Re: [Asterisk-Users] 7960 'multi-line' configuration
Henry, The *70 worked like a champ! Thanks a ton for your help!!! For a few hundred bucks I have something in my home office that exceeds the Avaya systems I've used at client sites! Pat Quoting Henry Devito [EMAIL PROTECTED]: Sorry for the late reply, I was out of town today. Turn call waiting on for your extension in [EMAIL PROTECTED] by dialing *70 I think. Look at the extensions.conf file for the code. I ran into the same problem when I was using [EMAIL PROTECTED] a while back. I've since migrated away from [EMAIL PROTECTED], but the codes are there in the extensions.conf file. - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 04, 2005 1:02 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I found where it's getting this flag in the agi script, but I'm not sure how to fix this, short of commenting out lines in the .agi file... Is [EMAIL PROTECTED] doing some funny with call waiting? Are the agi scripts something I should be messing around with or should I be looking elsewhere? Thanks! Pat Quoting Henry Devito [EMAIL PROTECTED]: It has something to do with the AGI script. Scroll down! - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 9:22 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I still can't get the multi-line magic to happen. When I get the second call, this is what appears on the CLI. Any ideas? Thanks! Pat dialparties.agi: Caller ID is not set -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 200 has call waiting disabled dialparties.agi: Max calls of 1 exceeded - deleting from dial ===this is the problem! dialparties.agi: Dial string is empty - nothing to do dialparties.agi: Was direct call, jumping to priority 23 -- AGI Script Executing Application: (NoOp) Options: () -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait(IAX2/[EMAIL PROTECTED]/23, 1) in new stack -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/23, [EMAIL PROTECTED]) in new stack -- Playing 'voicemail/default/200/unavail' (language 'en') == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' in macro 'exten-vm' == Spawn extension (ext-local, 200, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' ___ 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 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 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
Why switch from Asterisk@Home? was: Re: [Asterisk-Users] 7960 'multi-line' configuration
Thanks for the tip. I'll try it this evening when I get back to my office. Out of curiosity, what made you switch from [EMAIL PROTECTED] Pat Quoting Henry Devito [EMAIL PROTECTED]: Sorry for the late reply, I was out of town today. Turn call waiting on for your extension in [EMAIL PROTECTED] by dialing *70 I think. Look at the extensions.conf file for the code. I ran into the same problem when I was using [EMAIL PROTECTED] a while back. I've since migrated away from [EMAIL PROTECTED], but the codes are there in the extensions.conf file. - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 04, 2005 1:02 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I found where it's getting this flag in the agi script, but I'm not sure how to fix this, short of commenting out lines in the .agi file... Is [EMAIL PROTECTED] doing some funny with call waiting? Are the agi scripts something I should be messing around with or should I be looking elsewhere? Thanks! Pat Quoting Henry Devito [EMAIL PROTECTED]: It has something to do with the AGI script. Scroll down! - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 9:22 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I still can't get the multi-line magic to happen. When I get the second call, this is what appears on the CLI. Any ideas? Thanks! Pat dialparties.agi: Caller ID is not set -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 200 has call waiting disabled dialparties.agi: Max calls of 1 exceeded - deleting from dial ===this is the problem! dialparties.agi: Dial string is empty - nothing to do dialparties.agi: Was direct call, jumping to priority 23 -- AGI Script Executing Application: (NoOp) Options: () -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait(IAX2/[EMAIL PROTECTED]/23, 1) in new stack -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/23, [EMAIL PROTECTED]) in new stack -- Playing 'voicemail/default/200/unavail' (language 'en') == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' in macro 'exten-vm' == Spawn extension (ext-local, 200, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' ___ 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 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 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: Why switch from Asterisk@Home? was: Re: [Asterisk-Users] 7960'multi-line' configuration
Henry Devito wrote: --- I've since migrated away from [EMAIL PROTECTED], but the codes are there in the extensions.conf file. I am also cusrious to know as to what you have migrated to, from [EMAIL PROTECTED] Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick M. Gray, Jr. Sent: Thursday, May 05, 2005 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Why switch from [EMAIL PROTECTED] was: Re: [Asterisk-Users] 7960'multi-line' configuration Thanks for the tip. I'll try it this evening when I get back to my office. Out of curiosity, what made you switch from [EMAIL PROTECTED] Pat Quoting Henry Devito [EMAIL PROTECTED]: Sorry for the late reply, I was out of town today. Turn call waiting on for your extension in [EMAIL PROTECTED] by dialing *70 I think. Look at the extensions.conf file for the code. I ran into the same problem when I was using [EMAIL PROTECTED] a while back. I've since migrated away from [EMAIL PROTECTED], but the codes are there in the extensions.conf file. NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] 7960 'multi-line' configuration
