Re: [Asterisk-Users] Dialing Delay
Like manxpower said, set DigitTimeout to 2 seconds or whatever u want. Visit voip-info and look for urself. All whats happening is that it waiting to see if u will press another number (pattern matching) by default digitstimeout is set to 6 seconds you might want to change your dialplan as well. Example: exten = 123,1,Dial(Zap/1| 555) Since extension numbers have to be unique u can set an ext. # 123 to dial the 555 number. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialing Delay
Hello, When I dial out there is a long delay in dialing. Is this normal? Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing Delay
On Mon, 2005-01-24 at 07:42 -0800, David Shaw wrote: Hello, When I dial out there is a long delay in dialing. Is this normal? No it isn't normal. Examine/post relevant portions of config files and explain what interfaces you are using. Quick guess is the pattern match for your outbound calls is waiting for a timeout instead of matching a real specific pattern. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing Delay
David Shaw wrote: Hello, When I dial out there is a long delay in dialing. What are you dialing out to? What are you dialing out from? What does your config look like for the answer to the above questions? Is this normal? It varies. The dialing sequence from Stargate Command takes longer than the dialing sequence from Atlantis. I believe this is due to the true DHD at Atlantis, and notably upgraded Stargate. I am open for discussion. ;) As you can see, your question was rather vague, and was nowhere near specific enough to get the answer you were seeking. Unless, you just happenned to be wondering about the Stargate... -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing Delay
David Shaw wrote:
Hello, When I dial out there is a long delay in dialing. Is this normal?
For Analog FXS ports the delay is there because Asterisk has to dial the
DTMF to the line. For any other technology a delay will only happen if
you have a poorly designed dialplan.
Example:
exten = _XXX,1,Dial(Zap/1/${EXTEN}) ; Dial PSTN Local
exten = _,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN}) ; Dial extens on remote
How does Asterisk decide if you are dialing 1234 or 5551234? It can't.
It has to wait until the default DigitTimeout before completing the
call. This is an exmple of a poorly designed dialplan. This problem is
why PBXs require you do dial 9 for an outside line.
--Eric
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Re: [Asterisk-Users] Dialing Delay {Scanned}
Go easy on me I'm new.
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNKL1=Zap/1 ; Vonage line 1
TRUNKL2=Zap/2 ; Vonage line 2
TRUNKL3=Zap/3 ; Lingo line 1
TRUNKL4=Zap/4 ; Verizon home line
TRUNKMSD=1
[default]
;include = demo
exten = 300,1,Macro(stdexten,1234,SIP/300)
exten = 301,1,Macro(stdexten,1234,SIP/301)
;SpeedDial
exten = 01,1,Dial(${TRUNKL2}/XXX})
exten = 02,1,Dial(${TRUNKL2}/XXX})
exten = 03,1,Dial(${TRUNKL2}/XXX}) ;Mum Jim
exten = 05,1,Dial(${TRUNKL2}/XXX}) ;Kevin
exten = 100,1,Dial(SIP/100,30)
exten = 100,2,Voicemail,100
exten = 101,1,Dial(SIP/101,20)
exten = 101,2,Voicemail,101
exten = 102,1,Dial(SIP/102,20)
exten = 102,2,Voicemail,101
exten = 510,1,Dial(SIP/510,20)
exten = 510,2,Voicemail,510
exten = 8500,1,VoicemailMain
exten = _NXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _NXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _1NXXNXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _01144XX,1,Dial(${TRUNKL3}/${EXTEN})
[line1]
exten = s,1,Dial(SIP/101,20)
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Voicemail,101
exten = s,5,Hangup
[line2]
;exten = s,1,Dial(SIP/101,20)
exten = s,1,Answer
exten = s,2,Wait,1
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5,Authenticate(XX)
exten = s,6,DISA,no-password|default
;[line2
;exten = s,1,Dial(SIP/101,20)
;exten = s,2,Answer
;exten = s,3,Wait,1
;exten = s,4,Voicemail,101
;exten = s,5,Hangup
[line3]
exten = s,1,Dial(SIP/510,20)
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Voicemail,510
exten = s,5,Hangup
[line4]
exten = s,1,Dial(SIP/100,20)
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Voicemail,100
exten = s,5,Hangup
On Mon, 2005-01-24 at 07:42 -0800, David Shaw wrote:
Hello, When I dial out there is a long delay in dialing. Is this normal?
