Re: [asterisk-users] asterisk wait for traling digits

2007-08-08 Thread Michael Rice
This is part f the phones dial plan. Our aastra phones do the same 
thing. Most phones allow you to configure the dial plan on them.

satish patel wrote:
 i have only one single 16XX dialplan for reached to avaya system then 
 why i have to wait for more digit
 
 satish patel
 
 */Don Pobanz [EMAIL PROTECTED]/* wrote:
 
   satish patel said
  
   I have asterisk setup now what happend
   when i dial 4 digit number my asterisk wait for few digit why
   when i press # key it is dialing fast but without # wait for
   few number is there any configuration for dialplan
 
 This part of the dial plan looks like it should dial without the wait.
 Could there be another part of your dial plan that starts with '16'? If
 not have you reloaded extenions.conf either by restarting asterisk or
 doing an 'extensions reload'?
 
 Don Pobanz
 
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-- 
Michael Rice
Systems Administrator
Office: 210-366-2500
Ext.  : 231
Direct: 210-293-6231
McClelland and Hine, Inc.


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[asterisk-users] Wrinkled faxes or missing lines with Hylafax + IAXModem + Asterisk

2007-07-31 Thread Michael Rice
Hello.

We are running Asterisk 1.2.23 iaxmodem-0.2.1 and hylafax-4.3.3

When we send faxes the people who receive the faxes complain that they 
look wrinkled or smashed up. Sometimes they are missing random lines. 
Has anyone seen this happen, or know how to fix it?



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Re: [asterisk-users] Wrinkled faxes or missing lines with Hylafax + IAXModem + Asterisk

2007-07-31 Thread Michael Rice
We have a PRI and use a sangoma a101d to a PRI. The Asterisk and 
IAXModem are on the same box.
Here is a link to the output from cat /proc/interrupts

http://fluxbox.pastebin.ca/640841

I put it here since you recommended putting this question on the 
IAXModem list.

Thanks for any help

Lee Howard wrote:
 Michael Rice wrote:
 
 Hello.

 We are running Asterisk 1.2.23 iaxmodem-0.2.1 and hylafax-4.3.3

 When we send faxes the people who receive the faxes complain that they 
 look wrinkled or smashed up. Sometimes they are missing random lines. 
 Has anyone seen this happen, or know how to fix it?

 
 Well, firstly, this is probably best handled on the iaxmodem lists, but...
 
 How are you interfacing with the PSTN?  Zaptel, I presume?
 
 iaxmodem and Asterisk are on the same box?
 
 What does 'cat /proc/interrupts' say?
 
 Lee.
 
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[asterisk-users] Trouble getting sound from a call

2007-07-30 Thread Michael Rice
Having some issues with getting sound from a call.
I have 4 systems. 3 main systems which handle calls for our 3 locations. 
The 4th system is the central voice mail system. When an inbound call 
gets passed to someones voice mail its done with an IAX2 connection. The 
same happens after hours when we have our night mode set. If you dial 
the main number after hours you are passed straight to the voice mail 
server where I have an IVR set to answer/handle the calls:

[ivr-1]
include = heading-out
exten = h,1,Hangup
exten = s,1,Set(LOOPCOUNT=0)
exten = s,n,Set(__DIR-CONTEXT=default)
exten = s,n,Set(_IVR_CONTEXT_${CONTEXT}=${IVR_CONTEXT})
exten = s,n,Set(_IVR_CONTEXT=${CONTEXT})
exten = s,n,GotoIf($[${CDR(disposition)} = ANSWERED]?begin)
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n(begin),Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=60)
exten = s,n,Background(custom/mhi-main-greeting)
exten = s,n,WaitExten()
exten = #,1,Goto(app-directory,#,1)
exten = #,n,dbDel(${BLKVM_OVERRIDE})
exten = #,n,Set(__NODEST=)
exten = #,n,Goto(app-pbdirectory,pbdirectory,1)
exten = hang,1,Playback(vm-goodbye)
exten = hang,n,Hangup
exten = i,1,dbDel(${BLKVM_OVERRIDE})
exten = i,n,Set(__NODEST=)
exten = i,n,Goto(ivr-1,s,begin)
exten = t,1,dbDel(${BLKVM_OVERRIDE})
exten = t,n,Set(__NODEST=)
exten = t,n,Goto(app-blackhole,hangup,1)
exten = 0,1,Goto(incoming,252,1)

[heading-out]
include = call-sa-users
include = call-dal-users
include = call-hou-users