you can read the handbook. it lists all of the feature codes. http://asteriskathome.sourceforge.net/handbook/index.html --- Henry Devito [EMAIL PROTECTED] wrote: Sorry for the late reply, I was out of town today. Turn call waiting on for your extension in [EMAIL PROTECTED] by dialing *70 I think. Look at the extensions.conf file for the code. I ran into the same problem when I was using [EMAIL PROTECTED] a while back. I've since migrated away from [EMAIL PROTECTED], but the codes are there in the extensions.conf file. - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 04, 2005 1:02 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I found where it's getting this flag in the agi script, but I'm not sure how to fix this, short of commenting out lines in the .agi file... Is [EMAIL PROTECTED] doing some funny with call waiting? Are the agi scripts something I should be messing around with or should I be looking elsewhere? Thanks! Pat Quoting Henry Devito [EMAIL PROTECTED]: It has something to do with the AGI script. Scroll down! - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 9:22 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I still can't get the multi-line magic to happen. When I get the second call, this is what appears on the CLI. Any ideas? Thanks! Pat dialparties.agi: Caller ID is not set -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 200 has call waiting disabled dialparties.agi: Max calls of 1 exceeded - deleting from dial ===this is the problem! dialparties.agi: Dial string is empty - nothing to do dialparties.agi: Was direct call, jumping to priority 23 -- AGI Script Executing Application: (NoOp) Options: () -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait(IAX2/[EMAIL PROTECTED]/23, 1) in new stack -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/23, [EMAIL PROTECTED]) in new stack -- Playing 'voicemail/default/200/unavail' (language 'en') == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' in macro 'exten-vm' == Spawn extension (ext-local, 200, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' ___ 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 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 __ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail ___ 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: Why switch from Asterisk@Home? was: Re: [Asterisk-Users]7960'multi-line' configuration
There was no problem with [EMAIL PROTECTED] it was the customers' decision and not a very educated one if I may add. He was using this in a business and did not want to hear anything about software with the name @home. That and he wanted the OS platform to be RHE3. Well what can I say besides the customer is always right. - Original Message - From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 05, 2005 9:17 AM Subject: RE: Why switch from [EMAIL PROTECTED] was: Re: [Asterisk-Users]7960'multi-line' configuration Henry Devito wrote: --- I've since migrated away from [EMAIL PROTECTED], but the codes are there in the extensions.conf file. I am also cusrious to know as to what you have migrated to, from [EMAIL PROTECTED] Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick M. Gray, Jr. Sent: Thursday, May 05, 2005 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Why switch from [EMAIL PROTECTED] was: Re: [Asterisk-Users] 7960'multi-line' configuration Thanks for the tip. I'll try it this evening when I get back to my office. Out of curiosity, what made you switch from [EMAIL PROTECTED] Pat Quoting Henry Devito [EMAIL PROTECTED]: Sorry for the late reply, I was out of town today. Turn call waiting on for your extension in [EMAIL PROTECTED] by dialing *70 I think. Look at the extensions.conf file for the code. I ran into the same problem when I was using [EMAIL PROTECTED] a while back. I've since migrated away from [EMAIL PROTECTED], but the codes are there in the extensions.conf file. NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] 7960 'multi-line' configuration
Yes. Quoting Henry Devito [EMAIL PROTECTED]: Are you using asterisk @ home? - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 9:22 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I still can't get the multi-line magic to happen. When I get the second call, this is what appears on the CLI. Any ideas? Thanks! Pat dialparties.agi: Caller ID is not set -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 200 has call waiting disabled dialparties.agi: Max calls of 1 exceeded - deleting from dial dialparties.agi: Dial string is empty - nothing to do dialparties.agi: Was direct call, jumping to priority 23 -- AGI Script Executing Application: (NoOp) Options: () -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait(IAX2/[EMAIL PROTECTED]/23, 1) in new stack -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/23, [EMAIL PROTECTED]) in new stack -- Playing 'voicemail/default/200/unavail' (language 'en') == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' in macro 'exten-vm' == Spawn extension (ext-local, 200, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' ___ 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 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] 7960 'multi-line' configuration
