Re: [asterisk-users] [NEWBIE] Right dect to buy to use with asterisk
Hi, I made good experienes with Siemens Gigaset C610 IP. This model is about 90 Euro. Configuration via web interface. But encryption (SIPS/SRTP) is *not* possible with this phones. -Thorsten- Am 11.12.2013 11:30, schrieb Mario Giammarco: Hello, I need to setup this configuration: - asterisk as IVR; - dect phones. So basically I need a standard set of features: - each dect phone has its extension so I can call it directly; - handover of a call with R key; - if a call is not replied by someone ring all phones. I have little budget. I can choose to buy a fritz!box or a gigasect dect/ip base station. Which one should I buy? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [NEWBIE] Right dect to buy to use with asterisk
Hello Mario, nice to meet you on this mailing list! Gigaset phones are a very high quality/price ratio, so I'll suggest you to go with the dect ip models. Then you'll need to configure asterisk to act as IVR, configure a queue and a failover to ring all hunt list. Drop me a phone call and I'll be happy to help you Leandro 2013/12/11 Mario Giammarco mgiamma...@gmail.com Hello, I need to setup this configuration: - asterisk as IVR; - dect phones. So basically I need a standard set of features: - each dect phone has its extension so I can call it directly; - handover of a call with R key; - if a call is not replied by someone ring all phones. I have little budget. I can choose to buy a fritz!box or a gigasect dect/ip base station. Which one should I buy? Thanks, Mario -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie´s question about Asterisk...
Dean's link has references to Trixbox. TB has a bad, bad, very bad reputation for being very insecure. Alternatives to TB are FreePBX PBX in a Flash. All are Asterisk based and very easy to set up. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Collins Sent: Thursday, February 17, 2011 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]Newbie´s question about Asterisk... If you already have experience with linux asterisk will be easy for you. Other people will reply with official links but here is how I use Asterisk in my small home office www.cognation.net/asterisk Cheers, Dean From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francisco Javier Cintrón Olguín Sent: Thursday, February 17, 2011 7:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie´s question about Asterisk... Hi, My name is Francisco from México. Here, in my work we have a very very old panasonic PBX(12 years old). We are growing and we need to increase our external lines(from 3 to 4) and our internal lines(from 6 to 10). Besides we need voice mail and voice menu too. We asked for a quote to our panasonic dealer. The whole thing cost about 4,500 dollars. My boss just saw a thing called Asterisk this morning looking for options in Google. He asked my to investigate what this thing called Asterisk is and if we could save some money using it instead of the panasonic solution. So, here I am. I have some experience as linux sysadmin(we have 1 oracle linux server and 1 linux print server) nevertheless I don´t have any idea where and how to start this evaluation? Please Would you give us a clue where to see If Asterisk could work for us? Thanks for your kind help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie´s question about Asterisk...
i prefer to go with Elastix very easy to setup and maintain and reach UI rather than freePBX cheers Dhaval On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell te...@brummell.net wrote: Dean’s link has references to Trixbox. TB has a bad, bad, very bad reputation for being very insecure. Alternatives to TB are FreePBX PBX in a Flash. All are Asterisk based and very easy to set up. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dean Collins *Sent:* Thursday, February 17, 2011 7:29 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk... If you already have experience with linux asterisk will be easy for you. Other people will reply with official links but here is how I use Asterisk in my small home office www.cognation.net/asterisk Cheers, Dean -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Francisco Javier Cintrón Olguín *Sent:* Thursday, February 17, 2011 7:26 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Newbie´s question about Asterisk... Hi, My name is Francisco from México. Here, in my work we have a very very old panasonic PBX(12 years old). We are growing and we need to increase our external lines(from 3 to 4) and our internal lines(from 6 to 10). Besides we need voice mail and voice menu too. We asked for a quote to our panasonic dealer. The whole thing cost about 4,500 dollars. My boss just saw a thing called Asterisk this morning looking for options in Google. He asked my to investigate what this thing called Asterisk is and if we could save some money using it instead of the panasonic solution. So, here I am. I have some experience as linux sysadmin(we have 1 oracle linux server and 1 linux print server) nevertheless I don´t have any idea where and how to start this evaluation? Please Would you give us a clue where to see If Asterisk could work for us? Thanks for your kind help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie´s question about Asterisk...
Yes, I use Elastix myself too. Funny that I didn't mention that one! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Friday, February 18, 2011 6:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]Newbie´s question about Asterisk... i prefer to go with Elastix very easy to setup and maintain and reach UI rather than freePBX cheers Dhaval On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell te...@brummell.net wrote: Dean's link has references to Trixbox. TB has a bad, bad, very bad reputation for being very insecure. Alternatives to TB are FreePBX PBX in a Flash. All are Asterisk based and very easy to set up. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Collins Sent: Thursday, February 17, 2011 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]Newbie´s question about Asterisk... If you already have experience with linux asterisk will be easy for you. Other people will reply with official links but here is how I use Asterisk in my small home office www.cognation.net/asterisk Cheers, Dean From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francisco Javier Cintrón Olguín Sent: Thursday, February 17, 2011 7:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie´s question about Asterisk... Hi, My name is Francisco from México. Here, in my work we have a very very old panasonic PBX(12 years old). We are growing and we need to increase our external lines(from 3 to 4) and our internal lines(from 6 to 10). Besides we need voice mail and voice menu too. We asked for a quote to our panasonic dealer. The whole thing cost about 4,500 dollars. My boss just saw a thing called Asterisk this morning looking for options in Google. He asked my to investigate what this thing called Asterisk is and if we could save some money using it instead of the panasonic solution. So, here I am. I have some experience as linux sysadmin(we have 1 oracle linux server and 1 linux print server) nevertheless I don´t have any idea where and how to start this evaluation? Please Would you give us a clue where to see If Asterisk could work for us? Thanks for your kind help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie´s question about Asterisk...
Asterisk is open source and you can install in a normal PC itself and you can avail all the features that proprietary system has. If you want to integrate with any VoIP service then a PC with Asterisk is enough or else if you want to integrate with PSTN lines then you need FXO card to be installed, its a PCI card. Vendors like Sangoma, Digium (from Asterisk) were selling these cards. And for internal for your agents if you need analog hard phones then you need to have FXS card you can avail these FXO and FXS cards in combination. These cards will fit in your PCI slot of machine. Configuring these cards are also very easy. If it is VoIP then you dont need these cards simply install Asterisk in a PC and you are done. On Fri, Feb 18, 2011 at 5:38 PM, Terry Brummell te...@brummell.net wrote: Yes, I use Elastix myself too. Funny that I didn’t mention that one! *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA *Sent:* Friday, February 18, 2011 6:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk... i prefer to go with Elastix very easy to setup and maintain and reach UI rather than freePBX cheers Dhaval On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell te...@brummell.net wrote: Dean’s link has references to Trixbox. TB has a bad, bad, very bad reputation for being very insecure. Alternatives to TB are FreePBX PBX in a Flash. All are Asterisk based and very easy to set up. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dean Collins *Sent:* Thursday, February 17, 2011 7:29 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk... If you already have experience with linux asterisk will be easy for you. Other people will reply with official links but here is how I use Asterisk in my small home office www.cognation.net/asterisk Cheers, Dean -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Francisco Javier Cintrón Olguín *Sent:* Thursday, February 17, 2011 7:26 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Newbie´s question about Asterisk... Hi, My name is Francisco from México. Here, in my work we have a very very old panasonic PBX(12 years old). We are growing and we need to increase our external lines(from 3 to 4) and our internal lines(from 6 to 10). Besides we need voice mail and voice menu too. We asked for a quote to our panasonic dealer. The whole thing cost about 4,500 dollars. My boss just saw a thing called Asterisk this morning looking for options in Google. He asked my to investigate what this thing called Asterisk is and if we could save some money using it instead of the panasonic solution. So, here I am. I have some experience as linux sysadmin(we have 1 oracle linux server and 1 linux print server) nevertheless I don´t have any idea where and how to start this evaluation? Please Would you give us a clue where to see If Asterisk could work for us? Thanks for your kind help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie´s question about Asterisk...
I think I have 3 PSTN lines because I can connect a normal telephone to them all and make calls between each of them. We have 5 normal telephones and 1 panasonic. From what I got I need a PC and a of PCI card to interface to my 3 external lines and my 6 internal lines. For the PC I was planning to use the smallest PC posible like a HP Proliant Microserver but it doesn´t have space for this PCI card. Is there another way to interface to 3 external and 6 internal lines?? Thank you for your kind help On Fri, Feb 18, 2011 at 6:52 AM, Gopalakrishnan A.N sai...@gmail.comwrote: Asterisk is open source and you can install in a normal PC itself and you can avail all the features that proprietary system has. If you want to integrate with any VoIP service then a PC with Asterisk is enough or else if you want to integrate with PSTN lines then you need FXO card to be installed, its a PCI card. Vendors like Sangoma, Digium (from Asterisk) were selling these cards. And for internal for your agents if you need analog hard phones then you need to have FXS card you can avail these FXO and FXS cards in combination. These cards will fit in your PCI slot of machine. Configuring these cards are also very easy. If it is VoIP then you dont need these cards simply install Asterisk in a PC and you are done. On Fri, Feb 18, 2011 at 5:38 PM, Terry Brummell te...@brummell.netwrote: Yes, I use Elastix myself too. Funny that I didn’t mention that one! *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA *Sent:* Friday, February 18, 2011 6:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk... i prefer to go with Elastix very easy to setup and maintain and reach UI rather than freePBX cheers Dhaval On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell te...@brummell.net wrote: Dean’s link has references to Trixbox. TB has a bad, bad, very bad reputation for being very insecure. Alternatives to TB are FreePBX PBX in a Flash. All are Asterisk based and very easy to set up. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dean Collins *Sent:* Thursday, February 17, 2011 7:29 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk... If you already have experience with linux asterisk will be easy for you. Other people will reply with official links but here is how I use Asterisk in my small home office www.cognation.net/asterisk Cheers, Dean -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Francisco Javier Cintrón Olguín *Sent:* Thursday, February 17, 2011 7:26 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Newbie´s question about Asterisk... Hi, My name is Francisco from México. Here, in my work we have a very very old panasonic PBX(12 years old). We are growing and we need to increase our external lines(from 3 to 4) and our internal lines(from 6 to 10). Besides we need voice mail and voice menu too. We asked for a quote to our panasonic dealer. The whole thing cost about 4,500 dollars. My boss just saw a thing called Asterisk this morning looking for options in Google. He asked my to investigate what this thing called Asterisk is and if we could save some money using it instead of the panasonic solution. So, here I am. I have some experience as linux sysadmin(we have 1 oracle linux server and 1 linux print server) nevertheless I don´t have any idea where and how to start this evaluation? Please Would you give us a clue where to see If Asterisk could work for us? Thanks for your kind help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us
Re: [asterisk-users] Newbie´s question about Asterisk...
