Re: [asterisk-users] Variable stripping/removing part of string
It was a 1.8 but then we started to do a lot of development (ooh323) so
today it is Asterisk SVN-may-ooh323_ipv6_direct_rtp-r311741MS-/trunk.
Can hardly se that we have done any changes that would cause my problem.
-Ursprungligt meddelande-
From: Tilghman Lesher
Sent: Monday, April 11, 2011 9:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Variable stripping/removing part of string
On Monday 11 April 2011 00:25:35 magnu...@inputinterior.se wrote:
Now i am lost.
exten = 0424449631,n,NoOp(${CALLERID(name)})
exten = 0424449631,n,NoOp(${CUT(CALLERID(name),\(,2):0:-1})
-- Executing [0424449...@fax.inputinterior.se:4] NoOp(OOH323/Avaya2-8,
Martela (fax)) in new stack
-- Executing [0424449...@fax.inputinterior.se:5] NoOp(OOH323/Avaya2-8,
fax)) in new stack
But i am looking for the part before (, in my case: Martela
Oh, sorry. You were right before, then. As far as the :0:-1 nomenclature,
what version of Asterisk are you using? It was not supported before 1.4.
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Re: [asterisk-users] Variable stripping/removing part of string
I dont know if my mail client will keep formatting as I se it, but for me it
sure looks like one space.
-- Executing [0424449...@fax.inputinterior.se:4] NoOp(OOH323/Avaya2-109,
Martela (fax)) in new stack
xyz
-- Executing [0424449...@fax.inputinterior.se:5] NoOp(OOH323/Avaya2-109,
Martela ) in new stack
-Ursprungligt meddelande-
From: Tilghman Lesher
Sent: Monday, April 11, 2011 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Variable stripping/removing part of string
On Monday 11 April 2011 02:56:03 magnu...@inputinterior.se wrote:
It was a 1.8 but then we started to do a lot of development (ooh323) so
today it is Asterisk SVN-may-ooh323_ipv6_direct_rtp-r311741MS-/trunk.
Can hardly se that we have done any changes that would cause my
problem.
Are you sure there's only a single space separating the name from the
opening parenthesis? The :0:-1 nomenclature only removes a single
byte from the end, and if there was more than a single byte, that might
explain the difference. If that's the case, you may be forced to do a loop
to remove all trailing spaces, if that's still important:
While($[${foo:-1} = ])
Set(foo=${foo:0:-1})
EndWhile
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Re: [asterisk-users] Variable stripping/removing part of string
U were right, breaking it into two lines work.
exten = 0424449631,n,NoOp(${CALLERID(name)})
exten = 0424449631,n,Set(name=${CUT(CALLERID(name),\(,1)})
exten = 0424449631,n,NoOp(${name:0:-1})
-- Executing [0424449...@fax.inputinterior.se:4] NoOp(OOH323/Avaya2-150,
Martela (fax)) in new stack
-- Executing [0424449...@fax.inputinterior.se:5] Set(OOH323/Avaya2-150,
name=Martela ) in new stack
-- Executing [0424449...@fax.inputinterior.se:6] NoOp(OOH323/Avaya2-150,
Martela) in new stack
But still, dont understand why u cant do it on one line, but u cant always
understand everything.
Anyway, thx for pointing me to the correct direction.
-Ursprungligt meddelande-
From: Jeroen Eeuwes
Sent: Monday, April 11, 2011 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Variable stripping/removing part of string
Hi Magnus,
exten = 0424449631,n,NoOp(${CUT(CALLERID(name),\(,1):0:-1})
But that gave me “Martela “ so my way of doing it is wrong.
Any that can tell me what I am doing wrong or have any better suggestion
howto do it?
I think you are not able to do it in one step. Can you try something like
this:
exten = 0424449631,n,Set(TESTING=${CUT(CALLERID(name),\(,1)})
exten = 0424449631,n,NoOp(${TESTING:0:-1})
Best regards,
Jeroen Eeuwes
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Re: [asterisk-users] Variable stripping/removing part of string
Weired result:
exten = 0424449631,n,NoOp(${CALLERID(name)})
exten = 0424449631,n,NoOp(${${CUT(CALLERID(name),\(,1)}:0:-1})
-- Executing [0424449...@fax.inputinterior.se:4] NoOp(OOH323/Avaya2-248,
Martela (fax)) in new stack
-- Executing [0424449...@fax.inputinterior.se:5] NoOp(OOH323/Avaya2-248,
) in new stack
Now I understand even less. (But it was a nice try).
