Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten => 1637,1,Dial(SIP/100) Exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _NXXXXXXXXX,2,Congestion Exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _1NXXXXXXXXX,2,Congestion These extensions allow us to utilize our SIP provider - ONLY when being dialed from a regular telephone attached to a Cisco ATA-186. Our Cisco 7960 only allows us to dial 4 charachters before it tries dialing. So, I assume we need to implement 9, and the number. However, when I do this, the 9 gets passed on to our SIP provider, which tries to dial 9NXXXXXXXXXX, and all goes to hell. Question - is there a way to allow 9 in the dialing plan, without having it be passed to the sip provider. Regards, Phillip -- Phillip C. Jackson [pjackson@jacksongrp.com] ------------------------------------------------- This mail sent through IMP: http://horde.org/imp/
<quote who="Phillip Jackson">> So, I assume we need to implement 9, and the number. However, when I > do this, the 9 gets passed on to our SIP provider, which tries to dial > 9NXXXXXXXXXX, and all goes to hell. > > Question - is there a way to allow 9 in the dialing plan, without having > it be passed to the sip provider.exten => _9NXXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@sipdemo) -- END OF LINE
Phillip, exten => _9NXXXXX,1,StripMSD,1 Exten => _NXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _NXXXXX,2,Congestion Should work Dave -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Phillip Jackson Sent: 26 October 2003 23:35 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Extensions Problem Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten => 1637,1,Dial(SIP/100) Exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _NXXXXXXXXX,2,Congestion Exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _1NXXXXXXXXX,2,Congestion These extensions allow us to utilize our SIP provider - ONLY when being dialed from a regular telephone attached to a Cisco ATA-186. Our Cisco 7960 only allows us to dial 4 charachters before it tries dialing. So, I assume we need to implement 9, and the number. However, when I do this, the 9 gets passed on to our SIP provider, which tries to dial 9NXXXXXXXXXX, and all goes to hell. Question - is there a way to allow 9 in the dialing plan, without having it be passed to the sip provider. Regards, Phillip -- Phillip C. Jackson [pjackson@jacksongrp.com] ------------------------------------------------- This mail sent through IMP: http://horde.org/imp/ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
You may have a file called dialplan.xml being TFTP'd to your phone. It has a number of rules in it for helping the phone to determine when it has complete number. It may need some tuning to bring it in line with what you need. I've found that the phone appears to treat the contents of the file as a hash rather than as a sorted list. That is, certain rules that appear later in the file actually get used before rules earlier in the file. I think the rules get used in a 'shortest match first' scenario. Regards, Ray Burkholder http://www.oneunified.net 704 576 5101> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Phillip Jackson > Sent: October 26, 2003 18:35 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Extensions Problem > > > Hello again, > > Here's the next big issue, I thought I'd let you munch on. > We are utilizing > Cisco 7960's and the following entries in our extensions.conf file: > > Exten => 1637,1,Dial(SIP/100) > Exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) > Exten => _NXXXXXXXXX,2,Congestion > Exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) > Exten => _1NXXXXXXXXX,2,Congestion > > These extensions allow us to utilize our SIP provider - ONLY > when being dialed > from a regular telephone attached to a Cisco ATA-186. Our > Cisco 7960 only > allows us to dial 4 charachters before it tries dialing. So, > I assume we need > to implement 9, and the number. However, when I do this, the > 9 gets passed on > to our SIP provider, which tries to dial 9NXXXXXXXXXX, and > all goes to hell. > > Question - is there a way to allow 9 in the dialing plan, > without having it be > passed to the sip provider. > > Regards, > Phillip > > > -- > Phillip C. Jackson > [pjackson@jacksongrp.com] > > ------------------------------------------------- > This mail sent through IMP: http://horde.org/imp/ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > Scanned for viruses and dangerous content at > http://www.oneunified.net and is believed to be clean. >-- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean.
Hi, I've got 2 x100p's installed in my system. Both execute the same incoming contexts as follows: [inboundA] include => dialjon [inboundB] include => dialjon|09:00-16:30|Mon-Fri|*|* [dialjon] exten => s,1,answer exten => s,2,Dial(SIP/2000,15) exten => s,3,Playback(noone) exten => s,103,Goto(onphone,s,1) <snip> Am I right in saying: if no one answers at ext 2000 then s,3 is executed. if ext 2000 is busy then 103 is executed. If so then sometihng is wrong. If I'm already on a call, I want 103 to be executed however, this isn't happening. If a new call comes in (whilst I'm talking on ext 2000) I here it ringing on my handset. Can anyone point out where I've gone wrong ? TIA Jon
Your phone supports call waiting, so isn't giving out busy. I had the same problem with a budgetone 102, you can't turn this off on the phone but you can work round it by adding Incominglimit=1 Into the sip.conf entry for the phone>From: Jon Lawrence <jon@lawrence.org.uk> >To: asterisk-users@lists.digium.com >Date: Mon, 15 Mar 2004 15:29:01 +0000 >Subject: [Asterisk-Users] extensions problem >Reply-To: asterisk-users@lists.digium.com>Hi, >I've got 2 x100p's installed in my system. >Both execute the same incoming contexts as follows: >[inboundA] >include => dialjon >[inboundB] >include => dialjon|09:00-16:30|Mon-Fri|*|* > >[dialjon] >exten => s,1,answer >exten => s,2,Dial(SIP/2000,15) >exten => s,3,Playback(noone) >exten => s,103,Goto(onphone,s,1) ><snip>>Am I right in saying: >if no one answers at ext 2000 then s,3 is executed. >if ext 2000 is busy then 103 is executed.>If so then sometihng is wrong. If I'm already on a call, I want 103 tobe>executed however, this isn't happening. If a new call comes in (whilstI'm>talking on ext 2000) I here it ringing on my handset.>Can anyone point out where I've gone wrong ?>TIA >Jon