Re: [asterisk-users] queue.conf - Set(MONITOR_FILENAME=${})
Thanks for the answer. My actual solution is setting the name before entering the queque. I promise I'll think about using h extension, but I think there should be a way out with doing so many processing before and after entering a queue. Thanks again ML 2010/2/17 Warren Selby wcse...@selbytech.com On Wed, Feb 17, 2010 at 2:47 PM, Mariano Lecuona mlecu...@gmail.comwrote: Could anyone get the MONITOR_FILENAME set from the queue.conf with variables like: MEMBERINTERFACE is the interface name (eg. Agent/1234) MEMBERNAME is the member name (eg. Joe Soap) MEMBERCALLS is the number of calls that interface has taken, MEMBERLASTCALL is the last time the member took a call. MEMBERPENALTY is the penalty of the member MEMBERDYNAMIC indicates if a member is dynamic or not MEMBERREALTIME indicates if a member is realtime or not or QUEUENAME name of the queue QUEUEMAX maxmimum number of calls allowed QUEUESTRATEGY the strategy of the queue; QUEUECALLS number of calls currently in the queue QUEUEHOLDTIME current average hold time QUEUECOMPLETED number of completed calls for the queue QUEUEABANDONED number of abandoned calls QUEUESRVLEVEL queue service level QUEUESRVLEVELPERF current service level performance My version of asterisk is: 1.4.23.1 Thanks ML This doesn't directly answer your question, but you may look at setting a unique filename before entering the queue, then do some post processing in the h extension? -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue.conf - Set(MONITOR_FILENAME=${})
All,
I am trying to set a monitor file from the queue.conf as specified on
http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf In order to
avoid the default MONITOR_FILENAME format wich is:
agent-x-uniqueid.wav for example agent-10017-1266438575-26.wav
As you may now, when using the queue command you are not able to know which
agent will take the call, until someone take it, so its impossible to set
the MONITOR_FILENAME with the agent number before executing the exten =
s,x,QUEUE(${Q_NAME}${TIMEOUT})
Could anyone get the MONITOR_FILENAME set from the queue.conf with variables
like:
MEMBERINTERFACE is the interface name (eg. Agent/1234)
MEMBERNAME is the member name (eg. Joe Soap)
MEMBERCALLS is the number of calls that interface has taken,
MEMBERLASTCALL is the last time the member took a call.
MEMBERPENALTY is the penalty of the member
MEMBERDYNAMIC indicates if a member is dynamic or not
MEMBERREALTIME indicates if a member is realtime or not
or
QUEUENAME name of the queue
QUEUEMAX maxmimum number of calls allowed
QUEUESTRATEGY the strategy of the queue;
QUEUECALLS number of calls currently in the queue
QUEUEHOLDTIME current average hold time
QUEUECOMPLETED number of completed calls for the queue
QUEUEABANDONED number of abandoned calls
QUEUESRVLEVEL queue service level
QUEUESRVLEVELPERF current service level performance
My version of asterisk is: 1.4.23.1
Thanks
ML
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Re: [asterisk-users] app_dial.c: Unable to create channel oftype 'Zap' (cause 34 - Circuit/channel congestion)
As far as my experience, this problem occurs when the asterisk tries to take a new channel and teco does not count with any available channels. Contact your E1/T1 provider and work with them to search on the teco side. 2010/2/12 Tzafrir Cohen tzafrir.co...@xorcom.com On Fri, Feb 12, 2010 at 11:15:17AM +1100, Lee, John (Sydney) wrote: What is the output of 'cat /proc/dahdi/1' ? I did not record it but it just shows every channel as 'red alarm'. How many channels? E1 or T1? What do you have in /etc/zaptel.conf ? loadzone=au defaultzone=au # # For OnRamp 10 # span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17-31 dchan=16 # # Rhino 24-port Channel Bank # span=2,0,0,esf,b8zs fxols=32-55 So span 1 is E1 and span 2 is T1. Are you sure things weren't confused somehow? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can an agent Login to a queue and be paused
What Id did was on the dialplan, create an specifica extension for login
agents. Lets say Agent/10017, then
When dial 2110017 the agents is promts for Agent passwd.Then I have a macro
only for pausing agents depending on the meaning.