-Ursprungligt meddelande-
From: Chad Wallace
Sent: Tuesday, April 12, 2011 3:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Variable stripping/removing part of string
On Mon, 11 Apr 2011 12:58:39 +0200
magnu...@inputinterior.se wrote:
U were right, breaking it into two lines work.
exten = 0424449631,n,NoOp(${CALLERID(name)})
exten = 0424449631,n,Set(name=${CUT(CALLERID(name),\(,1)})
exten = 0424449631,n,NoOp(${name:0:-1})
-- Executing [0424449...@fax.inputinterior.se:4]
NoOp(OOH323/Avaya2-150, Martela (fax)) in new stack
-- Executing [0424449...@fax.inputinterior.se:5]
Set(OOH323/Avaya2-150, name=Martela ) in new stack
-- Executing [0424449...@fax.inputinterior.se:6]
NoOp(OOH323/Avaya2-150, Martela) in new stack
But still, dont understand why u cant do it on one line, but u cant
always understand everything.
Anyway, thx for pointing me to the correct direction.
Just a guess... try this:
exten = 0424449631,n,NoOp(${${CUT(CALLERID(name),\(,1)}:0:-1})
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[asterisk-users] Variable stripping/removing part of string
Hi!
I try to get rid of some part of CALLERID(name) but I cant realy figure out a
way to do it.
For example: CALLERID(name) = Martela (fax) I am just looking for the part
before “ (“ in my case “Martela”.
I can’t serch for “ “, could be many “ “, but only one “ (“, thought i could do
something like:
exten = 0424449631,n,NoOp(${CUT(CALLERID(name),\(,1):0:-1})
But that gave me “Martela “ so my way of doing it is wrong.
Any that can tell me what I am doing wrong or have any better suggestion howto
do it?
/Magnus--
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Re: [asterisk-users] Variable stripping/removing part of string
Now i am lost.
exten = 0424449631,n,NoOp(${CALLERID(name)})
exten = 0424449631,n,NoOp(${CUT(CALLERID(name),\(,2):0:-1})
-- Executing [0424449...@fax.inputinterior.se:4] NoOp(OOH323/Avaya2-8,
Martela (fax)) in new stack
-- Executing [0424449...@fax.inputinterior.se:5] NoOp(OOH323/Avaya2-8,
fax)) in new stack
But i am looking for the part before (, in my case: Martela
-Ursprungligt meddelande-
From: Tilghman Lesher
Sent: Monday, April 11, 2011 7:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Variable stripping/removing part of string
On Monday 11 April 2011 00:07:08 magnu...@inputinterior.se wrote:
Hi!
I try to get rid of some part of CALLERID(name) but I cant realy figure
out a way to do it. For example: CALLERID(name) = Martela (fax) I am
just looking for the part before “ (“ in my case “Martela”. I can’t
serch for “ “, could be many “ “, but only one “ (“, thought i could do
something like:
exten = 0424449631,n,NoOp(${CUT(CALLERID(name),\(,1):0:-1})
But that gave me “Martela “ so my way of doing it is wrong.
Any that can tell me what I am doing wrong or have any better suggestion
howto do it?