Pat, To my knowledge the only way to turn on and off the Call Waiting function is on-screen with the phone itself. There are quite a few of these 'little' features I wish would be configurable via the config file but don't seem to be... Best wishes, -Corey Great info! The only question I would have is on the call waiting setting. What should it be set to, and is the setting the one in the SIPX.conf file? Pat -- Corey S. McFadden ([EMAIL PROTECTED]) McFadden Associates - Technology Consultants phone 215-825-2121 x510 - web.csma.biz * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ 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] 7960 'multi-line' configuration
Corey S. McFadden wrote: Pat, To my knowledge the only way to turn on and off the Call Waiting function is on-screen with the phone itself. There are quite a few of these 'little' features I wish would be configurable via the config file but don't seem to be... # Call Waiting (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control) call_waiting: 2 ; Default 1 (Enable Call Waiting) ... a bunch of options are only listed if you browse through all the info on cisco.com. Regardless, there is the option to add to SIPDefault.cnf to make the phones do what you want in regards to Call Waiting, etc... -Chris ___ 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] 7960 'multi-line' configuration
I found where it's getting this flag in the agi script, but I'm not sure how to fix this, short of commenting out lines in the .agi file... Is [EMAIL PROTECTED] doing some funny with call waiting? Are the agi scripts something I should be messing around with or should I be looking elsewhere? Thanks! Pat Quoting Henry Devito [EMAIL PROTECTED]: It has something to do with the AGI script. Scroll down! - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 9:22 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I still can't get the multi-line magic to happen. When I get the second call, this is what appears on the CLI. Any ideas? Thanks! Pat dialparties.agi: Caller ID is not set -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 200 has call waiting disabled dialparties.agi: Max calls of 1 exceeded - deleting from dial ===this is the problem! dialparties.agi: Dial string is empty - nothing to do dialparties.agi: Was direct call, jumping to priority 23 -- AGI Script Executing Application: (NoOp) Options: () -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait(IAX2/[EMAIL PROTECTED]/23, 1) in new stack -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/23, [EMAIL PROTECTED]) in new stack -- Playing 'voicemail/default/200/unavail' (language 'en') == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' in macro 'exten-vm' == Spawn extension (ext-local, 200, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' ___ 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 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] 7960 'multi-line' configuration
Sorry for the late reply, I was out of town today. Turn call waiting on for your extension in [EMAIL PROTECTED] by dialing *70 I think. Look at the extensions.conf file for the code. I ran into the same problem when I was using [EMAIL PROTECTED] a while back. I've since migrated away from [EMAIL PROTECTED], but the codes are there in the extensions.conf file. - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 04, 2005 1:02 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I found where it's getting this flag in the agi script, but I'm not sure how to fix this, short of commenting out lines in the .agi file... Is [EMAIL PROTECTED] doing some funny with call waiting? Are the agi scripts something I should be messing around with or should I be looking elsewhere? Thanks! Pat Quoting Henry Devito [EMAIL PROTECTED]: It has something to do with the AGI script. Scroll down! - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 9:22 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I still can't get the multi-line magic to happen. When I get the second call, this is what appears on the CLI. Any ideas? Thanks! Pat dialparties.agi: Caller ID is not set -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 200 has call waiting disabled dialparties.agi: Max calls of 1 exceeded - deleting from dial ===this is the problem! dialparties.agi: Dial string is empty - nothing to do dialparties.agi: Was direct call, jumping to priority 23 -- AGI Script Executing Application: (NoOp) Options: () -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait(IAX2/[EMAIL PROTECTED]/23, 1) in new stack -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/23, [EMAIL PROTECTED]) in new stack -- Playing 'voicemail/default/200/unavail' (language 'en') == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' in macro 'exten-vm' == Spawn extension (ext-local, 200, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' ___ 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 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] 7960 multi-line configuration
On Mon, 2005-05-02 at 16:54 -0400, Patrick M. Gray, Jr. wrote: No errors, asterisk just immediately sends the other call to voicemail if there is already a call in progress. Try turning on Call waiting on your cisco phone. Cheers, Joris ___ 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] 7960 multi-line configuration
I setup my 7960 with line 1 as main, and 2 as a queue line. So if the line is busy, asterisk queues the call and it will continue to ring on line 2. Call waiting works too, but not as well as queueing... Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joris Vandalon Sent: Tuesday, May 03, 2005 2:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 7960 multi-line configuration On Mon, 2005-05-02 at 16:54 -0400, Patrick M. Gray, Jr. wrote: No errors, asterisk just immediately sends the other call to voicemail if there is already a call in progress. Try turning on Call waiting on your cisco phone. Cheers, Joris ___ 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] 7960 multi-line configuration
