Re: [Asterisk-Users] Grandstream Early Dial
On Wed, 2003-12-31 at 05:53, Tilghman Lesher wrote: What happens when you change the configuration of the GS phone to send DTMF via SIP INFO? I've just checked my voicemail with 1.0.4.30 and get the same multiple digits problem. sip.conf and GS config are both at info, for me this is a new problem voicemail has always worked perfectly with the GS. I can't go back to 1.0.3.81 to remove that variable. I updated from CVS on the 26th. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Early Dial
Hi, I have my GS set to in-audio for DTMF and as bellow for my sip.conf: - [7001] ; SIP Phone type=friend insecure=yes host=dynamic reinvite=no canreinvite=no nat=1 mailbox=7001 dtmfmode=inband callgroup=1 pickupgroup=1 disallow=all allow=ulaw allow=alaw allow=gsm I am using 1.0.4.26 and all is working fine. The only differance I have noticed since moving up to 1.0.4.x is the speaker volume is lower on speakerphone. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dave Cotton Sent: 31 December 2003 07:13 To: Asterisk List Subject: Re: [Asterisk-Users] Grandstream Early Dial On Wed, 2003-12-31 at 05:53, Tilghman Lesher wrote: What happens when you change the configuration of the GS phone to send DTMF via SIP INFO? I've just checked my voicemail with 1.0.4.30 and get the same multiple digits problem. sip.conf and GS config are both at info, for me this is a new problem voicemail has always worked perfectly with the GS. I can't go back to 1.0.3.81 to remove that variable. I updated from CVS on the 26th. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast gui client error
### connect to asterisk manager through telnet $t = new Net::Telnet (Port = 5038, Prompt = '/.*[\$%#] $/', Output_record_separator = '',); #$fh = $t-dump_log(./telnet_log.txt); # uncomment for telnet log $t-open($server_ip); i got error in this line $t-open($server_ip); my ip is 192.168.0.5 for asterisk and everyhings ok. the error i get is [EMAIL PROTECTED] astguiclient]# perl AST_SQL_update_channels.pl problem connecting to 192.168.0.5, port 5038: Connection refused at AST_SQL_update_channels.pl line 73 anyhting to do with port?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phone as intercom
Hello, all Sorry to correct you on this Matt, but I am currently doing this with the Polycom 600 phones. You need the latest version of both the SIP software and bootrom to do it, (and that stuff ain't easy to get) but it is workable. The latest version of software provides for distinctive ring tones just like the Cisco 7960's have. It also provides for auto answer. It's kind of tricky to do, but you can make your phones auto answer by setting the Alert-Info variable in asterisk and messing with the xml configuration files, sip.cfg and ipmid.cfg. In the sip.cfg file, look for the line with these variables: alertInfo voIpProt.SIP.alertinfo.1.value=Sales voIpProt.SIP.alertInfo.1.class=8... In this real-world example, whenever I set ALERT_INFO to Sales in Asterisk, the Polycom matches on that word and calls up class 8 in ipmid.cfg. In ipmid.cfg, my class 8 line looks like this: SALES se.rt.8.name=Sales se.rt.8.type=ring se.rt.8.ringer=11 se.rt.8.callWait=6 se.rt.8.mod=0 se.rt.8.type=ring tells the Polycom phone which type of ring to use - which in this case is a regular ring and se.rt.8.ringer=11 tells the phone to ring with ringtone 11 with is the Triplet. I use this one for signaling a new incoming sales call to one of my three sales guys. The secretary transfers it to the sales department and their lines ring with the Triplet. I feel like Pavlov whenever I hear it. The other ring types are visual, answer and ring-answer. The one you want is ring-answer. Here's how I do it: Again in sip.cfg (actually part of the same line listed above) ...voIpProt.SIP.alertinfo.2.value=Ring Answer voIpProt.SIP.alertInfo.2.class=4... and in ipmid.cfg (I just modified one of the existing ones to give me a High Double Trill ringtone) RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000... se.rt.4.ringer=7... The se.rt.4.timeout=1000 tells the Polycom to ring for 1000 milliseconds (one second) and then answer. I call it in Asterisk by setting the ALERT_INFO variable to Ring Answer whenever anybody pushes 8 plus the extension. It rings in to the extension and voila, I'm on speaker! By the way, for all you BOFH out there, you could actually use this feature as a somewhat surreptitious eavesdropping device by using a silent ring and a visual type. The phone would answer without any indication except on the console. I haven't tried this myself and if you do this, I don't want to know about it...unless I'm in your office at the time. Good Luck! I was going to put this in the Wiki myself, but maybe somebody will give me a late Christmas present. --John Baker On Tue, 2003-12-30 at 19:08, mattf wrote: Hello, It's all dependant upon the firmware of the phone(nothing to do with the PBX or SIP currently). The documentation of the Polycom VOIP phones shows no way of doing this currently but it is really just a matter of Polycom adding this feature to their firmware in the future which we are pushing for. People have gotten this to work with Cisco and Snom phones. MATT--- -Original Message- From: Sean Adams [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 6:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP phone as intercom (new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A Head Check
Greg Boehnlein wrote: Hello, I have been retained by a Building Management Company to install a combined Voice/Data solution for a Tennated Office Space. This space will rent offices, with telephone and internet service to inviduals or small groups of individuals. As fate would have it, the service will be provided in a building where we have a major Pop, with a DS-3 worth of ISDN PRI circuits, 345 megs of upstream bandwidth and diesel generator backup. We are providing everything for the solution, from the initial wiring to the ongoing maintenance of the PBX and Internet service. I have arranged for a single PRI to be broken out of our DS-3 w/ 100 inbound DID numbers assigned to it and have PICd it to the LD provider of our choice. I intend to plug this PRI into an Asterisk server w/ a Digium TE410P card, and deploy SNOM 200 IP phones to the desktops. We will be using a RedHawk power-injector system to provide power to the phones. Now.. This is our first deployment of Asterisk, and I need a head check here. Am I making the right decision? :) Sepcifically... 1. Are the SNOM 200 IP phones a good choice for standard users? Or should I consider Cisco? Price of the phone is not the important thing.. What is important is ease of use with minimal training and reliability! IMO Snom 200's are great, I have never had an issue with them and they are simple enough for a standard user while being feature packed enought from power users at the same time.. GS phones have been fine but may be a little too basic for an office and I have not tried the cisco's (Cant afford them). 2. Does anyone have reccomendations for a solid motherboard to use as the basis for the Asterisk server? Again, reliability and stability are the important issues here. I'm looking for a Dual CPU board (Athlon MP or P4) that will work flawlessly with the TE410P. I've used the Tyan Tiger MPX (2466) http://www.tyan.com/products/html/tigermpx.html with Dual MP processors with incredible success in the future. I'm considering building the box on that platform. My Advice would be to stick with intel processors (no flames please I know some of your love AMD's) becasue I have never heard of compatibility issues with Asterisk and Intel proc's and chipsets but I have heard some weird issues with VIA chipsets and AMD procs.. Also AFAIK you can't use the Athlon optimised kernel with Asterisk on an AMD which will probably mean you have to use the i386 kernel where on a P4/Xeon you can happily use the i686 kernel.. Not sure exactly what that means in terms of Asterisk performance but I will stay with the P4/Xeon. 3. I am also responsible for delivering inbound faxes to the DID numbers via Email. I.E. customer has a document faxed to them and they get it in Email as a tiff. I'm considering using Hylfax with a Multitech DID capable modem, but other suggestions are welcomed! My initail thought would be Hylafax but havent had enough experience in this area to comment. 4. I have built some cost for support from Digium and/or other Asterisk experts into the budget. Does Digium have paid support plans? What about other consultants out there? I'm just trying to make sure that I cover all the bases. This is got to be a bulletproof solution, and I'm departing from my comfort level with Altigen to give Asterisk a run for the money. We've got TONS of Linux experience here, and comfort with customizing code, so I am happy with what Asterisk gives me.. What else should I be worried about? Good luck.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Happy New Year!!
Hi all, Let me be the first to wish everyone, especially the Digium team, an awesome year in 2004.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] playback in [macro-stdexten] problem
Lance Arbuckle wrote:
I added the playback line to my [macro-stdexten] context but when I dail
an extension I don't get the please hold while I try that extension
message. It just dials the extexsion. Do I have a syntax problem
somewhere ?
exten = 8005,1,Macro(stdexten,8005,Zap/2)
exten = 8006,1,Macro(stdexten,8006,Sip/8006)
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Playback(transfer,skip)
exten = s,2,Dial(${ARG2},20) ; Ring the interface, 20 seconds
maximum
exten = s,3,Voicemail(u${ARG1}); If unavailable, send to
voicemail w/ unavail announce
exten = s,103,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
It's good practise to answer the line before you say something :-)
exten = s,1,answer
From the tips and tricks page on the Wiki:
http://www.voip-info.org/wiki-Asterisk+tips+answer-before-playback
/O
--
*** Olle E. Johansson, [EMAIL PROTECTED]
Mobile +46 70 593 68 51, Edvina AB, http://www.edvina.net
Runbovägen 10, 192 48 Sollentuna, Sweden
Phone: +46 8 594 78 810, Fax: +46 8 594 78 820
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MySQL CDRs
When using mysql cdrs, are all legs of a call session logged in the cdr table? i'm building an app that requires billing on both the incoming and outgoing (3rd-party transfers) legs. here's a snapshot of my cdr table: +-+-+-+-++--+--- --+ | calldate| src | dst | channel | dstchannel | duration | billsec | +-+-+-+-++--+--- --+ | 2003-12-31 16:19:08 | | s | vpb/1-3 | vpb/1-1| 69 | 67 | | 2003-12-31 16:20:25 | | s | vpb/1-3 | vpb/1-1| 48 | 47 | can someone verify if my assumptions below are correct? 1) these two records are for two call sessions, where both sessions had an inbound and an outbound leg 2) the 'duration' column only shows total time for the inbound leg if assumption #2 is correct, how would i be able to record the duration when the outbound leg is bridged in to the inbound leg? faiz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A Head Check
On Tue, 2003-12-30 at 23:47, Greg Boehnlein wrote: Hello, I have been retained by a Building Management Company to install a combined Voice/Data solution for a Tennated Office Space. This space will rent offices, with telephone and internet service to inviduals or small groups of individuals. As fate would have it, the service will be provided in a building where we have a major Pop, with a DS-3 worth of ISDN PRI circuits, 345 megs of upstream bandwidth and diesel generator backup. We are providing everything for the solution, from the initial wiring to the ongoing maintenance of the PBX and Internet service. I have arranged for a single PRI to be broken out of our DS-3 w/ 100 inbound DID numbers assigned to it and have PICd it to the LD provider of our choice. I intend to plug this PRI into an Asterisk server w/ a Digium TE410P card, and deploy SNOM 200 IP phones to the desktops. We will be using a RedHawk power-injector system to provide power to the phones. Now.. This is our first deployment of Asterisk, and I need a head check here. Am I making the right decision? :) Sepcifically... 1. Are the SNOM 200 IP phones a good choice for standard users? Or should I consider Cisco? Price of the phone is not the important thing.. What is important is ease of use with minimal training and reliability! 2. Does anyone have reccomendations for a solid motherboard to use as the basis for the Asterisk server? Again, reliability and stability are the important issues here. I'm looking for a Dual CPU board (Athlon MP or P4) that will work flawlessly with the TE410P. I've used the Tyan Tiger MPX (2466) http://www.tyan.com/products/html/tigermpx.html with Dual MP processors with incredible success in the future. I'm considering building the box on that platform. Oddly enough, any decent motherboard should be okay given that you will provide it appropriate cooling and proper power. I have a couple of abit boards with multiple 1+ year up times. One currently approaching 2 years. I also have 2 dells 2450s that exceeded a year with one approaching 2 years. We have a few supermicros, one of which is over a year of uptime now. I have had a couple other otherwise unremarkable machines make it past a year of uptime. All of those have been on huge powerware UPS with diesel backup to them and all the other tier 1 level colo facility amenities. So, while I am partial to the supermicros and dells, I have experience with no name systems being just as stable long term. 3. I am also responsible for delivering inbound faxes to the DID numbers via Email. I.E. customer has a document faxed to them and they get it in Email as a tiff. I'm considering using Hylfax with a Multitech DID capable modem, but other suggestions are welcomed! It has been mentioned here before that you can pick up a device that accepts a PRI and will do your fax reception for you. If you get one of those and hook it to a empty port of the TE410P then it will be better as you could accept several faxes at once. 4. I have built some cost for support from Digium and/or other Asterisk experts into the budget. Does Digium have paid support plans? What about other consultants out there? Digium has support, and there are several companies around here that will no doubt be contacting you shortly. I'm just trying to make sure that I cover all the bases. This is got to be a bulletproof solution, and I'm departing from my comfort level with Altigen to give Asterisk a run for the money. We've got TONS of Linux experience here, and comfort with customizing code, so I am happy with what Asterisk gives me.. What else should I be worried about? Maybe you should look into a couple of machines. One that does actual connections to the PSTN, and a couple of separate systems that then support the VoIP phones. I suggest this so that you can then have a very stable core machine that just routes calls, and several other machines that may need to be brought down from time to time for updates. This is the current method we use in my office. One super stable machine at the core that is rarely updated. A few machines to the side that any one may be downed and upgraded without affecting the others. This may be especially of interest to you if you are concerned with the sip phones functionality you may need to do somewhat regular updates. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL CDRs
Outbound calls are not logged seperatly in the CDR. So the duration is the time the hangup extension ends minus time someone dials in on *. Although I think it could be usefull to seperatly log an outgoing call-session when it was bridged... My .02 euro. with kind regards, Steven On Wed, 2003-12-31 at 09:48, Ahmad Faiz wrote: When using mysql cdrs, are all legs of a call session logged in the cdr table? i'm building an app that requires billing on both the incoming and outgoing (3rd-party transfers) legs. here's a snapshot of my cdr table: +-+-+-+-++--+--- --+ | calldate| src | dst | channel | dstchannel | duration | billsec | +-+-+-+-++--+--- --+ | 2003-12-31 16:19:08 | | s | vpb/1-3 | vpb/1-1| 69 | 67 | | 2003-12-31 16:20:25 | | s | vpb/1-3 | vpb/1-1| 48 | 47 | can someone verify if my assumptions below are correct? 1) these two records are for two call sessions, where both sessions had an inbound and an outbound leg 2) the 'duration' column only shows total time for the inbound leg if assumption #2 is correct, how would i be able to record the duration when the outbound leg is bridged in to the inbound leg? faiz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ser and Arterisk works together ?
