in new stack
== Spawn extension (scottsdale#queues-account, s, 3) exited non-zero on
'SIP/sipuser-0001'
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Friday, January 25, 2013 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Quoting error with gotoiftime
I'm getting the following error, and none of us can figure out why:
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror: ast_yyerror():
syntax error: syntax error, unexpected 'token', expecting $end; Input:
=
^
Here is the code that generates it:
[scottsdale#queues-account]
exten = s,1,GotoIfTime(8:00-16:55,mon-fri,*,*?queue)
exten = s,n,Goto(scottsdale#queues-closed,s,1)
exten = s,n(queue),ExecIf($[${prefix} =
]?Queue(azenglish):Queue(azspanish))
exten = s,n(callback),Playback(scottsdale-q/${prefix}callbackmessage)
exten = s,n,Voicemail(@scottsdale,s)
exten = s,n,Hangup
Here is the rest of the call progress surrounding it, which seems to be working
anyway:
-- Executing [2@scottsdale#queues-aax:1] Goto(SIP/televolve-1-1c7d,
scottsdale#queues-account,s,1) in new stack
-- Goto (scottsdale#queues-account,s,1)
-- Executing [s@scottsdale#queues-account:1]
GotoIfTime(SIP/televolve-1-1c7d, 8:00-16:55,mon-fri,*,*?queue) in new
stack
-- Goto (scottsdale#queues-account,s,3)
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:468 ast_yyerror: ast_yyerror():
syntax error: syntax error, unexpected 'token', expecting $end; Input:
=
^
[Jan 25 09:07:19] WARNING[19258]: ast_expr2.fl:472 ast_yyerror: If you have
questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [s@scottsdale#queues-account:3]
ExecIf(SIP/televolve-1-1c7d, ?Queue(azenglish):Queue(azspanish)) in
new stack
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- 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] Quoting error with gotoiftime
On Fri, Jan 25, 2013 at 9:20 AM, Eric Wieling ewiel...@nyigc.com wrote:
Looks to me like ${prefix} contains nothing but two quotes.
Which is as it should be unless they choose the Spanish option, but yeah,
maybe that's what is choking Asterisk.
We do this:
exten = _X.,n,Set(prefix=) ;Initialize variable used in
scottsdale#queues-aax
Then in the AAX if they choose 5:
exten = 5,1,Set(prefix=s-); SPANISH
That way we just carry along their preference for Spanish throughout the
system. Perhaps I need to initialize the variable as e-, but then would
have to rename everything.
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- 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] Quoting error with gotoiftime
Don't do that. Set(prefix=) You are setting the prefix to have two quotes.
You WANT prefix to be empty.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Friday, January 25, 2013 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Quoting error with gotoiftime
On Fri, Jan 25, 2013 at 9:20 AM, Eric Wieling ewiel...@nyigc.com wrote:
Looks to me like ${prefix} contains nothing but two quotes.
Which is as it should be unless they choose the Spanish option, but yeah, maybe
that's what is choking Asterisk.
We do this:
exten = _X.,n,Set(prefix=) ;Initialize variable used in scottsdale#queues-aax
Then in the AAX if they choose 5:
exten = 5,1,Set(prefix=s-); SPANISH
That way we just carry along their preference for Spanish throughout the
system. Perhaps I need to initialize the variable as e-, but then would have
to rename everything.
--
Carlos Alvarez
TelEvolve
602-889-3003
--
_
-- 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] Quoting error with gotoiftime
On Fri, Jan 25, 2013 at 9:22 AM, Eric Wieling ewiel...@nyigc.com wrote: What version does the error occur on? I suspect more recent versions of Asterisk removes extraneous quotes. This is in 1.8. Danny's test does support your theory. It looks like the var is being set as the quotes, rather than empty. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Queues and distributed device state over WAN
2013/1/25 Alec Davis siva...@paradise.net.nz
Oliver wrote:
snip
Before diving into this, I've got the following question :
- let say we have two Asterisk servers A and B,
- both are interconnected through PSTN (no SIP trunk)
- agent Alice's phone is connected (ie registered) to server A
- Alice's phone can be reached from server A phones dialing
Local/6789
- Alice's phone can also be reached from server B phones dialing
Local/00123456789
1. How do you configure both servers so that Alice's phone becomes
a
Queue Member from a server B given queue ?
Simply calling AddQueueMember on server B, passing
Local/00123456789
as interface value (ie AddQueueMember second argument) ?
2. Then, how should server A publish Alice's phone status ?
How should server B consume Alice's phone status and associate it
with the Queue member activity ? Using AddQueueMember stateinterface
argument ?
Before trying to get distributed device state going:
Device names across all Asterisk proxy's participating in
'distributed device state' need to be unique.
IE. You can't have 'SIP/cisco1' exist on server A for ALICE, and
SIP/cisco1 on server B for BOB.
You need to get XMPP distributed device state working.