I actually setup 6 registrations as separate lines. This allows me dialout selection, like line 5 for teliax, line 6 for voipjet, etc. I suspect you need different logons or all of your lines would ring at once. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris A. Icide Sent: Tuesday, May 03, 2005 1:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 7960 multi-line configuration -BEGIN PGP SIGNED MESSAGE- On 09:50 PM 5/2/2005, Matthew Boehm wrote: Hold up. So you have Phone #1. And all 6 lines register with the username of phone1 ? And you have phone #2; and all 6 lines register with username of phone2? And the phone only registers once? Interesting..I'm gonna test this. Sounds like it'd be a solution to 1 of my many problems. Yes, I have a 7960, and lines 1 through 6 are set to the same auth name and auth password. They all point at a single entry in the sip.conf table. The 7960 however only sends one register to the server. It just now has six presentations of that single entry (and actually can support 12 calls to that device if you allow call waiting) - -Chris -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQCVAwUBQncHJ+0LTNca2q41AQHf/QP7BB5ni6GOzc7JxvavF+ryg172gBtlIWku hmw5JkcinUBGKcRQ9paMXcZ+NRMokUFyljF+Yc1xLWPp4Gt1u/PCYmnU2tO/RIYg JAffPN5fVcA5zq5+uw/n0utwpUpo0VTzKPErcHonLJrr+ZF7MYxIiQ3NxHpQjAeR zOZw81xfyvU= =vHoj -END PGP SIGNATURE- ___ 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] 7960 multi-line configuration
Chris A. Icide wrote: -BEGIN PGP SIGNED MESSAGE- On 09:50 PM 5/2/2005, Matthew Boehm wrote: Hold up. So you have Phone #1. And all 6 lines register with the username of phone1 ? And you have phone #2; and all 6 lines register with username of phone2? And the phone only registers once? Interesting..I'm gonna test this. Sounds like it'd be a solution to 1 of my many problems. Yes, I have a 7960, and lines 1 through 6 are set to the same auth name and auth password. They all point at a single entry in the sip.conf table. The 7960 however only sends one register to the server. It just now has six presentations of that single entry (and actually can support 12 calls to that device if you allow call waiting) - -Chris All 6 lines can sign up with the same auth and secret and display. Works fine for us. -- respectfully, Joseph ___ 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] 7960 multi-line configuration
Matthew Boehm wrote: No, you can use the same username and secret for all 6 lines A 7940 or 7960 will just do the right thing That right thing being, roll over the new call to the second line if the first is busy, etc. There's no way. Asterisk doesn't support multiple logins from the same SIP username. Correct. Asterisk does not support multiple logins from the same SIP username. HOWEVER, if you configure the same SIP username/secret on all 6 lines of a Cisco (or Polycom) the PHONE will NOT register using the same username/secret more than once. It will see that more than one line is using the same username/secret and only register once. This is specific to the phone in question. ___ 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] 7960 multi-line configuration
My 7960 doesn't behave this way. With all usernames/display names/extensions the same, a second incoming call goes directly to voicemail. I'm on SIP firmware 7.1... could that be part of the problem or is it likely something in *? Thanks! Pat Quoting Henry Devito [EMAIL PROTECTED]: Usernames the same. The Cisco phone recognizes the usernames are the same and only registers once. - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Asterisk Users asterisk-users@lists.digium.com Sent: Monday, May 02, 2005 8:36 PM Subject: Re: [Asterisk-Users] 7960 multi-line configuration You can't use the same extension on multiple line buttons but you can Yes you can with the 7940's and 7960's. It works fine for a lot of people so calls will roll to the next button. I have this set up on several customer sites. You can use the same extension but the usernames must be different. Right? -Matthew ___ 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 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] 7960 multi-line configuration
On Tue, 2005-05-03 at 08:25 -0400, Patrick M. Gray, Jr. wrote: My 7960 doesn't behave this way. With all usernames/display names/extensions the same, a second incoming call goes directly to voicemail. I'm on SIP firmware 7.1... could that be part of the problem or is it likely something in *? 7.1 should not matter. Is call waiting turned on? Make sure that SIPDefault.cnf and your SIPMAC.cnf don't have some setting making it behave that way. What do you see if you telnet to the phone, and do a show config? Are the lines all set the same? You are not doing something with SetGroup/CheckGroup are you, in your dialplan? -- respectfully, Joseph === -= ** = ___ 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] 7960 multi-line configuration
Joseph, Thanks for your comments. I did have the phones in a group from some testing I was doing, which I have since removed. The phone reports call_waiting : 1 when I telnet in and check the config. Is this the correct setting or should it be set to 0? With the group config removed it still looks like I'm getting sent to voicemail rather than having secondary calls roll over. I'm working remotely and not near the phone, but if I call into my DID on two phones, one rings and the other goes immediately to voicemail. Again, thanks for your help! Pat Quoting Joseph [EMAIL PROTECTED]: On Tue, 2005-05-03 at 08:25 -0400, Patrick M. Gray, Jr. wrote: My 7960 doesn't behave this way. With all usernames/display names/extensions the same, a second incoming call goes directly to voicemail. I'm on SIP firmware 7.1... could that be part of the problem or is it likely something in *? 7.1 should not matter. Is call waiting turned on? Make sure that SIPDefault.cnf and your SIPMAC.cnf don't have some setting making it behave that way. What do you see if you telnet to the phone, and do a show config? Are the lines all set the same? You are not doing something with SetGroup/CheckGroup are you, in your dialplan? -- respectfully, Joseph === -= ** = ___ 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] 7960 multi-line configuration