Yes. P.S. Someone shoult set this sticky :) Jorge R. Constenla wrote: Hi, Anybody knows if Asterisk work fine with ser ? We are using SER (iptel) for VoIP and we want to use Asterisk for PSTN termination for inbound and outbound calls. Jorge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E100P configuration
Scott, Thanks a lot ! this is exaclty what I wanted. Both my E1's came up without problems. Best regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Tuesday, December 30, 2003 3:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] E100P configuration Hi- Not sure that I understand your question about grouping, but here is what I use for 2 E1's connected to a private switch (in addition to the other parameters) Note that I use the pri_cpe (customer premise equipment) setting. The defined channels act as one big group of 60 channels, if that's what you mean. Your telephone company will define the call distribution for your incoming calls: pridialplan=unknown context=incoming usecallerid=yes group=1 signalling=pri_cpe channel = 1-15,17-31 channel = 32-46,48-62 Regards, Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dawid Mielnik Sent: Tuesday, December 30, 2003 11:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] E100P configuration Hi ! I am trying to configure two E100P cards, but I am a bit confused with zapta.conf in what I am trying to achieve. The * will be connected to a pstn switch with two E1 PRI lines. The E1 lines will be used for incoming calls as well as outgoing calls. My problem now is what to put in zapta.conf, I would like to group all channels from both cards together (if that's possible). Does this make sense ? context=default switchtype=euroisdn signalling=pri_net ;pridialplan=national overlapdial=yes group=1 channel = 1-15,17-31,32-46,48-62 what does channel include ? all the channels d and b ? Thanks for your help. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ast gui client error
do you have the manager interface turned on? You need to make sure your /etc/asterisk/manager.conf file looks something like this: ; ; Asterisk Call Management support ; [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [testuser] secret = test ;deny=0.0.0.0/0.0.0.0 ;permit=192.168.0.1/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user MATT--- -Original Message- From: Chandra [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 2:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ast gui client error ### connect to asterisk manager through telnet $t = new Net::Telnet (Port = 5038, Prompt = '/.*[\$%#] $/', Output_record_separator = '',); #$fh = $t-dump_log(./telnet_log.txt); # uncomment for telnet log $t-open($server_ip); i got error in this line $t-open($server_ip); my ip is 192.168.0.5 for asterisk and everyhings ok. the error i get is [EMAIL PROTECTED] astguiclient]# perl AST_SQL_update_channels.pl problem connecting to 192.168.0.5, port 5038: Connection refused at AST_SQL_update_channels.pl line 73 anyhting to do with port?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phone as intercom
Cool, haven't looked that in depth into the new firmware(is that the 2.4.1 firmware?) I'll have to try that. I'll post your instructions on the Wiki page later today. Thanks, MATT--- -Original Message- From: John Baker [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 3:07 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP phone as intercom Hello, all Sorry to correct you on this Matt, but I am currently doing this with the Polycom 600 phones. You need the latest version of both the SIP software and bootrom to do it, (and that stuff ain't easy to get) but it is workable. The latest version of software provides for distinctive ring tones just like the Cisco 7960's have. It also provides for auto answer. It's kind of tricky to do, but you can make your phones auto answer by setting the Alert-Info variable in asterisk and messing with the xml configuration files, sip.cfg and ipmid.cfg. In the sip.cfg file, look for the line with these variables: alertInfo voIpProt.SIP.alertinfo.1.value=Sales voIpProt.SIP.alertInfo.1.class=8... In this real-world example, whenever I set ALERT_INFO to Sales in Asterisk, the Polycom matches on that word and calls up class 8 in ipmid.cfg. In ipmid.cfg, my class 8 line looks like this: SALES se.rt.8.name=Sales se.rt.8.type=ring se.rt.8.ringer=11 se.rt.8.callWait=6 se.rt.8.mod=0 se.rt.8.type=ring tells the Polycom phone which type of ring to use - which in this case is a regular ring and se.rt.8.ringer=11 tells the phone to ring with ringtone 11 with is the Triplet. I use this one for signaling a new incoming sales call to one of my three sales guys. The secretary transfers it to the sales department and their lines ring with the Triplet. I feel like Pavlov whenever I hear it. The other ring types are visual, answer and ring-answer. The one you want is ring-answer. Here's how I do it: Again in sip.cfg (actually part of the same line listed above) ...voIpProt.SIP.alertinfo.2.value=Ring Answer voIpProt.SIP.alertInfo.2.class=4... and in ipmid.cfg (I just modified one of the existing ones to give me a High Double Trill ringtone) RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000... se.rt.4.ringer=7... The se.rt.4.timeout=1000 tells the Polycom to ring for 1000 milliseconds (one second) and then answer. I call it in Asterisk by setting the ALERT_INFO variable to Ring Answer whenever anybody pushes 8 plus the extension. It rings in to the extension and voila, I'm on speaker! By the way, for all you BOFH out there, you could actually use this feature as a somewhat surreptitious eavesdropping device by using a silent ring and a visual type. The phone would answer without any indication except on the console. I haven't tried this myself and if you do this, I don't want to know about it...unless I'm in your office at the time. Good Luck! I was going to put this in the Wiki myself, but maybe somebody will give me a late Christmas present. --John Baker On Tue, 2003-12-30 at 19:08, mattf wrote: Hello, It's all dependant upon the firmware of the phone(nothing to do with the PBX or SIP currently). The documentation of the Polycom VOIP phones shows no way of doing this currently but it is really just a matter of Polycom adding this feature to their firmware in the future which we are pushing for. People have gotten this to work with Cisco and Snom phones. MATT--- -Original Message- From: Sean Adams [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 6:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP phone as intercom (new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Happy New Year!!
WipeOut wrote: Hi all, Let me be the first to wish everyone, especially the Digium team, an awesome year in 2004.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Same, from me as well. :) Happy new year to everybody, and let * become apache of VOIP!!! Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Happy New Year!!
Senad Jordanovic wrote: WipeOut wrote: Hi all, Let me be the first to wish everyone, especially the Digium team, an awesome year in 2004.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Same, from me as well. :) Happy new year to everybody, and let * become apache of VOIP!!! Ta SJ You mean its not already?? .. ;-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Happy New Year!!
WipeOut wrote: Senad Jordanovic wrote: WipeOut wrote: Hi all, Let me be the first to wish everyone, especially the Digium team, an awesome year in 2004.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Same, from me as well. :) Happy new year to everybody, and let * become apache of VOIP!!! Ta SJ You mean its not already?? .. ;-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Sure it is...but not yet :) Apache has: 1. full documentation 2. huge worldwide user base and presence 3. plenty of add-on software, web based interfaces etc 4. and lastly, people out there know about it. Anyway, a lot of people I spoke to lately did agree that 2004 will be VOIP boom year.. So... Lets see. :) Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Early Dial
On Tue, 30 Dec 2003, Tilghman Lesher wrote: On Tuesday 30 December 2003 22:16, Greg Boehnlein wrote: On Thu, 18 Dec 2003, Aaron Martin wrote: I have upgraded my grandstream phone from firmware 1.0.3.78 to 10.0.4.30 and now I am having problems with early dial. On the older firmware earlydial worked fine with my asterisk server, but now as soon as I have dialed the number I get a congested tone, and the number 4 flashes up on the LCD screen. Has anyone had this problem, and if so, how do I fix it? Early dial has never worked for me, and I just upgraded to the 1.0.4.30 load yesterday. Now, I am having DTMF recognition issues, making it impossible to check my voice mail. As an example, my extension is 100 and let's say my password is 1234. Here is what * captures: -- Executing VoiceMailMain(SIP/damin-3099, ) in new stack -- Playing 'vm-login' (language 'en') NOTICE[5126]: File chan_sip.c, Line 4667 (handle_response): Peer 'damin' is now REACHABLE! -- Playing 'vm-password' (language 'en') -- Incorrect password '111223' for user '11000' (context = any) -- Playing 'vm-incorrect' (language 'en') Not sure what to do, but I'm hoping that my SNOM 200 IP Phone will yield better results. What happens when you change the configuration of the GS phone to send DTMF via SIP INFO? I had that set originally. I get the same behavior no matter wether I use Send via SIP, RTP or INLINE AUDIO. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Early Dial
Greg Boehnlein wrote: On Tue, 30 Dec 2003, Tilghman Lesher wrote: On Tuesday 30 December 2003 22:16, Greg Boehnlein wrote: What happens when you change the configuration of the GS phone to send DTMF via SIP INFO? I had that set originally. I get the same behavior no matter wether I use Send via SIP, RTP or INLINE AUDIO. Make sure you change your dtmfmode= in your sip.conf to match the mode set on the phone.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phone as intercom
I've added it as a separate page: http://www.voip-info.org/wiki-Polycom+auto-answer+config linked from the Polycom phones page. Could you possibly send me a quick line or two(example code) on setting the ALERT_INFO variable in Asterisk? Thanks, MATT--- -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 7:57 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] SIP phone as intercom Cool, haven't looked that in depth into the new firmware(is that the 2.4.1 firmware?) I'll have to try that. I'll post your instructions on the Wiki page later today. Thanks, MATT--- -Original Message- From: John Baker [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 3:07 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP phone as intercom Hello, all Sorry to correct you on this Matt, but I am currently doing this with the Polycom 600 phones. You need the latest version of both the SIP software and bootrom to do it, (and that stuff ain't easy to get) but it is workable. The latest version of software provides for distinctive ring tones just like the Cisco 7960's have. It also provides for auto answer. It's kind of tricky to do, but you can make your phones auto answer by setting the Alert-Info variable in asterisk and messing with the xml configuration files, sip.cfg and ipmid.cfg. In the sip.cfg file, look for the line with these variables: alertInfo voIpProt.SIP.alertinfo.1.value=Sales voIpProt.SIP.alertInfo.1.class=8... In this real-world example, whenever I set ALERT_INFO to Sales in Asterisk, the Polycom matches on that word and calls up class 8 in ipmid.cfg. In ipmid.cfg, my class 8 line looks like this: SALES se.rt.8.name=Sales se.rt.8.type=ring se.rt.8.ringer=11 se.rt.8.callWait=6 se.rt.8.mod=0 se.rt.8.type=ring tells the Polycom phone which type of ring to use - which in this case is a regular ring and se.rt.8.ringer=11 tells the phone to ring with ringtone 11 with is the Triplet. I use this one for signaling a new incoming sales call to one of my three sales guys. The secretary transfers it to the sales department and their lines ring with the Triplet. I feel like Pavlov whenever I hear it. The other ring types are visual, answer and ring-answer. The one you want is ring-answer. Here's how I do it: Again in sip.cfg (actually part of the same line listed above) ...voIpProt.SIP.alertinfo.2.value=Ring Answer voIpProt.SIP.alertInfo.2.class=4... and in ipmid.cfg (I just modified one of the existing ones to give me a High Double Trill ringtone) RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000... se.rt.4.ringer=7... The se.rt.4.timeout=1000 tells the Polycom to ring for 1000 milliseconds (one second) and then answer. I call it in Asterisk by setting the ALERT_INFO variable to Ring Answer whenever anybody pushes 8 plus the extension. It rings in to the extension and voila, I'm on speaker! By the way, for all you BOFH out there, you could actually use this feature as a somewhat surreptitious eavesdropping device by using a silent ring and a visual type. The phone would answer without any indication except on the console. I haven't tried this myself and if you do this, I don't want to know about it...unless I'm in your office at the time. Good Luck! I was going to put this in the Wiki myself, but maybe somebody will give me a late Christmas present. --John Baker On Tue, 2003-12-30 at 19:08, mattf wrote: Hello, It's all dependant upon the firmware of the phone(nothing to do with the PBX or SIP currently). The documentation of the Polycom VOIP phones shows no way of doing this currently but it is really just a matter of Polycom adding this feature to their firmware in the future which we are pushing for. People have gotten this to work with Cisco and Snom phones. MATT--- -Original Message- From: Sean Adams [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 6:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP phone as intercom (new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Programming an unlocked ADSI phone?