So if the agent is successfully granted on the Login Context, that same
context goto pause macro.
Quick example:
[queues_logon]
; Agent Login Procedure
exten = _211,1,Answer()
exten = _211,n,NoCDR()
exten = _211,n,GotoIf($[${LEN(${AGENTBYCALLERID_${CALLERID(number)}})}
1 ]?4:5) ; Check that the physical extension is free
exten = _211,n,AgentCallbackLogin(${EXTEN:2},,${CALLERID(number)}) ;
Ask for agent password and log the agent on
exten = _211,n,Macro(agent_pause_reason,${EXTEN:2},30) ; Put Agents
into Initial Paused State on the Queue
exten = _211,n,Hangup()
[macro-agent_pause]
; ${ARG1} - Agent_nro
exten = s,1,PauseQueueMember(|Agent/${ARG1})
exten = s,n,MacroExit
2010/2/8 Lenz Emilitri lenz.lo...@gmail.com
I'm not sure if this works for newer versions of Asterisk, but on old ones,
you could pause an agent and THEN log him on, and he'd be paused.
l.
2010/2/4 Robert Grignon rgrig...@fleetone.com
I thought there was an option for this but cant find it
We have a busy callcenter and I would like the agents to log in and be
in a paused state upon login... Right now they login and they are
instantly receiving a call
Thanks for the input...
--
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Re: [asterisk-users] Can an agent Login to a queue and be paused
YEas actually a Web Gui is the best idea. My dialplan gives a backup
solution for login
2010/2/8 Robert Grignon rgrig...@fleetone.com
Not a bad idea... We use queuemetrics and the login is done via Web GUI.
I could easily just send it to pause upon login...
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mariano Lecuona
*Sent:* Monday, February 08, 2010 8:20 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Can an agent Login to a queue and be
paused
What Id did was on the dialplan, create an specifica extension for login
agents. Lets say Agent/10017, then
When dial 2110017 the agents is promts for Agent passwd.Then I have a macro
only for pausing agents depending on the meaning.
So if the agent is successfully granted on the Login Context, that same
context goto pause macro.
Quick example:
[queues_logon]
; Agent Login Procedure
exten = _211,1,Answer()
exten = _211,n,NoCDR()
exten = _211,n,GotoIf($[${LEN(${AGENTBYCALLERID_${CALLERID(number)}})}
1 ]?4:5) ; Check that the physical extension is free
exten = _211,n,AgentCallbackLogin(${EXTEN:2},,${CALLERID(number)}) ;
Ask for agent password and log the agent on
exten = _211,n,Macro(agent_pause_reason,${EXTEN:2},30) ; Put Agents
into Initial Paused State on the Queue
exten = _211,n,Hangup()
[macro-agent_pause]
; ${ARG1} - Agent_nro
exten = s,1,PauseQueueMember(|Agent/${ARG1})
exten = s,n,MacroExit
2010/2/8 Lenz Emilitri lenz.lo...@gmail.com
I'm not sure if this works for newer versions of Asterisk, but on old
ones, you could pause an agent and THEN log him on, and he'd be paused.
l.
2010/2/4 Robert Grignon rgrig...@fleetone.com
I thought there was an option for this but cant find it
We have a busy callcenter and I would like the agents to log in and be
in a paused state upon login... Right now they login and they are
instantly receiving a call
Thanks for the input...
--
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[asterisk-users] NVFaxDetect
Hi all, Do anyone has a detailed procedure for NV_application install? I have search as I was told, but I did no find any thing accurate. Thanks ML -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FAX over ISDN PRI
Hi,
All I just want to be able to detect the fax signail while doing an outbout
call taking advance of the out_dialout feature of asterisk. So for to have a
clear image on how i am doing it.