No it isn't normal.
Examine/post relevant portions of config files and explain what
interfaces you are using.
Quick guess is the pattern match for your outbound calls is waiting for
a timeout instead of matching a real specific pattern.
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] Dialing Delay {Scanned}
Here is my extensions.conf. Remember I'm a newbe.
Long delays on any out going call.
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNKL1=Zap/1
TRUNKL2=Zap/2
TRUNKL3=Zap/3
TRUNKL4=Zap/4 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[default]
;include = demo
exten = 300,1,Macro(stdexten,1234,SIP/300)
exten = 301,1,Macro(stdexten,1234,SIP/301)
;SpeedDial
exten = 01,1,Dial(${TRUNKL2}/X})
exten = 02,1,Dial(${TRUNKL2}/X})
exten = 03,1,Dial(${TRUNKL2}/X}) ;Mum Jim
exten = 05,1,Dial(${TRUNKL2}/X}) ;Kevin
exten = 100,1,Dial(SIP/100,30)
exten = 100,2,Voicemail,100
exten = 101,1,Dial(SIP/101,20)
exten = 101,2,Voicemail,101
exten = 102,1,Dial(SIP/102,20)
exten = 102,2,Voicemail,101
exten = 510,1,Dial(SIP/510,20)
exten = 510,2,Voicemail,510
exten = 8500,1,VoicemailMain
exten = _NXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _NXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _1NXXNXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _01144XX,1,Dial(${TRUNKL3}/${EXTEN})
[line1]
exten = s,1,Dial(SIP/101,20)
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Voicemail,101
exten = s,5,Hangup
[line2]
exten = s,1,Answer
exten = s,2,Wait,1
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5,Authenticate(X)
exten = s,6,DISA,no-password|default
[line3]
exten = s,1,Dial(SIP/510,20)
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Voicemail,510
exten = s,5,Hangup
[line4]
exten = s,1,Dial(SIP/100,20)
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Voicemail,100
exten = s,5,Hangup
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 24, 2005 8:32 AM
Subject: Re: [Asterisk-Users] Dialing Delay {Scanned}
David Shaw wrote:
Hello, When I dial out there is a long delay in dialing.
What are you dialing out to?
What are you dialing out from?
What does your config look like for the answer to the above questions?
Is this normal?
It varies. The dialing sequence from Stargate Command takes longer than
the dialing sequence from Atlantis. I believe this is due to the true
DHD at Atlantis, and notably upgraded Stargate. I am open for discussion.
;)
As you can see, your question was rather vague, and was nowhere near
specific enough to get the answer you were seeking. Unless, you just
happenned to be wondering about the Stargate...
--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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Re: [Asterisk-Users] Dialing Delay {Scanned}
David Shaw wrote:
exten = 510,1,Dial(SIP/510,20)
exten = 510,2,Voicemail,510
exten = 8500,1,VoicemailMain
exten = _NXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _NXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _1NXXNXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _01144XX,1,Dial(${TRUNKL3}/${EXTEN})
You have overlapping patterns. How does Asterisk know that you are
dialing 510 and not 510-1234? It doesn't. That's why people use
dial 9 and never start their extensions with 9.
Try:
exten = _9NXX,1,Dial(${TRUNKL4}/${EXTEN:1})
exten = _9NXX,2,Dial(${TRUNKL2}/${EXTEN:1})
exten = _91NXXNXX,1,Dial(${TRUNKL4}/${EXTEN:1})
exten = _91NXXNXX,2,Dial(${TRUNKL2}/${EXTEN:1})
exten = _901144XX,1,Dial(${TRUNKL3}/${EXTEN:1})
Also you do not want to just blindly dial the call again in priority
2. You want to find out what the status was of the previous Dial (See
show application dial, README.variables, and the stdexen macro in
the extensions.conf.sample.