When you push your services button, are there any available slots. If not, delete the 3rd or 4th ones Don't know if it matters, but I have: Comedian Mail Asterisk PBX available Aastra SL The only selectable ones are the Comedian Mail and the Asterisk PBX. You might try deleting them all and reloading. And # is the Transfer button. 8-) Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 9:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Update #2 According to the helpful folks at Sayson, this phone actually has two 'slots', and the ADSI stuff needs to be downloaded to both (the second one kicks in when the phone is idle for more than 1s). I had only downloaded asterisk.adsi to the first slot, which explains why I was seeing the default (PLEASE PROGRAM ME) triggered from slot 2. I downloaded to the second slot and all is well now ... at least the phone now identifies itself as I asked it to, and seems to be programmed ;-) It seems to me that asterisk.adsi is pretty vanilla - not much functionality. Has anyone built a better mousetrap that they'd care to share? My second problem remains ... when I connect to voicemail now I see: Comedian Mail download refused Services is full but I don't see any errors in Asterisk's console. What's up with that??? I am now seeing some really cool functionality once I get past this point. As I'm checking voicemail by navigating the menus, I see stuff like: Old Messages Message 1 of 5 Unknown Tue Dec 23 07:46:02 Sweet! -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Darren Nickerson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 4:18 PM Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Thanks for the reply Tim - I was beginning to think nobody used this stuff ;-) As you can tell, I'm a relative newcomer to ADSI - I'm really not sure what to expect once the phone gets programmed, but I would not expect to see (PLEASE PROGRAM ME) still, and I would have hoped it would not have broken voicemail so readily. I'm not using a channel bank at all ... I have a very simple setup using two x100p FXO cards and one TDM400P FXS card. As I mentioned below it does appear that SOMETHING was loaded into the phone, and it does appear to at least TRY to use ADSI when accessing voicemail. It's odd ... it's like everything worked but I'm left saying ... okay, now what? The phone isn't incredibly functional at this point - even if I do go into the services menu and select 'Asterisk PBX' this selection only persists until I use the phone once. Also, there aren't soft keys for anything useful like transferring a call ... how WOULD one do that with this phone anyway? -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tim Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 3:48 PM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? What kind of channelbank/FXS port are you connecting to? I've seen problems connecting to some of the older versions of the Adtran Total Access 750's. I wouldn't doubt there would be problems on other channelbanks with older firmwares. Of course, no firmware on CAC AB1's I have the AAstra 480, Adtran 750 Channelbank (updated firmware), T100P card, and it worked fine on the first try with current CVS. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Monday, December 29, 2003 10:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Update. As I mentioned in my first post (below), I had managed to get the phone to accept a download from asterisk, but it still said PLEASE PROGRAM ME after the download was completed. After further investigation it does appear I downloaded SOMETHING to the phone, because if I select the services button on the front of the phone, I get into a menu that says: Services Asterisk PBX Asterisk slot 2 available available If I select the first one in the list, the phone does change from PLEASE PROGRAM ME to ** Asterisk PBX** and there's a VMail softkey!! However, if I pick up the handset and replace it, I'm back to PLEASE PROGRAM ME again. Is this normal? My second problem is that when I dial voicemail from this handset now, which is ADSI-enabled, I see the following message on
Re: [Asterisk-Users] A Head Check
Greg, I have been retained by a Building Management Company to install a combined Voice/Data solution for a Tennated Office Space. This space will rent offices, with telephone and internet service to inviduals or small snip Now.. This is our first deployment of Asterisk, and I need a head check here. Am I making the right decision? :) Sepcifically... 1. Are the SNOM 200 IP phones a good choice for standard users? Or should I consider Cisco? Price of the phone is not the important thing.. What is important is ease of use with minimal training and reliability! I'd be careful with assumptions on the Snom 200. I've been trying to properly define two extns on this phone (with visual indication as to which extn is ringing) and have not been successful as yet. Have an open problem with snom right now. Same issues with v2.02t and v2.03e. Not sure what the problem is as yet, but three symptoms are highly visible: a. when a second (or more) extn is defined to the second (or more) button, the phone goes into a loop involving Register, 100 Trying, and 407 Proxy Authentication Required. Generates 1,000's of never- ending packets. (Two different snom support people are trying to replicate the issue, and both have initially been finger-pointing towards asterisk. Too early to know where the issue is. All other SIP phones function as expected with multiple extns.) b. when multiple lines are defined (with Key Mapping as suggested by the snom folks), the LED's for those keys remain lite at all times. No way to know which extension is actually ringing. c. distinctive ringing on a per-extn basis is apparently broken. Hopefully will know more about these issues by the end of this week. As with many other SIP components, documentation is below industry standards. The phone works fine with a single extn definition. Given the business environment you're talking about, there is a very high probability your customers will want two or more lines per phone. Bottom line: no way to visually see which extn is ringing, and therefore no way to answer the phone with a prearranged business greating. I've had 100% solid success/luck using the C7960 v6.0 code with lots of not-so-common addon funtions that may have value-add implications in your proposed implementation. (My past 20-year experience working for a telephone company and full understanding of shared-tennat services, I'd have to go with the Cisco phones if the decision had to be made today.) I've used the snom 200 for the better part of two months with several versions of their software. Snom seems to have a software quality control issue. It's likely due to lacking a structured software test plan, but don't know that for sure. I'd also be careful with assumptions regarding plugging PC's into the RJ45 switch jack on the back of various phones. There seems to be several unusual symptoms that have appeared on this list and I'm not sure which are still open (verses knowledge/skill level to identify the root-cause and associated resolutions). Given the relatively small percentage of folks using the jack, the only valid assumption is don't count on it in production. 3. I am also responsible for delivering inbound faxes to the DID numbers via Email. I.E. customer has a document faxed to them and they get it in Email as a tiff. I'm considering using Hylfax with a Multitech DID capable modem, but other suggestions are welcomed! Be careful with fax assumptions in terms of routing analog fax calls through T1's and asterisk, etc. It's certainly doable; just don't make any assumptions before hand. (Read: may require some additional implementation and/or testing hours to obtain exactly what you want from reliability perspective.) 4. I have built some cost for support from Digium and/or other Asterisk experts into the budget. Does Digium have paid support plans? What about other consultants out there? Their web site mentions such support. However, since asterisk has not matured to the point of supporting stable software releases (etc), support from a high-availability business perspective will require more then a Digium support contract (i.e, no published 24x7 plan today). (There are several people lurking on this list that can offer remote support. A contract with service-level penalities is probably worthy of consideration.) I'm just trying to make sure that I cover all the bases. This is got to be a bulletproof solution, and I'm departing from my comfort level with Altigen to give Asterisk a run for the money. We've got TONS of Linux experience here, and comfort with customizing code, so I am happy with what Asterisk gives me.. What else should I be worried about? I'd strongly recommend implementing two asterisk boxes on site; one as a primary production box and the second as a hot-spare identical backup (or some such combination). Obviously, the hot-spare backup could also be used to
[Asterisk-Users] grand stream phone and double nat
Hi I'm trying to configur a grandstream BT101 to connect to asterisk, both behind different NATs, I realise that a double Nat is a problem, I have tried using fwd forwarding to iaxtel as a solution but cannt seem to get them to connect as I think there is a codec problem as IAXTEL doesn't seem to accept alawor ulaw is this correct? Has anyone been able to connect a sip phone across a double NAT ? I realise this has been discussed before and sorry for that Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone, ideas for incoming call management for CRM system
Hi, we have implemented a first version of call support from a web based system for Asterisk (via the manager interface) and other, callto: and tel:, providers. Now I am looking at the other way around. If a call comes in, I want our web based system to automatically detect the number and present the call information to the user. Ideas anyone? I guess, I won't be able to get this done without some client specific programming, will I? All the best for 2004. Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programming an unlocked ADSI phone?
Yes, there are 2 available slots. When I push services, I see: Services Asterisk PBX Asterisk PBX available available As I mentioned, the only soft-key I seem to have is 'VMail', and when I select that I still get: Comedial Mail download refused Services is full And yet somehow there does seem to be some ADSI functionality later on, when it falls back to a voice prompt after the download is refused and I'm navigating the voice prompts, reading the messages. It troubles me that Asterisk isn't very chatty about the ADSI stuff ... when I access VMail all I see is: -- Starting simple switch on 'Zap/5-1' -- Executing VoiceMailMain(Zap/5-1, ) in new stack (here's where the ADSI stuff seems to happen on the phone, download refused etc) -- Playing 'vm-login' (language 'en') [snip rest of log] There's no indication that any ADSI transactions are going on here, but that tell-tale tone can be heard and the little rotating animated cursor on the phone means _something_ is definitely going on at the ADSI level. Am I missing a debug option that could show me more about what may be going wrong? I'll get with Sayson tech support today to see if they can make any sense of this. -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tim Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 9:59 AM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? When you push your services button, are there any available slots. If not, delete the 3rd or 4th ones Don't know if it matters, but I have: Comedian Mail Asterisk PBX available Aastra SL The only selectable ones are the Comedian Mail and the Asterisk PBX. You might try deleting them all and reloading. And # is the Transfer button. 8-) Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 9:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Update #2 According to the helpful folks at Sayson, this phone actually has two 'slots', and the ADSI stuff needs to be downloaded to both (the second one kicks in when the phone is idle for more than 1s). I had only downloaded asterisk.adsi to the first slot, which explains why I was seeing the default (PLEASE PROGRAM ME) triggered from slot 2. I downloaded to the second slot and all is well now ... at least the phone now identifies itself as I asked it to, and seems to be programmed ;-) It seems to me that asterisk.adsi is pretty vanilla - not much functionality. Has anyone built a better mousetrap that they'd care to share? My second problem remains ... when I connect to voicemail now I see: Comedian Mail download refused Services is full but I don't see any errors in Asterisk's console. What's up with that??? I am now seeing some really cool functionality once I get past this point. As I'm checking voicemail by navigating the menus, I see stuff like: Old Messages Message 1 of 5 Unknown Tue Dec 23 07:46:02 Sweet! -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Darren Nickerson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 4:18 PM Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Thanks for the reply Tim - I was beginning to think nobody used this stuff ;-) As you can tell, I'm a relative newcomer to ADSI - I'm really not sure what to expect once the phone gets programmed, but I would not expect to see (PLEASE PROGRAM ME) still, and I would have hoped it would not have broken voicemail so readily. I'm not using a channel bank at all ... I have a very simple setup using two x100p FXO cards and one TDM400P FXS card. As I mentioned below it does appear that SOMETHING was loaded into the phone, and it does appear to at least TRY to use ADSI when accessing voicemail. It's odd ... it's like everything worked but I'm left saying ... okay, now what? The phone isn't incredibly functional at this point - even if I do go into the services menu and select 'Asterisk PBX' this selection only persists until I use the phone once. Also, there aren't soft keys for anything useful like transferring a call ... how WOULD one do that with this phone anyway? -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tim Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 3:48 PM Subject:
Re: [Asterisk-Users] 7960 Register with 2 * Servers causes phone to reboot over and over
Justin, I have been trying to get my 7960 7960G to register with two seperate * servers. Asterisk box 1 is on out on the Internet running: Asterisk CVS-10/30/03-20:08:15 Asterisk box 2 is on the Lan running: Asterisk CVS-12/19/03-19:28:30 7960 is on the LAN running: P0S3-04-4-00 7960G is on the LAN running: P0S3-06-0-00 In sip.conf I have nat=yes to get the phones to register properly. And for a while both phones do actually work. However about every few mins they just restart! If I remove the second line that is registering with the server on the LAN they stop restarting Ideas? I recall this being mentioned in the past, but can't remember what the source was. Seems to me it was a Cisco bug back in the v3/v5 days. I'm running v6 code now but don't have two * systems to try it. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone, ideas for incoming call management for CRM system
Peer Oliver schmidt wrote: Hi, we have implemented a first version of call support from a web based system for Asterisk (via the manager interface) and other, callto: and tel:, providers. Now I am looking at the other way around. If a call comes in, I want our web based system to automatically detect the number and present the call information to the user. Ideas anyone? I guess, I won't be able to get this done without some client specific programming, will I? All the best for 2004. Best regards Peer Oliver Schmidt the internet company Your problem is that as you mentioned your app is web based and web based apps don't maintain a connection to the server, once the page is loaded the connection is terminated and so there is no way for the server to then send new data to the web browser when the call comes in.. you would have to somehow get the server to identify the call and then get the client to reload the web page in the browser.. This could have isues.. eg if the user is on a call and working on the web page and a second call comes in you don't want the page to be refreshed.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programming an unlocked ADSI phone?
And # is the Transfer button. 8-) Any way to map that to a soft button? How do I use # in a call if not? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] A Head Check
The cisco 7960 phone works great. Reliable and fully functional. The even have support, sort of. In a real business environment I couldn't imagine using anything else right now. The extra money you spend will be paid back immediately on the service calls you won't have to make. I have a bunch in service, the users love them. Why would you build your own computer? You can't beat something like a DELL server, especially a refurbished server. I just bought a refurbished 3.2GHz Dell server, 800MHz front side bus, with 128MB of memory and 60MB of disk for $598. If it breaks, Dell comes out and fixes it. Again, why try and save a few bucks at the expense of reliability? Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Wednesday, December 31, 2003 5:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] A Head Check Greg, I have been retained by a Building Management Company to install a combined Voice/Data solution for a Tennated Office Space. This space will rent offices, with telephone and internet service to inviduals or small snip Now.. This is our first deployment of Asterisk, and I need a head check here. Am I making the right decision? :) Sepcifically... 1. Are the SNOM 200 IP phones a good choice for standard users? Or should I consider Cisco? Price of the phone is not the important thing.. What is important is ease of use with minimal training and reliability! I'd be careful with assumptions on the Snom 200. I've been trying to properly define two extns on this phone (with visual indication as to which extn is ringing) and have not been successful as yet. Have an open problem with snom right now. Same issues with v2.02t and v2.03e. Not sure what the problem is as yet, but three symptoms are highly visible: a. when a second (or more) extn is defined to the second (or more) button, the phone goes into a loop involving Register, 100 Trying, and 407 Proxy Authentication Required. Generates 1,000's of never- ending packets. (Two different snom support people are trying to replicate the issue, and both have initially been finger-pointing towards asterisk. Too early to know where the issue is. All other SIP phones function as expected with multiple extns.) b. when multiple lines are defined (with Key Mapping as suggested by the snom folks), the LED's for those keys remain lite at all times. No way to know which extension is actually ringing. c. distinctive ringing on a per-extn basis is apparently broken. Hopefully will know more about these issues by the end of this week. As with many other SIP components, documentation is below industry standards. The phone works fine with a single extn definition. Given the business environment you're talking about, there is a very high probability your customers will want two or more lines per phone. Bottom line: no way to visually see which extn is ringing, and therefore no way to answer the phone with a prearranged business greating. I've had 100% solid success/luck using the C7960 v6.0 code with lots of not-so-common addon funtions that may have value-add implications in your proposed implementation. (My past 20-year experience working for a telephone company and full understanding of shared-tennat services, I'd have to go with the Cisco phones if the decision had to be made today.) I've used the snom 200 for the better part of two months with several versions of their software. Snom seems to have a software quality control issue. It's likely due to lacking a structured software test plan, but don't know that for sure. I'd also be careful with assumptions regarding plugging PC's into the RJ45 switch jack on the back of various phones. There seems to be several unusual symptoms that have appeared on this list and I'm not sure which are still open (verses knowledge/skill level to identify the root-cause and associated resolutions). Given the relatively small percentage of folks using the jack, the only valid assumption is don't count on it in production. 3. I am also responsible for delivering inbound faxes to the DID numbers via Email. I.E. customer has a document faxed to them and they get it in Email as a tiff. I'm considering using Hylfax with a Multitech DID capable modem, but other suggestions are welcomed! Be careful with fax assumptions in terms of routing analog fax calls through T1's and asterisk, etc. It's certainly doable; just don't make any assumptions before hand. (Read: may require some additional implementation and/or testing hours to obtain exactly what you want from reliability perspective.) 4. I have built some cost for support from Digium and/or other Asterisk experts into the budget. Does Digium have paid support plans? What about other consultants out there? Their web site mentions such support.