I think I have 3 PSTN lines because I can connect a normal telephone to them all and make calls between each of them. We have 5 normal telephones and 1 panasonic. From what I got I need a PC and a of PCI card to interface to my 3 external lines and my 6 internal lines. For the PC I was planning to use the smallest PC posible like a HP Proliant Microserver but it doesn´t have space for this PCI card. Is there another way to interface to 3 external and 6 internal lines?? Thank you for your kind help. On Fri, Feb 18, 2011 at 6:52 AM, Gopalakrishnan A.N sai...@gmail.comwrote: Asterisk is open source and you can install in a normal PC itself and you can avail all the features that proprietary system has. If you want to integrate with any VoIP service then a PC with Asterisk is enough or else if you want to integrate with PSTN lines then you need FXO card to be installed, its a PCI card. Vendors like Sangoma, Digium (from Asterisk) were selling these cards. And for internal for your agents if you need analog hard phones then you need to have FXS card you can avail these FXO and FXS cards in combination. These cards will fit in your PCI slot of machine. Configuring these cards are also very easy. If it is VoIP then you dont need these cards simply install Asterisk in a PC and you are done. On Fri, Feb 18, 2011 at 5:38 PM, Terry Brummell te...@brummell.netwrote: Yes, I use Elastix myself too. Funny that I didn’t mention that one! *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA *Sent:* Friday, February 18, 2011 6:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk... i prefer to go with Elastix very easy to setup and maintain and reach UI rather than freePBX cheers Dhaval On Fri, Feb 18, 2011 at 4:14 PM, Terry Brummell te...@brummell.net wrote: Dean’s link has references to Trixbox. TB has a bad, bad, very bad reputation for being very insecure. Alternatives to TB are FreePBX PBX in a Flash. All are Asterisk based and very easy to set up. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dean Collins *Sent:* Thursday, February 17, 2011 7:29 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users]Newbie´s question about Asterisk... If you already have experience with linux asterisk will be easy for you. Other people will reply with official links but here is how I use Asterisk in my small home office www.cognation.net/asterisk Cheers, Dean -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Francisco Javier Cintrón Olguín *Sent:* Thursday, February 17, 2011 7:26 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Newbie´s question about Asterisk... Hi, My name is Francisco from México. Here, in my work we have a very very old panasonic PBX(12 years old). We are growing and we need to increase our external lines(from 3 to 4) and our internal lines(from 6 to 10). Besides we need voice mail and voice menu too. We asked for a quote to our panasonic dealer. The whole thing cost about 4,500 dollars. My boss just saw a thing called Asterisk this morning looking for options in Google. He asked my to investigate what this thing called Asterisk is and if we could save some money using it instead of the panasonic solution. So, here I am. I have some experience as linux sysadmin(we have 1 oracle linux server and 1 linux print server) nevertheless I don´t have any idea where and how to start this evaluation? Please Would you give us a clue where to see If Asterisk could work for us? Thanks for your kind help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join
Re: [asterisk-users] Newbie´s question about Asterisk...
Yes, use a FXO device, like the AudioCodes MP-114. It is an external gateway that will allow you to interface your PSTN lines to Asterisk via IP. There are other brands out there but in my line of business we only use AudioCodes. From: asterisk-users-boun...@lists.digium.com on behalf of Francisco Javier Cintrón Olguín Sent: Fri 2/18/2011 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]Newbie´s question about Asterisk... Is there another way to interface to 3 external and 6 internal lines?? Thank you for your kind help winmail.dat-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie´s question about Asterisk...
(Please don't top-post and please trim posts that are no longer relevant.) On Fri, 18 Feb 2011, Francisco Javier Cintrón Olguín wrote: I think I have 3 PSTN lines because I can connect a normal telephone to them all and make calls between each of them. We have 5 normal telephones and 1 panasonic. From what I got I need a PC and a of PCI card to interface to my 3 external lines and my 6 internal lines. For the PC I was planning to use the smallest PC posible like a HP Proliant Microserver but it doesn´t have space for this PCI card. Is there another way to interface to 3 external and 6 internal lines?? There re USB and Ethernet devices (Xorcom, Sangoma, Sipura/Linksys/Cisco, and others) that can interface analog phones to your Asterisk server. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie´s question about Asterisk...
First of all, thank you for your help. I was seing Cisco and Linsys web sites and I just came across this 2 devices: Linksys SPA8000 8 phone ports, 1 port ethernet. Cisco SPA8800 4 phone ports, 4 lines, 1 port ethernet. I think they could work for us, because I need maximum 10 normal phones and 4 PSTN lines. Besides with these devices I could use my normal phones, so I would not need additional wiring. So far, this is the list to evaluate the costs of this Linux PBX solution: 1 HP Proliant Micro Server 1 Linksys SPA8000 1 Cisco SPA8800 1 UPS 1 switch 16 ports( we use a 8 port switch) 1 shelf rack 3 patch cords. I just have 3 doubts: 1. What do you think about Linksys SPA8000 and Linksys SPA8800, are they good solutions in my case? 2. What do you think about this list, am I missing something? 3. I am thinking to buy a switch with VLANS to have one VLAN for my PBX, what do you think about this, is it necessary? Thank you for your kind help. On Fri, Feb 18, 2011 at 11:22 AM, Steve Edwards asterisk@sedwards.comwrote: (Please don't top-post and please trim posts that are no longer relevant.) There re USB and Ethernet devices (Xorcom, Sangoma, Sipura/Linksys/Cisco, and others) that can interface analog phones to your Asterisk server. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie´s question about Asterisk...
If you already have experience with linux asterisk will be easy for you. Other people will reply with official links but here is how I use Asterisk in my small home office www.cognation.net/asterisk Cheers, Dean From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francisco Javier Cintrón Olguín Sent: Thursday, February 17, 2011 7:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie´s question about Asterisk... Hi, My name is Francisco from México. Here, in my work we have a very very old panasonic PBX(12 years old). We are growing and we need to increase our external lines(from 3 to 4) and our internal lines(from 6 to 10). Besides we need voice mail and voice menu too. We asked for a quote to our panasonic dealer. The whole thing cost about 4,500 dollars. My boss just saw a thing called Asterisk this morning looking for options in Google. He asked my to investigate what this thing called Asterisk is and if we could save some money using it instead of the panasonic solution. So, here I am. I have some experience as linux sysadmin(we have 1 oracle linux server and 1 linux print server) nevertheless I don´t have any idea where and how to start this evaluation? Please Would you give us a clue where to see If Asterisk could work for us? Thanks for your kind help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [newbie] Conference call
On Fri, 4 Feb 2011 10:54:56 +0330, Pezhman Lali l...@lopl.net wrote: Meetme is a default conference application, but you can try conference or konference http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference http://www.voip-info.org/wiki/view/Asterisk+cmd+Konference http://www.voip-info.org/wiki/view/Asterisk+cmd+Konferencethe installation for conference or konference are more easy Thanks for the links. I'll read up on Conference/Konference. BTW, am I correct in understanding that using Flash() in the dialplan is the programmatic equivalent of the flash hook (R key on European handsets) to put someone on hold and dialing a second call? What about combining the two calls into a conference call? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [newbie] Conference call
Dear, Meetme is a default conference application, but you can try conference or konference http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference http://www.voip-info.org/wiki/view/Asterisk+cmd+Konference http://www.voip-info.org/wiki/view/Asterisk+cmd+Konferencethe installation for conference or konference are more easy best On Thu, Feb 3, 2011 at 1:48 PM, Gilles codecompl...@free.fr wrote: Hello I've never used Asterisk for a three-person call, and would like to check that MeetMe is the way to do this. The ADSL modem provided by my ISP offers free calls to landlines/cellphones when using a handset connected to an RJ11 port on the modem. A three-person call can be set up by using the standard PBX sequence: 1. Using the handset, call party #1 2. Hit R key on handset, which puts party #1 on hold and gives a dialtone 3. Call party #2 4. Once both parties are off-hook, hit R+ 3 on handset to bridge both calls and have a conference call Is MeetMe the right way to do this in Asterisk, or should I look at some other way? Ideally, I'd rather go through a VOSP to avoid the extra digital/analog conversion added by going through the FXO module, but free calls are only available when using that port :-/ Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Question...