[call-dal-users]
exten = 101,1,Dial(IAX2/toPBX2/${EXTEN})
exten = 101,n,Hangup
exten = 102,1,Dial(IAX2/toPBX2/${EXTEN})
exten = 102,n,Hangup
exten = 103,1,Dial(IAX2/toPBX2/${EXTEN})
exten = 103,n,Hangup
exten = 104,1,Dial(IAX2/toPBX2/${EXTEN})
exten = 104,n,Hangup

[call-hou-users]
exten = 150,1,Dial(IAX2/toPBX3/${EXTEN})
exten = 150,n,Hangup
exten = 151,1,Dial(IAX2/toPBX3/${EXTEN})
exten = 151,n,Hangup
exten = 152,1,Dial(IAX2/toPBX3/${EXTEN})
exten = 152,n,Hangup
exten = 153,1,Dial(IAX2/toPBX3/${EXTEN})
exten = 153,n,Hangup

[call-sa-users]
exten = 200,1,Dial(IAX2/toPBX1/${EXTEN})
exten = 200,n,Hangup
exten = 201,1,Dial(IAX2/toPBX1/${EXTEN})
exten = 201,n,Hangup
exten = 202,1,Dial(IAX2/toPBX1/${EXTEN})
exten = 202,n,Hangup
exten = 203,1,Dial(IAX2/toPBX1/${EXTEN})
exten = 203,n,Hangup

[app-directory]
include = app-directory-custom
exten = #,1,Answer
exten = #,n,Wait(1)
exten = 
#,n,AGI(directory,${DIR-CONTEXT},heading-out,${DIRECTORY:0:1}${DIRECTORY_OPTS})
exten = #,n,Playback(vm-goodbye)
exten = #,n,Hangup
exten = i,1,Playback(privacy-incorrect)



If you know the persons extension who you want to call you can dial it 
and if they don't answer you get passed back to the voice mail system 
and the persons message is played, you can hear it play, and you are 
able to leave them a message. The problem comes if you hit # to enter 
the directory. Once you find the person you are looking for and you hit 
1 to dial them their phone rings, if they pick up you can talk to them 
fine and there are no audio problems. If they don't answer and you get 
passed back to the voice mail system I see the system answer the call

 -- Executing Goto(IAX2/sapeer-1, ivr-1|s|1) in new stack
 -- Goto (ivr-1,s,1)
 -- Executing Set(IAX2/sapeer-1, LOOPCOUNT=0) in new stack
 -- Executing Set(IAX2/sapeer-1, __DIR-CONTEXT=default) in new stack
 -- Executing Set(IAX2/sapeer-1, _IVR_CONTEXT_ivr-1=) in new stack
 -- Executing Set(IAX2/sapeer-1, _IVR_CONTEXT=ivr-1) in new stack
 -- Executing GotoIf(IAX2/sapeer-1, 0?begin) in new stack
 -- Executing Answer(IAX2/sapeer-1, ) in new stack
 -- Executing Wait(IAX2/sapeer-1, 1) in new stack
 -- Executing Set(IAX2/sapeer-1, TIMEOUT(digit)=3) in new stack
 -- Digit timeout set to 3
 -- Executing Set(IAX2/sapeer-1, TIMEOUT(response)=60) in new stack
 -- Response timeout set to 60
 -- Executing BackGround(IAX2/sapeer-1, 
custom/mhi-main-greeting) in new stack
 -- Playing 'custom/mhi-main-greeting' (language 'en')
   == CDR updated on IAX2/sapeer-1
 -- Executing Goto(IAX2/sapeer-1, app-directory|#|1) in new stack
 -- Goto (app-directory,#,1)
 -- Executing Answer(IAX2/sapeer-1, ) in new stack
 -- Executing Wait(IAX2/sapeer-1, 1) in new stack
 -- Executing AGI(IAX2/sapeer-1, directory|default|heading-out|) 
in new stack
 -- Launched AGI Script /var/lib/asterisk/agi-bin/directory
 -- Playing 'dir-intro-fn' (language 'en')
   ==  directory|default|heading-out|: Found 
/var/spool/asterisk/voicemail/default/231/greet.wav
   directory|default|heading-out|: -- Playing 'dir-instr' (language 'en')
 -- AGI Script directory completed, returning 0
 -- Executing Dial(IAX2/sapeer-1, IAX2/toPBX1/231) in new stack
 -- Called toPBX1/231
 -- Call accepted by 192.168.81.2 (format ulaw)
 -- Format for call is ulaw
 -- IAX2/toPBX1-2 is ringing
 -- IAX2/toPBX1-2 stopped sounds
 -- Accepting AUTHENTICATED call from 192.168.81.2:
 requested format = ulaw,