You're almost there. The issue is that CUT uses 1-based offsets, not
0-based offsets, so:
exten = 0424449631,n,NoOp(${CUT(CALLERID(name),\(,2):0:-1})
--
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Re: [asterisk-users] Checking status of a cell phone
Its not the Avaya that makes the call back, it is mobile. -Ursprungligt meddelande- From: Gilles Sent: Monday, March 28, 2011 1:57 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Checking status of a cell phone On Sat, 26 Mar 2011 14:58:30 +0100, magnu...@inputinterior.se wrote: Celluar Network - E1 - Avaya - OOH323 - Asterisk Thanks for the tip. So here's how it works: 1. The web app calls a script that uses AMI + Originate to send a call to the Avaya PBX 2. Avaya is able to check that a number (cellphone in this case) is busy and calls a different number in Asterisk to indicate the status through a value in the DB 3. The web script reads the value of DS/0733025975 and displays the status -- _ -- 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
[asterisk-users] Variable. AMI and dialplan
Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what. From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n From dialplan i can “access” the variable “x” and see the value “5” From dialplan i modify “x” to “8”. But from AMI i still se “x” as “5” not “8”. /Magnus-- _ -- 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] Variable. AMI and dialplan
I did use Action: Getvar when i read it back in AMI. On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote: Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what. From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n From dialplan i can “access” the variable “x” and see the value “5” From dialplan i modify “x” to “8”. But from AMI i still se “x” as “5” not “8”. /Magnus Maybe you need to perform a GetVar to read the new value of that channel variable -- Sherwood McGowan sherwood.mcgo...@gmail.com Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _ -- 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] Variable. AMI and dialplan
Could be, do u think its a bug or do u think I am doing totally wrong? I can easily reproduce it if any needs more info. -Ursprungligt meddelande- From: Sebastian Sent: Monday, March 28, 2011 3:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Variable. AMI and dialplan this may be related with: https://issues.asterisk.org/view.php?id=14662 El 28/03/2011 10:20, Sherwood McGowan escribió: Don't know then, that's all I've got far ya today mate, sorry On 3/28/2011 8:18 AM, magnu...@inputinterior.se wrote: I did use Action: Getvar when i read it back in AMI. On 3/28/2011 7:41 AM, magnu...@inputinterior.se wrote: Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what. From AMI, I set a variable Action: Setvar\r\nVariable:x\r\n\Value: 5\r\n\r\n From dialplan i can “access” the variable “x” and see the value “5” From dialplan i modify “x” to “8”. But from AMI i still se “x” as “5” not “8”. /Magnus Maybe you need to perform a GetVar to read the new value of that channel variable -- _ -- 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] Checking status of a cell phone
I was a little unclear, it is not the cell phone that does the call-back, it is the cell-phone-network. We can define 3 traffic-cases per cell-phone: 1) If cell-phone wont anser in x seconds call number a. 2) if cell-phone is busy call number b. 3) if cell-phone is unavailable call number c. From ami, a set db entry 0733025975 = 0 (Idle) from ami, make a short call (1 second) to 0733025975 wait 0.5 second check the db entry for 0733025975 when i wait for 0.5 second and my cell phone is busy, i will get a call to number b I catch that call in dialplan and set 0733025975 = 1 (InUse) Ofc, if cell-phone is unavailable, i will get call to number c I catch that call in dialplan and set 0733025975 = 4 (Unavailable) -Ursprungligt meddelande- From: Gilles Sent: Monday, March 28, 2011 10:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Checking status of a cell phone On Mon, 28 Mar 2011 14:12:09 +0200, magnu...@inputinterior.se wrote: Its not the Avaya that makes the call back, it is mobile. I thought the way you handled things, is that Asterisk would call your cellphone through the Avaya PBX just to check whether the cellphone is in_use/busy. At what point does the cellphone call Avaya or Asterisk back? -- _ -- 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
[asterisk-users] Checking status of a cell phone