Eric Wieling aka ManxPower wrote: Correct. Asterisk does not support multiple logins from the same SIP username. HOWEVER, if you configure the same SIP username/secret on all 6 lines of a Cisco (or Polycom) the PHONE will NOT register using the same username/secret more than once. It will see that more than one line is using the same username/secret and only register once. This is specific to the phone in question. Polycoms don't seem to handle this very well. (IP 500, IP 600). First, you can't disable call waiting, so each line has two call appearances. If I configure line 1 and line 2 with the same SIP username: I can dial out from call appearance 1 and call appearance 2 on line 1, And at the same time: I can dial out from call appearance 1 and call appearance 2 on line 2. That's all well and good. But when both call appearances on line 1 are busy the phone rejects new incoming calls - it doesn't roll down to line 2. Which means this isn't a useful way to use a Polycom as a receptionist phone. -- No virus found in this outgoing message. http://www.avg-antivirus.net/ Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.2 - Release Date: 5/2/2005 ___ 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] 7960 multi-line configuration
Joseph wrote: Is call waiting turned on? What do you see if you telnet to the phone, and do a show config? Are the lines all set the same? Sorry to barge in like this, here it's working like you're telling. If I do a show config, I see call_waiting: 1 and all the line's are set to the same SIP login. Chgeers, Kristof. ___ 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] 7960 'multi-line' configuration
Guys, I added some content to the Wiki on this feature. I don't think it's well documented anywhere. Please expand upon what I put in there if you have more details. http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx -Corey -- Corey S. McFadden ([EMAIL PROTECTED]) McFadden Associates - Technology Consultants phone 215-825-2121 x510 - web.csma.biz * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ 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] 7960 'multi-line' configuration
Great info! The only question I would have is on the call waiting setting. What should it be set to, and is the setting the one in the SIPX.conf file? Pat Quoting Corey S. McFadden [EMAIL PROTECTED]: Guys, I added some content to the Wiki on this feature. I don't think it's well documented anywhere. Please expand upon what I put in there if you have more details. http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx -Corey -- Corey S. McFadden ([EMAIL PROTECTED]) McFadden Associates - Technology Consultants phone 215-825-2121 x510 - web.csma.biz * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ 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] 7960 'multi-line' configuration
Corey S. McFadden wrote: I added some content to the Wiki on this feature. I don't think it's well documented anywhere. Please expand upon what I put in there if you have more details. http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+cisco+79xx Looks very nice! I have it working on a 7960 like described in the Call waiting section. I expect 2x 7940 phones and will try and use it on these also. If needed (but it looks complete to me) I'll add info to the wiki. Cheers. ___ 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] 7960 'multi-line' configuration
I still can't get the multi-line magic to happen. When I get the second call, this is what appears on the CLI. Any ideas? Thanks! Pat dialparties.agi: Caller ID is not set -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 200 has call waiting disabled dialparties.agi: Max calls of 1 exceeded - deleting from dial dialparties.agi: Dial string is empty - nothing to do dialparties.agi: Was direct call, jumping to priority 23 -- AGI Script Executing Application: (NoOp) Options: () -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait(IAX2/[EMAIL PROTECTED]/23, 1) in new stack -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/23, [EMAIL PROTECTED]) in new stack -- Playing 'voicemail/default/200/unavail' (language 'en') == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' in macro 'exten-vm' == Spawn extension (ext-local, 200, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' ___ 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] 7960 'multi-line' configuration
It has something to do with the AGI script. Scroll down! - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 9:22 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I still can't get the multi-line magic to happen. When I get the second call, this is what appears on the CLI. Any ideas? Thanks! Pat dialparties.agi: Caller ID is not set -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 200 has call waiting disabled dialparties.agi: Max calls of 1 exceeded - deleting from dial ===this is the problem! dialparties.agi: Dial string is empty - nothing to do dialparties.agi: Was direct call, jumping to priority 23 -- AGI Script Executing Application: (NoOp) Options: () -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait(IAX2/[EMAIL PROTECTED]/23, 1) in new stack -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/23, [EMAIL PROTECTED]) in new stack -- Playing 'voicemail/default/200/unavail' (language 'en') == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' in macro 'exten-vm' == Spawn extension (ext-local, 200, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' ___ 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] 7960 'multi-line' configuration