On Mon, 31 Jan 2011, Piotr Górski wrote: I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of free calls from each of 4 pstn lines... Can I configure Asterisk to call thru pstn line that has free minutes? For example Outgoing calls are going through PSTN 1 for 60 minutes. When I use all of these free minutes - outgoing calls go thru PSTN 2. When I use all free minutes from PSTN 2 outgoing calls go via PSTN3. You will need to keep track of the call duration for each channel in a persistent store -- something like MySQL. You may also want to read up on setting the absolute timeout on a channel so a caller won't consume all of your 'prepaid' (nothing is free) minutes and drive you into unexpected charges. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Looking For Login/Password
UIT DEVELOPMENT wrote: Sorry for such a silly question but I am VERY new to Linux, Asterisk, and so forth. I just downloaded and burned the AsteriskNOW ISO to CD and installed it. Everything went great. I removed the CD and rebooted and there is a prompt for me to login. I hate to ask but after searching for a few hours, what on earth is the initial login and the password?!? Wasn't this supplied by you on install? I have no experience with this product, and two Google searches give somewhat different answers. If you log in through freepbx, then the answer is here: http://www.asterisk.org/AsteriskNOW-1.5-QuickStart Console Login as user root uses the password supplied by you when you did the initial install. Best of luck Find the PDF book Asterisk the Future of Telephony, free for downloading for help with configuration. Though it doesn't address your specific product, there are many good concepts there. John Novack And, please, where is this information so that I may read further about the installation and so forth. Thank you so much! Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Checked by AVG - www.avg.com Version: 9.0.722 / Virus Database: 270.14.119/2585 - Release Date: 12/24/09 03:11:00 -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Looking For Login/Password
John, Thank you! That was it. I was trying admin, login...I should have searched google more. Thank you and thanks for the link - I've got lots of reading ahead of me this evening! Mike On Fri, Dec 25, 2009 at 6:00 PM, John Novack jnov...@stromberg-carlson.org wrote: UIT DEVELOPMENT wrote: Sorry for such a silly question but I am VERY new to Linux, Asterisk, and so forth. I just downloaded and burned the AsteriskNOW ISO to CD and installed it. Everything went great. I removed the CD and rebooted and there is a prompt for me to login. I hate to ask but after searching for a few hours, what on earth is the initial login and the password?!? Wasn't this supplied by you on install? I have no experience with this product, and two Google searches give somewhat different answers. If you log in through freepbx, then the answer is here: http://www.asterisk.org/AsteriskNOW-1.5-QuickStart Console Login as user root uses the password supplied by you when you did the initial install. Best of luck Find the PDF book Asterisk the Future of Telephony, free for downloading for help with configuration. Though it doesn't address your specific product, there are many good concepts there. John Novack And, please, where is this information so that I may read further about the installation and so forth. Thank you so much! Mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Checked by AVG - www.avg.com Version: 9.0.722 / Virus Database: 270.14.119/2585 - Release Date: 12/24/09 03:11:00 -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie
Hi, I'd say Linphone configuration. I suggest to check Linphone configuration and also asterisk debug - one of those will give you an answer (most likely asterisk debug as it shows you what it receives ... if it doesn't receive anything then Linxphone fails) -- razu On 11/19/2009 11:48 PM, Michael Hausl wrote: Hi, I just started with Asterisk as I am very unhappy with the functionality of my current PBX at home. I try to understand everything and play around, but it is not as easy as I thought. So please be patient if this is a too easy question for You. I installed Asterisk 1.4.26.3 on a Debian Lenny with IP 192.168.2.147 My extension.conf looks like this: [default] exten = 1001,1,Answer() exten = 1001,n,Playback(hello-world) exten = 1001,n,Hangup() exten = 2000,1,Dial(SIP/2000) My sip.conf: [general] port=5060 bindaddr=0.0.0.0 [2000] type=friend secret=1234 host=dynamic I configured a Linphone client on my Ubuntu system. It registers at the Asterisk server. When I type console dial 1001 it works, when I type console dial 2000 on the CLI, my Linphone client rings. When I call sip:1...@192.168.2.147 from Linphone nothing happens at all, even when I set the verbose level to 5 I get no output in CLI. Where is my fault? Thanks for Your help in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
You can tee your CLI screen (google for it) so your output is in a file that you can use more|less|vi or some other controlled viewing method on. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Shaw Sent: Tuesday, November 17, 2009 10:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] newbie question Hi All, When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. Thanks, Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote: Hi All, When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. No. But you can either: 1. Use a terminal that has a long enough scroll-back buffer (or screen inside one that doesn't) 2. Run from the external shell prompt: asterisk -rx 'help whatever' | less -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote: Hi All, When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. On Tue, 17 Nov 2009, Tzafrir Cohen wrote: No. But you can either: 1. Use a terminal that has a long enough scroll-back buffer (or screen inside one that doesn't) 2. Run from the external shell prompt: asterisk -rx 'help whatever' | less Or, you can use the script command to capture the output to a file so you can refer to it as needed. script is also useful to capture the console log to a file when you are trying to debug a call and your console output looks like a broken fire hydrant. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
Option #2 is really the best option unless you need real time viewing of your help information (IMO). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Tuesday, November 17, 2009 10:59 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] newbie question On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote: Hi All, When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. No. But you can either: 1. Use a terminal that has a long enough scroll-back buffer (or screen inside one that doesn't) 2. Run from the external shell prompt: asterisk -rx 'help whatever' | less -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Bill Shaw Sent: Tuesday, November 17, 2009 11:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] newbie question When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. Another option is to use 'screen' and use the integrated scroll back buffer. I'm pretty lazy so most of my servers have established screen sessions with consoles, logs, mysql, etc. already running that I simply reconnect to. sl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
On Tue, Nov 17, 2009 at 09:09:39AM -0800, Steve Edwards wrote: On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote: Hi All, [snip] 2. Run from the external shell prompt: asterisk -rx 'help whatever' | less Or, you can use the script command to capture the output to a file so you can refer to it as needed. I find screen helpful here you can set the scroll back buffer to a large number and you can detach the running screen from the console to reattach some time later. my default scroll back buffer is set to around 1000 usually enough to capture what I need, plus you can cut paste between screens script is also useful to capture the console log to a file when you are trying to debug a call and your console output looks like a broken fire hydrant. -- It's amazing I won. I was running against peace, prosperity, and incumbency. - George W. Bush 06/14/2001 speaking to Swedish Prime Minister Goran Perrson, unaware that a live television camera was still rolling. signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. my default scroll back buffer is set to around 1000 usually enough to capture what I need, plus you can cut paste between screens You could also make it much simpler and just set your verbosity very low or just turn it off, so there are very few messages coming across your screen. Unless you're on a really busy machine, you should be able to read most of the help screens. core set verbose 0 - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie question
On Tue, 17 Nov 2009, Noah Miller wrote: You could also make it much simpler and just set your verbosity very low or just turn it off, so there are very few messages coming across your screen. Unless you're on a really busy machine, you should be able to read most of the help screens. core set verbose 0 Unfortunately, when your boss comes in and says Why did this just* happen?, those logs are kind of handy. I like a lot of logging on production systems. I funnel everything from every server to a single loghost via syslog. First thing every morning, a cron job bzip2s the previous days syslog file and saves it as syslog.bz2-$(date +%d) so I always have 30 days logs on tap and don't have to worry about deleting old log files. *) Sometime in the last 30 days. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to copy track from CD for MOH without getting Junk at beginning of frame ...
On Wed, 19 Aug 2009, Lee, John (Sydney) wrote: I was copying tracks from CD into mp3 files so that I could use it in Asterisk 1.4.21.2 MOH. Are there any Asterisk+Audio expert that can offer me some advice? Don't use MP3. Why would you want to burn CPU cycles decompressing the same stuff over and over? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to copy track from CD for MOH without getting Junk at beginning of frame ...
Steve Edwards wrote: On Wed, 19 Aug 2009, Lee, John (Sydney) wrote: I was copying tracks from CD into mp3 files so that I could use it in Asterisk 1.4.21.2 MOH. Are there any Asterisk+Audio expert that can offer me some advice? Don't use MP3. Why would you want to burn CPU cycles decompressing the same stuff over and over? Yep, agreed. Convert the file to the native codec(s) in which it will be played. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to copy track from CD for MOH without getting Junk at beginning of frame ...
Yep, agreed. Convert the file to the native codec(s) in which it will be played. Alex, could you please elaborate on this? I am no audio guy. On Media player, I can rip it into mp3 or wav or windows media audio. Which one should I use? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to copy track from CD for MOH without getting Junk at beginning of frame ...
Probably none of the ones you list, though I believe wav files are uncompressed. Use SOX http://sox.sourceforge.net/ under Linux, Windows or OSX and RIP/Convert the files to match the codec you are using for calls. If you are accepting calls that use the GSM codec then have a set of MOH files encoded as .gsm, if you are accepting calls that use the g.723 codec then encode your MOH files as g.723, if using speex, use speex, etc... use files already encoded in the formats in which you originate and terminate calls. That way the processor isn't repeating the process of transcoding on every call! Eric Fort FortConsulting On Wed, Aug 19, 2009 at 5:25 PM, Lee, John (Sydney) john@compuware.comwrote: Yep, agreed. Convert the file to the native codec(s) in which it will be played. Alex, could you please elaborate on this? I am no audio guy. On Media player, I can rip it into mp3 or wav or windows media audio. Which one should I use? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to copy track from CD for MOH without getting Junk at beginning of frame ...
On Thu, 20 Aug 2009, Lee, John (Sydney) wrote: Convert the file to the native codec(s) in which it will be played. Alex, could you please elaborate on this? I am no audio guy. On Media player, I can rip it into mp3 or wav or windows media audio. Which one should I use? Neither. If your channels use gsm|ulaw|g729|whatever, encode your sound files (prompts, music on hold, everything) in that format. If you have your sound files encoded with the same codec as the codec your channels are using, Asterisk does not need to transcode so the cost is minimized. The workflow is to rip the cd to disk and then encode to the desired encodings. cdparanoia is a great ripper. cdda2wav is also common. sox is probably the most commonly used tool for encoding. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to find the serial number ofDigium card?
Lee, John (Sydney) wrote: Thanks Tilghman. I learnt it the hard way - I never imagined I need to jot down the serial number of a PCI card :-( I've had a linecard that's been unregistered now for 4 years or more, because it's in a production server. It does of course mean that I didn't get any HPEC licenses. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to find the serial number ofDigium card?
Hello John , On Mon, 17 Aug 2009, Lee, John (Sydney) wrote: Thanks Tilghman. I learnt it the hard way - I never imagined I need to jot down the serial number of a PCI card :-( If you still have the paper work from the box that came to you . The stock agent , if you are lucky , may have written the serial number on the sheet . I have had them do this at various fulfillment centers . Hth , JimL -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Monday, 17 August 2009 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie: How to find the serial number ofDigium card? On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote: Does anyone know how to find the serial number of Digium card without opening the machine? I was trying to call for support at Digium and they asked me for the serial number. You cannot. The serial number is not anywhere in the firmware, only on a sticker on the card itself. -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkSystem Engineer | 2133McCullam Ave | Give me Linux | | bab...@baby-dragons.com | Fairbanks, AK. 99701 | only on AXP | +--+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to find the serial number of Digium card?
On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote: Does anyone know how to find the serial number of Digium card without opening the machine? I was trying to call for support at Digium and they asked me for the serial number. You cannot. The serial number is not anywhere in the firmware, only on a sticker on the card itself. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to find the serial number ofDigium card?