Hi, I am looking for a way to check the status of a cell phone. Found one way that worked for me and would like to have some feedback or suggestion of improvments. Below example is for a “Swedish” cell phone, dont know if it works in the same way for other countries. I could define “redirecting” numbers for 3 traffic cases when u dial my mobile (073-302 59 75): NOT_INUSE call forward to A INUSE call forward to B in my case 010-602 4975 UNAVAILABLE call forward to C in my case 010-602 4976 From manager: Action: Originate\r\nChannel: OOH323/00733025975@Avaya\r\nExten: 0106024000\r\nContext: inputinterior.se\r\nPriority: 1\r\nTimeout: 1000\r\nCallerID: 106024000\r\n\r\n DBPut\r\nFamily: DS\r\nKey: 0733025975\r\nVal: NOT_INUSE\r\n\r\n Wait a second... Action: DBGet\r\nFamily: DS\r\nKey: 0733025975\r\n\r\n In the dialplan: exten = 0106024975,1,Set(DB(DS/0733025975)=INUSE) exten = 0106024975,n,Hangup() exten = 0106024976,1,Set(DB(DS/0733025975)=UNAVAILABLE) exten = 0106024976,n,Hangup() Just a short call to my cell phone, to se if i get anything back, my cell phone doesn’t even ring. Wait a second if the call is redirected, then check to se if the status has changed from NOT_INUSE to something else. Dont know if it is a stupid idea, but it worked on my cell phone, and the switchboard girls was very happy to be able “to ask” my cell phone “what I am doing” Most of the day i am INUSE so they dont need to transfer calls to me ehen they know I am INUSE. Ofc there is some delay from asking to getting the answer, but as the girls said, we could live with the delay, 2seconds compared to be “blind” is nothing.wlEmoticon-smile[1].png-- _ -- 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] Checking status of a cell phone
Setup as below: Celluar Network - E1 - Avaya - OOH323 - Asterisk It works like this, some1 (we can call her Åsa) wants to know if i am avaiable (my cell phone 073-302 59 75 is NOT_INUSE) She have a web-app (just a simple form), where she enter my extension and hits enter. The web-app originates the call as i wrote and waits for the status then ofc presents it to Åsa. I was writing the app (probably the worst written code i have done so dont ask me to post it) late thursday and let Åsa use it on Friday. And yes, the AMI code was enough, everytime she should transfer a call to me or just call me , she used the web-app first, and she was very happy. When she saw that I was INUSE she sent me a mail that mr X has been looking for me, i got the mail while I was talking in the phone so I know that she used the web-app to determine my status, not just transfering the call. -Ursprungligt meddelande- From: Gilles Sent: Saturday, March 26, 2011 11:37 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Checking status of a cell phone On Sat, 26 Mar 2011 10:50:19 +0100, magnu...@inputinterior.se wrote: I am looking for a way to check the status of a cell phone. Found one way that worked for me and would like to have some feedback or suggestion of improvments. I'd like to check I understood: Your Asterisk server is connected to a landline and can call your cellephone (073-302 59 75). When a call comes in from the landline, Asterisk checks whether your cellphone is available and redirects the call; If not available, it calls a landline number (010-602 4975). If this landline number is not available, it tries a third number (010-602 4976)? Is the AMI code below enough to check if the cellphone is available/in-use? Action: Originate Channel: OOH323/00733025975@Avaya\r\nExten: 0106024000 Context: inputinterior.se Priority: 1 Timeout: 1000 CallerID: 106024000 DBPut Family: DS Key: 0733025975 Val: NOT_INUSE -- _ -- 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
[asterisk-users] Automatically unpause a paused queue memeber - bad idea?
I have some cases when I want to pause a queue member and automatically unpause the queue member after a specified time. Right now I am doing it by a AMI script, but was thinking if it is possible to add a parameter to PauseQueueMember like, PauseQueueMember([queuename],interface[,options[,reason[,time]]]) where time will be how long (in seconds) the interface will be paused. before brought back. Maybe it is a bad idea, I dont know, what do you think? -- _ -- 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] AddQueueMember and stateinterface question
Hi,
I have missed something so I wonder if someone could explain for me?
0424449647 desktop phone
0106024647 DECT phone
0424449630 Helsingborg queue
extensions.conf
---
[support]
exten = 0424449647,hint,SIP/0424449647SIP/0106024647
exten = 0424449647,1,Dial(SIP/0424449647SIP/0106024647,15,rtT)
[inputinterior.se]
exten = 0/0424449647,1,Answer()
exten = 0/0424449647,n,RemoveQueueMember(Helsinborg,Local/0424449647@support)
exten = 0/0424449647,n,Hangup()
;
exten = 1/0424449647,1,Answer()
exten = 1/0424449647,n,RemoveQueueMember(Helsinborg,Local/0424449647@support)
exten =
1/0424449647,n,AddQueueMember(Helsinborg,Local/0424449647@support,1,,Lisbeth
Mingert Nilsson,SIP/0424449647)
exten = 1/0424449647,n,Hangup()
;
exten = 0424449630,1,Answer()
exten =
0424449630,n,ExecIf($[${QUEUE_MEMBER(Helsingborg,logged)}=0]?Queue(Goteborg,rtT))
exten = 0424449630,n,Queue(Helsingborg,nrtT)
If i dial 0424449630 both desktop and DECT phone rings (if 0424449647 is logged
in ofc)
If desktop phone is answering, everything is fine:
Lisbeth Mingert Nilsson (Local/0424449647@support) with penalty 1 (dynamic) (In
use) has taken no calls yet
But if DECT phone is a answering:
Lisbeth Mingert Nilsson (Local/0424449647@support) with penalty 1 (dynamic)
(Not in use) has taken 1 calls (last was 136 secs ago)
I am looking for a way to monitor both phones.
I hought i could do something like:
exten =
1/0424449647,n,AddQueueMember(Helsinborg,Local/0424449647@support,1,,Lisbeth
Mingert Nilsson,SIP/0424449647SIP/0106024647)
But that didn't work.