I followed
https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMP
P+PubSub
You need a reasonbly reliable WAN links to the jabber server support the
XMPP updates between servers.
Asterisk segfaults if it can't contact the jabber server!!! See
https://issues.asterisk.org/jira/browse/ASTERISK-18078
Then:
With Alice reqistered on Server A as SIP/cisco1
With Server B hosting the queue named 'queue1'.
;(on Server B)
queues.conf:
[queue1]
;what makes this work with distributed states is the 'State Interface'
parameter
...
member = Local/00123456789,0,ALICE,SIP/cisco1
Alice will need a number to ring to login/logout of queue1 hosted on Server
B;
Dialplan Example: on server B:
...
exten = s,n,Set(queuename=queue1)
exten = s,n,Set(interface=Local/00123456789)
exten = s,n,Set(penalty=0)
exten = s,n,Set(stateinterface=SIP/cisco1)
exten =
s,n(queue-add),AddQueueMember(${queuename},${interface},${penalty},options,,
${stateinterface})
And to remove the member;
...
exten = s,n(queue-remove),RemoveQueueMember(${queuename},${interface})
Alec Davis
I've not tried to publish device state with XMPP yet but I've discovered
this issue https://issues.asterisk.org/jira/browse/ASTERISK-18078
I'm planning to install my XMPP server on the same machine as one asterisk
server so hopefully, I won't be hit by the issue above but have you met
this issue ?
Could you get around ?
--
_
-- 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] How to implement priority queuing within a single queue ?
Am 25.01.2013 um 17:22 schrieb Olivier: Hi, Let say that in a call center, callers are recognized and categorized in 4 priority levels (priority 1 for Very Very Important Personalities, 2 for VIP, and so on) before entering a Queue. How can you make sure a priority 2 caller is answered before priority 3 callers, for instance ? I can think of several solutions but none really pleases me : 1. Have 4 different queues, set penalty value and let each caller enter one queue depending on its own priority. I don't like this solution because I foresee editing stats for 4 queues instead of one is harder. Just set the Queue_PRIO for that specific caller-type before you send them all into the same queue: exten = s,n,Set(QUEUE_PRIO=10) exten = s,n,Queue(test,tC,,,180) 2. Iterate over each call waiting in the queue and insert new call with Queue's position argument accordingly valued. I don't like this one because I'm afraid coding this won't be so easy. What would you suggest ? 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 Michael http://www.mksolutions.info smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] Quoting error with gotoiftime
Where possible you should have a VM to try these things as needed. Where not, it isn't too difficult to duplicate the contexts and do something like this [default] . . Exten = 1260,1,answer Exten = 1260,n,goto(test-context,s,1) . From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Friday, January 25, 2013 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Quoting error with gotoiftime On Fri, Jan 25, 2013 at 9:27 AM, Eric Wieling ewiel...@nyigc.com wrote: Don't do that. Set(prefix=) You are setting the prefix to have two quotes. You WANT prefix to be empty. I'll give that a try during non-production hours. Odd that the same code works in earlier versions and later, but not this one. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] How to implement priority queuing within a single queue ? [SOLVED]
2013/1/25 Michael Keuter li...@mksolutions.info Am 25.01.2013 um 17:22 schrieb Olivier: Hi, Let say that in a call center, callers are recognized and categorized in 4 priority levels (priority 1 for Very Very Important Personalities, 2 for VIP, and so on) before entering a Queue. How can you make sure a priority 2 caller is answered before priority 3 callers, for instance ? I can think of several solutions but none really pleases me : 1. Have 4 different queues, set penalty value and let each caller enter one queue depending on its own priority. I don't like this solution because I foresee editing stats for 4 queues instead of one is harder. Just set the Queue_PRIO for that specific caller-type before you send them all into the same queue: exten = s,n,Set(QUEUE_PRIO=10) exten = s,n,Queue(test,tC,,,180) That's exactly what I was looking for. The strange thing is I couldn't find it mentioned in Queue app doc, if I'm not mistaken (but that's another story). Thank you very much. 2. Iterate over each call waiting in the queue and insert new call with Queue's position argument accordingly valued. I don't like this one because I'm afraid coding this won't be so easy. What would you suggest ? 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 Michael http://www.mksolutions.info -- _ -- 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] Quoting error with gotoiftime
On Fri, Jan 25, 2013 at 9:31 AM, Danny Nicholas da...@debsinc.com wrote: Where possible you should have a VM to try these things as needed. Where not, it isn’t too difficult to duplicate the contexts and do something like this [default] I do have a test VM, but I also have a maintenance window for this customer later tonight for other things, so I was being lazy. Poor excuse. Tested it, and it works, thanks guys! -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] How to implement priority queuing within a single queue ? [SOLVED]