I have my .call that I move the /var/spool/asterisk/ouotgoing like:
(numbers were changed to preserve privacy)
Channel: Local/9...@dial_out/n
CallerID: Fax Test
MaxRetries: 0
RetryTime: 300
WaitTime: 60
Archive: yes
Context: dial_go
Extension: s
Priority: 1
Set: Q_NAME=511
Set: ANI=
extension.conf
[dial_out]
;
exten = _X.,1,Macro(recordcall,${Q_NAME},${CALLERID(number)}) ;; this is a
custom macro that I created for personal recording puporse
exten = _X.,n,Dial(DAHDI/g1/${EXTEN:0}) ;; this is en sime truck dial
macro that dial into the SPANS - I know it is OK for sure
exten = _X.,n,Hangup()
[dial_go]
;
exten = s,1,Answer
exten = s,n,Wait(2)
exten = s,n,Goto(${Q_NAME},1)
; If none of above happen, send to queue
exten = _s-.,1,Goto(${Q_NAME},1)
;
exten = fax,1,QueueLog(${Q_NAME}|${UNIQUEID}|NONE|FAXDETECTED|${ANI})
exten = fax,n,Hangup()
;
exten = _512,1,Goto(voicemenu-custom-1,s,1)
;
exten = _5[01]X,1,Queue(${EXTEN}${Q_TIMEOUT})
exten = _5[01]X,n,Hangup()
;
chan_dahdi.conf
faxdetect = incoming
I have successfully detected and Answer Machine with AMD application, but as
I experience some dificulties to detect fax I have removed on AMD structure
form the dial plan, only to focus on fax detection. Here is the asterisk
console output when placing a call to fax destination.
[Jan 30 17:49:14] -- Attempting call on Local/9990...@dial_out/n for
s...@dial_go:1 (Retry 1)
[Jan 30 17:49:14] -- Executing [9990...@dial_out:1]
Macro(Local/9990...@dial_out-dda8,2, recordcall|511|9990909) in new
stack
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:1]
GotoIf(Local/9990...@dial_out-dda8,2, 1?5:2) in new stack
[Jan 30 17:49:14] -- Goto (macro-recordcall,s,5)
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:5]
Set(Local/9990...@dial_out-dda8,2,
FILEREC=2010-01-30-17-49-14-SRC-511-DST-9990909) in new stack
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:6]
Set(Local/9990...@dial_out-dda8,2,
FILE_PATH=2010/01/30/2010-01-30-17-49-14-SRC-511-DST-9990909) in new stack
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:7]
Set(Local/9990...@dial_out-dda8,2,
CDR(userfield)=2010-01-30-17-49-14-SRC-511-DST-9990909.wav) in new stack
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:8]
MixMonitor(Local/9990...@dial_out-dda8,2,
/opt/rec/2010/01/30/2010-01-30-17-49-14-SRC-511-DST-9990909.wav|b) in new
stack
[Jan 30 17:49:14] -- Executing [...@macro-recordcall:9]
MacroExit(Local/9990...@dial_out-dda8,2, ) in new stack
[Jan 30 17:49:14] -- Executing [9990...@dial_out:2]
Macro(Local/9990...@dial_out-dda8,2, dialerdial|DAHDI/g1/9990909|511) in
new stack
[Jan 30 17:49:14] -- Executing [...@macro-dialerdial:1]
Dial(Local/9990...@dial_out-dda8,2, DAHDI/g1/9990909) in new stack
[Jan 30 17:49:14] -- Requested transfer capability: 0x00 - SPEECH
[Jan 30 17:49:14] -- Called g2/9990909
[Jan 30 17:49:14] == Begin MixMonitor Recording
Local/9990...@dial_out-dda8,2
[Jan 30 17:49:14] -- DAHDI/32-1 is proceeding passing it to
Local/9990...@dial_out-dda8,2
[Jan 30 17:49:15] -- DAHDI/32-1 is ringing
[Jan 30 17:49:21] -- DAHDI/32-1 answered Local/9990...@dial_out-dda8,2
[Jan 30 17:49:21] -- Executing [...@dial_go:1]
Answer(Local/9990...@dial_out-dda8,1, ) in new stack
[Jan 30 17:49:21] -- Executing [...@dial_go:2]
Wait(Local/9990...@dial_out-dda8,1, 2) in new stack
[Jan 30 17:49:23] -- Executing [...@dial_go:3]
Goto(Local/9990...@dial_out-dda8,1, 511|1) in new stack
[Jan 30 17:49:23] -- Goto (dial_go,511,1)
[Jan 30 17:49:23] -- Executing [...@dial_go:1]
Queue(Local/9990...@dial_out-dda8,1, 511180) in new stack
[Jan 30 17:49:23] -- Started music on hold, class 'default', on channel
'Local/9990...@dial_out-dda8,1'
[Jan 30 17:49:23] -- outgoing agentcall, to agent '10017', on
'Local/3...@default-d45d,1'