Thanks Tilghman. I learnt it the hard way - I never imagined I need to jot down the serial number of a PCI card :-( -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Monday, 17 August 2009 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie: How to find the serial number ofDigium card? On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote: Does anyone know how to find the serial number of Digium card without opening the machine? I was trying to call for support at Digium and they asked me for the serial number. You cannot. The serial number is not anywhere in the firmware, only on a sticker on the card itself. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] newbie questions
On Sat, 20 Jun 2009, C. Savinovich wrote: Let me see if I get you: you inserted the installation CD, then you restarted the computer, and now you want to know what to do next? How about: 1) Turn off the computer. 2) Read the installation guide for the CD. 3) Install the software. 4) Read ATOF to get a clue to the scope of what Asterisk can do. 5) Get frustrated trying to do really cool things within the GUI. 6) Format the drive. 7) Install CentOS. 8) Install Asterisk from source. 9) Learn to configure the configuration files by hand. But then, I gladly admit to being a command line weenie. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie, Question on making a PSTN call..
I understand the desire to try, but you are trying too hard. Getting a soft modem to work with Asterisk is. like trying to push a string up a 10 foot pipe. At the least, buy an inexpensive FXO device from someone like Grandstream and use it via Ethernet to work with Asterisk. If you have greater ambitions, buy any appropriate piece of hardware and start with that. Otherwise, You are going to have a lot of string in that pipe, before you see any come out the top. You won't get help on this because no one really knows how to do it or if it will work at all. I am trying to help, by getting you to try a better way. Good luck. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shiva Kumar Sent: Tuesday, June 16, 2009 12:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie, Question on making a PSTN call.. Need help pls..Anyone? On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote: Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. On my Ubuntu: Next up, I connected my phone(NOKIA N73) to my computer, ensured that I am able to connect to internet on my ubuntu. wvdial works good too. Again, I am unsure how to get asterisk to connect to this modem so that I can use my soft phones to make a call. Need help. Thanks in Advance. -- Shivku, http://blog.shivku.com -- Shivku, http://blog.shivku.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie, Question on making a PSTN call..
On Mon, 15 Jun 2009, Shiva Kumar wrote: Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. Go out and buy specific hardware. OpenVox are really cheap these days. Well under £100 for a card with an FXO interface now. Gordon___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie, Question on making a PSTN call..
Need help pls..Anyone? On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote: Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. On my Ubuntu: Next up, I connected my phone(NOKIA N73) to my computer, ensured that I am able to connect to internet on my ubuntu. wvdial works good too. Again, I am unsure how to get asterisk to connect to this modem so that I can use my soft phones to make a call. Need help. Thanks in Advance. -- Shivku, http://blog.shivku.com -- Shivku, http://blog.shivku.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie trying to make calls outside via digium card and POTS line
On Mon, Mar 30, 2009 at 5:16 PM, Bruce Thayre br...@mipscomputation.comwrote: Up to this point, all i have set up are two SIP phones, my POTS phone, and 1 ring group. My POTS line is connected to channel 1, and my POTS phone is connected on channel 3. Perhaps my understanding of how the calls are handled is flawed, but it seems to me that: 1. I dial a number on my POTS phone 2. Using the number, asterisk should match it against the dialing rules i have set 3. Having matched the number to an outbound dialing rule, it routes the call to the outside trunk and bingo bango i'm talking on the phone with someone outside my office However in this situation, it doesn't seem to work. And lines like [18585300...@from-internal:6] Congestion(Zap/3-1, 20) in new stack are a mystery to me. If any additional information is needed just let me know what, and i'll post it. Any help would be greatly appreciated as i'm kind of stuck on at this point. Thanks http://lists.digium.com/mailman/listinfo/asterisk-users Hi Bruce, I can't be sure without looking at your dialplan, but based on your description it looks like you are routing calls out the wrong port. Asterisk is trying to dial 1-858-530-0400 on port 3 of your Digium card. You've stated that your POTS line is plugged into port 1, so there's likely an error in your dial command. Do you have Dial(ZAP/3-1... instead of Dial(ZAP/1-1... ? AR -- Alex Robar alex.ro...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie trying to make calls outside via digiumcard and POTS line
Show us your dialplan. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Thayre Sent: Monday, March 30, 2009 4:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie trying to make calls outside via digiumcard and POTS line Hello, This is my first asterisk installation, and having read up on the documentation, and trying several tutorials i'm unable to get my outbound route working. I'm certain it's an issue with my configuration and my inexperience with asterisk. So i have my POTS phone connected to my digium card, and when i make a call, I receive the cannot be completed as dialed message. The log for the event in question is: [Mar 30 10:51:22] VERBOSE[1944] logger.c: -- Starting simple switch on 'Zap/3-1' [Mar 30 10:51:30] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:1] ResetCDR(Zap/3-1, ) in new stack [Mar 30 10:51:30] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:2] NoCDR(Zap/3-1, ) in new stack [Mar 30 10:51:30] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:3] Wait(Zap/3-1, 1) in new stack [Mar 30 10:51:31] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:4] Playback(Zap/3-1, silence/1cannot-complete-as-dialedcheck-number-dial-again|noanswer) in new stack [Mar 30 10:51:31] VERBOSE[1944] logger.c: -- Zap/3-1 Playing 'silence/1' (language 'en') [Mar 30 10:51:32] VERBOSE[1944] logger.c: -- Zap/3-1 Playing 'cannot-complete-as-dialed' (language 'en') [Mar 30 10:51:34] VERBOSE[1944] logger.c: -- Zap/3-1 Playing 'check-number-dial-again' (language 'en') [Mar 30 10:51:37] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:5] Wait(Zap/3-1, 1) in new stack [Mar 30 10:51:38] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:6] Congestion(Zap/3-1, 20) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: == Spawn extension (from-internal, 18585300400, 6) exited non-zero on 'Zap/3-1' [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@from-internal:1] Macro(Zap/3-1, hangupcall) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:1] ResetCDR(Zap/3-1, w) in new stack [Mar 30 10:51:39] DEBUG[1944] app_macro.c: Executed application: ResetCDR [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:2] NoCDR(Zap/3-1, ) in new stack [Mar 30 10:51:39] DEBUG[1944] app_macro.c: Executed application: NoCDR [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:3] GotoIf(Zap/3-1, 1?skiprg) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Goto (macro-hangupcall,s,6) [Mar 30 10:51:39] DEBUG[1944] app_macro.c: Executed application: GotoIf [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:6] GotoIf(Zap/3-1, 1?skipblkvm) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Goto (macro-hangupcall,s,9) [Mar 30 10:51:39] DEBUG[1944] app_macro.c: Executed application: GotoIf [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:9] GotoIf(Zap/3-1, 1?theend) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Goto (macro-hangupcall,s,11) [Mar 30 10:51:39] DEBUG[1944] app_macro.c: Executed application: GotoIf [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:11] Hangup(Zap/3-1, ) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Zap/3-1' in macro 'hangupcall' [Mar 30 10:51:39] VERBOSE[1944] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Zap/3-1' [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Hungup 'Zap/3-1' [Mar 30 10:51:43] DEBUG[2857] chan_zap.c: Message status for 4...@default changed from -1 to 0 on 3 Up to this point, all i have set up are two SIP phones, my POTS phone, and 1 ring group. My POTS line is connected to channel 1, and my POTS phone is connected on channel 3. Perhaps my understanding of how the calls are handled is flawed, but it seems to me that: 1. I dial a number on my POTS phone 2. Using the number, asterisk should match it against the dialing rules i have set 3. Having matched the number to an outbound dialing rule, it routes the call to the outside trunk and bingo bango i'm talking on the phone with someone outside my office However in this situation, it doesn't seem to work. And lines like [18585300...@from-internal:6] Congestion(Zap/3-1, 20) in new stack are a mystery to me. If any additional information is needed just let me know what, and i'll post it. Any help would be greatly appreciated as i'm kind of stuck on at this point. Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options
Re: [asterisk-users] Newbie trying to make calls outside via digiumcard and POTS line
