RE: [asterisk-users] Help with IAX Trunk
Yes. That was the solution. Not sure why that 'r' is there in the first place David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Saturday, December 02, 2006 11:57 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Help with IAX Trunk On 09:48, Sat 02 Dec 06, Dave Morrow wrote: H.interesting thought. Not sure how to do it though... I found this this morning. I think it might be the answer I seek http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=5303forum =2 Probably yeah. The r option in the dial command will not pass early media but instead generates it's own. I find the r flag for dial and queue the wrong thing to do. In dial it will disable stuff like 'this call will cost you 300 euro a minute and that's something I really wanna hear. In queue() it will kill the periodic announcements. annoying as well. I removed them from everywhere in my extensions.conf and my system is much more usable. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --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 ___ --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
[asterisk-users] * key on Linksys SPA-841
I wonder if anyone has experienced an issue I have found with the Linksys SPA-841 phone. On my Asterisk (Trixbox 2), to login to a queue, a user must enter the queue number, followed by the * key. This works fine on my Companies mix of phones, with the exception of the Linksys (Sipura) SPA-841. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com http://www.autodatasolutions.com/ Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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
[asterisk-users] Help with IAX Trunk
Hi all. I have an IAX trunk between 2 Asterisk servers. Everything is working correctly dialing between the servers as well as through the PSTN (a T1 connected to one of the servers). The second Asterisk server routes all calls to the PSTN via the first server. Calls to local 10-digit, and toll free calls are working properly. My long distance provider requires that a billing code be entered after dialing a long distance call. From the directly attached Asterisk server, these calls work when the user enters their PIN after dialing. From the second server (connected via an IAX trunk), I never get the tone to enter the long distance PIN..all I get is a steady ringtone. Has anyone encountered this or know how to fix it? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com http://www.autodatasolutions.com/ Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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
RE: [asterisk-users] Help with IAX Trunk
Unfortunately, the codes are private for the individual. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Saturday, December 02, 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with IAX Trunk Dave Morrow wrote: My long distance provider requires that a billing code be entered after dialing a long distance call. From the directly attached Asterisk server, these calls work when the user enters their PIN after dialing. From the second server (connected via an IAX trunk), I never get the tone to enter the long distance PIN..all I get is a steady ringtone. Instead of having the user enter the billing code, maybe you could program it to be sent via the dial plan? Or, is the code different each time? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 ___ --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
RE: [asterisk-users] Help with IAX Trunk
H.interesting thought. Not sure how to do it though... I found this this morning. I think it might be the answer I seek http://www.trixbox.org/modules/newbb/viewtopic.php?topic_id=5303forum=2 David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Saturday, December 02, 2006 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with IAX Trunk Dave Morrow wrote: Unfortunately, the codes are private for the individual. Then I would suggest that you prompt the user for that code, before the actual dial. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 ___ --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
[asterisk-users] One extension to ring on multiple outside lines
I have a need to have a single extension actually ring on 2 phone lines which are not extensions (they are analog phone lines). Does anyone know a suitable extensions.conf config for this? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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
RE: [asterisk-users] One extension to ring on multiple outside lines
Yes, to some extent it is what I want, but I want it to dial outside lines (ie. 800-555-1212 and 800-666-3434) insteand of a Zap channel. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: Friday, July 28, 2006 6:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] One extension to ring on multiple outside lines - Original Message - From: Dave Morrow [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 11:34:37 -0300 Subject: [asterisk-users] One extension to ring on multiple outside lines I have a need to have a single extension actually ring on 2 phone lines which are not extensions (they are analog phone lines). Does anyone know a suitable extensions.conf config for this? Sure! exten = 145,1,Dial(Zap/1Zap/2) That line would dial both Zap/1 and Zap/2 whenever someone called 145. The first one to answer gets the call. Is that what you were looking for? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com http://www.autodatasolutions.com/ Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! Joshua Colp Digium ___ --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 ___ --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
RE: [Asterisk-Users] features.conf *1 Call Recording
Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call RecordingOK. You lost me.David MorrowTechnical Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attachedmaterial is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom theyare addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording2006/5/10, Dave Morrow [EMAIL PROTECTED] : I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success.Is there a trick to this?May be a problem with the way you are sending the dialtones. Try sending as data.--Alejandro Vargas___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
RE: [Asterisk-Users] features.conf *1 Call Recording
All I see when I press *1 is -- Attempting native bridge of SIP/8001-252e and SIP/3020-5171 I still cannot make this work. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] features.conf *1 Call Recording Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call RecordingOK. You lost me.David MorrowTechnical Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attachedmaterial is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom theyare addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users
RE: [Asterisk-Users] features.conf *1 Call Recording
It's quite strange. When I press *1 I do not hear a tone indicated that it's even trying to record. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] features.conf *1 Call Recording Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call RecordingOK. You lost me.David MorrowTechnical Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attachedmaterial is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom theyare addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording2006
RE: [Asterisk-Users] features.conf *1 Call Recording
I have one Sipura SPA-841 which is configured to use dtmfmode=info and one Cisco 7905 which is using the default signalling (I believe this is rfc2833) I have also set relaxdtmf=yes in sip.conf I've tried pressing *1 on both phones (they are both on my desk) and both behave the same. ;; Sample Parking configuration; [general]parkext = 700 ; What ext. to dial to parkparkpos = 701-720 ; What extensions to park calls oncontext = parkedcalls ; Which context parked calls are in;parkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds);transferdigittimeout = 3 ; Number of seconds to wait between digits when transfering a call;courtesytone = beep ; Sound file to play to the parked caller ; when someone dials a parked call;xfersound = beep ; to indicate an attended transfer is complete;xferfailsound = beeperr ; to indicate a failed transfer;adsipark = yes ; if you want ADSI parking announcements;findslot = next ; Continue to the 'next' parking space. Defaults to 'first' available;pickupexten = *8 ; Configure the pickup extension. Default is *8featuredigittimeout = 2000 ; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap]blindxfer = #1 ; Blind transferdisconnect = *0 ; Disconnectautomon = *1 ; One Touch Recordatxfer = *2 ; Attended transfer [applicationmap];testfeature = #9,callee,Playback,tt-monkeys ;Play tt-monkes to ;callee if #9 was pressed ~~~ David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 12:55 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording hi Dave i get the following log on *CLI -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4 -- Playing 'beep' (language 'en') -- User hit '*1' to record call. filename: wav|auto-1147452537-200-204|m -- Playing 'beep' (language 'en') -- User hit '*1' to stop recording call. -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4what are you using as SIP client ( imean softphone/ analog phone + ATA / IPphone ) ? if you are using a softphone and that doesnot have a dtmf signaling then asterisk will not be able to recognize that you are pressing.--Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: It's quite strange. When I press *1 I do not hear a tone indicated that it's even trying to record. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dave MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] features.conf *1 Call Recording Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non
[Asterisk-Users] Dial Command Reference for SIP channel
Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between Dial(SIP/100|20|Ttr,,wW) and Dial(SIP/100,,wW) Specifically, what is the |20|Ttr ? I cannot seem to find any reference which would indicate this is even a valid format for the SIP channel. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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
RE: [Asterisk-Users] features.conf *1 Call Recording
I found the issue. It was my Dial command! In my dialplan I had Dial(SIP/100|20|Ttr,,wW) as this was something I gleaned from a sample config for call forwarding. I removed the |20|Ttr andnow the call recording works! Anyone know what the |20|Ttr did anyhow? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: Dave Morrow Sent: Friday, May 12, 2006 10:41 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] features.conf *1 Call Recording It's quite strange. When I press *1 I do not hear a tone indicated that it's even trying to record. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] features.conf *1 Call Recording Yes. I did. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote: Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto: [EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf :[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo
RE: [Asterisk-Users] features.conf *1 Call Recording