[Asterisk-Users] Current database abstraction effort ?
Dear all, I read across Asterisk's lists archives, and found out various discussions about how nice it would be to have a (SQL) database abstraction layer enabling the use of various SQL backends, for various purposes inside of Asterisk. As far as I see, there is no such thing yet, although there are various efforts (on CDR and voicemail, mostly) to use external databases. As I'm planning some work that will require external database support (in order to be fully dynamic and potentially shared by different Asterisk servers), I have to solve this issue first. It's definetly not a major issue, but before starting to work on it I'd like to know what the community has to say about it. I found a few references to the way the FreeRadius people did it, and I would probably make something similar, although there a a few things in their design that I would make otherwise. Finally, Asterisk uses the db1 library to store some local dynamic data. The db.h interface could be kept as-is and optionnally work with this new database backend for tasks that do not require full SQL syntax (although a there shall be a way to specify if the caller is expecting its data to be local or shared amongst servers). -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP phone as intercom
Wow! Thanks John for the detailed information. This is such an awesome system... and great support here, too. On Dec 31, 2003, at 12:07 AM, John Baker wrote: Hello, all Sorry to correct you on this Matt, but I am currently doing this with the Polycom 600 phones. You need the latest version of both the SIP software and bootrom to do it, (and that stuff ain't easy to get) but it is workable. The latest version of software provides for distinctive ring tones just like the Cisco 7960's have. It also provides for auto answer. It's kind of tricky to do, but you can make your phones auto answer by setting the Alert-Info variable in asterisk and messing with the xml configuration files, sip.cfg and ipmid.cfg. In the sip.cfg file, look for the line with these variables: alertInfo voIpProt.SIP.alertinfo.1.value=Sales voIpProt.SIP.alertInfo.1.class=8... In this real-world example, whenever I set ALERT_INFO to Sales in Asterisk, the Polycom matches on that word and calls up class 8 in ipmid.cfg. In ipmid.cfg, my class 8 line looks like this: SALES se.rt.8.name=Sales se.rt.8.type=ring se.rt.8.ringer=11 se.rt.8.callWait=6 se.rt.8.mod=0 se.rt.8.type=ring tells the Polycom phone which type of ring to use - which in this case is a regular ring and se.rt.8.ringer=11 tells the phone to ring with ringtone 11 with is the Triplet. I use this one for signaling a new incoming sales call to one of my three sales guys. The secretary transfers it to the sales department and their lines ring with the Triplet. I feel like Pavlov whenever I hear it. The other ring types are visual, answer and ring-answer. The one you want is ring-answer. Here's how I do it: Again in sip.cfg (actually part of the same line listed above) ...voIpProt.SIP.alertinfo.2.value=Ring Answer voIpProt.SIP.alertInfo.2.class=4... and in ipmid.cfg (I just modified one of the existing ones to give me a High Double Trill ringtone) RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer se.rt.4.timeout=1000... se.rt.4.ringer=7... The se.rt.4.timeout=1000 tells the Polycom to ring for 1000 milliseconds (one second) and then answer. I call it in Asterisk by setting the ALERT_INFO variable to Ring Answer whenever anybody pushes 8 plus the extension. It rings in to the extension and voila, I'm on speaker! By the way, for all you BOFH out there, you could actually use this feature as a somewhat surreptitious eavesdropping device by using a silent ring and a visual type. The phone would answer without any indication except on the console. I haven't tried this myself and if you do this, I don't want to know about it...unless I'm in your office at the time. Good Luck! I was going to put this in the Wiki myself, but maybe somebody will give me a late Christmas present. --John Baker On Tue, 2003-12-30 at 19:08, mattf wrote: Hello, It's all dependant upon the firmware of the phone(nothing to do with the PBX or SIP currently). The documentation of the Polycom VOIP phones shows no way of doing this currently but it is really just a matter of Polycom adding this feature to their firmware in the future which we are pushing for. People have gotten this to work with Cisco and Snom phones. MATT--- -Original Message- From: Sean Adams [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 6:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP phone as intercom (new asterisk user - currently setting up Polycom IP600 phones) Does anyone know if it's possible to make a sip phone instantly pick up on speakerphone when a particular call comes in? Eg so that you can quickly bother someone across the office without making them reach for their phone? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 Register with 2 * Servers causes phone to reboot over and over
Strange.. I have my 7960 registered with 3 diffrent asterisk servers... its ROCK SOLID!! version 6 firmware. bkw On Wed, 31 Dec 2003, Rich Adamson wrote: Justin, I have been trying to get my 7960 7960G to register with two seperate * servers. Asterisk box 1 is on out on the Internet running: Asterisk CVS-10/30/03-20:08:15 Asterisk box 2 is on the Lan running: Asterisk CVS-12/19/03-19:28:30 7960 is on the LAN running: P0S3-04-4-00 7960G is on the LAN running: P0S3-06-0-00 In sip.conf I have nat=yes to get the phones to register properly. And for a while both phones do actually work. However about every few mins they just restart! If I remove the second line that is registering with the server on the LAN they stop restarting Ideas? I recall this being mentioned in the past, but can't remember what the source was. Seems to me it was a Cisco bug back in the v3/v5 days. I'm running v6 code now but don't have two * systems to try it. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Happy New Year!!
Happy New Year Wipeout and all the other Asterisk Users around the globe from the Shore Linux Solutions Team Special kudos and a Happy New Year to the Digium Team P.S. Wipeout in 2004 are you going to do a Fedora Howto like your RH9 Howto of 2003? AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Happy New Year!!
P.S. Wipeout in 2004 are you going to do a Fedora Howto like your RH9 Howto of 2003? Why would you need a howto in the first place? I never did! :P bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Happy New Year!!
Quite frankly, I don't need a howto, I have it running on my Fedora core 1 system as well as my RH 9 system. I just thought it might be a good idea for newbies or other people not very familiar with Linux or asterisk, needed packages, dependencies, etc. Personally I think thorough documentation at all levels are a good thing as it often gives one an option before asking high minded pricks for assistance, advice or answers to questions. Mr. West I've been called arrogant, egotistical, self centered and even an asshole but I think i have enough sense to realize that not everyone is on the same level just because you didn't need a howto guide doesn't mean others don't or won't. By the way, two closing points. First of all someone recommended you to me the other day. Termed you as a competent asterisk consultant. I must say at this point, my opinion has been clouded by your lack of social ettiquette. Secondly, that bottom portion of the mail was intended for Wipeout. AJ On Wed, 31 Dec 2003, Brian West wrote: P.S. Wipeout in 2004 are you going to do a Fedora Howto like your RH9 Howto of 2003? Why would you need a howto in the first place? I never did! :P bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] after hours - is this logic ok ?
Ok, first off, Asterisk is the coolest piece of software I have EVER had
the pleasure of using in my 8 years of running linux !! and I know I
haven't even scratched the surface feature wise.
Before I get too excited, I wanted to get all you experts to look at the
how I implemented my after hours test. The goal is to prevent the phone
from ringing afer certain hours, just go to VM.
Bacically, when a call comes from the PSTN, I use these includes to
either set a key in the DB or not.
include = day|8:00-21:00|mon-fri|*|*
include = day|9:00-21:00|sat-sun|*|*
; if we're not open, we're closed
include = night
[day]
exten = s,2,DBput(FEATURE/DAY=yes)
exten = s,3,Goto(s,10)
[night]
exten = s,2,NoOp
exten = s,3,Goto(s,10)
And then in my stdexten macro, I test for the existence of the key in
the database. If the key exists, it must be daytime so delete the key
and allow the calls to ring the extension. If the key does not exist,
it must be night and since we don't want the phones to ring we jump to
the unavailable VM. I've tested this and it works as I expect but my
only concern is how this would hold up in a busy environment where many
calls are being processed. Could one asterisk thread delete the
database key before another thread had gotton the oportunity to test for
the key ?
[macro-stdexten]
exten = s,1,NoOp
other testing crap deleted
exten = s,10,DBget(foo=FEATURE/DAY); is it day time ?
exten = s,11,DBdel(FEATURE/DAY); yes, delete the key
exten = s,12,Goto(s,201) ; and ring the phone
exten = s,111,Goto(s,204) ; no, goto uVM
exten = s,201,answer
exten = s,202,Playback(transfer,skip)
exten = s,203,Dial(${ARG2},5) ; Ring the interface, 20 seconds
maximum
exten = s,204,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
exten = s,205,Playback(vm-goodbye)
exten = s,206,Wait(1)
exten = s,207,Hangup
exten = s,304,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
exten = s,305,Playback(vm-goodbye)
exten = s,306,Wait(1)
exten = s,307,Hangup
--
.~.
/V\Lance C. Arbuckle
// \\
/( )\
^'~'^
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Grandstream Early Dial
I've just checked my voicemail with 1.0.4.30 and get the same multiple digits problem. sip.conf and GS config are both at info, for me this is a new problem voicemail has always worked perfectly with the GS. This has come up many times in this list, with no consensus for a solution. According to Grandstream, the multiple digit problem arises from a difference in the interpretation of the SIP standard. I'm not sure I really understand this, so no flames please, but, paraphrasing a conversation I had with GS, apparently they retransmit the digit as long as the key is pressed and expect asterisk to know that it is a re-transmission by examining other data in the packet. Asterisk does not handle the SIP packet in the way GS expects, resulting in multiple digit transmission. This flaw (?) is avoided by setting DTMF to INBAND. Why this behaviour is not repeatable on everyones installations escapes me. However, I have noticed one thing that may be a clue. I have one phone that is older hardware (redial button instead of send and an unused battery compartment on the bottom). This phone behaves differently than all the other, later, models. For example, it is the only phone on which the flash button actually works to answer the alternate line (eg when an incoming call waiting call arrives). All phones are using 3.81 firmware. Early dial has never worked for me, and I just upgraded to the 1.0.4.30 load yesterday. Now, I am having DTMF recognition issues, making it impossible to check my voice mail. This is an acknowleged bug on the GS. They have connected to my * server and acknowleged the problem. A fix has been promised but not yet delivered. Until then, the only solution is to turn early dial off and let the phone send the entire dial string in one packet. Since this does not affect later single digit transmission for IVR's, etc, the only consequence is the irritating delay between the last entered digit and the actual placing of the call. But, you can always hit the send key. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Happy New Year!!
Where can I find that Howto? I'm new to Asterisk and am looking for all the doc I can find. TIA, Eric On Wednesday 31 December 2003 12:29, [EMAIL PROTECTED] wrote: Happy New Year Wipeout and all the other Asterisk Users around the globe from the Shore Linux Solutions Team Special kudos and a Happy New Year to the Digium Team P.S. Wipeout in 2004 are you going to do a Fedora Howto like your RH9 Howto of 2003? AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Happy New Year!!
Hi all! Mr. West I've been called arrogant, egotistical, self centered and even an asshole but I think i have enough sense to realize that ... Heat up that flame, turn it into a nice firework and celebrate - 2004 may be closer than you think! :-)) Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Happy New Year!!
Where can I find that Howto? I'm new to Asterisk and am looking for all the doc I can find. TIA, Eric Eric, You will find at at: http://members.lycos.co.uk/wipe_out/asterisk/ Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Programming an unlocked ADSI phone?