Are you using asterisk @ home? - Original Message - From: Patrick M. Gray, Jr. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 9:22 PM Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration I still can't get the multi-line magic to happen. When I get the second call, this is what appears on the CLI. Any ideas? Thanks! Pat dialparties.agi: Caller ID is not set -- dialparties.agi: Added extension 200 to extension map -- dialparties.agi: Extension 200 cf is disabled -- dialparties.agi: Extension 200 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 200 has call waiting disabled dialparties.agi: Max calls of 1 exceeded - deleting from dial dialparties.agi: Dial string is empty - nothing to do dialparties.agi: Was direct call, jumping to priority 23 -- AGI Script Executing Application: (NoOp) Options: () -- AGI Script dialparties.agi completed, returning 0 -- Executing Wait(IAX2/[EMAIL PROTECTED]/23, 1) in new stack -- Executing VoiceMail(IAX2/[EMAIL PROTECTED]/23, [EMAIL PROTECTED]) in new stack -- Playing 'voicemail/default/200/unavail' (language 'en') == Spawn extension (macro-exten-vm, s, 6) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' in macro 'exten-vm' == Spawn extension (ext-local, 200, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/23' ___ 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] 7960 multi-line configuration
I'm trying to configure a 7960 in a small-office setting to function like the large avaya-type PBX systems I have used. Basically each station has a DID number, and 4 or more lines on the phone. If you're on line 1 and someone calls, line 2 rings rather than a call waiting beep. Similarly you can conference two lines, etc. In google'ing around a bit, it seems I should be able to assign the same extension to several of the SIP lines on the 7960, and asterisk should automatically roll calls to a vacant line. When I configure the 7960 in this manner, any second call to a given DID just goes to voicemail. Is this the best practice way to configure the 7960, or should I assign 6 different lines to the 7960 and create a call group? Thanks! Pat ___ 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] 7960 multi-line configuration
Patrick M. Gray, Jr. wrote: In google'ing around a bit, it seems I should be able to assign the same extension to several of the SIP lines on the 7960, and asterisk should I don't think that is possible, at least not the way one thinks it would work. I have also done some reading on this, maybe this thread gives a solution: http://lists.digium.com/pipermail/asterisk-users/2004-March/039271.html But, I am also curious on how other people have solved this, especially with using AMP for example. Cheers.. ___ 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] 7960 multi-line configuration
On Mon, 2005-05-02 at 21:51 +0200, Kristof Hardy wrote: Patrick M. Gray, Jr. wrote: In google'ing around a bit, it seems I should be able to assign the same extension to several of the SIP lines on the 7960, and asterisk should Works fine for us. Do you get errors? -- respectfully, Joseph === -= ** = ___ 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] 7960 multi-line configuration
No errors, asterisk just immediately sends the other call to voicemail if there is already a call in progress. There's a busy message or something on the CLI (I'm not near the server right now so I can't get the exact text) and then the voicemail macro runs. Did you do anything funky when assigning the lines to the Cisco? I basically copied the config for a single extension to 4 of the 6 lines. When I do a sip show peers I only see one listing... do you get multiple listings for each extension (i.e. a separate registration for each instance of the line?) Thanks, Pat Quoting Joseph [EMAIL PROTECTED]: On Mon, 2005-05-02 at 21:51 +0200, Kristof Hardy wrote: Patrick M. Gray, Jr. wrote: In google'ing around a bit, it seems I should be able to assign the same extension to several of the SIP lines on the 7960, and asterisk should Works fine for us. Do you get errors? -- respectfully, Joseph === -= ** = ___ 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] 7960 multi-line configuration
You can't use the same extension on multiple line buttons but you can use different extensions on different line buttons. Just curious, why do you need the same line to appear on multiple line buttons. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Kristof Hardy wrote: Patrick M. Gray, Jr. wrote: In google'ing around a bit, it seems I should be able to assign the same extension to several of the SIP lines on the 7960, and asterisk should I don't think that is possible, at least not the way one thinks it would work. I have also done some reading on this, maybe this thread gives a solution: http://lists.digium.com/pipermail/asterisk-users/2004-March/039271.html But, I am also curious on how other people have solved this, especially with using AMP for example. Cheers.. ___ 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] 7960 multi-line configuration
Scott Henderson wrote: You can't use the same extension on multiple line buttons but you can use different extensions on different line buttons. Actually you can, and the 7960 does the 'right thing'.. surprised me too. Tony ___ 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] 7960 multi-line configuration