Thanks for the response. How would I change the DTMF transfer mode? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Fabio Sent: Thursday, May 11, 2006 6:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] features.conf *1 Call Recording if you ar using SIP clients, try changing DTMF transfer mode. For test use sip debug on your * console, then place a call and watch the output. In INFO or rfc2833 mode you must see the codes like SIP messages. If you are using inband transfer mode (DTMF codes are transferred like sounds) you don't see the codes. Also, try adjusting featuredigittimeout in features.conf: [general] featuredigittimeout = 2000 ; 2 seconds because the default 500ms is a very short time. Fabio Olaechea 3Tech SRL Calle 48 Nro 632, Of. 67. La Plata, CP B1900AMZ Buenos Aires, Argentina. Tel. +54 221 445 0244 Ext. 301 Fax. +54 221 445 0245 www.trestech.com.ar -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Dave Morrow Enviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] features.conf *1 Call Recording OK. You lost me. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] features.conf *1 Call Recording 2006/5/10, Dave Morrow [EMAIL PROTECTED]: I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? May be a problem with the way you are sending the dialtones. Try sending as data. -- Alejandro Vargas ___ --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 ___ --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 ___ --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 ___ --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
[Asterisk-Users] features.conf *1 Call Recording
Hi all. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=automon [default] exten=123,2,Dial(SIP/3000,,wW);wWallowone-touchrecording During the call, I press *1 but it records nothing. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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
RE: [Asterisk-Users] features.conf *1 Call Recording
OK. You lost me. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Vargas Sent: Wednesday, May 10, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] features.conf *1 Call Recording 2006/5/10, Dave Morrow [EMAIL PROTECTED]: I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? May be a problem with the way you are sending the dialtones. Try sending as data. -- Alejandro Vargas ___ --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 ___ --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
[Asterisk-Users] REPOST: features.conf *1 Call Recording
Hi all. I posted this earlier but nevergot any advice that helped. If anyone knows how to get this going, I'd appreciate some advice. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=automon [default] exten=123,2,Dial(SIP/3000,,wW);wWallowone-touchrecording During the call, I press *1 but it records nothing. My phones are all Sipura SPA841 SIP phones and I amrunning the latest* build. David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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
[Asterisk-Users] RE: Install/Upgrade
Hi all, I was just wondering ifanyone knows of any gotchas with respect to upgrading Asterisk to the latest 1.2.7 ? Is the procedure the same? Config files remain intact? Just untar/make install? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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
[Asterisk-Users] Most Popular FREE SoftPhone for Windows
Hi all. I am trying to find out what the most popular soft phone for Windows is for use with Asterisk. SIP or IAX? David Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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
[Asterisk-Users] Call Transfer
Title: Call Transfer Can anyone point me in the right direction. My users (all using Sipura SPA-841 phones) need the ability to transfer a call to another number. How can I setup a dial plan to do this? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodatasolutions.com NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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
RE: [Asterisk-Users] Linksys SPA-841 Missing Calls
Thanks all for the replies. I've narrowed it down to the phones dislike for my older 3COM switch. I noticed on the weekend that when these missed calls occur, if I ping the phone, the first few packets are dropped..almost like it's gone to sleep.. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Sent: Saturday, December 03, 2005 1:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Linksys SPA-841 Missing Calls I experienced a similar situation with the SPA-841, it turned out to be that the calls I was missing didn't have caller ID (outside calls with caller ID Blocked), found that the SPA841 phone has an option to ignore calls without caller ID. Turned this option off and it fixed the problem. Sorry, I no longer use the SPA841 and I can't remember the exact menu setting on the SPA841 that fixed it, so you will have to go through the manual. c Message: 1 Date: Fri, 02 Dec 2005 21:43:01 -0800 From: Wolfgang S. Rupprecht [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Linksys SPA-841 Missing Calls To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Might the SPA-841 be crashing and rebooting? With the current firmware (v. 3.1.4) I often see my phone hang and flash all its lights Really? For me the 841 is a quite stable phone. Out of the 15 we have in the office neither one crashed in the past 3 months. And they are used heavily. The phone has weaknesses, but stability in my opinion is not one of them. Phone info: Software Version: 3.1.4(a) Hardware Version: 1.0.0(1813) Elapsed Time: 50 days and 09:48:10 I only have 1 phone so it is hard to tell if the crashing is a hardware or software problem. I never noticed the phone having problems previous to this. I did resync asterisk to HEAD a month ago. Thats also about the time the phone started crashing (or at least I started noticing it). Come to think of it, I've been running the current firmware in the phone since July 20th. The only think that changed in recently was asterisk. I wonder if there is something the newer asterisk is doing that the phone really hates... Asterisk CVS HEAD built by [EMAIL PROTECTED] on a amd64 running OpenBSD on 2005-11-02 00:58:42 UTC Software Version: 3.1.4(a) Hardware Version: 1.0.0(700b) Elapsed Time: 1 day and 05:54:03 (crashed during a call) People have been reporting a finicky ethernet connector, so maybe that is the reason the phone does not answer to any traffic? Yea, this phone has that problem too. ;-) Some cables just don't work. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --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 ___ --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