Have you tried deleting those services and adding them back in by checking voicemail 1st (dial 8500) then dial your 8600 for AdsiProg You should be able to just hit the # during the call, but you will also have to make sure you have the |Tt defined in your extensions.conf file as well. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 10:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Yes, there are 2 available slots. When I push services, I see: Services Asterisk PBX Asterisk PBX available available As I mentioned, the only soft-key I seem to have is 'VMail', and when I select that I still get: Comedial Mail download refused Services is full And yet somehow there does seem to be some ADSI functionality later on, when it falls back to a voice prompt after the download is refused and I'm navigating the voice prompts, reading the messages. It troubles me that Asterisk isn't very chatty about the ADSI stuff ... when I access VMail all I see is: -- Starting simple switch on 'Zap/5-1' -- Executing VoiceMailMain(Zap/5-1, ) in new stack (here's where the ADSI stuff seems to happen on the phone, download refused etc) -- Playing 'vm-login' (language 'en') [snip rest of log] There's no indication that any ADSI transactions are going on here, but that tell-tale tone can be heard and the little rotating animated cursor on the phone means _something_ is definitely going on at the ADSI level. Am I missing a debug option that could show me more about what may be going wrong? I'll get with Sayson tech support today to see if they can make any sense of this. -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tim Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 9:59 AM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? When you push your services button, are there any available slots. If not, delete the 3rd or 4th ones Don't know if it matters, but I have: Comedian Mail Asterisk PBX available Aastra SL The only selectable ones are the Comedian Mail and the Asterisk PBX. You might try deleting them all and reloading. And # is the Transfer button. 8-) Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 9:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Update #2 According to the helpful folks at Sayson, this phone actually has two 'slots', and the ADSI stuff needs to be downloaded to both (the second one kicks in when the phone is idle for more than 1s). I had only downloaded asterisk.adsi to the first slot, which explains why I was seeing the default (PLEASE PROGRAM ME) triggered from slot 2. I downloaded to the second slot and all is well now ... at least the phone now identifies itself as I asked it to, and seems to be programmed ;-) It seems to me that asterisk.adsi is pretty vanilla - not much functionality. Has anyone built a better mousetrap that they'd care to share? My second problem remains ... when I connect to voicemail now I see: Comedian Mail download refused Services is full but I don't see any errors in Asterisk's console. What's up with that??? I am now seeing some really cool functionality once I get past this point. As I'm checking voicemail by navigating the menus, I see stuff like: Old Messages Message 1 of 5 Unknown Tue Dec 23 07:46:02 Sweet! -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Darren Nickerson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 4:18 PM Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Thanks for the reply Tim - I was beginning to think nobody used this stuff ;-) As you can tell, I'm a relative newcomer to ADSI - I'm really not sure what to expect once the phone gets programmed, but I would not expect to see (PLEASE PROGRAM ME) still, and I would have hoped it would not have broken voicemail so readily. I'm not using a channel bank at all ... I have a very simple setup using two x100p FXO cards and one TDM400P FXS card. As I mentioned below it does appear that SOMETHING was loaded into the phone, and it does appear to at least TRY to use ADSI when accessing voicemail. It's odd ... it's like everything worked but I'm left saying ... okay, now what? The phone isn't incredibly
[Asterisk-Users] Asterisk Web Dialer
I am putting together a solution that will employ the Digium TE410P with one T1 going out the PSTN and the other front-ending a PBX. The idea is that based on a URL, Asterisk will dial an employee behind the PBX. When the employee picks up, Asterisk will dial the customer (detailed in the URL). I am assuming Asterisk can work with Apache (through AGI maybe) to dial the employee and then connect to the customer via info in the URL (or related through some sort of DB lookup). Another requirement will be to record the phone call as well. I have worked a bit with Asterisk and am very happy with what it can do -- and would prefer to stay with Asterisk. The question is, can Asterisk handle what my requirements are or would this better be served by Bayonne? -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax: (407)682-3455 Cell: (321)229-5311 [EMAIL PROTECTED] www.adstelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programming an unlocked ADSI phone?
Update #3. Sayson tell me that this is likely a result of Comedian Mail not knowing anything about the phone's slots or lock codes, causing the download to fail. When I was programming the slots originally, I needed to change the following information in asterisk.: SECURITY _AST; Security code FDN 0x000f ; Descriptor number to information Sayson provided me for each of the slots (1 2 of 4 available). Slot 1 is activated by telephony events, and slot 2 is a self-loading slot that activates when the phone is idle for more than 1 second. I downloaded asterisk.adsi to each slot. Looking at app_voicemail.c there's clearly a lot of ADSI intelligence, ... but it's not clear to me how one configures Comedian Mail to know about a 'slot' Descriptor number and Security Code. -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Darren Nickerson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 11:01 AM Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Yes, there are 2 available slots. When I push services, I see: Services Asterisk PBX Asterisk PBX available available As I mentioned, the only soft-key I seem to have is 'VMail', and when I select that I still get: Comedial Mail download refused Services is full And yet somehow there does seem to be some ADSI functionality later on, when it falls back to a voice prompt after the download is refused and I'm navigating the voice prompts, reading the messages. It troubles me that Asterisk isn't very chatty about the ADSI stuff ... when I access VMail all I see is: -- Starting simple switch on 'Zap/5-1' -- Executing VoiceMailMain(Zap/5-1, ) in new stack (here's where the ADSI stuff seems to happen on the phone, download refused etc) -- Playing 'vm-login' (language 'en') [snip rest of log] There's no indication that any ADSI transactions are going on here, but that tell-tale tone can be heard and the little rotating animated cursor on the phone means _something_ is definitely going on at the ADSI level. Am I missing a debug option that could show me more about what may be going wrong? I'll get with Sayson tech support today to see if they can make any sense of this. -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tim Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 9:59 AM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? When you push your services button, are there any available slots. If not, delete the 3rd or 4th ones Don't know if it matters, but I have: Comedian Mail Asterisk PBX available Aastra SL The only selectable ones are the Comedian Mail and the Asterisk PBX. You might try deleting them all and reloading. And # is the Transfer button. 8-) Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 9:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Update #2 According to the helpful folks at Sayson, this phone actually has two 'slots', and the ADSI stuff needs to be downloaded to both (the second one kicks in when the phone is idle for more than 1s). I had only downloaded asterisk.adsi to the first slot, which explains why I was seeing the default (PLEASE PROGRAM ME) triggered from slot 2. I downloaded to the second slot and all is well now ... at least the phone now identifies itself as I asked it to, and seems to be programmed ;-) It seems to me that asterisk.adsi is pretty vanilla - not much functionality. Has anyone built a better mousetrap that they'd care to share? My second problem remains ... when I connect to voicemail now I see: Comedian Mail download refused Services is full but I don't see any errors in Asterisk's console. What's up with that??? I am now seeing some really cool functionality once I get past this point. As I'm checking voicemail by navigating the menus, I see stuff like: Old Messages Message 1 of 5 Unknown Tue Dec 23 07:46:02 Sweet! -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Darren Nickerson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 4:18 PM Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone?
RE: [Asterisk-Users] Programming an unlocked ADSI phone?
After looking a litter deeper in the code, it looks as though the Comedian Mail will only load in the 1st slot of your ADSI phone and the asterisk loads in the 2nd. Soif you delete the 1st slot that you had listed as Asterisk, then dial the 8500 extension it will hopefully work. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Tim Thompson Sent: Wednesday, December 31, 2003 1:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? Have you tried deleting those services and adding them back in by checking voicemail 1st (dial 8500) then dial your 8600 for AdsiProg You should be able to just hit the # during the call, but you will also have to make sure you have the |Tt defined in your extensions.conf file as well. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 10:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Yes, there are 2 available slots. When I push services, I see: Services Asterisk PBX Asterisk PBX available available As I mentioned, the only soft-key I seem to have is 'VMail', and when I select that I still get: Comedial Mail download refused Services is full And yet somehow there does seem to be some ADSI functionality later on, when it falls back to a voice prompt after the download is refused and I'm navigating the voice prompts, reading the messages. It troubles me that Asterisk isn't very chatty about the ADSI stuff ... when I access VMail all I see is: -- Starting simple switch on 'Zap/5-1' -- Executing VoiceMailMain(Zap/5-1, ) in new stack (here's where the ADSI stuff seems to happen on the phone, download refused etc) -- Playing 'vm-login' (language 'en') [snip rest of log] There's no indication that any ADSI transactions are going on here, but that tell-tale tone can be heard and the little rotating animated cursor on the phone means _something_ is definitely going on at the ADSI level. Am I missing a debug option that could show me more about what may be going wrong? I'll get with Sayson tech support today to see if they can make any sense of this. -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tim Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 9:59 AM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? When you push your services button, are there any available slots. If not, delete the 3rd or 4th ones Don't know if it matters, but I have: Comedian Mail Asterisk PBX available Aastra SL The only selectable ones are the Comedian Mail and the Asterisk PBX. You might try deleting them all and reloading. And # is the Transfer button. 8-) Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 9:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Update #2 According to the helpful folks at Sayson, this phone actually has two 'slots', and the ADSI stuff needs to be downloaded to both (the second one kicks in when the phone is idle for more than 1s). I had only downloaded asterisk.adsi to the first slot, which explains why I was seeing the default (PLEASE PROGRAM ME) triggered from slot 2. I downloaded to the second slot and all is well now ... at least the phone now identifies itself as I asked it to, and seems to be programmed ;-) It seems to me that asterisk.adsi is pretty vanilla - not much functionality. Has anyone built a better mousetrap that they'd care to share? My second problem remains ... when I connect to voicemail now I see: Comedian Mail download refused Services is full but I don't see any errors in Asterisk's console. What's up with that??? I am now seeing some really cool functionality once I get past this point. As I'm checking voicemail by navigating the menus, I see stuff like: Old Messages Message 1 of 5 Unknown Tue Dec 23 07:46:02 Sweet! -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Darren Nickerson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 4:18 PM Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Thanks for the reply Tim - I was beginning to think nobody used this stuff ;-) As you can tell, I'm a relative newcomer to ADSI - I'm really not sure what to expect once the
Re: [Asterisk-Users] Programming an unlocked ADSI phone?
Tim, Thanks for your continued participation in this thread. The truth is, it's not clear to me how to delete a service ... the services menu only allows me to 'Select' or 'Quit'. It's also not clear to me how you managed to get Comedian Mail downloaded to Slot 1 without unlocking it with a code. When you did the ADSIProg originally (presumably you only did this once), did you download to both slots, or just the second one? -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tim Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 2:42 PM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? Have you tried deleting those services and adding them back in by checking voicemail 1st (dial 8500) then dial your 8600 for AdsiProg You should be able to just hit the # during the call, but you will also have to make sure you have the |Tt defined in your extensions.conf file as well. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 10:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Yes, there are 2 available slots. When I push services, I see: Services Asterisk PBX Asterisk PBX available available As I mentioned, the only soft-key I seem to have is 'VMail', and when I select that I still get: Comedial Mail download refused Services is full And yet somehow there does seem to be some ADSI functionality later on, when it falls back to a voice prompt after the download is refused and I'm navigating the voice prompts, reading the messages. It troubles me that Asterisk isn't very chatty about the ADSI stuff ... when I access VMail all I see is: -- Starting simple switch on 'Zap/5-1' -- Executing VoiceMailMain(Zap/5-1, ) in new stack (here's where the ADSI stuff seems to happen on the phone, download refused etc) -- Playing 'vm-login' (language 'en') [snip rest of log] There's no indication that any ADSI transactions are going on here, but that tell-tale tone can be heard and the little rotating animated cursor on the phone means _something_ is definitely going on at the ADSI level. Am I missing a debug option that could show me more about what may be going wrong? I'll get with Sayson tech support today to see if they can make any sense of this. -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tim Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 9:59 AM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? When you push your services button, are there any available slots. If not, delete the 3rd or 4th ones Don't know if it matters, but I have: Comedian Mail Asterisk PBX available Aastra SL The only selectable ones are the Comedian Mail and the Asterisk PBX. You might try deleting them all and reloading. And # is the Transfer button. 8-) Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 9:29 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Update #2 According to the helpful folks at Sayson, this phone actually has two 'slots', and the ADSI stuff needs to be downloaded to both (the second one kicks in when the phone is idle for more than 1s). I had only downloaded asterisk.adsi to the first slot, which explains why I was seeing the default (PLEASE PROGRAM ME) triggered from slot 2. I downloaded to the second slot and all is well now ... at least the phone now identifies itself as I asked it to, and seems to be programmed ;-) It seems to me that asterisk.adsi is pretty vanilla - not much functionality. Has anyone built a better mousetrap that they'd care to share? My second problem remains ... when I connect to voicemail now I see: Comedian Mail download refused Services is full but I don't see any errors in Asterisk's console. What's up with that??? I am now seeing some really cool functionality once I get past this point. As I'm checking voicemail by navigating the menus, I see stuff like: Old Messages Message 1 of 5 Unknown Tue Dec 23 07:46:02 Sweet! -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Darren Nickerson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 30, 2003 4:18 PM
Re: [Asterisk-Users] Asterisk Web Dialer
I am putting together a solution that will employ the Digium TE410P with one T1 going out the PSTN and the other front-ending a PBX. The idea is that based on a URL, Asterisk will dial an employee behind the PBX. When the employee picks up, Asterisk will dial the customer (detailed in the URL). I am assuming Asterisk can work with Apache (through AGI maybe) to dial the employee and then connect to the customer via info in the URL (or related through some sort of DB lookup). Another requirement will be to record the phone call as well. You could do it through either the Asterisk manager interface or have a CGI scrip t in your web front end create an auto call file that dials the employee and runs a second Dial command upon answer. I have worked a bit with Asterisk and am very happy with what it can do -- and would prefer to stay with Asterisk. The question is, can Asterisk handle what my requirements are or would this better be served by Bayonne? Asterisk is better. Hands down. No questions. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New to asterisk? RUN... don't walk.
As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
There are many people on this list that are more than happy to help you with a problem if you know how to ask the question. But if you've tried to keep up with this mailing list over any amount of time, you will see how quickly it becomes frustrating when people ask the same questions over and over again. Hence being pointed to the how-to's and the wiki. Do they answer every question? No. But they cover the most frequently asked. Do they answer the question exactly like you'd hope? Maybe not. It requires some thinking. There are two things you need to remember here. One, no one on this list gets paid to help you or anyone else with your problems. They do so because they choose to. Two, Asterisk is open source. It doesn't cost you a penny. If you want stable and friendly ass-kissing support personnel, you need to look at a commercial solution. Sean -Original Message- From: Me [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 3:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. No we aren't anti-n00b... Have you tried the IRC channel? Its usually more helpful for the newcomers. I have personally helped many people get started with examples and other such things. As far as asterisk segfaulting you might have hardware problems... I recommend you join the IRC channel and ask some questions. But you also have to be prepaired to read a little because we can't hold your hand thru it all just like with any other open source software solution out there. Asterisk is fairly easy to understand once you see how it all fits together. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Happy New Year!!