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Re: [Asterisk-Users] Dialing Delay {Scanned}
On Mon, 2005-01-24 at 13:47 -0600, Eric Wieling wrote:
David Shaw wrote:
exten = 510,1,Dial(SIP/510,20)
exten = 510,2,Voicemail,510
exten = 8500,1,VoicemailMain
exten = _NXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _NXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _1NXXNXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _01144XX,1,Dial(${TRUNKL3}/${EXTEN})
You have overlapping patterns. How does Asterisk know that you are
dialing 510 and not 510-1234? It doesn't. That's why people use
dial 9 and never start their extensions with 9.
Try:
exten = _9NXX,1,Dial(${TRUNKL4}/${EXTEN:1})
exten = _9NXX,2,Dial(${TRUNKL2}/${EXTEN:1})
exten = _91NXXNXX,1,Dial(${TRUNKL4}/${EXTEN:1})
exten = _91NXXNXX,2,Dial(${TRUNKL2}/${EXTEN:1})
exten = _901144XX,1,Dial(${TRUNKL3}/${EXTEN:1})
Also you do not want to just blindly dial the call again in priority
2. You want to find out what the status was of the previous Dial (See
show application dial, README.variables, and the stdexen macro in
the extensions.conf.sample.
You also missed what appears to be another set of problems.
IAXINFO=guest ; IAXtel username/password
TRUNKL1=Zap/1 ; Vonage line 1
TRUNKL2=Zap/2 ; Vonage line 2
TRUNKL3=Zap/3 ; Lingo line 1
TRUNKL4=Zap/4 ; Verizon home line
Specifically, it appears that he is using a TDM card to dial an external
adapter for Vonage. So that means you enter the digits yourself, then
asterisk enters them on the analog line to the adapter, then it has to
signal it out to the far side. So there is a possibility that there is
also a pattern match problem on the vonage SIP device
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] Dialing Delay {Scanned}
Thanks Everyone for the help. I want to change over to Broadvoice and dump
the ATAs from Vonage and Lingo. That should help with dialing delays. (I
hope)
Thanks, David
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 24, 2005 12:26 PM
Subject: Re: [Asterisk-Users] Dialing Delay {Scanned}
On Mon, 2005-01-24 at 13:47 -0600, Eric Wieling wrote:
David Shaw wrote:
exten = 510,1,Dial(SIP/510,20)
exten = 510,2,Voicemail,510
exten = 8500,1,VoicemailMain
exten = _NXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _NXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _1NXXNXX,1,Dial(${TRUNKL4}/${EXTEN})
exten = _1NXXNXX,2,Dial(${TRUNKL2}/${EXTEN})
exten = _01144XX,1,Dial(${TRUNKL3}/${EXTEN})
You have overlapping patterns. How does Asterisk know that you are
dialing 510 and not 510-1234? It doesn't. That's why people use
dial 9 and never start their extensions with 9.
Try:
exten = _9NXX,1,Dial(${TRUNKL4}/${EXTEN:1})
exten = _9NXX,2,Dial(${TRUNKL2}/${EXTEN:1})
exten = _91NXXNXX,1,Dial(${TRUNKL4}/${EXTEN:1})
exten = _91NXXNXX,2,Dial(${TRUNKL2}/${EXTEN:1})
exten = _901144XX,1,Dial(${TRUNKL3}/${EXTEN:1})
Also you do not want to just blindly dial the call again in priority
2. You want to find out what the status was of the previous Dial (See
show application dial, README.variables, and the stdexen macro in
the extensions.conf.sample.
You also missed what appears to be another set of problems.