[Asterisk-Users] Music on Hold Error
Title: Music on Hold Error Can anyone help with; Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Dec 2 12:11:56 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player Dec 2 12:20:16 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Dec 2 12:20:16 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player Dec 2 12:28:36 WARNING[2562]: res_musiconhold.c:421 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Dec 2 12:28:36 WARNING[2562]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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
[Asterisk-Users] Linksys SPA-841 Missing Calls
Title: Linksys SPA-841 Missing Calls Hi all, I have been plagued by an issue with my SPA-841 phones. The issue is that frequently, usually after a period of inactivity on the phone, an incoming call will be missed by the phone. The call works, cause the caller ends up at voicemail, but the phone never rings. I've managed to trap one of these missed calls in Asterisk, the log is below. Can anyone make sense of it? Retransmitting #3 (no NAT) to 172.16.140.114:5060: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.199.100:5060;branch=z9hG4bK4bf4b4ef;rport From: asterisk sip:[EMAIL PROTECTED];tag=as10f82be7 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 81 Messages-Waiting: yes Message-Account: sip:asterisk@ Voice-Message: 1/2 (0/0) --- Retransmitting #4 (no NAT) to 172.16.140.114:5060: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.199.100:5060;branch=z9hG4bK4bf4b4ef;rport From: asterisk sip:[EMAIL PROTECTED];tag=as10f82be7 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 81 Messages-Waiting: yes Message-Account: sip:asterisk@ Voice-Message: 1/2 (0/0) --- Retransmitting #5 (no NAT) to 172.16.140.114:5060: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.199.100:5060;branch=z9hG4bK4bf4b4ef;rport From: asterisk sip:[EMAIL PROTECTED];tag=as10f82be7 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 81 Messages-Waiting: yes Message-Account: sip:asterisk@ Voice-Message: 1/2 (0/0) [=== Please enter your reply above this line ===] David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --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
RE: [Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk
Title: Linksys SPA-841 Disconnects from Asterisk Thanks for the reply, however, I am already running the latest 3.14a It seems it may have something to do with the "Registration expires" setting on these phones. This value is set at the default 3600. After this interval, the phone de-registers and does not re-register with the Asterisk server. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex TerneroSent: Thursday, November 24, 2005 5:51 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk I don t have problems, after upgrade the firmware to the latest version. Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Thursday, November 24, 2005 3:49 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk Hi all, I wonder if anyone out there has experienced an issue I am having with my Sipura / Linksys SPA-841 phones. They work fine generally, but occasionally, incoming calls are missed. It's like the SIP registration is expiring. Does anyone know how to force the phone to re-register automatically? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Linksys SPA-841 Disconnects from Asterisk
Title: Linksys SPA-841 Disconnects from Asterisk Thanks for the reply, however, I am already running the latest 3.14a It seems it may have something to do with the "Registration expires" setting on these phones. This value is set at the default 3600. After this interval, the phone de-registers and does not re-register with the Asterisk server. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin LawetzSent: Thursday, November 24, 2005 3:59 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk Check in you console or your logs when this happens. I'm guessing it's a Stale Nonce If this is the case, Sipura supposedly fixed the bug on it's most recent firmware (At least for the SPA-1001 and SPA-2100, but I'm guessing the SPA-841 also) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: November 24, 2005 3:49 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk Hi all, I wonder if anyone out there has experienced an issue I am having with my Sipura / Linksys SPA-841 phones. They work fine generally, but occasionally, incoming calls are missed. It's like the SIP registration is expiring. Does anyone know how to force the phone to re-register automatically? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Linksys SPA-841 Disconnects from Asterisk
Title: Linksys SPA-841 Disconnects from Asterisk Hi all, I wonder if anyone out there has experienced an issue I am having with my Sipura / Linksys SPA-841 phones. They work fine generally, but occasionally, incoming calls are missed. It's like the SIP registration is expiring. Does anyone know how to force the phone to re-register automatically? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sipura SPA-841 Disconnects from Asterisk
Title: Sipura SPA-841 Disconnects from Asterisk Hi all, I am hoping to find someone who has run into this issue with the Sipura SPA-841 phone. Although my phones appear to be working fine, occasionally, they do not ring. When this happens, if I make a call on the phone, it seems to reconnect and start accepting incoming calls again. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sipura SPA-841 Second Line Help
Thanks, that's exactly what I have done...
But I am still trying to find out what the Shared line appearance is
on these phones?
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net
* PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE
*
NEW !!! Tel: (519) 963-3020
Fax: (519) 451-6615
Poor planning on your part does not necessarily constitute an
emergency on my part!