[Jan 30 17:49:23] -- Executing [3...@default:1]
Dial(Local/3...@default-d45d,2, SIP/3601) in new stack
[Jan 30 17:49:23] -- Called 3601
[Jan 30 17:49:23] -- SIP/3601-096664f0 is ringing
[Jan 30 17:49:23] -- Agent/10017 is ringing
Thanks to all
Mariano
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Re: [asterisk-users] FAX over ISDN PRI
thanks.. 2010/1/30 Kevin P. Fleming kpflem...@digium.com Mariano Lecuona wrote: All I just want to be able to detect the fax signail while doing an outbout call taking advance of the out_dialout feature of asterisk. So for to have a clear image on how i am doing it. The faxdetect functionality in Asterisk is not intended to detect answering FAX machines; it is for detection of calling FAX machines. The open source NVFaxDetect application may be able to do what you want. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6
You are probabably right on that. My comments were taking into account the actual documentation on how to implement AgentCallBackLogin() from the doc. That is why i thought of a macro. Ref: doc/queues-with-callback-members.txt Thanks 2009/11/3 Lenz Emilitri lenz.lo...@gmail.com IMHO, one of the major advantages of ACBL is that the set of queues is statically defined, so at the dialplan level you do not have to keep track of the set of queues an agent is enabled on. l. 2009/11/3 Mariano Lecuona mlecu...@gmail.com My mental plan orginilly was: 1.- Creating a macro that acceps ARGs like. a.- agent b.- queue/s In the macro we could have the voice respose for the loging. I am using on 1.4 the following procedure. * the agents call to 21Agentid to loging, and it is promt just for the passwd * the agents call to 22Agentid to logoff using the same philosofy we could implement some easy marco that only ask for the password and: 1.- sets the astdb 2.- sets the globals AGENTBYCALLERID_X= 3.- adds the agent to the queues. Let me work deeper on this idea and see what comes up. ML 2009/11/2 Lenz Emilitri lenz.lo...@gmail.com We were thinking about doing something similar as well. A lot of people are asking for this. If there is anybody else interested, we could share the load I was thinking about creating a context like @agents, so that when you do the log-on you basically add Local/1...@agents as a member of the queue. When you ring it, it basically looks up for the actual device in AstDB and dials it like: Queue - (member) Local/1...@agents - (astdb) SIP/234 I think that we should be able to forward channel state as well (using hints? I've never done it) so that app_queue does not try dialling agents that are busy. I was thinking about storing queue-agent associations into config strings, and/or AstDB, and/or http over curl. And yes, ideally it should work fine on 1.4's as well Things that should be working from version one: - logging compatible with older asterisk's - authentication using Voicemail -.plug and play on most systems - channel states - pause/unpause with pause codes - ...you tell me Anybody interested? l. 2009/10/30 Mariano Lecuona mlecu...@gmail.com Hi all, I would like to know if there is any application replacement for the AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what I've read that the call back login agent can be done using a smart dialplan as showed on the docs. But I cannot thinks a flexible dialplan for a dinamic reassignation of agents to different queues every day. Thanks in advance. Mariano ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6