Nope I should have been a bit more clear in my response.. but its one of those days The howto should be generalized... Not just for X or Y distro... its not helpful and cuses duplication of documentation efforts. I would tend to disagree with your statement that it is not helpful. Writing any HOWTO requires some effort on the author's part, and the effort does help a number of users. I agree with you that a HOWTO for Asterisk, for example, should be generalized so that it can be helpful to a wider audience. An existing HOWTO for a particular operating system can serve as a good basis for a general HOWTO. I hope that the previously named author does not interpret this as trivializing their contribution to the community because I'm sure many first-time Asterisk users are glad to find step-by-step instructions that cover such a readily available and widely-used flavor of Linux. -- Tony Kava Network Administrator Pottawattamie County, Iowa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. Wow... I think this is our first troll... Not much of one at that, either. For the sake of the archives, newbies should be looking in the following areas: 1. the handbook. www.asterisk.org/index.php?menu=support. It's down under the google logo. 2. there are TONS of other resources on that page. Use them. 3. IRC (also mentioned on that page): irc.freenode.net, #asterisk 4. this mailing list's ARCHIVES. http://lists.digium.com/pipermail/asterisk-users/ you can search the archives by using google and including site:lists.digium.com in your search. The reason many of us here seem newbie-hostile is because we answer the SAME FREAKING BASIC QUESTIONS OVER AND OVER AND OVER. Personally I blame asterisk.org's webmasters for not cleaning up that hideous documentation page and making it CLEAR where the handbook is and where other very common resources are, but nevertheless it gets very tedious to hear the same bitch and moans from people who wouldn't lift a finger to solve their own problems. So yes, you in particular, should run from asterisk. As a general rule no open source project tolerates people who refuse to try and help themselves first. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP phone as intercom
Sure, here's my extension for paging on the intercom:
[ext-intercom-one];
exten = _87XXX,1,SetVar(ALERT_INFO=Ring Answer)
exten = _87XXX,2,Dial(SIP/${EXTEN:1},20,r)
exten = _87XXX,103,Congestion
exten = _87XXX,104,Congestion
exten = t,1,Hangup
Internally, I use a four digit extension here, starting with 7. When I
preface it with 8, it calls this extension, which sets the ALERT_INFO
variable and makes the phone work its auto-answer magic. If I dial my
partner with 7002, for example, it'll ring normally, go to voicemail,
etc. If I dial him with 87002, it beeps his office and his phone
automatically answers.
There's probably a better way to do this, but I've only had these phones
for a couple of weeks.
In answer to your other question about the bootrom, I think you need at
least the 2.4.0 bootrom to run the latest SIP software with these
features. I'm running 2.4.1. The SIP software version you must have to
do this is 1.1.0.
If you go to polycom's website and download the manual for these phones
and the 1.1.0 release notes, you can find out how to do all sorts of
tricks by manipulating the cfg files. (Sadly, you can only get the
manuals from Polycom. You have to get the bootrom and software from
your vendor)
John Baker
On Wed, 2003-12-31 at 08:24, mattf wrote:
I've added it as a separate page:
http://www.voip-info.org/wiki-Polycom+auto-answer+config linked from the
Polycom phones page.
Could you possibly send me a quick line or two(example code) on setting the
ALERT_INFO variable in Asterisk?
Thanks,
MATT---
-Original Message-
From: mattf [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 7:57 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] SIP phone as intercom
Cool, haven't looked that in depth into the new firmware(is that the 2.4.1
firmware?) I'll have to try that.
I'll post your instructions on the Wiki page later today.
Thanks,
MATT---
-Original Message-
From: John Baker [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 3:07 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] SIP phone as intercom
Hello, all
Sorry to correct you on this Matt, but I am currently doing this with
the Polycom 600 phones. You need the latest version of both the SIP
software and bootrom to do it, (and that stuff ain't easy to get) but it
is workable.
The latest version of software provides for distinctive ring tones just
like the Cisco 7960's have. It also provides for auto answer. It's
kind of tricky to do, but you can make your phones auto answer by
setting the Alert-Info variable in asterisk and messing with the xml
configuration files, sip.cfg and ipmid.cfg.
In the sip.cfg file, look for the line with these variables:
alertInfo voIpProt.SIP.alertinfo.1.value=Sales
voIpProt.SIP.alertInfo.1.class=8...
In this real-world example, whenever I set ALERT_INFO to Sales in
Asterisk, the Polycom matches on that word and calls up class 8 in
ipmid.cfg.
In ipmid.cfg, my class 8 line looks like this:
SALES se.rt.8.name=Sales se.rt.8.type=ring se.rt.8.ringer=11
se.rt.8.callWait=6 se.rt.8.mod=0
se.rt.8.type=ring tells the Polycom phone which type of ring to use -
which in this case is a regular ring and se.rt.8.ringer=11 tells the
phone to ring with ringtone 11 with is the Triplet.
I use this one for signaling a new incoming sales call to one of my
three sales guys. The secretary transfers it to the sales department
and their lines ring with the Triplet. I feel like Pavlov whenever I
hear it.
The other ring types are visual, answer and ring-answer. The one you
want is ring-answer.
Here's how I do it: Again in sip.cfg (actually part of the same line
listed above)
...voIpProt.SIP.alertinfo.2.value=Ring Answer
voIpProt.SIP.alertInfo.2.class=4...
and in ipmid.cfg (I just modified one of the existing ones to give me a
High Double Trill ringtone)
RING_ANSWER se.rt.4.name=Ring Answer se.rt.4.type=ring-answer
se.rt.4.timeout=1000... se.rt.4.ringer=7...
The se.rt.4.timeout=1000 tells the Polycom to ring for 1000
milliseconds (one second) and then answer.
I call it in Asterisk by setting the ALERT_INFO variable to Ring
Answer whenever anybody pushes 8 plus the extension. It rings in to
the extension and voila, I'm on speaker!
By the way, for all you BOFH out there, you could actually use this
feature as a somewhat surreptitious eavesdropping device by using a
silent ring and a visual type. The phone would answer without any
indication except on the console. I haven't tried this myself and if
you do this, I don't want to know about it...unless I'm in your office
at the time.
Good Luck! I was going to put this in the Wiki myself, but maybe
somebody will give me a late Christmas present.
--John Baker
On Tue, 2003-12-30 at 19:08, mattf wrote:
Hello,
It's all dependant upon the firmware of the phone(nothing to do with the
PBX
or
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
If you are a person who likes all things easy, and if you don't need to know nothing to be better professional, well, run now, and let us continue our journey. Who cares? People likes you don't help to our community. Regards, Gus - Original Message - From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 5:37 PM Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. ... There are many people on this list that are more than happy to help you with a problem if you know how to ask the question. But if you've tried to keep up with this mailing list over any amount of time, you will see how quickly it becomes frustrating when people ask the same questions over and over again. Hence being pointed to the how-to's and the wiki. Do they answer every question? No. But they cover the most frequently asked. I agree. Lately there have been some harsh responses to common questions. 'Tis the season, as they say. I've personally been able to do everything I wanted to do at home with Asterisk, and I've never once encountered a segmentation fault or any real crash. I have not found Asterisk to be riddled with bugs, and I've seen that the project continues to improve rapidly. Initially I found everything I needed to setup a home Asterisk system by Google searches, and reading the documentation that is available. I must state that when you decide to experiment with a project that is somewhat bleeding edge you should expect to a lot of research on your own, and you should start out with a decent amount of patience. That being said, I must admit that the Asterisk-Users mailing list has been very helpful and responsive to a couple of questions I have asked in the past. I am no Asterisk guru, but I believe I have a good handle on things, and I learn more everyday by following this list. If people make a reasonable effort to find the easy answers and provide ample details when asking the not-so-trivial questions then they will find this list, and the greater Asterisk users community, is not such a terrible place after all. -- Tony Kava Network Administrator Pottawattamie County, Iowa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
Dear newbies, As a newcomer to woodworking, you will not be welcomed with open arms. First, you will find no documentation on how to make your completely custom ceiling-height cabinets perfectly the first time that your wife will appreciate. Second, if you ask any woodworker for assistance, you will be treated like a fool and your new cabinets will be set aflame and you will be instructed to experiment with your tool and learn your craft. This worthless waste of time will only develop you into a competent woodworker able to make anything you wish. You should go to the furniture store or ask an already competent person to take care of your cabinetry for you as you have neither the desire or intelligence. Lastly, your raw material is so bug-ridden, all your handiwork will prove fruitless. We should all leave it up to the experts. With a carpentry community so anti-n00b, don't expect your handbuilt cabinets to be fixed for free by other people with their own problems who have graciously given their time and knowledge to the rest of us. You might actually be expected to fix it yourself. Here's the deal: Asterisk is free. If we go with * we will save $50k. It does almost anything. I can make it open my garage door. My installation records all conversations and then archives them as timestamped stereo MP3s. Our VB windows application can dial out with a click. All for free. It's not done. We are not at v1.0. Mr Spencer is a busy guy. It might not solve 'your' problem. We contracted the AgentCallbackLogin Queue stuff. That part works great. If you want it modified or fixed, pay for it or do it yourself. If you change your own oil, do your own plumbing, have more that 3 computers at home, or have [EMAIL PROTECTED] running, you are either a do-it-yourselfer or a geek. Asterisk might be for you. On the other hand, if you can't change a lightbulb or don't know what a dipstick is and have lots of money, then pay someone for a phone system. But please stop whining. I have 3 kids. Gettin' tired of it. Good day. - Original Message - From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 2:37 PM Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail was scanned for viruses using BitDefender Sent by 602Pro LAN SUITE - http://www.software602.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
There are many reliable Asterisk installations, some running hundreds of phones. It's easy to stop at a CVS version and build a very stable system. We have many happy customers running Asterisk. I certainly prefer it to my Cisco Call Manager installations. It's already a much better product than Call Manager. I have always been very well received in this group. The help has always been extraordinary. I had one problem with my Cisco phones that Cisco couldn't figure out in a week, someone here had the answer in a few minutes. Asterisk is a new product, but it obsoletes the legacy products. The documentation problem will disappear shortly as there will be a book about Asterisk for beginning users in January. Warm Regards and Good Luck, Paul Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Cheesman Sent: Wednesday, December 31, 2003 12:47 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] New to asterisk? RUN... don't walk. There are many people on this list that are more than happy to help you with a problem if you know how to ask the question. But if you've tried to keep up with this mailing list over any amount of time, you will see how quickly it becomes frustrating when people ask the same questions over and over again. Hence being pointed to the how-to's and the wiki. Do they answer every question? No. But they cover the most frequently asked. Do they answer the question exactly like you'd hope? Maybe not. It requires some thinking. There are two things you need to remember here. One, no one on this list gets paid to help you or anyone else with your problems. They do so because they choose to. Two, Asterisk is open source. It doesn't cost you a penny. If you want stable and friendly ass-kissing support personnel, you need to look at a commercial solution. Sean -Original Message- From: Me [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 3:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
Dear newbies, As a newcomer to woodworking, you will not be welcomed with open arms. First, you will find no documentation on how to make your completely custom ceiling-height cabinets perfectly the first time that your wife will appreciate. Second, if you ask any woodworker for assistance, you will be treated like a fool and your new cabinets will be set aflame and you will be instructed to experiment with your tool and learn your craft. This worthless waste of time will only develop you into a competent woodworker able to make anything you wish. You should go to the furniture store or ask an already competent person to take care of your cabinetry for you as you have neither the desire or intelligence. Lastly, your raw material is so bug-ridden, all your handiwork will prove fruitless. We should all leave it up to the experts. With a carpentry community so anti-n00b, don't expect your handbuilt cabinets to be fixed for free by other people with their own problems who have graciously given their time and knowledge to the rest of us. You might actually be expected to fix it yourself. Certainly great! You make me laugh so much... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
As a new asterisk user myself, I would agree with you that the learning curve is steep, but that was my expectation coming into this. I took the time to browse the list archives before signing up so no surprises there. There are some real experts here and they obviously help those who ask interesting questions that aren't answered elsewhere. I would agree that this list would be better without the retarded flame wars, and furthermore, trolls the likes of you. If you don't want to read the information that's available, or if what you expect is total hand-holding - someone else to install and configure your phone system for you, then asterisk is a great choice but you need to hire someone to do that. Or you can go with a commercial phone system and pay thousands for a basic system with 1/10th the features. Regarding the stability problem you're having - clearly that's not the norm. I wouldn't suggest that anyone expect that behavior. I certainly haven't seen any crashes. On Dec 31, 2003, at 12:37 PM, Me wrote: As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. __ Do you Yahoo!? Find out what made the Top Yahoo! Searches of 2003 http://search.yahoo.com/top2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 31 December 2003 03:24 pm, asterisk wrote: Here's the deal: It does almost anything. I can make it open my garage door. My installation records all conversations and then archives them as timestamped stereo MP3s. Our VB windows application can dial out with a click. All for free. No argument here. I think 80% of us n00bs can get by with the docs as-is (all I ask is to not be attacked), although if listserv gets repeated questions, maybe it's a symptom. Thing is, a novice or journeyman can't really fix the docs to the best technical info; takes a master, who is understandably doing more important things. Looks to me at this point, that asterisk has the potential of being (is?) one of the great open-source projects. Kudos. BTW, is anyone participating in the ENum trial? With Asterisk? -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iEYEARECAAYFAj/zR9MACgkQnQ18+PFcZJvGuQCfSjwr0WQhy3l9tUH9tgjL8L0K laEAnRsFlpC+kcU81c+imhB7WOpZJw3u =X/ME -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New to asterisk? RUN... don't walk.