This message has originated from Autodata Solutions. The attached
material is the Confidential and Proprietary Information of Autodata
Solutions. This email and any files transmitted with it are confidential
and intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please delete
this message and notify the Autodata system administrator at
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wilson
Pickett
Sent: Sunday, November 20, 2005 6:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sipura SPA-841 Second Line Help
My requirement is to have an extension 9000 ring on all of the phones'
second line.
I've configured this extension in asterisk (extensions.conf and
sip.conf) as I would any other extension.
Inside the SPA-841 interface, I configured Ext 2 with the appropriate
SIP information and set the line appearance to Shared
You could also have a separate account for each phone's second line and
configure the extension 9000 to ring ${SPA1}${SPA2}${SPA3} etc (you
didn't say how many phones. Obviously for over 10 this may not be a good
idea) ___
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[Asterisk-Users] Sipura SPA-841 Second Line Help
Title: Sipura SPA-841 Second Line Help Hi all, I recently purchased Sipura SPA-841 phones for a group of users. While the phones are functioning great, I am having some troubles configuring one aspect. Hopefully someone will know what I am doing wrong. On each of the phones, I have configured Line 1 as a private line. That's working fine. My requirement is to have an extension 9000 ring on all of the phones' second line. I've configured this extension in asterisk (extensions.conf and sip.conf) as I would any other extension. Inside the SPA-841 interface, I configured Ext 2 with the appropriate SIP information and set the line appearance to Shared In the Phone tab, I set the line appearance to shared. With all this configured. It doesn't work. L2 on each of the phones simply flashes red and does not work. Any help would be greatly appreciated! David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sipura SPA-841 Second Line Help
Thanks Jerry. It's working now. One question comes to mind though. In order for it to work, I had to set the second line to private. When would one use shared then with these units? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Friday, November 18, 2005 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sipura SPA-841 Second Line Help You cannot have multiple devices registering with the same name in asterisk. Only the most recent to register will actually receive the call. Create a new registration ie 9001, 9002, etc on each of the phones then have 9000 ring all of them. On Nov 18, 2005, at 7:32 AM, Dave Morrow wrote: Hi all, I recently purchased Sipura SPA-841 phones for a group of users. While the phones are functioning great, I am having some troubles configuring one aspect. Hopefully someone will know what I am doing wrong. On each of the phones, I have configured Line 1 as a private line. That's working fine. My requirement is to have an extension 9000 ring on all of the phones' second line. I've configured this extension in asterisk (extensions.conf and sip.conf) as I would any other extension. Inside the SPA-841 interface, I configured Ext 2 with the appropriate SIP information and set the line appearance to Shared In the Phone tab, I set the line appearance to shared. With all this configured. It doesn't work. L2 on each of the phones simply flashes red and does not work. Any help would be greatly appreciated! David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net * PLEASE NOTE THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Extension Ring on Multiple Phones
Nonetheless .. Thanks everyone for the responses! I think I have it now! You guys are great! David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Tuesday, November 08, 2005 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Extension Ring on Multiple Phones I guess I should have read up further before I posted a response. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Tuesday, November 08, 2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones Like instead of exten = s,1,Dial(SIP/110,20,tr) you must mean exten = s,1,Dial(SIP/110SIP/112,20,tr) ? Just append all extensions you wish to ring, separated by ampersands (). The first one to answer will be winner. That's what I think you're asking, at least. Moj Dave Morrow wrote: Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] _http://www.autodata.net_ Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at_ [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]_ ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] CVS HEAD - app_muxmon
Title: CVS HEAD - app_muxmon I just upgraded to the latest CVS HEAD and found that the install reported app_muxmon.so as being incompatible for this version of Asterisk. Had to remove it from /var/lib/asterisk/modules in order to get asterisk started. Just an FYI David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Extension Ring on Multiple Phones
Title: Extension Ring on Multiple Phones Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Options for 3-way or Conference Calling
Title: Options for 3-way or Conference Calling Hi all, I wonder if someone could lend a little insight into the best way to configure either 3-way calling or conference calling. My goal is to keep this as simple for my users as it was with our legacy PBX. On our old phone system, a user could simply, during a call, press a Conference button on their phone to bring in a third party to a call. Can this be accomplished with Asterisk? My phones are all SIP devices (Cisco and Sipura). David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] How to do Call Forwarding