My mental plan orginilly was: 1.- Creating a macro that acceps ARGs like. a.- agent b.- queue/s In the macro we could have the voice respose for the loging. I am using on 1.4 the following procedure. * the agents call to 21Agentid to loging, and it is promt just for the passwd * the agents call to 22Agentid to logoff using the same philosofy we could implement some easy marco that only ask for the password and: 1.- sets the astdb 2.- sets the globals AGENTBYCALLERID_X= 3.- adds the agent to the queues. Let me work deeper on this idea and see what comes up. ML 2009/11/2 Lenz Emilitri lenz.lo...@gmail.com We were thinking about doing something similar as well. A lot of people are asking for this. If there is anybody else interested, we could share the load I was thinking about creating a context like @agents, so that when you do the log-on you basically add Local/1...@agents as a member of the queue. When you ring it, it basically looks up for the actual device in AstDB and dials it like: Queue - (member) Local/1...@agents - (astdb) SIP/234 I think that we should be able to forward channel state as well (using hints? I've never done it) so that app_queue does not try dialling agents that are busy. I was thinking about storing queue-agent associations into config strings, and/or AstDB, and/or http over curl. And yes, ideally it should work fine on 1.4's as well Things that should be working from version one: - logging compatible with older asterisk's - authentication using Voicemail -.plug and play on most systems - channel states - pause/unpause with pause codes - ...you tell me Anybody interested? l. 2009/10/30 Mariano Lecuona mlecu...@gmail.com Hi all, I would like to know if there is any application replacement for the AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what I've read that the call back login agent can be done using a smart dialplan as showed on the docs. But I cannot thinks a flexible dialplan for a dinamic reassignation of agents to different queues every day. Thanks in advance. Mariano ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Real replacement for AgentCallBackLogin() on Asterisk 1.6
Hi all, I would like to know if there is any application replacement for the AgentCallBackLogin() from asterisk 1.4 on asterisk 1.6. I know, from what I've read that the call back login agent can be done using a smart dialplan as showed on the docs. But I cannot thinks a flexible dialplan for a dinamic reassignation of agents to different queues every day. Thanks in advance. Mariano ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AEX800P on HP Prolaint ML115 kernel panic
Take a look at this document. This may help you on trouble shoot your kernel panic. http://www.novavox.co.uk/docs/install-guides/novavox-asterisk-card-installation-issues.pdf 2009/10/30 David Shauger sollost...@gmail.com Setting up a new Asterisk server with Centos 5.2 and Asterisk 1.4.23 using Dahdi and getting a kernel panic - not syncing: Fatal exception error during boot. Anyone have thoughts on what I can do to rectify this or is this card not compatible with this machine? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI not detecting RINGING Status on the Channel
I have plugges only 2 lines. That's why the rest is in RED [r...@pbx ~]# lsdahdi ### Span 1: WCTDM/8 YSTDM8xx REV E Board 9 (MASTER) 1 FXOFXSLS (In use) 2 FXOFXSLS (In use) 3 FXOFXSKS (In use) RED 4 FXOFXSKS (In use) RED 5 FXOFXSKS (In use) RED 6 FXOFXSKS (In use) RED 7 FXOFXSKS (In use) RED 8 FXOFXSKS (In use) RED 2009/10/27 Tzafrir Cohen tzafrir.co...@xorcom.com On Mon, Oct 26, 2009 at 09:02:10PM -0300, Mariano Lecuona wrote: For some reason I am not able to set loopstart instead of kewlstart: Console out put: [Oct 26 20:58:40] == Parsing '/etc/asterisk/chan_dahdi.conf': [Oct 26 20:58:40] Found [Oct 26 20:58:40] -- Registered channel 1, FXS Loopstart signalling [Oct 26 20:58:40] -- Registered channel 2, FXS Loopstart signalling [Oct 26 20:58:40] ERROR[23050]: chan_dahdi.c:7677 mkintf: Signalling requested on channel 3 is FXS Loopstart but line is in FXS Kewlstart signalling [Oct 26 20:58:40] ERROR[23050]: chan_dahdi.c:11294 build_channels: Unable to register channel '1-8' pbx*CLI module load chan_dahdi.so any ideas? What is the output of lsdahdi ? Have you edited /etc/dahdi/system.conf ? To apply changes there, run dahdi_cfg . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI not detecting RINGING Status on the Channel
