Hi all, Iv' got a problem taking lines to call from SIP to PSTN. I have to press # after 9 to get ringtone, otherwise I would have to wait above 15 seconds. [out] exten => 9,1,Dial,Zap/g1/9 exten => 9,2,Hangup exten => 9,102,Congestion The problem occurs when the user doesn't complete the call, and hangup after pressing only 9. If these events occur twice consecutively, Asterisk attempts to native bridge between 2 channels. I think the problem is that # is being used like a transfer trigger. But when I deactivate these feature, I have to wait 15 second after press 9 no get line. What can I do?? What should I do to get line without spend this time? Pablo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060719/597eb326/attachment.htm
You must have other dialplan entries that start with 9.
How does asterisk know you are dialing "9" or one of your other
dialplan entries that starts with "9"?
I has to wait for the digit timeout.
I am curious what this "9" is used to connect to? Are you trying to
get dialtone from another PBX?
--
--
Steven
http://www.glimasoutheast.org
"Pablo Mora" <pablo@espoltel.net> wrote in message
news:000001c6ab37$0e70c3e0$c5954145@yusuke...
Hi all,
Iv' got a problem taking lines to call from SIP to PSTN. I have to press #
after 9 to get ringtone, otherwise I would have to wait above 15 seconds.
[out]
exten => 9,1,Dial,Zap/g1/9
exten => 9,2,Hangup
exten => 9,102,Congestion
The problem occurs when the user doesn't complete the call, and hangup
after pressing only 9. If these events occur twice consecutively, Asterisk
attempts to native bridge between 2 channels.
I think the problem is that # is being used like a transfer trigger. But when
I deactivate these feature, I have to wait 15 second after press 9 no get line.
What can I do?? What should I do to get line without spend this time?
Pablo
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Eric "ManxPower" Wieling
2006-Jul-19 06:56 UTC
[asterisk-users] Don't Hit # after 9 to get PSTN line
Turn off 3-way calling on your SIP device. Set the dialplan on your SIP device to not wait 15 seconds after pressing 9. Pablo Mora wrote:> Hi all, > > > > Iv' got a problem taking lines to call from SIP to PSTN. I have to press # > after 9 to get ringtone, otherwise I would have to wait above 15 seconds. > > > > > > [out] > > exten => 9,1,Dial,Zap/g1/9 > > exten => 9,2,Hangup > > exten => 9,102,Congestion > > > > The problem occurs when the user doesn't complete the call, and hangup after > pressing only 9. If these events occur twice consecutively, Asterisk > attempts to native bridge between 2 channels. > > > > I think the problem is that # is being used like a transfer trigger. But > when I deactivate these feature, I have to wait 15 second after press 9 no > get line. > > > > What can I do?? What should I do to get line without spend this time? > > > > Pablo > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery.
Pablo Mora
2006-Jul-19 07:11 UTC
[asterisk-users] Re: Don't Hit # after 9 to get PSTN line
Really don't. Dialplan is very simple, please take a look [incoming] exten => s,1,Answer exten => s,2,Background(prueba-pbx) exten => s,3,Set(TIMEOUT(response)=5) exten => 1001,1,Dial,SIP/1001|20 exten => 1001,2,Hangup exten => 1001,102,Congestion,3 exten => 1002,1,Dial,SIP/1002|20 exten => 1002,2,Hangup exten => 1002,102,Congestion,3 exten => 1003,1,Dial,SIP/1003|20 exten => 1003,2,Hangup [sip] include => out exten => 1001,1,Dial(SIP/1001,20) exten => 1001,2,Hangup exten => 1001,102,Congestion,3 exten => 1002,1,Dial(SIP/1002,20) exten => 1002,2,Hangup exten => 1002,102,Congestion,3 exten => 1003,1,Dial(SIP/1003,20) exten => 1003,2,Hangup [out] exten => 9,1,Dial,Zap/g1/9 exten => 9,2,Hangup exten => 9,102,Congestion And yes, I'm trying asterisk behind and Ericsson MD110 PBX, and when I hit 9 I ask for an internal line and re-send 9 to get an external line. Thanks Pablo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060719/4357e4d3/attachment.htm
I really don't understand what you say. I've been searching in my SIP device (Innomedia 3308), and there isn't any option to disable 3-way calling. Do you refer to sip.conf??? Pablo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060719/13f96fba/attachment.htm
Pablo Mora
2006-Jul-19 14:46 UTC
[asterisk-users] Re: Don't Hit # after 9 to get PSTN line
Steven,
I've been searching that you say, but certainly I don't know where to
search
or those lines isn't there.
I found these:
Configuring VoIP DigitMap
dialing pattern
- empty -
Configure FXS Setting Parameters
Ringing Timeout = 180 second
Ringing Cadence = 0
Ringing Repetition = 0
Dial Tone Timeout = 16 seconds
Echo Cancellation: Yes
Prefix Digit = NULL
Configuring SIP Settings
Current SIP Proxy Servers = 192.168.42.3
Use Outbound Proxy = No
Current Local SIP Port = 5060
Response Code for Retry Registration =
Retry Registration Interval = 0 seconds
Current SIP Domain =
Current Exponential Backoff = 500 ms
Current Exponential Cap = 2000 ms
Current Non-INVITE retry = 4 times
Current INVITE msg retry = 4 times
Current REGISTER expiration = 3600 seconds
Current Session Timer = 0 seconds
Current Bullet Interval = 0 seconds
Current Number of Codecs = 1
Current Codec List = G729A
Digitmap Partial Match Timeout = 16
Digitmap Critical Timeout = 4
Cancel Call Waiting Invoke String = *72
Call Transfer Invoke String = *90
CID Block Invoke String = *67
CID Display Invoke String = *82
Call Park Invoke String = *98
Call Retrieve Invoke String = *99
Outside Line Access Number = 9
Use User-Agent Header = Yes
Set Jitter Buffer Adaptive = Yes
Use SIP INFO for DTMF = No
Re-registration Credential Enable = No
Current SIP PING Interval = 0 seconds
Current SIP PING Proxy Require Header =
Current SIP External IP address =
Use SIP INFO for Flash Event = No
So, what do you think??
Pablo
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