/Magnus--
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Re: [asterisk-users] MEMBERINTERFACE and MEMBERNAME questions
Hmm,
First i must correct myself, MEMBERINTERFACE seems to be NULL, not the “device”
that called in, my bad reading.
Did some changes:
queues.conf
---
[Kinna]
keepstats=yes
ringinuse=no
setinterfacevar=yes
setqueuevar=yes
strategy=rrmemory
timeout=5
wrapuptime=120
extensions.conf
exten = 0320209030,1,Answer()
exten =
0320209030,n,ExecIf($[${QUEUE_MEMBER(Kinna,logged)}=0]?Queue(Goteborg,rtT))
exten = 0320209030,n,Queue(Kinna,nrtT)
exten = 0320209030,n,NoOp(${MEMBERINTERFACE})
exten = 0320209030,n,NoOp(${MEMBERNAME})
exten = 0320209030,n,NoOp(${QUEUENAME})
exten = 0320209030,n,Queue(Goteborg,rtT)
exten = 0320209030,n,Hangup()
Same call flows as below:
== Using SIP RTP CoS mark 5
-- Executing [0320209...@inputinterior.se:1]
Answer(SIP/0317998985-0050, ) in new stack
-- Executing [0320209...@inputinterior.se:2]
ExecIf(SIP/0317998985-0050, 0?Queue(Goteborg,rtT)) in new stack
-- Executing [0320209...@inputinterior.se:3] Queue(SIP/0317998985-0050,
Kinna,nrtT) in new stack
== Using SIP RTP CoS mark 5
-- SIP/0317998972-0051 is ringing
-- SIP/0317998972-0051 is ringing
-- SIP/0317998972-0051 is ringing
-- SIP/0317998972-0051 is ringing
-- Nobody picked up in 5000 ms
-- Exiting on time-out cycle
-- Executing [0320209...@inputinterior.se:4] NoOp(SIP/0317998985-0050,
) in new stack
-- Executing [0320209...@inputinterior.se:5] NoOp(SIP/0317998985-0050,
) in new stack
-- Executing [0320209...@inputinterior.se:6] NoOp(SIP/0317998985-0050,
Kinna) in new stack
-- Executing [0320209...@inputinterior.se:7] Queue(SIP/0317998985-0050,
Goteborg,rtT) in new stack
QUEUENAME is working the way i am excpecting but MEMBERINTERFACE and MEMBERNAME
is not, or am I wrong?
From: magnu...@inputinterior.se
Sent: Sunday, February 20, 2011 8:05 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MEMBERINTERFACE and MEMBERNAME questions
Hi!
Did play around with queues and need some help. I thought that MEMBERINTERFACE
and MEMBERNAME should be set to the “device” the call was queued to not the
device that called the queue, or do i miss something?
Running: Asterisk 1.8.2.3 built by root @ sip on a i686 running Linux on
2011-01-31 13:38:23 UTC
0317998985 calls Kinna (0320209030)
Tomas Ekman (SIP/0317998972) receives the call but don’t answer.
When the queue “timeout” I would like to get the name of the device that didn’t
answered, in my case: SIP/0317998972.
${MEMBERINTERFACE} gives me the name of the device that called in.
queue show Kinna
Kinna has 0 calls (max unlimited) in 'rrmemory' strategy (4s holdtime, 2s
talktime), W:0, C:1, A:13, SL:0.0% within 0s
Members:
Tomas Ekman (SIP/0317998972) with penalty 1 (dynamic) (Not in use) has
taken no calls yet
No Callers
queues.conf
---
[general]
;
autofill=yes
keepstats=yes
setinterfacevar=yes
;
[Kinna]
retry=5
ringinuse=no
strategy=rrmemory
timeout=20
wrapuptime=120
extensions.conf
---
exten = 0320209030,1,Answer()
exten =
0320209030,n,ExecIf($[${QUEUE_MEMBER(Kinna,logged)}=0]?Queue(Goteborg,rtT))
exten = 0320209030,n,Queue(Kinna,nrtT)
exten = 0320209030,n,NoOp(${MEMBERINTERFACE})
exten = 0320209030,n,NoOp(${MEMBERNAME})
exten = 0320209030,n,Queue(Goteborg,rtT)
exten = 0320209030,n,Hangup()
CLI
== Using SIP RTP CoS mark 5
-- Executing [0320209...@inputinterior.se:1]
Answer(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:2]
ExecIf(SIP/0317998985-0033, 0?Queue(Goteborg,rtT)) in new stack
-- Executing [0320209...@inputinterior.se:3]
Queue(SIP/0317998985-0033, Kinna,nrtT) in new stack
== Using SIP RTP CoS mark 5
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- Nobody picked up in 2 ms
-- Exiting on time-out cycle
-- Executing [0320209...@inputinterior.se:4]
NoOp(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:5]
NoOp(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:6]
Queue(SIP/0317998985-0033, Goteborg,rtT) in new stack
== Spawn extension (inputinterior.se, 0320209030, 6) exited non-zero on
'SIP/0317998985-0033'
Could any help me understand what I am doing wrong?