Hello, I am not a veteran here, but would like to share my thoughts on this subject. True, * is opensource and freely available, but it is not a computer program that you download and run. It is a very versatile telecommunication product you would otherwise pay at least 100 K to buy from a telecom vendor, if not more based on modules and usage, license hash-codes etc. Even to try * one would need some pre requisite knowledge in telecom, if not many years in the field. I work for a large telecom company and my specialty is voice over broadband (or xDSL). I worked with asterisk for couple of months now and I am amazed to see areas of telecom that * touch upon with. Starting from Linux, to SIP, H323, DSL technologies (PPP, PPPoE, PPPoA, DHCP, NAT), Call routing(Dial Plan), IVR, Transcoding, STUN are few areas that one would have to master even thinking about *. True one would know the syntax, and howtos etc, but also would have to have the ability to troubleshoot. For last two-three months in this list, I have not seen any newbi posting a sip trace (from a ethereal or a TCP dump) and asking a question about it. I have seen many question for instance, asking syntax of h.323 dial, but never seen a question asked on a h323 trace. I think, having * openly available is like keeping an airplane openly available in a airfield, so that anybody can try flying. Tell me how many of us would go try and fly that airplane if we do not know how to fly :) Point that I want to make here is simple, please try to understand what * is all about. If you like it's features and would like it to run in a production environment try to get some professional help. If you are learning these technologies for fun then get educated, use tools available to troubleshoot. Hooking up couple of phones and making a call is far from knowing *. Asterisk is a great product (thanks Mark and many others) and if you know what you are doing, you can do wonders with it. Don't put it down, because you do not have the background to understand it or work with it. Cheers SW Message: 4 Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Reply-To: [EMAIL PROTECTED] As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to asterisk? RUN... don't walk.
Well said. - Original Message - From: SW [EMAIL PROTECTED] To: [EMAIL PROTECTED] Digium. Com [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 2:13 PM Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Hello, I am not a veteran here, but would like to share my thoughts on this subject. True, * is opensource and freely available, but it is not a computer program that you download and run. It is a very versatile telecommunication product you would otherwise pay at least 100 K to buy from a telecom vendor, if not more based on modules and usage, license hash-codes etc. Even to try * one would need some pre requisite knowledge in telecom, if not many years in the field. I work for a large telecom company and my specialty is voice over broadband (or xDSL). I worked with asterisk for couple of months now and I am amazed to see areas of telecom that * touch upon with. Starting from Linux, to SIP, H323, DSL technologies (PPP, PPPoE, PPPoA, DHCP, NAT), Call routing(Dial Plan), IVR, Transcoding, STUN are few areas that one would have to master even thinking about *. True one would know the syntax, and howtos etc, but also would have to have the ability to troubleshoot. For last two-three months in this list, I have not seen any newbi posting a sip trace (from a ethereal or a TCP dump) and asking a question about it. I have seen many question for instance, asking syntax of h.323 dial, but never seen a question asked on a h323 trace. I think, having * openly available is like keeping an airplane openly available in a airfield, so that anybody can try flying. Tell me how many of us would go try and fly that airplane if we do not know how to fly :) Point that I want to make here is simple, please try to understand what * is all about. If you like it's features and would like it to run in a production environment try to get some professional help. If you are learning these technologies for fun then get educated, use tools available to troubleshoot. Hooking up couple of phones and making a call is far from knowing *. Asterisk is a great product (thanks Mark and many others) and if you know what you are doing, you can do wonders with it. Don't put it down, because you do not have the background to understand it or work with it. Cheers SW Message: 4 Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Reply-To: [EMAIL PROTECTED] As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: New to asterisk? RUN... don't walk.
Me wrote: As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. H! I: 0) Was a Newbie 1) Had no (or little) Linux experience. 2) Found * with Google 3) Read 4) Read 5) Read 6) Installed and got * running 7) Read, Read, Read 8) Bought, installed, set up channel bank 9) Asked my first real question of the list - was happy with reply 10) Read, Read, Read 11) Bought, installed, set up 20 SIP Phones 12) Read, Read, Read 13) Very happy with result. Is there a message here? I'm not brilliant, not a linux/asterisk guru. Just patient, determined and willing to try a lot of stuff so that I know what to ask and when. Do I have any advice? Yes: Read, Read, Read. It really does work - and by the way, so does asterisk. Since July of this year I have never had a seg fault, never had asterisk freeze or crash, and since I've gone fully online in September, there has not been a single problem related to asterisk. And while the documentation is, well, scattered, it is nevertheless out there. And, for what it's worth, if you ask a question about the more arcane, poorly documented stuff, no matter how dumb (well, almost), you will never get flamed. Finally, I rather prefer taking a few lumps from the * community than having the experience I had with our local Cisco people, who took so long to call back with information about their products (which when it finally arrived, was wrong anyway - I found the correct information on the * list!), that in the meantime I had discovered *, bought my hardware and set up a functioning system. So - Shutup, take your lumps and do your homework. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Happy New Year!!
First of all regarding my social ettiquette, it has nothing to do with my lack of it but more to do with my philosophy of chew them up, spit them out regardless of the venue they choose. I'll play on field, You lay it, I'll play it. As far as your thought of a generalized howto, it happens to be a pretty good idea. As far as your figuring most people should have gotten what you said, I guess you figured wrong again. The world doesn't always work quite the way we figure it in our little brains. Regarding you caring what others think, I think the mere fact of you responding to the first post and even this one speak for themselves. Other than the previously stated, you and I probably see the world much alike. I tell it how I see it, don't sugar coat, am very blunt and to the point; however even in doing so I try to consider the diversity of views and objectives. As far as what you do good for asterisk, keep up the good work. As far as how you interact with people, hell it might change as you go through different stages of life. By the way, Happy New Year to you Brian. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Happy New Year!!
Also to add a bit to what I said earlier, the reason I thought RedHat / Fedora was a good howto was the mere fact that RedHat / Fedora has been known to present it's own distinct installation problems because of packages / pachage dependencies. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programming an unlocked ADSI phone?
I've tried it several times, and your ADSI clearing tip didn't check out on my phone ... in particular: Hit options Hit Mute or Flash When I hit options, then the mute button (I don't have a flash button) I'm still left in the options screen. -d -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: PBX [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 6:42 PM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? I wanted to give you some guidance on the configuration of the phone Here is sniplet of configuration Aastra 390 and 480 Phones... In an ADSI script for the 1st Slot: ; ; Asterisk default ADSI script ; ; ; Begin with the preamble requirements ; DESCRIPTION Asterisk PBX ; Name of vendor VERSION 0x00 ; Version of stuff ;SECURITY _AST ; Security code SECURITY 0x9BDBF7AC; Security code FDN 0x000F ; Descriptor number In an ADSI script for the 2nd Slot: ; ; Asterisk default ADSI script ; ; ; Begin with the preamble requirements ; DESCRIPTION Asterisk PBX ; Name of vendor VERSION 0x00 ; Version of stuff ;SECURITY _AST ; Security code SECURITY 0x78921D49; Security code FDN 0x85EFD9DA ; Descriptor number You wouldn't need to program any other slots than these 2, and the 1st slot is the only one that MUST contain programming. This is because the first slot is triggered when the phone rings or when the phone is placed off-hook. The second slot (the Self Launching slot) is triggered when the phone has had no activity for a certain amount of time. Programming in this slot can be identical to slot 1, or it can be completely different, such as for advertising purposes. How to clear the ADSI Scripts from the phone Hit options. Choose Time/Date Set the time to Jan 1 12:00am Hit done Hit done Hit options Hit Mute or Flash A display giving the CPE ID and other stuff will appear QUICKLY press the # key Hope this helps. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Nickerson Posted At: Wednesday, December 31, 2003 2:58 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Programming an unlocked ADSI phone? Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Tim, Thanks for your continued participation in this thread. The truth is, it's not clear to me how to delete a service ... the services menu only allows me to 'Select' or 'Quit'. It's also not clear to me how you managed to get Comedian Mail downloaded to Slot 1 without unlocking it with a code. When you did the ADSIProg originally (presumably you only did this once), did you download to both slots, or just the second one? -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tim Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 2:42 PM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? Have you tried deleting those services and adding them back in by checking voicemail 1st (dial 8500) then dial your 8600 for AdsiProg You should be able to just hit the # during the call, but you will also have to make sure you have the |Tt defined in your extensions.conf file as well. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 10:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Yes, there are 2 available slots. When I push services, I see: Services Asterisk PBX Asterisk PBX available available As I mentioned, the only soft-key I seem to have is 'VMail', and when I select that I still get: Comedial Mail download refused Services is full And yet somehow there does seem to be some ADSI functionality later on, when it falls back to a voice prompt after the download is refused and I'm navigating the voice prompts, reading the messages. It troubles me that Asterisk isn't very chatty about the ADSI stuff ... when I access VMail all I see is: -- Starting simple switch on 'Zap/5-1' -- Executing VoiceMailMain(Zap/5-1, ) in new stack (here's where the ADSI stuff seems to happen on the phone, download refused etc) -- Playing 'vm-login' (language 'en') [snip rest of log] There's no indication that any ADSI transactions are going on here, but that tell-tale tone can be heard and the little rotating animated cursor on the phone means _something_ is
Re: [Asterisk-Users] after hours - is this logic ok ?
- Original Message -
From: Lance Arbuckle [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 1:54 PM
Subject: [Asterisk-Users] after hours - is this logic ok ?
Ok, first off, Asterisk is the coolest piece of software I have EVER had
the pleasure of using in my 8 years of running linux !! and I know I
haven't even scratched the surface feature wise.
Before I get too excited, I wanted to get all you experts to look at the
how I implemented my after hours test. The goal is to prevent the phone
from ringing afer certain hours, just go to VM.
Bacically, when a call comes from the PSTN, I use these includes to
either set a key in the DB or not.
include = day|8:00-21:00|mon-fri|*|*
include = day|9:00-21:00|sat-sun|*|*
; if we're not open, we're closed
include = night
[day]
exten = s,2,DBput(FEATURE/DAY=yes)
exten = s,3,Goto(s,10)
[night]
exten = s,2,NoOp
exten = s,3,Goto(s,10)
And then in my stdexten macro, I test for the existence of the key in
the database. If the key exists, it must be daytime so delete the key
and allow the calls to ring the extension. If the key does not exist,
it must be night and since we don't want the phones to ring we jump to
the unavailable VM. I've tested this and it works as I expect but my
only concern is how this would hold up in a busy environment where many
calls are being processed. Could one asterisk thread delete the
database key before another thread had gotton the oportunity to test for
the key ?
[macro-stdexten]
exten = s,1,NoOp
other testing crap deleted
exten = s,10,DBget(foo=FEATURE/DAY); is it day time ?
exten = s,11,DBdel(FEATURE/DAY); yes, delete the key
exten = s,12,Goto(s,201) ; and ring the phone
exten = s,111,Goto(s,204) ; no, goto uVM
exten = s,201,answer
exten = s,202,Playback(transfer,skip)
exten = s,203,Dial(${ARG2},5) ; Ring the interface, 20 seconds
maximum
exten = s,204,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
exten = s,205,Playback(vm-goodbye)
exten = s,206,Wait(1)
exten = s,207,Hangup
exten = s,304,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
exten = s,305,Playback(vm-goodbye)
exten = s,306,Wait(1)
exten = s,307,Hangup
Although I can't provide you with an example, I think you might find it
easier to simplify the turn up/down logic.
When a call comes in, check day/night, as you already do. If it's day, set
the key. (If the key's already there, resetting it shouldn't hurt.) If it's
night, delete the key if it exists.
Then your extensions should test for the key and do the right thing if it is
or isn't there, but not touch the key themselves.
Andrew Thompson http://aktzero.com/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] after hours - is this logic ok ?
- Original Message -
From: Lance Arbuckle [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 1:54 PM
Subject: [Asterisk-Users] after hours - is this logic ok ?
Ok, first off, Asterisk is the coolest piece of software I have EVER had
the pleasure of using in my 8 years of running linux !! and I know I
haven't even scratched the surface feature wise.
Before I get too excited, I wanted to get all you experts to look at the
how I implemented my after hours test. The goal is to prevent the phone
from ringing afer certain hours, just go to VM.
Bacically, when a call comes from the PSTN, I use these includes to
either set a key in the DB or not.
include = day|8:00-21:00|mon-fri|*|*
include = day|9:00-21:00|sat-sun|*|*
; if we're not open, we're closed
include = night
[day]
exten = s,2,DBput(FEATURE/DAY=yes)
exten = s,3,Goto(s,10)
[night]
exten = s,2,NoOp
exten = s,3,Goto(s,10)
And then in my stdexten macro, I test for the existence of the key in
the database. If the key exists, it must be daytime so delete the key
and allow the calls to ring the extension. If the key does not exist,
it must be night and since we don't want the phones to ring we jump to
the unavailable VM. I've tested this and it works as I expect but my
only concern is how this would hold up in a busy environment where many
calls are being processed. Could one asterisk thread delete the
database key before another thread had gotton the oportunity to test for
the key ?
[macro-stdexten]
exten = s,1,NoOp
other testing crap deleted
exten = s,10,DBget(foo=FEATURE/DAY); is it day time ?
exten = s,11,DBdel(FEATURE/DAY); yes, delete the key
exten = s,12,Goto(s,201) ; and ring the phone
exten = s,111,Goto(s,204) ; no, goto uVM
exten = s,201,answer
exten = s,202,Playback(transfer,skip)
exten = s,203,Dial(${ARG2},5) ; Ring the interface, 20 seconds
maximum
exten = s,204,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
exten = s,205,Playback(vm-goodbye)
exten = s,206,Wait(1)
exten = s,207,Hangup
exten = s,304,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
exten = s,305,Playback(vm-goodbye)
exten = s,306,Wait(1)
exten = s,307,Hangup
Although I can't provide you with an example, I think you might find it
easier to simplify the turn up/down logic.
When a call comes in, check day/night, as you already do. If it's day, set
the key. (If the key's already there, resetting it shouldn't hurt.) If it's
night, delete the key if it exists.
Then your extensions should test for the key and do the right thing if it is
or isn't there, but not touch the key themselves.
Andrew Thompson http://aktzero.com/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] after hours - is this logic ok ?
Andrew Thompson wrote:
- Original Message -
From: Lance Arbuckle [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 1:54 PM
Subject: [Asterisk-Users] after hours - is this logic ok ?
Ok, first off, Asterisk is the coolest piece of software I have EVER had
the pleasure of using in my 8 years of running linux !! and I know I
haven't even scratched the surface feature wise.
Before I get too excited, I wanted to get all you experts to look at the
how I implemented my after hours test. The goal is to prevent the phone
from ringing afer certain hours, just go to VM.