Joseph wrote: On Mon, 2005-05-02 at 21:51 +0200, Kristof Hardy wrote: Patrick M. Gray, Jr. wrote: In google'ing around a bit, it seems I should be able to assign the same extension to several of the SIP lines on the 7960, and asterisk should Works fine for us. Do you get errors? If you have 6 of these phones, then that means you have 36 SIP username/password combinations right? -Matthew ___ 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] 7960 multi-line configuration
Tony Hoyle wrote: Scott Henderson wrote: You can't use the same extension on multiple line buttons but you can use different extensions on different line buttons. Actually you can, and the 7960 does the 'right thing'.. surprised me too. Tony Please explain in more detail Tony. I've got tons of 7960's and we only use the first button because asterisk doesn't support multiple SIp registrations. -Matthew ___ 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] 7960 multi-line configuration
You can configure multiple button to use multiple extensions. You do this in the phones configure file, here is a quick example. In this example the phone is the receptionist for two companies and it also has an individual line appearance for the the receptionist DID line. Keep in mind that if you configure the voice mail box in the sip.conf fill then you will see a voice mail indicator for each line as well. You will need to create entries in the sip.conf file for each extension as well as dial plan entires in extensions.conf but then all should be well # SIP Configuration Generic File # Line 1 line1_name: CRV_Reception line1_authname: "crv_reception" line1_password: "crv_reception" # Line 2 line2_name: "KP_Reception" line2_authname: "kp_reception" line2_password: "kp_reception" # Line 3 line3_name: "Colleen" line3_authname: "colleen" line3_password: "colleen" # Line 4 line4_name: "Line 4" line4_authname: "UNPROVISIONED" line4_password: "UNPROVISIONED" # Line 5 line5_name: "Line 5" line5_authname: "UNPROVISIONED" line5_password: "UNPROVISIONED" # Line 6 line6_name: "Line 6 line6_authname: "UNPROVISIONED" line6_password: "UNPROVISIONED" Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Matthew Boehm wrote: Tony Hoyle wrote: Scott Henderson wrote: You can't use the same extension on multiple line buttons but you can use different extensions on different line buttons. Actually you can, and the 7960 does the 'right thing'.. surprised me too. Tony Please explain in more detail Tony. I've got tons of 7960's and we only use the first button because asterisk doesn't support multiple SIp registrations. -Matthew ___ 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] 7960 multi-line configuration
Well that proves my point; If you have 6 7960's and want 6 lines on each phone, you will need 36 username entries in your sip.conf. ouch.. -Matthew Scott Henderson wrote: You can configure multiple button to use multiple extensions. You do this in the phones configure file, here is a quick example. In this example the phone is the receptionist for two companies and it also has an individual line appearance for the the receptionist DID line. Keep in mind that if you configure the voice mail box in the sip.conf fill then you will see a voice mail indicator for each line as well. You will need to create entries in the sip.conf file for each extension as well as dial plan entires in extensions.conf but then all should be well # SIP Configuration Generic File # Line 1 line1_name: CRV_Reception line1_authname: crv_reception line1_password: crv_reception # Line 2 line2_name: KP_Reception line2_authname: kp_reception line2_password: kp_reception # Line 3 line3_name: Colleen line3_authname: colleen line3_password: colleen # Line 4 line4_name: Line 4 line4_authname: UNPROVISIONED line4_password: UNPROVISIONED # Line 5 line5_name: Line 5 line5_authname: UNPROVISIONED line5_password: UNPROVISIONED # Line 6 line6_name: Line 6 line6_authname: UNPROVISIONED line6_password: UNPROVISIONED Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK Matthew Boehm wrote: Tony Hoyle wrote: Scott Henderson wrote: You can't use the same extension on multiple line buttons but you can use different extensions on different line buttons. Actually you can, and the 7960 does the 'right thing'.. surprised me too. Tony Please explain in more detail Tony. I've got tons of 7960's and we only use the first button because asterisk doesn't support multiple SIp registrations. -Matthew ___ 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 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] 7960 multi-line configuration
Matthew Boehm wrote: Well that proves my point; If you have 6 7960's and want 6 lines on each phone, you will need 36 username entries in your sip.conf. ouch.. That's just the way it is. Some people have 1,000 or more entries in sip.conf. It's not like they change all that often. ___ 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] 7960 multi-line configuration
- Original Message - From: Scott Henderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 02, 2005 4:01 PM Subject: Re: [Asterisk-Users] 7960 multi-line configuration You can't use the same extension on multiple line buttons but you can Yes you can with the 7940's and 7960's. It works fine for a lot of people so calls will roll to the next button. I have this set up on several customer sites. use different extensions on different line buttons. ___ 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] 7960 multi-line configuration