Title: How to do Call Forwarding Hi all. I am attempting to setup a dial plan which will allow me to forward an extension using the handset. I have followed the instructions in http://www.voip-info.org/wiki/index.php?page=Asterisk%20call%20forwarding however it does not work correctly. Does anyone have some expertise they could lend. Not sure if it matters, but when I setup as in these instructions, and attempt to call forward my phone, asterisk logs when in fact I am attempting to forward to extension 8001 ; == Spawn extension (default, *21*800, 4) exited non-zero on 'SIP/8001-9be7' David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] How to do Call Forwarding
I was hoping there would be something considerably more simple. For example, on my legacy PBX, all I need do is press the Call Fwd button on my phone, followed by an extension. Something similar (like *72#ext) would be nice. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley Sent: Wednesday, October 26, 2005 1:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to do Call Forwarding i have implemented something using mysql.. and thus i have a phone-features page that allows me to login / authenticate using the voicemail-users table for the pin and extension.. and then set the destination number, and then turn it on or off. then in the dialplan, mysql kicks in and checks to see if there is an enabled destination number and re-routes the incoming call WITH their caller id to that new destination.. works great... contact me off line if you would like some information. ./Ben Hi all. I am attempting to setup a dial plan which will allow me to forward an extension using the handset. I have followed the instructions in http://www.voip-info.org/wiki/index.php?page=Asterisk%20call%20forward ing however it does not work correctly. Does anyone have some expertise they could lend. Not sure if it matters, but when I setup as in these instructions, and attempt to call forward my phone, asterisk logs when in fact I am attempting to forward to extension 8001 ; == Spawn extension (default, *21*800, 4) exited non-zero on 'SIP/8001-9be7' David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Help with Dial Plan
Thanks Steve, the 'w's worked great. I managed to tune it down to them
only hearing a please wait out of the greeting..
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615
Poor planning on your part does not necessarily constitute an
emergency on my part!
This message has originated from Autodata Solutions. The attached
material is the Confidential and Proprietary Information of Autodata
Solutions. This email and any files transmitted with it are confidential
and intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please delete
this message and notify the Autodata system administrator at
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, October 19, 2005 4:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with Dial Plan
On Wed, 19 Oct 2005, Dave Morrow wrote:
Thanks Steve. It almost works, but never dials the extension. Also,
is there a way I could mute the line while the remote attendant comes
on?
Oops sorry - the dangers of posting without testing.
The ,s are wrong - they should be w. Each w is 1/2 second of waiting.
So that makes it:
exten = _6XXX,1,Dial(Zap/gX/1234567890,60,D(${EXTEN}))
As for the muting - bit of a loss about that one.
Steve
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[Asterisk-Users] Call Forwarding
Title: Call Forwarding Hi all. I am attempting to setup a dial plan which will allow me to forward an extension. I have followed the instructions in http://www.voip-info.org/wiki/index.php?page=Asterisk%20call%20forwarding however it does not work correctly. Does anyone have some expertise they could lend. Not sure if it matters, but when I setup as in these instructions, and attempt to call forward my phone, asterisk logs when in fact I am attempting to forward to extension 8001 ; == Spawn extension (default, *21*800, 4) exited non-zero on 'SIP/8001-9be7' David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Help with Dial Plan
Title: Help with Dial Plan Hi all. So far this list is proving it's worth, even on my first day using it! I hope that someone might know an easy solution to this one. I would like to create a dial plan which will allow me to have all extensions 6XXX cause a dial-out of my T1 interface (TE110P) to a local number, wait for an answer, wait 2 seconds and then enter the extension. Can I do this in a dial plan somehow? This will allow me to pseudo-integrate a legacy telephone switch (whose extensions are all 6XXX) to my Asterisk system for direct extension dialing. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Help with Dial Plan
Thanks Steve. It almost works, but never dials the extension. Also, is
there a way I could mute the line while the remote attendant comes on?
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615
Poor planning on your part does not necessarily constitute an
emergency on my part!
This message has originated from Autodata Solutions. The attached
material is the Confidential and Proprietary Information of Autodata
Solutions. This email and any files transmitted with it are confidential
and intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please delete
this message and notify the Autodata system administrator at
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, October 19, 2005 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Help with Dial Plan
On Wed, 19 Oct 2005, Dave Morrow wrote:
Hi all. So far this list is proving it's worth, even on my first day
using it! I hope that someone might know an easy solution to this
one.
I would like to create a dial plan which will allow me to have all
extensions 6XXX cause a dial-out of my T1 interface (TE110P) to a
local number, wait for an answer, wait 2 seconds and then enter the
extension.