Bacically, when a call comes from the PSTN, I use these includes to
either set a key in the DB or not.
include = day|8:00-21:00|mon-fri|*|*
include = day|9:00-21:00|sat-sun|*|*
; if we're not open, we're closed
include = night
[day]
exten = s,2,DBput(FEATURE/DAY=yes)
exten = s,3,Goto(s,10)
[night]
exten = s,2,NoOp
exten = s,3,Goto(s,10)
And then in my stdexten macro, I test for the existence of the key in
the database. If the key exists, it must be daytime so delete the key
and allow the calls to ring the extension. If the key does not exist,
it must be night and since we don't want the phones to ring we jump to
the unavailable VM. I've tested this and it works as I expect but my
only concern is how this would hold up in a busy environment where many
calls are being processed. Could one asterisk thread delete the
database key before another thread had gotton the oportunity to test for
the key ?
[macro-stdexten]
exten = s,1,NoOp
other testing crap deleted
exten = s,10,DBget(foo=FEATURE/DAY); is it day time ?
exten = s,11,DBdel(FEATURE/DAY); yes, delete the key
exten = s,12,Goto(s,201) ; and ring the phone
exten = s,111,Goto(s,204) ; no, goto uVM
exten = s,201,answer
exten = s,202,Playback(transfer,skip)
exten = s,203,Dial(${ARG2},5) ; Ring the interface, 20 seconds
maximum
exten = s,204,Voicemail(u${ARG1}) ; If unavailable, send to
voicemail w/ unavail announce
exten = s,205,Playback(vm-goodbye)
exten = s,206,Wait(1)
exten = s,207,Hangup
exten = s,304,Voicemail(b${ARG1}) ; If busy, send to voicemail w/
busy announce
exten = s,305,Playback(vm-goodbye)
exten = s,306,Wait(1)
exten = s,307,Hangup
Although I can't provide you with an example, I think you might find it
easier to simplify the turn up/down logic.
When a call comes in, check day/night, as you already do. If it's day, set
the key. (If the key's already there, resetting it shouldn't hurt.) If it's
night, delete the key if it exists.
Then your extensions should test for the key and do the right thing if it is
or isn't there, but not touch the key themselves.
Andrew Thompson http://aktzero.com/
ok, thanks for the suggestion. I'm not sure why I decided to unset the
key from within the stdexten anyway. I think I've been staring at this
too long :) or maybe all the radiation from these 3 21 monitors has
finally cooked my brain cell.
So, the contexts would be more like this:
[day]
exten = s,2,DBput(FEATURE/DAY=yes)
exten = s,3,Goto(s,10)
[night]
exten = s,2,DBdel(FEATURE/DAY) ;if we got here it must be night
time so remove the key
exten = s,3,Goto(s,10)
[macro-stdexten]
exten = s,1,NoOp
other testing crap deleted
exten = s,10,DBget(foo=FEATURE/DAY); is it day time ? if key
exists, goto n+1, otherwise n+101
exten = s,11,Goto(s,201) ; yes, well let's ring the phones
exten = s,111,Goto(s,204) ; no, goto uVM
--
.~.Triad Internet Systems, Inc.
/V\Lance C. Arbuckle
// \\ 3315 Anderson Drive
/( )\ Winston-Salem, NC 27127
^'~'^ 336-771-2090
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Java?
On Wed, 31 Dec 2003 21:19:10 +0200 Stephen Karrington [EMAIL PROTECTED] wrote: We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. snip Apache + Mysql + PHP ( Ming + Actionscript ) + Asterisk is good. Dynamic effective,Easy coding and Fast response :-) --- Masakazu Nakano. Dairiten.com - an open source VoIP and Ubiquitous Portal site in Japan. http://www.dairiten.com/modules/news/ powered by xoops at http://www.xoops.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Grandstream Early Dial
I've never had early dial working, however, I resolved my multiple digit issue by simply putting both the GS phones and asterisk in INFO mode. This worked on both 10.0.3.81 firmware on the budgetone and the ATA286, as well as 10.0.4.30 firmware. I'm not saying I don't believe you, but doubelcheck your lines in asterisk to be dtmfmode=info and the gs devices are on SIP INFO method, and your DTMF Payload type is 101. Just my $.02 -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen R. Besch Sent: Wednesday, December 31, 2003 12:59 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Grandstream Early Dial I've just checked my voicemail with 1.0.4.30 and get the same multiple digits problem. sip.conf and GS config are both at info, for me this is a new problem voicemail has always worked perfectly with the GS. This has come up many times in this list, with no consensus for a solution. According to Grandstream, the multiple digit problem arises from a difference in the interpretation of the SIP standard. I'm not sure I really understand this, so no flames please, but, paraphrasing a conversation I had with GS, apparently they retransmit the digit as long as the key is pressed and expect asterisk to know that it is a re-transmission by examining other data in the packet. Asterisk does not handle the SIP packet in the way GS expects, resulting in multiple digit transmission. This flaw (?) is avoided by setting DTMF to INBAND. Why this behaviour is not repeatable on everyones installations escapes me. However, I have noticed one thing that may be a clue. I have one phone that is older hardware (redial button instead of send and an unused battery compartment on the bottom). This phone behaves differently than all the other, later, models. For example, it is the only phone on which the flash button actually works to answer the alternate line (eg when an incoming call waiting call arrives). All phones are using 3.81 firmware. Early dial has never worked for me, and I just upgraded to the 1.0.4.30 load yesterday. Now, I am having DTMF recognition issues, making it impossible to check my voice mail. This is an acknowleged bug on the GS. They have connected to my * server and acknowleged the problem. A fix has been promised but not yet delivered. Until then, the only solution is to turn early dial off and let the phone send the entire dial string in one packet. Since this does not affect later single digit transmission for IVR's, etc, the only consequence is the irritating delay between the last entered digit and the actual placing of the call. But, you can always hit the send key. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.554 / Virus Database: 346 - Release Date: 12/20/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.554 / Virus Database: 346 - Release Date: 12/20/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
Well, since everyone else is top-quoting on this message, so will I :P I'm no veteran either. As a matter of fact, I have had ZERO prior knowledge to the telcom industry or more than 'user level' experience with telecommunications in general. I decided that I wanted to expand my knowledge, and actually LEARN a few things, so I jumped into asterisk. I was, and quite frankly, IMO, still AM a 'n00b' to *. However, after playing around, and learning what things do, by reading the documentation that IS there, searching the archives, and just trolling the list and IRC, I have learned more in the last 4-5 months of having * than a lot of people I've noticed have learned in a lifetime of experience.I now have a fully functional (well, minus MOH, because mpg123 isn't yet compiled on my new box), * implementation, serving myself and my roommates strictly over VoIP, and a couple ata's and a Internet PhoneJack card. I love it. And I'm STILL learning to this date. Asterisk is not something you can expect everyone to just drop what their doing and help you with. Sure, it can be frustrating, but if you are so dense that you can't sit down an play with it and learn what happens when you type something in the cli, or change a few things in your dialplan, then get out, I agree. If you liked taking apart mom's hairdryer as a kid and seeing how it worked, and then later on, rewired up a few things to do what you wanted them to, or even took a hex editor to command.com in msdos to change what it says to suit your taste (mucho guilty on that one.. lol), then you will have no problem finding out what you can and can't change simply by editing files, and trying things out. Take off your training wheels, and just TRY IT. - Josh R. [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of SW Sent: Wednesday, December 31, 2003 4:13 PM To: [EMAIL PROTECTED] Digium. Com Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Hello, I am not a veteran here, but would like to share my thoughts on this subject. True, * is opensource and freely available, but it is not a computer program that you download and run. It is a very versatile telecommunication product you would otherwise pay at least 100 K to buy from a telecom vendor, if not more based on modules and usage, license hash-codes etc. Even to try * one would need some pre requisite knowledge in telecom, if not many years in the field. I work for a large telecom company and my specialty is voice over broadband (or xDSL). I worked with asterisk for couple of months now and I am amazed to see areas of telecom that * touch upon with. Starting from Linux, to SIP, H323, DSL technologies (PPP, PPPoE, PPPoA, DHCP, NAT), Call routing(Dial Plan), IVR, Transcoding, STUN are few areas that one would have to master even thinking about *. True one would know the syntax, and howtos etc, but also would have to have the ability to troubleshoot. For last two-three months in this list, I have not seen any newbi posting a sip trace (from a ethereal or a TCP dump) and asking a question about it. I have seen many question for instance, asking syntax of h.323 dial, but never seen a question asked on a h323 trace. I think, having * openly available is like keeping an airplane openly available in a airfield, so that anybody can try flying. Tell me how many of us would go try and fly that airplane if we do not know how to fly :) Point that I want to make here is simple, please try to understand what * is all about. If you like it's features and would like it to run in a production environment try to get some professional help. If you are learning these technologies for fun then get educated, use tools available to troubleshoot. Hooking up couple of phones and making a call is far from knowing *. Asterisk is a great product (thanks Mark and many others) and if you know what you are doing, you can do wonders with it. Don't put it down, because you do not have the background to understand it or work with it. Cheers SW Message: 4 Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Reply-To: [EMAIL PROTECTED] As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. ___ Asterisk-Users mailing
Re: [Asterisk-Users] A Head Check
Steven Critchfield wrote: On Tue, 2003-12-30 at 23:47, Greg Boehnlein wrote: 3. I am also responsible for delivering inbound faxes to the DID numbers via Email. I.E. customer has a document faxed to them and they get it in Email as a tiff. I'm considering using Hylfax with a Multitech DID capable modem, but other suggestions are welcomed! It has been mentioned here before that you can pick up a device that accepts a PRI and will do your fax reception for you. If you get one of those and hook it to a empty port of the TE410P then it will be better as you could accept several faxes at once. (Disclaimer: I work for a company that sells something exactly like Steven is recommended) HylaFAX with an EICON Diva Server or Brooktrout TR1043 T1/PRI fax board would seem to fit the bill nicely. They're available in various port densities, depending upon the inbound fax volume you anticipate. If you go the MultiModemDID route, make sure you understand the exact nature of the analog DID line that device requires, and understand that this will limit you to a concurrency of only one incoming fax per modem. For more info, see http://www.hylafax.org/ or my employer's URL (below). -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Programming an unlocked ADSI phone?
Hi Darren (and anyone interested in this issue), Just an FYI that the factory reset procedure that wipes out the programming in these phones is different for the 390 than the 480 phone. If any of you need this procedure, or information on ADSI programming or the 390/480 phones in general for this market (Asterisk), please feel free to contact me directly (Sorry but I rarely have the chance to keep up with the discussions here). Anyway, I'm off home now. I wish you all a wonderful New Year! Regards and Best Wishes, Cheryl Millossi Sayson Technologies, Ltd. 604-629-5014 (Tel) 604-732-8726 (Fax) __ Confidentiality Notice The information contained in this communication is confidential and/or proprietary business or technical data. If you are not the intended recipient, you are hereby notified that any dissemination, copying or distribution of this communication, or the taking of any action in reliance on the contents of this communication, is strictly prohibited. If you have received this communication in error, please immediately notify us by telephone (604-730-1842) or electronically by return message, and delete or destroy all copies of this communication. -Original Message- From: Darren Nickerson [mailto:[EMAIL PROTECTED] Sent: December 31, 2003 4:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? I've tried it several times, and your ADSI clearing tip didn't check out on my phone ... in particular: Hit options Hit Mute or Flash When I hit options, then the mute button (I don't have a flash button) I'm still left in the options screen. -d -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: PBX [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 6:42 PM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? I wanted to give you some guidance on the configuration of the phone Here is sniplet of configuration Aastra 390 and 480 Phones... In an ADSI script for the 1st Slot: ; ; Asterisk default ADSI script ; ; ; Begin with the preamble requirements ; DESCRIPTION Asterisk PBX ; Name of vendor VERSION 0x00 ; Version of stuff ;SECURITY _AST ; Security code SECURITY 0x9BDBF7AC; Security code FDN 0x000F ; Descriptor number In an ADSI script for the 2nd Slot: ; ; Asterisk default ADSI script ; ; ; Begin with the preamble requirements ; DESCRIPTION Asterisk PBX ; Name of vendor VERSION 0x00 ; Version of stuff ;SECURITY _AST ; Security code SECURITY 0x78921D49; Security code FDN 0x85EFD9DA ; Descriptor number You wouldn't need to program any other slots than these 2, and the 1st slot is the only one that MUST contain programming. This is because the first slot is triggered when the phone rings or when the phone is placed off-hook. The second slot (the Self Launching slot) is triggered when the phone has had no activity for a certain amount of time. Programming in this slot can be identical to slot 1, or it can be completely different, such as for advertising purposes. How to clear the ADSI Scripts from the phone Hit options. Choose Time/Date Set the time to Jan 1 12:00am Hit done Hit done Hit options Hit Mute or Flash A display giving the CPE ID and other stuff will appear QUICKLY press the # key Hope this helps. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Nickerson Posted At: Wednesday, December 31, 2003 2:58 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Programming an unlocked ADSI phone? Subject: Re: [Asterisk-Users] Programming an unlocked ADSI phone? Tim, Thanks for your continued participation in this thread. The truth is, it's not clear to me how to delete a service ... the services menu only allows me to 'Select' or 'Quit'. It's also not clear to me how you managed to get Comedian Mail downloaded to Slot 1 without unlocking it with a code. When you did the ADSIProg originally (presumably you only did this once), did you download to both slots, or just the second one? -Darren -- Darren Nickerson Senior Sales Support Engineer iFAX Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 ext 8106 office +1.215.243.8335 fax - Original Message - From: Tim Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 31, 2003 2:42 PM Subject: RE: [Asterisk-Users] Programming an unlocked ADSI phone? Have you tried deleting those services and adding them back in by checking voicemail 1st (dial 8500) then dial your 8600 for AdsiProg You should be able to just hit the # during the call, but you will also