Thank you for the prompt input! My extension.conf can be viewed here: http://dpaste.com/21356/ I'm currently doing the configuration through the GUI bundled with the trixbox distro, and i'm not entirely sure where it stores all of the changes as i haven't seen the changes to extension.conf that i would expect. Should there be additional files i post that will offer more information? And to Alex: Yes you are correct, the POTS line is in port 1, the POTS phone is on port 3. I'm not sure where it's getting the idea that i want to dial on port 3. That part of asterisk (how it routes from one port to another on the digium card) is something i still do not understand well. Thanks for the input...and hopefully patience:) Danny Nicholas wrote: Show us your dialplan. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Thayre Sent: Monday, March 30, 2009 4:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Newbie trying to make calls outside via digiumcard and POTS line Hello, This is my first asterisk installation, and having read up on the documentation, and trying several tutorials i'm unable to get my outbound route working. I'm certain it's an issue with my configuration and my inexperience with asterisk. So i have my POTS phone connected to my digium card, and when i make a call, I receive the cannot be completed as dialed message. The log for the event in question is: [Mar 30 10:51:22] VERBOSE[1944] logger.c: -- Starting simple switch on 'Zap/3-1' [Mar 30 10:51:30] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:1] ResetCDR(Zap/3-1, ) in new stack [Mar 30 10:51:30] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:2] NoCDR(Zap/3-1, ) in new stack [Mar 30 10:51:30] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:3] Wait(Zap/3-1, 1) in new stack [Mar 30 10:51:31] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:4] Playback(Zap/3-1, silence/1cannot-complete-as-dialedcheck-number-dial-again|noanswer) in new stack [Mar 30 10:51:31] VERBOSE[1944] logger.c: -- Zap/3-1 Playing 'silence/1' (language 'en') [Mar 30 10:51:32] VERBOSE[1944] logger.c: -- Zap/3-1 Playing 'cannot-complete-as-dialed' (language 'en') [Mar 30 10:51:34] VERBOSE[1944] logger.c: -- Zap/3-1 Playing 'check-number-dial-again' (language 'en') [Mar 30 10:51:37] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:5] Wait(Zap/3-1, 1) in new stack [Mar 30 10:51:38] VERBOSE[1944] logger.c: -- Executing [18585300...@from-internal:6] Congestion(Zap/3-1, 20) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: == Spawn extension (from-internal, 18585300400, 6) exited non-zero on 'Zap/3-1' [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@from-internal:1] Macro(Zap/3-1, hangupcall) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:1] ResetCDR(Zap/3-1, w) in new stack [Mar 30 10:51:39] DEBUG[1944] app_macro.c: Executed application: ResetCDR [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:2] NoCDR(Zap/3-1, ) in new stack [Mar 30 10:51:39] DEBUG[1944] app_macro.c: Executed application: NoCDR [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:3] GotoIf(Zap/3-1, 1?skiprg) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Goto (macro-hangupcall,s,6) [Mar 30 10:51:39] DEBUG[1944] app_macro.c: Executed application: GotoIf [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:6] GotoIf(Zap/3-1, 1?skipblkvm) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Goto (macro-hangupcall,s,9) [Mar 30 10:51:39] DEBUG[1944] app_macro.c: Executed application: GotoIf [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:9] GotoIf(Zap/3-1, 1?theend) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Goto (macro-hangupcall,s,11) [Mar 30 10:51:39] DEBUG[1944] app_macro.c: Executed application: GotoIf [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Executing [...@macro-hangupcall:11] Hangup(Zap/3-1, ) in new stack [Mar 30 10:51:39] VERBOSE[1944] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Zap/3-1' in macro 'hangupcall' [Mar 30 10:51:39] VERBOSE[1944] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Zap/3-1' [Mar 30 10:51:39] VERBOSE[1944] logger.c: -- Hungup 'Zap/3-1' [Mar 30 10:51:43] DEBUG[2857] chan_zap.c: Message status for 4...@default changed from -1 to 0 on 3 Up to this point, all i have set up are two SIP phones, my POTS phone, and 1 ring group. My POTS line is connected to channel 1, and my POTS phone is connected on channel 3. Perhaps my understanding of how the calls are handled is flawed, but it
Re: [asterisk-users] Newbie query: how to write priority n+101
Geoff Lane wrote: On Thursday, February 5, 2009, Mark Michelson wrote: I've tried it and you're correct. So it looks like the docs need a bug report - any idea how I go about that? If you're using the 2nd edition of the book, check the preface, page xix for contact information. Thanks - errata reported. And I've since fixed it in the SVN repo. Thanks for the report! Leif Madsen. http://www.asteriskdocs.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie query: how to write priority n+101
On Thu, 2009-02-05 at 22:09 +, Geoff Lane wrote: Thanks. For info, *TFOT says: PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to either SUCCESS or FAILURE. If Caller ID is received on the channel, PrivacyManager() does nothing. I've tried it and you're correct. So it looks like the docs need a bug report - any idea how I go about that? Woops! Just goes to show that the guys who wrote Asterisk: The Future of Telephony are human too. Also, thanks for sending an errata to O'Reilly... the problem will be fixed in future printings. -- Jared Smith Digium, Inc. | Training Manager ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie query: how to write priority n+101
On Thu, 2009-02-05 at 22:01 +0100, Philipp Kempgen wrote: Mark Michelson schrieb: Actually, jumping to priority n + 101 is a thing of the past And in addition extensions.conf is a thing of the past. ;-) extensions.ael is cleaner and easier to maintain for most purposes. In the same vein, you may want to look at extensions.lua too, if you are using 1.6. Really extensions.conf is still a perfectly viable way to build your dialplan and will probably remain so for some time. -- Matthew Nicholson Digium, Inc. | Software Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie query: how to write priority n+101
On Thu, 2009-02-05 at 22:01 +0100, Philipp Kempgen wrote: Mark Michelson schrieb: Actually, jumping to priority n + 101 is a thing of the past And in addition extensions.conf is a thing of the past. ;-) snip How about .. dialplan.conf .;-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie query: how to write priority n+101
Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
Sorry for a question that I'm guessing is obvious to most of you.
I'm trying to revamp my dialplan. When I first created it, I had
something like:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten = s,2,Dial(${rgMain},${RINGTIME},t)
exten = s,3,VoiceMail(m...@default)
exten = s,103,VoiceMail(m...@default)
Now I want to play around to add things like the privacy manager and
blacklist handling, which all goes before priority 2 in the above. The
Dial() application jumps to the priority 101 more than its own
priority (i.e. n+101) if it times out. But how can I specify this if
I'm numbering priorities as 1,n,n,n,n?
(BTW, the reason for priority 3 in the above extension is that in an
earlier version of Asterisk, Dial() sometimes jumped to the next
priority rather than one 101 more).
TIA,
Actually, jumping to priority n + 101 is a thing of the past, and this will
only
occur now if you pass the 'j' option to Dial. Dial will just go to the next
priority on a timeout now, and the DIALSTATUS channel variable will be set to
NOANSWER I suspect that if you enable verbose console logging, you'll
actually
see that priority 3 is what is being executed and not priority 103.
Check out the UPGRADE.txt file in Asterisk 1.4. In the Applications section,
you'll see:
* In previous Asterisk releases, many applications would jump to priority n+101
to indicate some kind of status or error condition. This functionality was
marked deprecated in Asterisk 1.2. An option to disable it was provided with
the default value set to 'on'. The default value for the global priority
jumping option is now 'off'.
Mark Michelson
___
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Re: [asterisk-users] Newbie query: how to write priority n+101
On Thursday, February 5, 2009, Mark Michelson wrote: Actually, jumping to priority n + 101 is a thing of the past, and this will only occur now if you pass the 'j' option to Dial. Dial will just go to the next priority on a timeout now, and the DIALSTATUS channel variable will be set to NOANSWER I suspect that if you enable verbose console logging, you'll actually see that priority 3 is what is being executed and not priority 103. Many thanks for that. The change makes sense and also makes dialplan logic somewhat easier! Thanks again, -- Geoff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie query: how to write priority n+101
Mark Michelson schrieb: Actually, jumping to priority n + 101 is a thing of the past And in addition extensions.conf is a thing of the past. ;-) extensions.ael is cleaner and easier to maintain for most purposes. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie query: how to write priority n+101
On Thursday, February 5, 2009, Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most purposes.
Oh-oh ... I don't think I can keep up with the rate of change ;-)
BTW, on a related note, I'm having some trouble with Privacy Manager
that I'd appreciate some insight with. In one priority, I'm calling
PrivacyManager(2,8). In the next priority, I've got:
GotoIf($[${PRIVACYMGRSTATUS} = FAILURE]?withheld,1)
Which I almost cribbed straight out of *TFOT. However, when I call
from a withheld number I get the two expected challenges and when I
deliberately fail (i.e. I haven't entered a valid number), execution
continues with the priority following the GotoIf rather than jumping
to the first priority of the withheld extension.
What am I doing wrong?
TIA,
--
Geoff
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Re: [asterisk-users] Newbie query: how to write priority n+101
Philipp Kempgen wrote: And in addition extensions.conf is a thing of the past. ;-) extensions.ael is cleaner and easier to maintain for most purposes. *gack* Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie query: how to write priority n+101
On Thursday 05 February 2009 15:37:19 Geoff Lane wrote:
On Thursday, February 5, 2009, Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most purposes.
Oh-oh ... I don't think I can keep up with the rate of change ;-)
BTW, on a related note, I'm having some trouble with Privacy Manager
that I'd appreciate some insight with. In one priority, I'm calling
PrivacyManager(2,8). In the next priority, I've got:
GotoIf($[${PRIVACYMGRSTATUS} = FAILURE]?withheld,1)
Which I almost cribbed straight out of *TFOT. However, when I call
from a withheld number I get the two expected challenges and when I
deliberately fail (i.e. I haven't entered a valid number), execution
continues with the priority following the GotoIf rather than jumping
to the first priority of the withheld extension.
What am I doing wrong?
The correct string is FAILED, not FAILURE.
--
Tilghman
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Re: [asterisk-users] Newbie query: how to write priority n+101
On Thursday, February 5, 2009, Tilghman Lesher wrote: The correct string is FAILED, not FAILURE. Thanks. For info, *TFOT says: PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to either SUCCESS or FAILURE. If Caller ID is received on the channel, PrivacyManager() does nothing. I've tried it and you're correct. So it looks like the docs need a bug report - any idea how I go about that? Thanks again, -- Geoff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie query: how to write priority n+101
Tilghman Lesher schrieb:
On Thursday 05 February 2009 15:37:19 Geoff Lane wrote:
BTW, on a related note, I'm having some trouble with Privacy Manager
that I'd appreciate some insight with. In one priority, I'm calling
PrivacyManager(2,8). In the next priority, I've got:
GotoIf($[${PRIVACYMGRSTATUS} = FAILURE]?withheld,1)
Which I almost cribbed straight out of *TFOT.
The correct string is FAILED, not FAILURE.
Whenever you experience a problem like this just have a look
at what the variable contains. Sometimes the documentation is
wrong.