It is only registering with asterisk once. ASterisk doesn't know that it is a second button on the phone, the sip software on the phone makes the call roll to the next free button. - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 02, 2005 5:08 PM Subject: Re: [Asterisk-Users] 7960 multi-line configuration Tony Hoyle wrote: Scott Henderson wrote: You can't use the same extension on multiple line buttons but you can use different extensions on different line buttons. Actually you can, and the 7960 does the 'right thing'.. surprised me too. Tony Please explain in more detail Tony. I've got tons of 7960's and we only use the first button because asterisk doesn't support multiple SIp registrations. -Matthew ___ 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] 7960 multi-line configuration
You can't use the same extension on multiple line buttons but you can Yes you can with the 7940's and 7960's. It works fine for a lot of people so calls will roll to the next button. I have this set up on several customer sites. You can use the same extension but the usernames must be different. Right? -Matthew ___ 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] 7960 multi-line configuration
-BEGIN PGP SIGNED MESSAGE- On 06:36 PM 5/2/2005, Matthew Boehm wrote: You can't use the same extension on multiple line buttons but you can Yes you can with the 7940's and 7960's. It works fine for a lot of people so calls will roll to the next button. I have this set up on several customer sites. You can use the same extension but the usernames must be different. Right? No, you can use the same username and secret for all 6 lines A 7940 or 7960 will just do the right thing That right thing being, roll over the new call to the second line if the first is busy, etc. - -Chris -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQCVAwUBQnbg1O0LTNca2q41AQEXPwP8DIZZo3W2vEZjVzmFUHa4J6uOHI/WB1JH FD8STuiXMb0g+KmABe89DDtR521q4Yqr4D230GxDKqNm4WzFmd5n39MAiIVTrgam sbrAiCGU6qw+15jer2KnuD5HAWLuJuHgcKG5tdrhogdLXBPQ8wI/Ng12lyTLLv6K BKY7RM3G9o8= =pd7W -END PGP SIGNATURE- ___ 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] 7960 multi-line configuration
Usernames the same. The Cisco phone recognizes the usernames are the same and only registers once. - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Asterisk Users asterisk-users@lists.digium.com Sent: Monday, May 02, 2005 8:36 PM Subject: Re: [Asterisk-Users] 7960 multi-line configuration You can't use the same extension on multiple line buttons but you can Yes you can with the 7940's and 7960's. It works fine for a lot of people so calls will roll to the next button. I have this set up on several customer sites. You can use the same extension but the usernames must be different. Right? -Matthew ___ 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] 7960 multi-line configuration
No, you can use the same username and secret for all 6 lines A 7940 or 7960 will just do the right thing That right thing being, roll over the new call to the second line if the first is busy, etc. There's no way. Asterisk doesn't support multiple logins from the same SIP username. You're telling me you have the same exact SIPmac.cnf file loaded into every one of your 7960's? You have 6 separate username/passwords and all 6 of those are registered on all of your Cisco 7960 phones? And that you can send a call to a specific extension on a specific phone? I'd have to see that to believe it because every developer on the -dev list will tell you that asterisk doesn't support simultaneous SIP logins. -Matthew ___ 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] 7960 multi-line configuration
Hold up. So you have Phone #1. And all 6 lines register with the username of phone1 ? And you have phone #2; and all 6 lines register with username of phone2? And the phone only registers once? Interesting..I'm gonna test this. Sounds like it'd be a solution to 1 of my many problems. -Matthew From: Henry Devito [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 2 May 2005 22:50:24 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 7960 multi-line configuration Usernames the same. The Cisco phone recognizes the usernames are the same and only registers once. - Original Message - From: Matthew Boehm [EMAIL PROTECTED] To: Asterisk Users asterisk-users@lists.digium.com Sent: Monday, May 02, 2005 8:36 PM Subject: Re: [Asterisk-Users] 7960 multi-line configuration You can't use the same extension on multiple line buttons but you can Yes you can with the 7940's and 7960's. It works fine for a lot of people so calls will roll to the next button. I have this set up on several customer sites. You can use the same extension but the usernames must be different. Right? -Matthew ___ 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 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] 7960 multi-line configuration
-BEGIN PGP SIGNED MESSAGE- On 09:50 PM 5/2/2005, Matthew Boehm wrote: Hold up. So you have Phone #1. And all 6 lines register with the username of phone1 ? And you have phone #2; and all 6 lines register with username of phone2? And the phone only registers once? Interesting..I'm gonna test this. Sounds like it'd be a solution to 1 of my many problems. Yes, I have a 7960, and lines 1 through 6 are set to the same auth name and auth password. They all point at a single entry in the sip.conf table. The 7960 however only sends one register to the server. It just now has six presentations of that single entry (and actually can support 12 calls to that device if you allow call waiting) - -Chris -BEGIN PGP SIGNATURE- Version: PGP 8.1 iQCVAwUBQncHJ+0LTNca2q41AQHf/QP7BB5ni6GOzc7JxvavF+ryg172gBtlIWku hmw5JkcinUBGKcRQ9paMXcZ+NRMokUFyljF+Yc1xLWPp4Gt1u/PCYmnU2tO/RIYg JAffPN5fVcA5zq5+uw/n0utwpUpo0VTzKPErcHonLJrr+ZF7MYxIiQ3NxHpQjAeR zOZw81xfyvU= =vHoj -END PGP SIGNATURE- ___ 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