Am 25.01.2013 um 17:39 schrieb Olivier: 2013/1/25 Michael Keuter li...@mksolutions.info Am 25.01.2013 um 17:22 schrieb Olivier: Hi, Let say that in a call center, callers are recognized and categorized in 4 priority levels (priority 1 for Very Very Important Personalities, 2 for VIP, and so on) before entering a Queue. How can you make sure a priority 2 caller is answered before priority 3 callers, for instance ? I can think of several solutions but none really pleases me : 1. Have 4 different queues, set penalty value and let each caller enter one queue depending on its own priority. I don't like this solution because I foresee editing stats for 4 queues instead of one is harder. Just set the Queue_PRIO for that specific caller-type before you send them all into the same queue: exten = s,n,Set(QUEUE_PRIO=10) exten = s,n,Queue(test,tC,,,180) That's exactly what I was looking for. The strange thing is I couldn't find it mentioned in Queue app doc, if I'm not mistaken (but that's another story). Thank you very much. I found it here a while ago: http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List https://wiki.asterisk.org/wiki/display/AST/Various+application+variables Michael http://www.mksolutions.info smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] Queues and distributed device state over WAN
I've not tried to publish device state with XMPP yet but I've discovered this issue https://issues.asterisk.org/jira/browse/ASTERISK-18078 I'm planning to install my XMPP server on the same machine as one asterisk server so hopefully, I won't be hit by the issue above but have you met this issue ? Could you get around ? I installed Tigase on the asterisk server hosting the queues, our main office. Yes I have experienced https://issues.asterisk.org/jira/browse/ASTERISK-18078 But for us, we have a fibre link between 3 offices, so hardly ever see the problem, only when I reboot the server with XMPP do the other 2 asterisk's segfault. The work around for us is: Don't reboot XMPP/Asterisk server during critical periods, however they have a cron script checking to see whether asterisk is alive, and if not restart asterisk. Prior to having the luxury of private 10Mb fibre links, we had to rely on internet ADSL VPN links between our offices, no good for voice, but reliably enough for device state updates. Alec -- _ -- 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] Queues and distributed device state over WAN
On 01/25/2013 01:59 PM, Alec Davis wrote: I've not tried to publish device state with XMPP yet but I've discovered this issue https://issues.asterisk.org/jira/browse/ASTERISK-18078 I'm planning to install my XMPP server on the same machine as one asterisk server so hopefully, I won't be hit by the issue above but have you met this issue ? Could you get around ? I installed Tigase on the asterisk server hosting the queues, our main office. Yes I have experienced https://issues.asterisk.org/jira/browse/ASTERISK-18078 But for us, we have a fibre link between 3 offices, so hardly ever see the problem, only when I reboot the server with XMPP do the other 2 asterisk's segfault. The work around for us is: Don't reboot XMPP/Asterisk server during critical periods, however they have a cron script checking to see whether asterisk is alive, and if not restart asterisk. Prior to having the luxury of private 10Mb fibre links, we had to rely on internet ADSL VPN links between our offices, no good for voice, but reliably enough for device state updates. Not that this is an excuse or a valid workaround for everyone, but I believe that issue won't apply if you're using Asterisk 11 and res_xmpp. res_jabber: yup, totally still a problem. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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] How to implement priority queuing within a single queue ?
1. Have 4 different queues, set penalty value and let each caller enter one queue depending on its own priority. Penalty isn't anything to do with the caller, it's to do with the agent. We set round robin for our queues. With penalty=0 for the main members of a queue, to service most of the calls. Penalty 50 for a trainee. Penalty 99 for the supervisor of the queue. This way the supervisor isn't bothered with the bulk of the calls, only when the queue is busy does the supervisor get any calls. Of course this can be done by analysis later, but in realtime the supervisor can assess why they got the call - assuming they are in close proximinity. Large queues, different story. We only have 5 members in a queue, so it works for us. Alec Davis -- _ -- 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] Queues and distributed device state over WAN
Not that this is an excuse or a valid workaround for everyone, but I believe that issue won't apply if you're using Asterisk 11 and res_xmpp. res_jabber: yup, totally still a problem. Hmm. We're using Asterisk 11, but I still think res_jabber. Why havn't I changed to res_xmpp, I have no answer for that. Alec -- _ -- 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] asterisk 11's app_page options
I have just upgraded to asterisk 11 from 1.8 I have noticed that my Page command: exten = 1,1,Page(SIP/101,diqA(local/intercom)) does not play the local/intercom sound to the conference. according to the doc at https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Page , it seems like it still should. is there something i need to do to make this work how i expect it? my confbridge.conf is vanilla; i dont see anything that needs changing. also, when the conference ends, the CLI shows: [Jan 25 23:50:52] ERROR[3746][C-000a]: confbridge/conf_state.c:47 conf_invalid_event_fn: Invalid event for confbridge user '' [Jan 25 23:50:52] ERROR[3745][C-000a]: confbridge/conf_state.c:47 conf_invalid_event_fn: Invalid event for confbridge user '' any way to hush/fix that? Thanks, -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