PrivacyManager(...);
Verbose(1,### PRIVACYMGRSTATUS: ${PRIVACYMGRSTATUS});
GotoIf(...);
Philipp Kempgen
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Re: [asterisk-users] Newbie query: how to write priority n+101
Is that true? I was under the impression that .ael was still in use at your own risk mode. AEL certainly looks like a real programming language, but I wasn`t willing to test it out with my dialplan last time I made serious changes. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Thursday, February 05, 2009 16:01 To: Asterisk Users Subject: Re: [asterisk-users] Newbie query: how to write priority n+101 Mark Michelson schrieb: Actually, jumping to priority n + 101 is a thing of the past And in addition extensions.conf is a thing of the past. ;-) extensions.ael is cleaner and easier to maintain for most purposes. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie query: how to write priority n+101
Geoff Lane wrote: On Thursday, February 5, 2009, Tilghman Lesher wrote: The correct string is FAILED, not FAILURE. Thanks. For info, *TFOT says: PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to either SUCCESS or FAILURE. If Caller ID is received on the channel, PrivacyManager() does nothing. I've tried it and you're correct. So it looks like the docs need a bug report - any idea how I go about that? Thanks again, If you're using the 2nd edition of the book, check the preface, page xix for contact information. For those monitoring the mailing list who do not have a copy of the book, the following web page is listed as containing errata, examples, and any additional information: http://www.oreilly.com/catalog/9780596510480 Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie query: how to write priority n+101
On Thursday, February 5, 2009, Mark Michelson wrote: I've tried it and you're correct. So it looks like the docs need a bug report - any idea how I go about that? Thanks again, If you're using the 2nd edition of the book, check the preface, page xix for contact information. Thanks - errata reported. -- Geoff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in Cisco Phone
I run chan_sccp at home. It works well, supports the park function, but does not make use of the conference button. I haven't used the chan_skinny, so I don't know how it compares. With chan_sccp, if you make a change to the configuration, you need to reload the module, thus taking down all phones running sccp. That's fine if there are only a couple of phones, but would be a problem if it is a big office. Mike -Original Message- From: Sam Tam [mailto:samtam...@gmail.com] Sent: Friday, January 23, 2009 12:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Newbie in Cisco Phone Hi I am no expert in the cisco phone Do you have time to help Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Federico Santulli Sent: Saturday, January 24, 2009 12:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: tam...@gmail.com Subject: Re: [asterisk-users] Newbie in Cisco Phone you can try chan_sccp at www.chan-sccp.org it supports most of ccm features and all kind of cisco phones with skinny firmware. Take a look ;) If you need support you can write me back. Federico - Original Message - From: Sam Tam samtam...@gmail.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, January 23, 2009 8:56 AM Subject: Re: [asterisk-users] Newbie in Cisco Phone Well does it matter if the asterisk server is not located in the same network? I am willing to spend a bit of cash to get someone help me to set it up . Since I need it quite done before end of this month Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van Baak Sent: Friday, January 23, 2009 3:35 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie in Cisco Phone On 05:39, Fri 23 Jan 09, Sam Tam wrote: Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam Asterisk does have chan_skinny. Featureset is not as good as CCM, but it's handling my phones and some customers phones as well. Check it out before returning the phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in Cisco Phone
On 15:56, Fri 23 Jan 09, Sam Tam wrote: Well does it matter if the asterisk server is not located in the same network? No, I used to have my phones at home and my asterisk in Denmark in a colocating facility. I am willing to spend a bit of cash to get someone help me to set it up . Since I need it quite done before end of this month If it's ok for you that I'm not in the same country I'm willing to help you a bit. The next 8 to 9 hours are for my boss, but after that I can help. Contact me off-list if you want. Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van Baak Sent: Friday, January 23, 2009 3:35 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie in Cisco Phone On 05:39, Fri 23 Jan 09, Sam Tam wrote: Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam Asterisk does have chan_skinny. Featureset is not as good as CCM, but it's handling my phones and some customers phones as well. Check it out before returning the phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in Cisco Phone
you can try chan_sccp at www.chan-sccp.org it supports most of ccm features and all kind of cisco phones with skinny firmware. Take a look ;) If you need support you can write me back. Federico - Original Message - From: Sam Tam samtam...@gmail.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, January 23, 2009 8:56 AM Subject: Re: [asterisk-users] Newbie in Cisco Phone Well does it matter if the asterisk server is not located in the same network? I am willing to spend a bit of cash to get someone help me to set it up . Since I need it quite done before end of this month Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van Baak Sent: Friday, January 23, 2009 3:35 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie in Cisco Phone On 05:39, Fri 23 Jan 09, Sam Tam wrote: Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam Asterisk does have chan_skinny. Featureset is not as good as CCM, but it's handling my phones and some customers phones as well. Check it out before returning the phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in Cisco Phone
Hi I am no expert in the cisco phone Do you have time to help Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Federico Santulli Sent: Saturday, January 24, 2009 12:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: tam...@gmail.com Subject: Re: [asterisk-users] Newbie in Cisco Phone you can try chan_sccp at www.chan-sccp.org it supports most of ccm features and all kind of cisco phones with skinny firmware. Take a look ;) If you need support you can write me back. Federico - Original Message - From: Sam Tam samtam...@gmail.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, January 23, 2009 8:56 AM Subject: Re: [asterisk-users] Newbie in Cisco Phone Well does it matter if the asterisk server is not located in the same network? I am willing to spend a bit of cash to get someone help me to set it up . Since I need it quite done before end of this month Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van Baak Sent: Friday, January 23, 2009 3:35 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie in Cisco Phone On 05:39, Fri 23 Jan 09, Sam Tam wrote: Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam Asterisk does have chan_skinny. Featureset is not as good as CCM, but it's handling my phones and some customers phones as well. Check it out before returning the phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit
Re: [asterisk-users] Newbie in Cisco Phone
The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/p s8759/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying... So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in Cisco Phone
Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in Cisco Phone
Asterisk's Skinny support is very rudimentary and doesn't include the CCM provisioning stuff. Short answer - not really. Not unless you want to go through a *whole* lot of work. Sam Tam wrote: Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in Cisco Phone
And I assume no one know when they will have a SIP firmware for it too right? Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Friday, January 23, 2009 5:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone Asterisk's Skinny support is very rudimentary and doesn't include the CCM provisioning stuff. Short answer - not really. Not unless you want to go through a *whole* lot of work. Sam Tam wrote: Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in Cisco Phone
On 05:39, Fri 23 Jan 09, Sam Tam wrote: Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam Asterisk does have chan_skinny. Featureset is not as good as CCM, but it's handling my phones and some customers phones as well. Check it out before returning the phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in Cisco Phone
Well does it matter if the asterisk server is not located in the same network? I am willing to spend a bit of cash to get someone help me to set it up . Since I need it quite done before end of this month Sam -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michiel van Baak Sent: Friday, January 23, 2009 3:35 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie in Cisco Phone On 05:39, Fri 23 Jan 09, Sam Tam wrote: Yes I know too. Is there anyway to make it work with asterisk without using Callmanager? Sam Asterisk does have chan_skinny. Featureset is not as good as CCM, but it's handling my phones and some customers phones as well. Check it out before returning the phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Aarons (US) Sent: Friday, January 23, 2009 5:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in Cisco Phone The 7936G/ 7937G Data Sheet says SCCP only which is a shame. It really is a great sounding phone. I have several customers with them as SCCP. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps875 9/product_data_sheet0900aecd806e021a.html From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sam Tam Sent: Thursday, January 22, 2009 3:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Newbie in Cisco Phone Hello all I have used some low end cisco phones in the past and had no problem setting up SIP on it. But today, I have made a big mistake. Buying Cisco Conference phone without even looking whether it supports SIP on not. And yes it is the nice 7937G that I am talking about. Damn this is annoying. So wondering is there anything I can do to make it work with Asterisk or am I good to send back to exchange another item? Sam Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: Cannot conference with 10 digit 3rd party
Lee, John (Sydney) wrote: Calling all Polycom gurus: I am using Polycom IP601 phones with Asterisk 1.4.21.2 In all Polycom phones, I set the following in sip.cfg. dialplan dialplan.impossibleMatchHandling=2 /dialplan (I leave the digitmap unchanged because I thought setting impossibleMatchHandling will ignore the bitmap) ...so that I could dial any number by entering a variable-size telephone number and then hit the send or dial key. This works quite well except when I am doing conferencing. It goes like this: I dialled the 1st party and was answered. Then I press conf key and then enter the 3rd party. I can keep entering until it reaches the 10th digit and then the 10-digit number is automatically dialled. Any thoughts? I don't think the 2 works quite that way. From what I read in the admin guide the impossibleMatchHandling lets you tell the phone how it should handle numbers that are dialed that do NOT match the dial plan. Your numbers that are longer than 10 digits probably match one of the entries in the phone's dialplan so as soon as it matches it sends the number to asterisk. You will either need to wipe out the phone dialplan and replace it with a generic X.T or add a digit map for the number you are dialing that is greater than 10 digits long. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie AEL2: Syntax for Hint
Steve, I downloaded the latest Asterisk version (see below). *CLI core show version Asterisk 1.4.21.2 built by root @ machine1 on a i686 running Linux on 2008-09-11 06:10:06 UTC If I code: Hint(Custom:light1) It will pass aelparse but when it runs, it says Hint is an unknown application on the console. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Murphy Sent: Thursday, 11 September 2008 2:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie AEL2: Syntax for Hint On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote: I am struggling to find out how to code hint in AEL2. I did hint(Custom:light1) and it keeps complaining about the : (colon). It works fine for SIP device like hint(SIP/439). Anyone who has tried it before? Yes, a while back I upgraded AEL to handle both ':' and '' inside the hint parens. This should work on 1.4 on up. What version of asterisk are you using? 1.2? murf -- Steve Murphy Software Developer Digium ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie AEL2: Syntax for Hint
On Thu, 2008-09-11 at 17:41 +1000, Lee, John (Sydney) wrote:
Steve, I downloaded the latest Asterisk version (see below).
*CLI core show version
Asterisk 1.4.21.2 built by root @ machine1 on a i686 running Linux on
2008-09-11 06:10:06 UTC
If I code:
Hint(Custom:light1)
It will pass aelparse but when it runs, it says Hint is an unknown
application on the console.
Try :
context BLF {
hint(Sip/1000) 1000 = NoOp();
};
Works for me
Eric Dantie
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Re: [asterisk-users] Newbie AEL2: Syntax for Hint
context BLF {
hint(Sip/1000) 1000 = NoOp();
};
Works for me
Thanks Eric.
I did not experience any problem in hint with SIP. The problem is if you use
it with Custom.
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Re: [asterisk-users] Newbie AEL2: Syntax for Hint
Lee, John (Sydney) wrote:
I am struggling to find out how to code hint in AEL2.
I did hint(Custom:light1) and it keeps complaining about the : (colon).
It works fine for SIP device like hint(SIP/439).
Anyone who has tried it before?