/Magnus
--
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[asterisk-users] MEMBERINTERFACE and MEMBERNAME questions
Hi!
Did play around with queues and need some help. I thought that MEMBERINTERFACE
and MEMBERNAME should be set to the “device” the call was queued to not the
device that called the queue, or do i miss something?
Running: Asterisk 1.8.2.3 built by root @ sip on a i686 running Linux on
2011-01-31 13:38:23 UTC
0317998985 calls Kinna (0320209030)
Tomas Ekman (SIP/0317998972) receives the call but don’t answer.
When the queue “timeout” I would like to get the name of the device that didn’t
answered, in my case: SIP/0317998972.
${MEMBERINTERFACE} gives me the name of the device that called in.
queue show Kinna
Kinna has 0 calls (max unlimited) in 'rrmemory' strategy (4s holdtime, 2s
talktime), W:0, C:1, A:13, SL:0.0% within 0s
Members:
Tomas Ekman (SIP/0317998972) with penalty 1 (dynamic) (Not in use) has
taken no calls yet
No Callers
queues.conf
---
[general]
;
autofill=yes
keepstats=yes
setinterfacevar=yes
;
[Kinna]
retry=5
ringinuse=no
strategy=rrmemory
timeout=20
wrapuptime=120
extensions.conf
---
exten = 0320209030,1,Answer()
exten =
0320209030,n,ExecIf($[${QUEUE_MEMBER(Kinna,logged)}=0]?Queue(Goteborg,rtT))
exten = 0320209030,n,Queue(Kinna,nrtT)
exten = 0320209030,n,NoOp(${MEMBERINTERFACE})
exten = 0320209030,n,NoOp(${MEMBERNAME})
exten = 0320209030,n,Queue(Goteborg,rtT)
exten = 0320209030,n,Hangup()
CLI
== Using SIP RTP CoS mark 5
-- Executing [0320209...@inputinterior.se:1]
Answer(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:2]
ExecIf(SIP/0317998985-0033, 0?Queue(Goteborg,rtT)) in new stack
-- Executing [0320209...@inputinterior.se:3]
Queue(SIP/0317998985-0033, Kinna,nrtT) in new stack
== Using SIP RTP CoS mark 5
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- SIP/0317998972-0034 is ringing
-- Nobody picked up in 2 ms
-- Exiting on time-out cycle
-- Executing [0320209...@inputinterior.se:4]
NoOp(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:5]
NoOp(SIP/0317998985-0033, ) in new stack
-- Executing [0320209...@inputinterior.se:6]
Queue(SIP/0317998985-0033, Goteborg,rtT) in new stack
== Spawn extension (inputinterior.se, 0320209030, 6) exited non-zero on
'SIP/0317998985-0033'
Could any help me understand what I am doing wrong?