I am using an 8 port tdm card and also I implemented a dialer using a .call file generator. As you know on the .call you specify the channel to call and then the contex/extension/priority to let dial plan continue when the call is bridge. My actual problem is that when the call process starts, asterisk (DAHDI) sets the channel as answered when the truth is that on the other side the channel has not started to ring yet. My felling is that the DAHDI driver cannot detect /understand the signaling of the analog line. Here is the evidence and configuration settings. I live in Argentina BTW Console output. [Oct 22 23:42:03] -- Attempting call on Local/84776...@outgoing_campaign/n for 5...@queues:2 (Retry 1) [Oct 22 23:42:03] -- Executing [84776...@outgoing_campaign:1] Macro(Local/84776...@outgoing_campaign-d2c3,2, recordcall|501|84776114)in new stack [Oct 22 23:42:03] -- Executing [...@macro-recordcall:1] GotoIf(Local/84776...@outgoing_campaign-d2c3,2, 1?5:2) in new stack [Oct 22 23:42:03] -- Goto (macro-recordcall,s,5) [Oct 22 23:42:03] -- Executing [...@macro-recordcall:5] Set(Local/84776...@outgoing_campaign-d2c3,2, FILEREC=2009-10-22-23-42-03-SRC-501-DST-84776114) in new stack [Oct 22 23:42:03] -- Executing [...@macro-recordcall:6] Set(Local/84776...@outgoing_campaign-d2c3,2, FILE_PATH=2009/10/22/2009-10-22-23-42-03-SRC-501-DST-84776114) in new stack [Oct 22 23:42:03] -- Executing [...@macro-recordcall:7] Set(Local/84776...@outgoing_campaign-d2c3,2, CDR(userfield)=2009-10-22-23-42-03-SRC-501-DST-84776114.wav) in new stack [Oct 22 23:42:03] -- Executing [...@macro-recordcall:8] MixMonitor(Local/84776...@outgoing_campaign-d2c3,2, /opt/rec/2009/10/22/2009-10-22-23-42-03-SRC-501-DST-84776114.wav|b) in new stack [Oct 22 23:42:03] -- Executing [...@macro-recordcall:9] MacroExit(Local/84776...@outgoing_campaign-d2c3,2, ) in new stack [Oct 22 23:42:03] -- Executing [84776...@outgoing_campaign:2] Macro(Local/84776...@outgoing_campaign-d2c3,2, dialtrunk-failover|DAHDI/R1/4776114|DAHDI/R2/4776114|DAHDI/R3/4776114|DAHDI/R4/4776114|span_1|span_2|span_3|span_4) in new stack [Oct 22 23:42:03] == Begin MixMonitor Recording Local/84776...@outgoing_campaign-d2c3,2 [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:1] GotoIf(Local/84776...@outgoing_campaign-d2c3,2, 0?1-fmsetcid|1) in new stack [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:2] GotoIf(Local/84776...@outgoing_campaign-d2c3,2, 0?1-setgbobname|1) in new stack [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:3] Set(Local/84776...@outgoing_campaign-d2c3,2, CALLERID(num)=) in new stack [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:4] GotoIf(Local/84776...@outgoing_campaign-d2c3,2, 0?1-dial|1) in new stack [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:5] Set(Local/84776...@outgoing_campaign-d2c3,2, CALLERID(all)=) in new stack [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:6] Goto(Local/84776...@outgoing_campaign-d2c3,2, 1-dial|1) in new stack [Oct 22 23:42:03] -- Goto (macro-dialtrunk-failover,1-dial,1) [Oct 22 23:42:03] -- Executing [1-d...@macro-dialtrunk-failover:1] Dial(Local/84776...@outgoing_campaign-d2c3,2, DAHDI/R1/4776114|90|tT) in new stack [Oct 22 23:42:03] -- Called R1/4776114 [Oct 22 23:42:05] -- DAHDI/4-1 answered Local/84776...@outgoing_campaign-d2c3,2 [Oct 22 23:42:05] Channel Local/84776...