I just whipped this up to test and it works for me in 1.4.21.2:
context nightmode {
// When you dial 1000, toggle the state of Custom:nightmode
hint(Custom:nightmode) 1000 = {
NoOP(${DEVSTATE(Custom:nightmode)});
if (${DEVSTATE(Custom:nightmode)}==UNKNOWN ||
${DEVSTATE(Custom:nightmode)}==NOT_INUSE)
Set(DEVSTATE(Custom:nightmode)=BUSY);
else
Set(DEVSTATE(Custom:nightmode)=NOT_INUSE);
}
}
You'll need the DEVSTATE backport in order to use this example. See the
links at the bottom of this page:
http://www.voip-info.org/wiki/view/Asterisk+func+Devstate
-Dave
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Re: [asterisk-users] Newbie AEL2: Syntax for Hint
On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote: I am struggling to find out how to code hint in AEL2. I did hint(Custom:light1) and it keeps complaining about the : (colon). It works fine for SIP device like hint(SIP/439). Anyone who has tried it before? Yes, a while back I upgraded AEL to handle both ':' and '' inside the hint parens. This should work on 1.4 on up. What version of asterisk are you using? 1.2? murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie AEL2: Syntax for Hint
*CLI core show version Asterisk 1.4.13 built by root @ machine1 on a i686 running Linux on 2008-09-10 06:46:17 UTC Thanks Steve. What syntax should I use then? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Murphy Sent: Thursday, 11 September 2008 2:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie AEL2: Syntax for Hint On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote: I am struggling to find out how to code hint in AEL2. I did hint(Custom:light1) and it keeps complaining about the : (colon). It works fine for SIP device like hint(SIP/439). Anyone who has tried it before? Yes, a while back I upgraded AEL to handle both ':' and '' inside the hint parens. This should work on 1.4 on up. What version of asterisk are you using? 1.2? murf -- Steve Murphy Software Developer Digium ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: ACD AgentLogin display on phone
I played with the Polycom login/logout function about a year ago, and it looked brilliant. I could never get it to work, but at the time I had both Polycom and Digium agree that it would be worth getting running. I ran out of time on that project, and have never re-visited it. But it would be a great feature to get working! PaulH Lee, John (Sydney) wrote: I have been coding my own IVR for ACD (aka queue) using Polycom phones using AEL2. In particular, I have coded my own AgentCallbackLogin because a) cmd AgentCallbackLogin() is buggy and will not be supported by dev anymore b) I can put in features like hotdesking and additional validation like prohibiting repeated logins and current phone already logged on by other agent and so forth. Having said that, that still leaves one feature not available which is a visible display on the Polycom phone that an agent has already logged on to the phone. I searched the mailing list up and low and there were some sketchy notes about bweschke had developed a patch which could understand the acd-login-logout of Polycom phones. However, I hope someone can answer the following questions for me. a) Is bweschke's patch available in the current version or do we have to download and install it separately? b) Does bweschke's patch only interface with the AgentLogin() command? In other words, after we enabled the acd-login-logout parameters in the Polycom config files and we pressed the key on the phone, will the phone then basically initiate an AgentLogin() command to the Asterisk server? And does the light beside the key shows red to signify that an agent has logged on successfully. c) I have coded my own Agent Login and Logout extension and it would be great if the softkey could call my own agent login and logout extension (this bit is easy) and then showing the red light if it is a successful login (hard?). Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser
We use http://www.areski.net/asterisk-stat-v2/about.php http://www.micpc.com/qloganalyzer/ on Asterisk 1.2, don't know how well they work with later versions regards, Drew Mark Hamilton wrote: Doesn't Queuemetrics run on a license basis? Anything else that's probably open source and free? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faraz Khan Sent: August 13, 2008 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser queuemetrics Lee, John (Sydney) wrote: I am trying to look for a software (open source or proprietory) that could do reporting on both queue and CDR in Asterisk 1.4.* Could someone give me some suggestions? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser
Doesn't Queuemetrics run on a license basis? Anything else that's probably open source and free? Does anyone have any comments/experience about using asteriskguru queue statistics? http://www.asteriskguru.com/tutorials/installation_guide.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser
One of the Asterisk people down here in Melb set it up for the company they used to work for, and I played with it once and it seemed to be usable. PaulH Lee, John (Sydney) wrote: Doesn't Queuemetrics run on a license basis? Anything else that's probably open source and free? Does anyone have any comments/experience about using asteriskguru queue statistics? http://www.asteriskguru.com/tutorials/installation_guide.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Queue: Code your own queuefor AgentCallBackLogin
There's actually a document included with the source code which will take you through setting up an agent callback system. You can find it in 'doc/queues-with-callback-members.txt'. The 'AgentCallBackLogin' application has some issues, and since you can do the same thing with your dialplan, you're better off doing so. I have basically re-written most of my major dial plan extensions using AEL2 and I think they work pretty well although writing a proper agent login and logout and subsequent incoming call handling is not easy for layman in programming. I am not sure about the take up rate of AEL2 but the 2007 comments on this link does not seem too promising. http://www.voip-info.org/wiki/index.php?page_id=2929tk=5d05138dc792171e f5a0comments_page=1 From a programmer's perspective, AEL2 looks much more like a language than AEL and is easier to code and read. BTW, is there any plan when AEL will be retired? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
I am really grateful to all the experts on the mailing list who gave me some very good advice on this problem which I experienced in China. I think we have fixed the problem and the card is no longer reporting any problems. We are able to dial out successfully and we will continue to test. Here are my findings of the problem with working with PRI in China. 1) MFC/R2 is the biggest distraction so far when I tested the line. When the line did not work, I was preoccupied with thinking whether the line is Euro-ISDN or MFC/R2. As I found out, the line which I ordered from NETCOM is definitely Euro-ISDN. As a matter of fact, the search on this mailing list about installs in China has returned almost 100% ISDN. So, the zaptel.conf should be pretty straightforward with no tweaking at all. 2) Communication with the telco is the other problem. When we placed the order, we just said we wanted an E1. They delivered the E1 no problem but we forgot to specify that the interface must be RJ48 and NOT RJ45. As you might know, RJ45 and RJ48 mean 2 totally different interfaces. RJ48 is for ISDN and voice connection and RJ45 is for data. So, after my local contact talked to the engineers, they came and replace the converter and plug the RJ48 into the Digium card and everything worked fine. Hope this helps with any future installs in China! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Excelent!! but may be better if you send to the list the zaptel.conf and zapata.conf Regards, Luis Morales On Thu, Aug 21, 2008 at 10:19 PM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: I am really grateful to all the experts on the mailing list who gave me some very good advice on this problem which I experienced in China. I think we have fixed the problem and the card is no longer reporting any problems. We are able to dial out successfully and we will continue to test. Here are my findings of the problem with working with PRI in China. 1) MFC/R2 is the biggest distraction so far when I tested the line. When the line did not work, I was preoccupied with thinking whether the line is Euro-ISDN or MFC/R2. As I found out, the line which I ordered from NETCOM is definitely Euro-ISDN. As a matter of fact, the search on this mailing list about installs in China has returned almost 100% ISDN. So, the zaptel.conf should be pretty straightforward with no tweaking at all. 2) Communication with the telco is the other problem. When we placed the order, we just said we wanted an E1. They delivered the E1 no problem but we forgot to specify that the interface must be RJ48 and NOT RJ45. As you might know, RJ45 and RJ48 mean 2 totally different interfaces. RJ48 is for ISDN and voice connection and RJ45 is for data. So, after my local contact talked to the engineers, they came and replace the converter and plug the RJ48 into the Digium card and everything worked fine. Hope this helps with any future installs in China! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser
queuemetrics Lee, John (Sydney) wrote: I am trying to look for a software (open source or proprietory) that could do reporting on both queue and CDR in Asterisk 1.4.* Could someone give me some suggestions? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser
Doesn't Queuemetrics run on a license basis? Anything else that's probably open source and free? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faraz Khan Sent: August 13, 2008 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser queuemetrics Lee, John (Sydney) wrote: I am trying to look for a software (open source or proprietory) that could do reporting on both queue and CDR in Asterisk 1.4.* Could someone give me some suggestions? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
On 7/31/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Thu, Jul 31, 2008 at 05:36:14PM +1000, Lee, John (Sydney) wrote: Yes, I tried all sorts of cables and ended up getting the local contact to complain to NETCOM. An engineer came and swapped the Fast Ethernet to E1 converter. Hmmm. Whose side is Fast Ethernet, and whose side is E1? Are you trying to take the E1 that they've *converted into 100BT* for you and plug it into an E1 port? Since this thread is still going I thought I'd chime in again. With our working CNC setup in Kunming, they provide some kind of router which breaks a single incoming fibre in to both 100BT and an E1 line that plugs in to the Sangoma card. zaptel_hardware output is: pci::04:06.0 wanpipe- 1923:0300 Sangoma Technologies Corp. A101 single-port T1/E1 /etc/asterisk/zapata.conf: ; Sangoma A102 port 1 [slot:6 bus:4 span:1] wanpipe1 switchtype=5ess context=incoming-kunming group=0 signalling=pri_cpe channel =1-15,17-31 One thing that caused issues when setting up for the first time was the fact that dialling out without setting the correct 'caller ID' would yield errors. So, make sure in your dialplan you do this, or outgoing testing may inexplicably fail. A line like: exten = s,n,Set(CALLERID(number)=02222) Also, if you have not set up an incoming context calling in over the analog network will generate an error tone from the network, rather than anything more obvious. In this case somewhere in asterisk's logfiles you can see unknown extension or an error of that sort that appears each time an incoming attempt is made, but there are no other clues. So make sure your incoming contexts are set up! Best of luck. Regards, Walter Stanish Owner / Director Occident Systems (+86 15808 700 801) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
if after you tried both straight through crossover cables and it still give you RED alarm. just tell them you can't get any clocking signal. they'll probably send someone on site and test the line. Yes, I tried all sorts of cables and ended up getting the local contact to complain to NETCOM. An engineer came and swapped the Fast Ethernet to E1 converter. Now we use a normal RJ45 cable to connect the converter to TE412P card. The lights turns green but changes to yellow and green again. dmesg shows a continuous stream of: wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 ...and I am using the following in zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 ... I have changed the timing source from 1 to 0 to 2 but it doesn't make any difference. Any thoughts? p.s. note that T1/E1 crossover cable pin out is not the same as ethernet crossover cable. Do you mean RJ48? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