IAXINFO=guest ; IAXtel
username/password
TRUNKL1=Zap/1 ; Vonage line 1
TRUNKL2=Zap/2 ; Vonage line 2
TRUNKL3=Zap/3 ; Lingo line 1
TRUNKL4=Zap/4 ; Verizon home line
Specifically, it appears that he is using a TDM card to dial an external
adapter for Vonage. So that means you enter the digits yourself, then
asterisk enters them on the analog line to the adapter, then it has to
signal it out to the far side. So there is a possibility that there is
also a pattern match problem on the vonage SIP device
--
Steven Critchfield [EMAIL PROTECTED]
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[Asterisk-Users] Dialing delay when using Zap channels
Good day, I've got around to installing an X100P card in my computer to try out asterisk. I noticed (and people who were testing with me also noticed) that when dialing from my SIP soft phone to the PSTN, the ringer tone changes after 2-3 seconds, precisely when the Zap channel takes over the call. Is it possible to eliminate the first ringing? Is there a reason to this delay-before-choosing-a-channel? Thanks, -- === Mathieu Nantel - RHCE,BOFH Ecopia BioSciences Systems Manager (514) 336-2724 x434 [EMAIL PROTECTED] === [*] Please avoid sending me Word/Excel/PowerPoint attachments: this assumes that I run MS Office, which is not always the case. See: http://www.fsf.org/philosophy/no-word-attachments.html === ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing delay when using Zap channels
Remove the evil r option from the Dial line in Asterisk. Mathieu Nantel wrote: Good day, I've got around to installing an X100P card in my computer to try out asterisk. I noticed (and people who were testing with me also noticed) that when dialing from my SIP soft phone to the PSTN, the ringer tone changes after 2-3 seconds, precisely when the Zap channel takes over the call. Is it possible to eliminate the first ringing? Is there a reason to this delay-before-choosing-a-channel? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialing delay question.
John Bittner wrote: Hello. I have been working on getting my asterisk box to connect to a lucent definity PBX using a T100p. I connected it to a t1 port on the lucent Let me start by saying I have not worked on a lucent definity. Having said that, I'll tell you my thoughts, and maybe they're things you have not yet considered. call goes through. On the asterisk console I see the partial phone numbers when this happens. It looks like the lucent is set to an immediate-type mode. This is not very surprising, as that is how my Nortel MICS acts. It shouldn't be a really big deal. How does this type of setup work. Is this a lucent issue or an asterisk issue. Does the lucent pbx just open a channel, send digits I think that's what is happening. I would try increasing the timeout in your context that receives calls from the lucent. I'm not sure right off-hand how to do this, but when you find the command, I'd appreciate it, as I'll probably have to fight the same battle in a couple of days. Having said that, I'm looking at my Nortel, and can set how long the time outs are. Maybe you can do something like that for the Lucent? I'd also take a good, long look at my extensions.conf file. How does your outgoing look? Can Asterisk attempt to dial a phone number that is not appropriately long? i.e. Do you have an extension _9. or _X. or something like that? If so, try being more specific, like _1XX, or better yet, use N's where you can. This might keep Asterisk from attempting to dial before it's time. Final question: does the Lucent T1 card also support PRI? That's what we're using between our Nortel and *, and so far it seems to work well. Any help will be appreciated. Hope this helps. And please, hold the flames to a minimum. I told you I didn't know for sure what to try, but thought these might get you on the right path. Thanks, David Gomillion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialing delay question.
Hello. I have been working on getting my asterisk box to connect to a lucent definity PBX using a T100p. I connected it to a t1 port on the lucent box and ran a T1 crossover cable to the t100p. I have asterisk setup with a voicepulse account for outbound and inbound dialing. On the lucent pbx I matched the configs of a working channelized MCI t1. The only thing asterisk is doing is acting as a voip gateway. Everything works ok with only one issue. When someone makes a call outbound and hesitates on dialing the numbers, asterisk only picks up a partial number. It doesn't wait until the person dialing finished dialing the full number. If the person dials the number without any delay the call goes through. On the asterisk console I see the partial phone numbers when this happens. How does this type of setup work. Is this a lucent issue or an asterisk issue. Does the lucent pbx just open a channel, send digits and them the asterisk box has dialplan time out or is the lucent pbx to wait until it sees all the numbers before sending it to the asterisk box. Any help will be appreciated. Thanks John Bittner Zapata.conf [channels] ; ; T100P plugged into Lucent Definity ; context=voicepulse-outgoing signalling=em_w group=1 channel =1-24 rxwink=300 usecallerid=yes callwaiting=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no callerid=asreceived ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