@outgoing_campaign-d2c3,1 was answered. [Oct 22 23:42:05] -- Executing [...@queues:2] Queue(Local/84776...@outgoing_campaign-d2c3,1, 501) in new stack [Oct 22 23:42:05] -- Started music on hold, class 'default', on channel 'Local/84776...@outgoing_campaign-d2c3,1' [Oct 22 23:42:05] -- outgoing agentcall, to agent '10009', on 'Local/1...@default-4a9e,1' [Oct 22 23:42:05] -- Executing [1...@default:1] Dial(Local/1...@default-4a9e,2, SIP/1000) in new stack [Oct 22 23:42:05] -- Called 1000 [Oct 22 23:42:07] -- SIP/1000-0895df08 is ringing [Oct 22 23:42:07] -- Agent/10009 is ringing ** PLease see las line with [Oct 22 23:42:05] when the output shows that Called/ and then says DAHDI/4-1 answered. [r...@pbx ~]# cat /asterisk/chan_dahdi.conf [trunkgroups] [channels] language=ar context=DID_trunk_1 signalling=fxs_ks callwaiting=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no relaxdtmf=yes usedistinctiveringdetection=yes usecallingpres=yes busydetect=yes callprogress=yes rxgain=2.0 txgain=2.0 ; group=1 channel = 1-8 callgroup=1 pickupgroup=1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI not detecting RINGING Status on the Channel
For some reason I am not able to set loopstart instead of kewlstart: Console out put: [Oct 26 20:58:40] == Parsing '/etc/asterisk/chan_dahdi.conf': [Oct 26 20:58:40] Found [Oct 26 20:58:40] -- Registered channel 1, FXS Loopstart signalling [Oct 26 20:58:40] -- Registered channel 2, FXS Loopstart signalling [Oct 26 20:58:40] ERROR[23050]: chan_dahdi.c:7677 mkintf: Signalling requested on channel 3 is FXS Loopstart but line is in FXS Kewlstart signalling [Oct 26 20:58:40] ERROR[23050]: chan_dahdi.c:11294 build_channels: Unable to register channel '1-8' pbx*CLI module load chan_dahdi.so any ideas? I am still tring to set some other parameters like: cidsignalling= cidstart= Thanks ML 2009/10/26 Danny Nicholas da...@debsinc.com It’s not the DAHDI driver; it’s the POTS service you are (presumably) using. The DAHDI driver works fine with PRI/E1 interfaces, but POTS requires “human” knowledge (it can’t tell if a line is ringing/answered, etc). The only “reasonable” solution I can suggest for this scenario is a polarity/silence detect to keep you from processing many minutes of silence and other garbage. LoopStart instead of KewlStart has been suggested in some instances (possibly applicable to you since you are non-US). -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mariano Lecuona *Sent:* Monday, October 26, 2009 2:59 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] DAHDI not detecting RINGING Status on the Channel I am using an 8 port tdm card and also I implemented a dialer using a .call file generator. As you know on the .call you specify the channel to call and then the contex/extension/priority to let dial plan continue when the call is bridge. My actual problem is that when the call process starts, asterisk (DAHDI) sets the channel as answered when the truth is that on the other side the channel has not started to ring yet. My felling is that the DAHDI driver cannot detect /understand the signaling of the analog line. Here is the evidence and configuration settings. I live in Argentina BTW Console output. [Oct 22 23:42:03] -- Attempting call on Local/84776...@outgoing_campaign/n for 5...@queues:2 (Retry 1) [Oct 22 23:42:03] -- Executing [84776...@outgoing_campaign:1] Macro(Local/84776...@outgoing_campaign-d2c3,2, recordcall|501|84776114)in new stack [Oct 22 23:42:03] -- Executing [...@macro-recordcall:1] GotoIf(Local/84776...@outgoing_campaign-d2c3,2, 1?5:2) in new stack [Oct 22 23:42:03] -- Goto (macro-recordcall,s,5) [Oct 22 23:42:03] -- Executing [...@macro-recordcall:5] Set(Local/84776...