On Thu, Jul 31, 2008 at 12:31 PM, Uros Djokic [EMAIL PROTECTED] wrote: Hi, Ensure that in file indications.conf you have [general] contry=cn ; not usa ! or if you are in Australia shortcut for Australia Regards, Uros -- Use Free Software http://www.fsf.org/ --- Four essential software freedoms: 1) To study source code 2) To copy program 3) To modify source code 4) To redistribute modified program under condition that new user has all 4 freedoms. Richard M. Stallman -- Use Free Software http://www.fsf.org/ --- Four essential software freedoms: 1) To study source code 2) To copy program 3) To modify source code 4) To redistribute modified program under condition that new user has all 4 freedoms. Richard M. Stallman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Hi, Ensure that in file indications.conf you have [general] contry=cn ; not usa ! Regards, Uros -- Use Free Software http://www.fsf.org/ --- Four essential software freedoms: 1) To study source code 2) To copy program 3) To modify source code 4) To redistribute modified program under condition that new user has all 4 freedoms. Richard M. Stallman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee, John (Sydney) Sent: Thursday, July 31, 2008 3:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1 if after you tried both straight through crossover cables and it still give you RED alarm. just tell them you can't get any clocking signal. they'll probably send someone on site and test the line. Yes, I tried all sorts of cables and ended up getting the local contact to complain to NETCOM. An engineer came and swapped the Fast Ethernet to E1 converter. Now we use a normal RJ45 cable to connect the converter to TE412P card. The lights turns green but changes to yellow and green again. dmesg shows a continuous stream of: wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! timing source auto card 0! wct4xxp: Clearing yellow alarm on span 1 ...and I am using the following in zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 ... I have changed the timing source from 1 to 0 to 2 but it doesn't make any difference. Any thoughts? Sounds like you're making progress. I would try the above span definition without the crc4. That might do the trick. Regards, - Brad ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Sounds like you're making progress. I would try the above span definition without the crc4. That might do the trick. Thanks Brad. I already tried it without crc4 but it makes no difference. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Ensure that in file indications.conf you have [general] country=cn ; not usa ! or if you are in Australia shortcut for Australia Uros, that was a good reminder. However, I don't think it is related to this problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Make experiment.Make loopback Rj-45. (wire 1 from pin 1 to pin 4 wire 2 from pin 2 to pin 5). Then put it in card and if card is OK you should see green led.You should also see dozens of ALARMS notices or warnings on asterisk CLI. Also check pinout http://www.goonda.org/archive/docs/pinout.html Pinout should be 1,2,4,5 (on card side). Call telco. Make them check line with tester (from their point to isdn) to ensure line is ok. What is color of Fritz led ? (green,red or yellow ?) What is color of card's led ? (green ? red ?) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
On Thu, Jul 31, 2008 at 05:36:14PM +1000, Lee, John (Sydney) wrote: Yes, I tried all sorts of cables and ended up getting the local contact to complain to NETCOM. An engineer came and swapped the Fast Ethernet to E1 converter. Hmmm. Whose side is Fast Ethernet, and whose side is E1? Are you trying to take the E1 that they've *converted into 100BT* for you and plug it into an E1 port? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Dan Austin wrote: John wrote: Thanks Steve for your suggestions. In China you will generally get either MFC/R2 or EuroISDN. MFC/R2 is much more common. This is exactly my current problem. NETCOM in Shanghai just told my local contact it is an E1 and that's it. I have no idea whether it is MFC/R2 or EuroISDN and so there is a lot of trial and error, not to mention about communicating with the telco. Is there anyway I could find out from zaptel what the line signal is? International installs are always fun. I have had some luck getting a local employee to relay my questions about provisioning, but all to often the response is 'We use the standard settings...'. At that point I resort to trial and error. I have setup a circuit in Shanghai, it is an E1, CRC4/HDB3 with the telco switch being/or compatible with ATT 5ESS. You should be able to get Netcom to tell you if the circuit is ISDN or not. Asking if it is a PRI will just confuse them, but they do understand the question 'ISDN or not ISDN' The only oddity with EuroISDN is that it often provided without CRC4. That doesn't make a lot of sense, but there it is. MFC/R2 seems to be universally provided without CRC4 in China. That's great info, Steve. Just to comment - this is a great thread. I am expecting that the answer will either be quite interesting or quite odd. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
You don't need to install it. Just run kernel/xpp/utils/genzaptelconf directly from the source directory. Thanks Tzafrir. My local contact is away today and so I could not get him to plug the line to port 4. So, it is still in port 1. Here is the output after running genzaptelconf. # /usr/src/zaptel-1.4.11/kernel/xpp/utils/genzaptelconf # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 B8ZS/ESF RED == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 B8ZS/ESF RED == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 B8ZS/ESF RED Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
emist wrote: My best guess from looking at that is that its a driver bug. The last thing that happens before the lockup seems to be an ioctl call to the device. That was a bug that should have been resolved by 1.4.11 (he subsequently updated and it was resolved). Matthew Fredrickson Digium, Inc Hope it helps, Igor H. Lee, John (Sydney) wrote: This time, I am trying to remotely install Asterisk in China. I was told that an E1 line has been installed and so I plug it into port 1 of a TE412P. On the box, first of all, I just installed Zaptel 1.4.10.1. # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] # vi zaptel.conf [...] span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 *** However, I received a red alarm in zttool and the LED on the TE412P card is also red. *** I have made sure that the jumper is closed for port 1 on the TE412P card and so it could not be the jumper problem. ### Because this is the first time I install Asterisk in China and I was wondering if their E1 is different from the Euro E1. ### However, I went into dmesg and I discovered the following. ### Could it really be a zaptel bug? I saw on a similar few on the digium bug list but I cannot be 100% sure. Any thoughts? About to enter spanconfig! Done with spanconfig! Registered tone zone 33 (China) About to enter startup! TE4XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 128 channels BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681] Pid: 4681, comm:ztcfg EIP: 0060:[f8cba1df] CPU: 2 EIP is at init_vpm450m+0x32d/0x34a [wct4xxp] EFLAGS: 0286Tainted: G (2.6.18-92.1.6.el5 #1) EAX: EBX: f76ae8f0 ECX: 0019 EDX: ESI: f7997400 EDI: 0286 EBP: f76838c0 DS: 007b ES: 007b CR0: 8005003b CR2: b7f01007 CR3: 37bc1000 CR4: 06d0 [f8ca1b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8ca5ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042609c] release_console_sem+0x17e/0x1b8 [c046d53a] cache_alloc_refill+0x14b/0x450 [f8956f61] zt_ioctl+0x273/0x144f [zaptel] [c04d7d45] generic_make_request+0x248/0x258 [c045ae3c] __do_page_cache_readahead+0x69/0x1c6 [c0484a5b] __d_lookup+0x98/0xdb [c047c110] do_lookup+0x53/0x166 [c047e7e4] do_path_lookup+0x20e/0x25e [c047c389] permission+0xa2/0xb5 [c04e2d06] kobject_get+0xf/0x13 [c046f7fa] __dentry_open+0xea/0x1ab [c046f91f] nameidata_to_filp+0x19/0x28 [c046f959] do_filp_open+0x2b/0x31 [c048029b] do_ioctl+0x47/0x5d [c04804fb] vfs_ioctl+0x24a/0x25c [c0471bbe] __fput+0x13f/0x167 [c0480555] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Completed startup! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Lee, John (Sydney) wrote: The test for that is simple: head -n 1 /proc/zaptel/* Let's look at all four spans. Not just the first one. Thanks Tzafrir. # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 So I am quite sure that port 1 is plugged in properly. As I am dealing with telecom in China, I think I might have stepped onto the MFC R/2 bombshell but I have no idea whether the signalling is ISDN or R2. I tried the suggestion on http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 but the red alarm is still on. If it is really R2, then maybe I need to buy an E100P card instead of TE412P. No, you should be fine with a TE412. Just make sure that your line is plugged in correctly and your span= line is correct for the line settings. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Queue: Code your own queue for AgentCallBackLogin
There's actually a document included with the source code which will take you through setting up an agent callback system. You can find it in 'doc/queues-with-callback-members.txt'. The 'AgentCallBackLogin' application has some issues, and since you can do the same thing with your dialplan, you're better off doing so. :M Lee, John (Sydney) wrote: I am trying to build a simple queue with several agents using AgentCallBackLogin. From what I read on the Internet and tried briefly, it seems to suggest that I should be coding my own queue system for AgentCallBackLogin using AEL2 instead of using the AgentCallBackLogin command because it is buggy and will no longer be supported. Is this true? I don't seem to see too much literature on the Internet about using AEL2 or are people still waiting until we are forced to use AEL2? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Lee, John (Sydney) wrote: i've installed several Asterisk systems in Shanghai Beijing. Thanks Edwin. The remote site is in Shanghai and NETCOM is the telco. Do you know if their E1 line is MFC/R2 or EuroISDN? i'm not sure if they provide MFC/R2. but we always ordered PRI from them. as far as switch type. seems like nobody in CNC can give us a definite answer, but we have success using EuroISDN swicth type. red alarm usually means there's no clocking signal. check all your cables (crossover vs straight through) As far as the cable goes, this is a bit complicated. The way it works is the telco delivers a fibre optic cable to the floor and the fibre terminates on a fibre optic multiplexer. Then the multiplexer is connected to a Fast Ethernet to E1 converter which has a RJ45 port. We then connect this RJ45 port to the TE412P port. Anyway what you said is still a good point - I will try replacing the straight through cable with a crossover and give it a go. if the cable's good. call phone company and complain. in my experience 9 out of 10 time we have to call phone company and complain. How should we complain? Are there any technical details we need to show them? It is a different country though. if after you tried both straight through crossover cables and it still give you RED alarm. just tell them you can't get any clocking signal. they'll probably send someone on site and test the line. p.s. note that T1/E1 crossover cable pin out is not the same as ethernet crossover cable. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Queue: Code your own queue for AgentCallBackLogin
Lee, John (Sydney) wrote: I am trying to build a simple queue with several agents using AgentCallBackLogin. From what I read on the Internet and tried briefly, it seems to suggest that I should be coding my own queue system for AgentCallBackLogin using AEL2 instead of using the AgentCallBackLogin command because it is buggy and will no longer be supported. Is this true? I don't seem to see too much literature on the Internet about using AEL2 or are people still waiting until we are forced to use AEL2? I have used addqueuemember and that works quite well with current versions of Asterisk and the old dialplan (ie: not ael2) I have the example code somewhere if you would like a copy. PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
I think it can't hurt to try a different release. Let me know how it goes. Thanks Igor. I just upgraded zaptel to 1.4.11. However, I am still seeing red in the alarm in zttool and the LED on port 1 also shows red. --- cat /proc/zaptel/1 is also showing Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 RED 1 TE4/0/1/1 Clear RED 2 TE4/0/1/2 Clear RED Zaptel started up fine and dmesg below does not show the error message. I am just wondering whether this China E1 could be using MFC R/2? How do I know it is? Stopped TE4XXP, Turned off DMA TE4XXP: Disabling interrupts since there are no active spans Unregistered Tormenta2 Registered Tormenta2 PCI Found TE4XXP at base address fc4ffc00, remapped to f89f4c00 TE4XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x35fe0400 Reg 1: 0x35fe Reg 2: 0x Reg 3: 0x Reg 4: 0x0101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1100 Reg 8: 0x010200ff Reg 9: 0x00fd0001 Reg 10: 0x004a TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE410P (4th Gen) usbcore: registered new driver wcusb Wildcard USB FXS Interface driver registered About to enter spanconfig! Done with spanconfig! About to enter startup! TE4XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 1: Primary Sync Source VPM400: Not Present VPM450: echo cancellation for 128 channels VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Completed startup! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