Can I do this in a dial plan somehow? This will allow me to
pseudo-integrate a legacy telephone switch (whose extensions are all
6XXX) to my Asterisk system for direct extension dialing.
exten = _6XXX,1,Dial(Zap/gX/1234567890,60,D(,,${EXTEN}))
where:
gX needs to become the group of the channels of your T1, 1234567890 is
the number of your legacy system.
60 is the dial timeout
You may need to adjust the number of commas to get the right delay.
Hope that helps,
Steve
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[Asterisk-Users] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to the [default-incoming] context. It is the [default-incoming] context that I am unsure of how to set. If anyone could provide some insight, it would be much appreciated.get me started at least. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan I do not use any DID, all calls come in on the same number 111222 so what I would like to do is simply prompt the caller to enter the extension they wish to reach, then redirect to that extension in the [default] context. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: Wednesday, October 19, 2005 11:34 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan This is how I do it. [default-incoming]exten = 2691,1,Goto(extensions,3212,1)exten = 2692,1,Goto(extensions,3204,1)exten = 2693,1,Goto(extensions,3207,1)exten = 2694,1,Goto(extensions,3212,1)exten = 2695,1,Goto(extensions,3205,1)exten = 2696,1,Goto(extensions,3208,1)exten = 2697,1,Goto(extensions,1105,1)exten = 3211,1,Goto(extensions,,1)exten = 3223,1,Goto(extensions,3207,1) You will have to know how many digits are being sent, in my case it is four. So for example, someone dials the DID xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to the context "extensions" (you would replace with default) and the extension "3212" in the first priority. If more or less digits are being sent by the telco, you will have to adjust the exten = to match. Sometimes they send three. Thanks, Steve Totaro - Original Message ----- From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:26 AM Subject: [Asterisk-Users] Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to the [default-incoming] context. It is the [default-incoming] context that I am unsure of how to set. If anyone could provide some insight, it would be much appreciated.get me started at least. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.344 / Virus Database: 267.12.1/136 - Release Date: 10/15/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Newbie Question: Help with incoming dial plan
Title: Newbie Question: Help with incoming dial plan Thanks Steve, this works like a charm! Might I ask how I setup that Directory? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: Wednesday, October 19, 2005 11:51 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan add this context [default-incoming]exten = 111222,1,Goto(default-incoming,s,1) exten = s,1,Answerexten = s,2,DigitTimeout(10)exten = s,3,ResponseTimeout(20)exten = s,4,Background(swelcome)exten = t,1,Hangupinclude = extensions add this to your extensions context ;directory appexten = 9,1,Directory(default-extensions) ; exten for recording greetings/menusexten = 12,1,Authenticate(1234|)exten = 12,2,Wait(2)exten = 12,3,Record(/var/lib/asterisk/sounds/welcome:gsm)exten = 12,4,Wait(2)exten = 12,5,Playback(/var/lib/asterisk/sounds/welcome)exten = 12,6,Wait(2)exten = 12,7,Hangup Reload and dial 12 with the password of 1234 and record your greeting and then hangup. If you mess up just do it over. Thanks, Steve - Original Message - From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:41 AM Subject: RE: [Asterisk-Users] Newbie Question: Help with incoming dial plan I do not use any DID, all calls come in on the same number 111222 so what I would like to do is simply prompt the caller to enter the extension they wish to reach, then redirect to that extension in the [default] context. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asteriskSent: Wednesday, October 19, 2005 11:34 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Newbie Question: Help with incoming dial plan This is how I do it. [default-incoming]exten = 2691,1,Goto(extensions,3212,1)exten = 2692,1,Goto(extensions,3204,1)exten = 2693,1,Goto(extensions,3207,1)exten = 2694,1,Goto(extensions,3212,1)exten = 2695,1,Goto(extensions,3205,1)exten = 2696,1,Goto(extensions,3208,1)exten = 2697,1,Goto(extensions,1105,1)exten = 3211,1,Goto(extensions,,1)exten = 3223,1,Goto(extensions,3207,1) You will have to know how many digits are being sent, in my case it is four. So for example, someone dials the DID xxx-xxx-2691 the dialplan matches on the first line and then sends the cal to the context "extensions" (you would replace with default) and the extension "3212" in the first priority. If more or less digits are being sent by the telco, you will have to adjust the exten = to match. Sometimes they send three. Thanks, Steve Totaro - Original Message ----- From: Dave Morrow To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, October 18, 2005 11:26 AM Subject: [Asterisk-Users] Newbie Question: Help with incoming dial plan Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning. Since all inbound calls come through my T1, I would like to setup a dial plan that handles the incoming call and tells the caller to enter the extension they wish to reach. All of my real extensions are in the [default] context, and the Zaptel is configured to go to t
[Asterisk-Users] Forwarding Extensions using dialplan