/Magnus--
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[asterisk-users] SendFAX dialplan example
Hi! I am playing with SendFAX but cant really figure out how it is working. I have a “fax” /var/spool/asterisk/tmp/fax.tiff that i would like to send to a “physical” fax at numer 0317998901. Can some1 write me a simple dialplan that just “grab” the file and send it to 0317998901? /Magnus-- _ -- 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] Fax stopped working when upgrading to 1.8.2
Vladimir, Sorry, I was a little “unclear”, Asterisk SVN-trunk-r280589M was not a 1.8 release. It was compiled before 1.8 was released. May did a lot of patches that he comitted to the trunk version and then when I got everything to work, I didnt recompile until I tried 1.8.2. But I will dig into the logs and se if i can find anything. Magnus, I finally got to testing the patch myself. Apparently it did not work for me. That means that if you are affected by the same issue it is not fixed yet. The current manifestation an incoming OOH_323 call fails in 30 seconds. After reading your initial message more carefully I realized the problem you experience must be different as the ooh_323 issue affects all releases in 1.8 branch and you state that it worked for you when you built from the svn trunk. I would suggest you analyze the full log and make sure the fax application does not complain of any problems, then check the h323_log and make sure there are no complaints of codecs incompatibility. If you are not utilizing T.38 then only alaw and ulaw will support fax. -Vladimir On 1/14/2011 12:32 AM, magnu...@inputinterior.se wrote: Did apply the patch and did a recompile, no difference, fax still not working. But I did notice one thing, when I was standing at a fax attched to PSTN and trying to send a fax to a fax attached to the Asterisk: The PSTN fax never switched to saying “Sending...” in the display just “Dialing”, but I can “hear” the Asterisk fax i answering. When I went back to Trunk version and did the same, I saw the fax display going from “Dialing” to “Sending” to “Sending OK”. I am sorry to say that I am not smart enough to know what trace I should start looking at, any knows? From: Vladimir Mikhelson Sent: Thursday, January 13, 2011 5:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: magnu...@inputinterior.se Subject: Re: [asterisk-users] Fax stopped working when upgrading to 1.8.2 Magnus, Can it be the same as I experienced https://issues.asterisk.org/view.php?id=18542 ? Do not be confused by the ticket subject, it reflects the symptoms as they looked originally You can try the patch if applicable and if you know how to compile Addons in 1.8 separately or if you have a capacity to compile the whole thing. -Vladimir On 1/13/2011 6:31 AM, magnu...@inputinterior.se wrote: Gentlemen, We have a setup as below: PSTN – E1 – Avaya – OOH323 trunk – Asterisk – SPA-2102 – Fax machine Running Asterisk SVN-trunk-r280589M, fax working as a clock. I decided to leave “trunk” and go a stable version so I upgraded to 1.8.2. Didn’t change any config files, everything worked as before except fax. I wonder if there are any known issues or things that I have missed to do in some config file. Did a downgrade to SVN-trunk-r280589M and fax started to work again. /Magnus -- _ -- 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-usersimage/png-- _ -- 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] Fax stopped working when upgrading to 1.8.2
Gentlemen, We have a setup as below: PSTN – E1 – Avaya – OOH323 trunk – Asterisk – SPA-2102 – Fax machine Running Asterisk SVN-trunk-r280589M, fax working as a clock. I decided to leave “trunk” and go a stable version so I upgraded to 1.8.2. Didn’t change any config files, everything worked as before except fax. I wonder if there are any known issues or things that I have missed to do in some config file. Did a downgrade to SVN-trunk-r280589M and fax started to work again. /Magnus-- _ -- 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] Fax stopped working when upgrading to 1.8.2
Did apply the patch and did a recompile, no difference, fax still not working. But I did notice one thing, when I was standing at a fax attched to PSTN and trying to send a fax to a fax attached to the Asterisk: The PSTN fax never switched to saying “Sending...” in the display just “Dialing”, but I can “hear” the Asterisk fax i answering. When I went back to Trunk version and did the same, I saw the fax display going from “Dialing” to “Sending” to “Sending OK”. I am sorry to say that I am not smart enough to know what trace I should start looking at, any knows? From: Vladimir Mikhelson Sent: Thursday, January 13, 2011 5:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: magnu...@inputinterior.se Subject: Re: [asterisk-users] Fax stopped working when upgrading to 1.8.2 Magnus, Can it be the same as I experienced https://issues.asterisk.org/view.php?id=18542 ? Do not be confused by the ticket subject, it reflects the symptoms as they looked originally You can try the patch if applicable and if you know how to compile Addons in 1.8 separately or if you have a capacity to compile the whole thing. -Vladimir On 1/13/2011 6:31 AM, magnu...@inputinterior.se wrote: Gentlemen, We have a setup as below: PSTN – E1 – Avaya – OOH323 trunk – Asterisk – SPA-2102 – Fax machine Running Asterisk SVN-trunk-r280589M, fax working as a clock. I decided to leave “trunk” and go a stable version so I upgraded to 1.8.2. Didn’t change any config files, everything worked as before except fax. I wonder if there are any known issues or things that I have missed to do in some config file. Did a downgrade to SVN-trunk-r280589M and fax started to work again. /Magnus -- _ -- 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-userswlEmoticon-sadsmile[1].png-- _ -- 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