@outgoing_campaign-d2c3,2, FILEREC=2009-10-22-23-42-03-SRC-501-DST-84776114) in new stack [Oct 22 23:42:03] -- Executing [...@macro-recordcall:6] Set(Local/84776...@outgoing_campaign-d2c3,2, FILE_PATH=2009/10/22/2009-10-22-23-42-03-SRC-501-DST-84776114) in new stack [Oct 22 23:42:03] -- Executing [...@macro-recordcall:7] Set(Local/84776...@outgoing_campaign-d2c3,2, CDR(userfield)=2009-10-22-23-42-03-SRC-501-DST-84776114.wav) in new stack [Oct 22 23:42:03] -- Executing [...@macro-recordcall:8] MixMonitor(Local/84776...@outgoing_campaign-d2c3,2, /opt/rec/2009/10/22/2009-10-22-23-42-03-SRC-501-DST-84776114.wav|b) in new stack [Oct 22 23:42:03] -- Executing [...@macro-recordcall:9] MacroExit(Local/84776...@outgoing_campaign-d2c3,2, ) in new stack [Oct 22 23:42:03] -- Executing [84776...@outgoing_campaign:2] Macro(Local/84776...@outgoing_campaign-d2c3,2, dialtrunk-failover|DAHDI/R1/4776114|DAHDI/R2/4776114|DAHDI/R3/4776114|DAHDI/R4/4776114|span_1|span_2|span_3|span_4) in new stack [Oct 22 23:42:03] == Begin MixMonitor Recording Local/84776...@outgoing_campaign-d2c3,2 [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:1] GotoIf(Local/84776...@outgoing_campaign-d2c3,2, 0?1-fmsetcid|1) in new stack [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:2] GotoIf(Local/84776...@outgoing_campaign-d2c3,2, 0?1-setgbobname|1) in new stack [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:3] Set(Local/84776...@outgoing_campaign-d2c3,2, CALLERID(num)=) in new stack [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:4] GotoIf(Local/84776...@outgoing_campaign-d2c3,2, 0?1-dial|1) in new stack [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:5] Set(Local/84776...@outgoing_campaign-d2c3,2, CALLERID(all)=) in new stack [Oct 22 23:42:03] -- Executing [...@macro-dialtrunk-failover:6] Goto(Local/84776...@outgoing_campaign-d2c3,2, 1-dial|1) in new stack [Oct 22 23:42:03] -- Goto (macro-dialtrunk-failover,1-dial,1) [Oct 22 23:42:03] -- Executing [1-d...@macro-dialtrunk-failover:1] Dial(Local/84776...@outgoing_campaign-d2c3,2, DAHDI/R1/4776114|90|tT) in new stack [Oct 22 23:42:03] -- Called R1
[asterisk-users] SIP Trunk groups
Hey all,
I have 2 GSM to Voip gateways and probably we will grow up to 4 more
gateways. I already created a macro to make failover happen between
gateways, but can imagine that everytime I add a new gateway I will need to
modify the macro. The initial intention of this macro was to failover
between different techonolgies.
So I was hoping to create a Sip Trunk group using the same idea as
truckgroup under dahdi but for sip trunks.
Is that possible?, have you ever done this before?
My Idea is:
sip_trunk1 = SIP/gateway1
sip_trunk2 = SIP/gateway2
sip_trunk3 = SIP/gateway3
gsm_trunkgoup = sip_trunk1 ; sip_trunk2 ; sip_trunk3
[user]
exten = _0.,1,wait()
exten = _0.,n,Dial(gsm_trunkgoup/${ exten:1},30)
exten = _0.,n,Hangup
Thanks,
--
--
*Mariano Lecuona*
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[asterisk-users] SIP Trunk groups
He all,
I have 2 GSM to Voip gateways and probably we will grow up to 4 more
gateways. I already created a macro to make failover happen between
gateways, but can imagine that everytime I add a new gateway I will need to
modify the macro. The initial intention of this macro was to failover
between different techonolgies.
So I was hoping to create a Sip Trunk group using the same idea as
truckgroup under dahdi but for sip trunks.
Is that possible?, have you ever done this before?
My Idea is:
sip_trunk1 = SIP/gateway1
sip_trunk2 = SIP/gateway2
sip_trunk3 = SIP/gateway3
gsm_trunkgoup = sip_trunk1 ; sip_trunk2 ; sip_trunk3
[user]
exten = _0.,1,wait()
exten = _0.,n,Dial(gsm_trunkgoup/${exten:1},30)
exten = _0.,n,Hangup
Thanks,
--
--
Mariano Lecuona
___
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