Title: Forwarding Extensions using dialplan Hi all. So far this list is proving it's worth, even on my first day using it! I hope that someone might know an easy solution to this one. I would like to create a dial plan which will allow me to have all extensions 6XXX cause a dial-out of my T1 interface to a local number, wait for an answer, wait 2 seconds and then enter the extension. Can I do this in a dial plan somehow? This will allow me to pseudo-integrate a legacy telephone switch (whose extensions are all 6XXX) to my Asterisk system for direct extension dialing. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Soft Phone
Title: Soft Phone Can anyone recommend a good soft phone that's easy to configure under Asterisk and works well on a typical Windows XP system? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[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
[Asterisk-Users] AMP or Asterisk
Title: AMP or Asterisk Hi all. I have been using Asterisk for sometime now and have recently come across AMP for the first time. I am wondering if someone could enlighten me a little as to the advantages and disadvantages to using AMP as opposed to the do-it-yourself Asterisk? Is this documented someplace? Any advise would be greatly appreciated. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Freedom is just another word for nothing left to lose- Janis Joplin This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[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
[Asterisk-Users] Minimum Setup
Title: Minimum Setup Hi all, I have asterisk installed and working just fine with a couple of Cisco IP Phones. I am now ready to pilot connectivity to PSTN and am wondering what hardware would be recommended to make minimum connectivity to the public telephone network. I am think ISDN as I would like a few external lines to be accessible. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily consitute an emergency on my part. This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[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] AMP Anyone?
Thanks for the information Dennis, it is much appreciated. I think I am going to start from scratch (with AMP) also. It's just a bit of a pain is all. Do you have any expertise in regards to keeping current with * when new versions come out? How does this impact AMP? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily consitute an emergency on my part. This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dennis Boylan Sent: Wednesday, January 12, 2005 9:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AMP Anyone? On Tue, Jan 11, 2005 at 09:56:39PM -0500, Dave Morrow wrote: Hi all, I have been using Asterisk for a while now, and loving it. Just about to update to 1.0 (running like 0.93) I was wondering if anyone has any expertise in the implementation of AMP onto an existing Asterisk install? The instructions for it all deal with a fresh install of Asterisk, and I would hate to be forced to re-configure. Any advise would be greatly appreciated. A couple of notes here. I'd setup my first * server manually. I found some issues which required a faster box, so I rebuilt my * server and installed AMP. The rebuild did not copy any of the existing * configuration, but using the web interface, I duplicated most of my existing configuration after correcting my mistakes from the first install. The idea of migrating an existing * install to AMP is ok, but has many hurdles that have to be crossed. The biggest thing is that a lot of the configuration is now in MySQL. So, you would need a script to extract portions of the existing * configuration and inserting it into the database or duplicate your configuration using the web interface. - Dennis David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily consitute an emergency on my part. ___ 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 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] AMP Anyone?
Cool thanks. I got the system running today, and you were right, it was pretty easy. There are several things I would like to change. What files can be changed manually without AMP clobbering them? David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily consitute an emergency on my part. This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dennis Boylan Sent: Wednesday, January 12, 2005 6:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AMP Anyone? On Wed, Jan 12, 2005 at 11:59:00AM -0500, Dave Morrow wrote: Thanks for the information Dennis, it is much appreciated. I think I am going to start from scratch (with AMP) also. It's just a bit of a pain is all. Do you have any expertise in regards to keeping current with * when new versions come out? How does this impact AMP? I've updated both AMP and asterisk multiple times. Other than the overwriting of my hooks (includes get killed), it went seemlessly. I have the issue where I want to have some extensions which don't have voicemail. So, I've added them via include files. - Dennis David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 ___ 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 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] AMP Anyone?
Title: AMP Anyone? Hi all, I have been using Asterisk for a while now, and loving it. Just about to update to 1.0 (running like 0.93) I was wondering if anyone has any expertise in the implementation of AMP onto an existing Asterisk install? The instructions for it all deal with a fresh install of Asterisk, and I would hate to be forced to re-configure. Any advise would be greatly appreciated. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily consitute an emergency on my part. This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[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
