[asterisk-users] Asterisk CTI interface to control legacy PBX
Hello, I am looking for a way to control another legacy PBX from Asterisk using a CTI interface. Are you aware of any legacy PBX CTI control card that can be controlled by Asterisk? I have an Avaya PBX with CTI interface and researching if I can connect Asterisk to this. :-) Thanks for any hints. -- - David Hajek Daktela - VoipObchod http://www.daktela.com/ http://www.voipobchod.cz/shop/ Tel: +420-226213305 GSM: +420-604352968 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CTI interface to control legacy PBX
I need Asterisk to tell Avaya which calls we need to record. Avaya is using their NICE call recording suite. Thanks - David Hajek Daktela - VoipObchod http://www.daktela.com/ http://www.voipobchod.cz/shop/ Tel: +420-226213305 GSM: +420-604352968 C F wrote: You could by crating an application that sits between the Avaya CTI and listens to Asterisk manager interface. What exactly are you trying to accomplish on the Avaya? On 7/22/07, David Hajek [EMAIL PROTECTED] wrote: Hello, I am looking for a way to control another legacy PBX from Asterisk using a CTI interface. Are you aware of any legacy PBX CTI control card that can be controlled by Asterisk? I have an Avaya PBX with CTI interface and researching if I can connect Asterisk to this. :-) Thanks for any hints. -- - David Hajek Daktela - VoipObchod http://www.daktela.com/ http://www.voipobchod.cz/shop/ Tel: +420-226213305 GSM: +420-604352968 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue - how to provide a caller ringing tone when some agent become available
Hi, I'm a little bit stuck with Queue app. I'm putting callers into the queue and have them hear music on hold when all (static) agents are busy. This is easy. But when agent become available I want the caller to hear a ringing tone (with message that his call has been routed to the support representative). Is this somehow doable? Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff does not work with TDM400P
Title: bristuff does not work with TDM400P Hi- we are having issues with quadBRI card which does not work together with TDM400P. We've tried to hunt the problem and here is the scenario: 1) starting asterisk with tdm400P and two FXS modules (two phones) 2) pickup first phone and dial the second one works great 3) hangup 4) pickup second phone and tried to dial the first phone - no luck - asterisk does not recognize DTMF of the dialing numbers and does not initiate call 5) restarting asterisk 6) go back to 4 and works! 7) go back to 2 and does not work again - same asterisk does not recognize DTMF of the dialing numbers Vanilla asterisk works just fine. The above scenario works even quadBRI card is removed it must be problem of bristuff patches. Do you have any hints what can be wrong? We've tried latest bristuff-0.3.X series. Thanks. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a general questionto clear up my understanding.
Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI cards, HFC,and bristuff - a general questionto clear up my understanding. I have 2 different instalations with 1 Billion HFC Card (1port), and 1 TDM400. Asterisk 1.0.10+bristuff+florz patch. Only issue is that you must load all modules (wcfxs, zaphfc) before runing ztcfg, otherwise nothing works. Everything works ok, even faxing. Julian. On 3/30/06, Chris Earle [EMAIL PROTECTED] wrote: What? After hours of searching for anything to help me, I found this comment about zaptel cards in systems with bristuff-cards (junghanns for me in this case) I havent' seen any other reports of this sort of behaviour --- can anyone confirm whether they've got a QuadBRI and TDM400P card working together in one machine? thanks :-S Zoa [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] snip We stopped with the bristuff as bristuff will break any other zaptel cards in the same system. (pri seems logical, why the tdm card also broke is unknown to me). snip ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding.
Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again but only one direction. -David From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Filip Drągowski Sent: Friday, March 31, 2006 1:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding. Asterisk 1.2.4 + bristuff-0.3.0-PRE-1l + libpri-1.2.2 + zapte-1.2.3 + *CLI zap show status Description Alarms IRQ bpviol CRC4 quadBRI PCI ISDN Card 1 Span 1 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 2 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 3 [TE] (caˇ OK 0 0 0 quadBRI PCI ISDN Card 1 Span 4 [TE] (caˇ OK 0 0 0 Wildcard TDM400P REV E/F Board 1 OK 0 0 0 and it seems to work. Only sip phones connected to *PBX (by gateways or ethernet) as i remember installation process: 1. bristuff patching asterisk/libpri/zaptel 2. libpri/zaptel/asterisk install 3. zaptel/quozap/wctdm modules installation Runs on Debian 3.1. kernel 2.6.15.4 but i have problem: when SIP hardphone hangup connection (SIP/ - Zap/) asterisk don't send Q.931 DISCONNECT message, and i don't have any idea how to fight with that. Filip D. Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can'tget it to work.-David-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Julian J.M.Sent: Friday, March 31, 2006 1:44 AMTo: Chris Earle; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: BRI cards, HFC,and bristuff - ageneral questionto clear up my understanding.I have 2 different instalations with 1 Billion HFC Card (1port), and 1TDM400. Asterisk 1.0.10+bristuff+florz patch.Only issue is that you must load all modules (wcfxs, zaphfc) beforeruning ztcfg, otherwise nothing works.Everything works ok, even faxing.Julian.On 3/30/06, Chris Earle [EMAIL PROTECTED] wrote: What? After hours of searching for anything to help me, I found thiscomment about zaptel cards in systems with bristuff-cards (junghanns for me in this case)I havent' seen any other reports of this sort of behaviour --- can anyone confirm whether they've got a QuadBRI and TDM400P card working together in one machine?thanks :-SZoa [EMAIL PROTECTED] wrote in messagenews:[EMAIL PROTECTED]... snip We stopped with the bristuff as bristuff will break any other zaptelcards in the same system. (pri seems logical, why the tdm card alsobroke is unknown to me). snip ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users__This email has been scanned by the MessageLabs Email Security System.For more information please visit http://www.messagelabs.com/email_--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --___Filip DrągowskiMobile: +48(0)500 054045E-mail: [EMAIL PROTECTED]___ONTP.NET Tomasz KarczewskiAleja Wojska Polskiego 33 pokoj 12265-077 Zielona Góra, PolandMobile: +48(0)501 653395Office: +48(0)68 4141018Fax: +48(0)68 4141017http://www.ontp.net___ __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk perms in manager.conf
Any reason whz additional classess are necessary for AstTapi? How to make that secure? ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Tichy Sent: Wednesday, March 22, 2006 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Asterisk perms in manager.conf On Wed, Mar 22, 2006 at 05:54:27AM -0500, David Hajek wrote: [public] secret = private deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/255.255.0.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Lets say I want some users to use dial through manager interface. But don't want to allow them to run asterisk commands? read = write = call That is sufficient, but if you use AstTapi to dial from outlook authorization for additional classes is necessary. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk perms in manager.conf
Title: Asterisk perms in manager.conf Hi, can someone sched a light what exactly mean the read write permissions in manager.conf? [public] secret = private deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/255.255.0.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Lets say I want some users to use dial through manager interface. But dont want to allow them to run asterisk commands? Whats the recommended solution? Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] digium certification for Europe
Title: digium certification for Europe Im little bit confused which Digium hardware is certificated for use in Europe. It looks like new cards are certificated, like TE4XX series. What about TE110 or TDM400P? Can someone confirm that? Thanks, David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP phone with many speed dial buttons
Title: IP phone with many speed dial buttons Hello, Im looking for IP phones with at least 10 or so speed dial buttons. Can you recommend something which works with Asterisk and does not cost fortune? An option can be analog phone combined with ATA adapter. So hints for good analog phones (EU) are welcomed as well. Thank you, -- David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 320 and message retrieve key
Title: Snom 320 and message retrieve key Hi, I found some issues with Snom 320 message retrieve key. This button works only when the MWI does not blink! If MWI blinks and I do press retrieve button I get Unknown on display and busy tone. From the sip debug it looks like that Snom does not send credentials to Asterisk which responds with 407 Proxy Auth required. I have loaded Snom with latest 5 firmware. No change. Im using Asterisk 1.0.9 and have not tried 1.2.X. Looks like this issue is related to http://bugs.digium.com/view.php?id=4801? Does someone get Snom 320 retrieve button working with Asterisk 1.0.9? Thanks, - David Hajek ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voice over atlantic
Hi, have you tried speex? I'm going to give it a shot. I think Speex should be better then gsm. Thanks. -David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk groups Sent: Friday, September 09, 2005 1:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic Hi David, I just looked at my iax.conf on one of my boxes in Argentina and actually there are no jitterbuffer settings indicated so I'm assuming it is using Asterisk defaults. We are experimenting with G.729 on these IAX trunks also and I just realized I have no accurate means of measuring bandwidth consumption vis-a-vis GSM/G.729. I think I'll pose that question to the group in another message to see what recommendations and best practices are out there. Or, do some research. Best of luck. On Thu, 2005-09-08 at 17:49 -0400, David Hajek wrote: Nice. Thanks. What Asterisk version? Can you lookup jitterbuffer settings? Thanks a lot. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice over atlantic
Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? Thanks for any hints. -- David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voice over atlantic
Nice. Thanks. What Asterisk version? Can you lookup jitterbuffer settings? Thanks a lot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk groups Sent: Thursday, September 08, 2005 7:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voice over atlantic David, I have IAX trunks running between the US and S. America using the GSM codec and jitterbuffer=yes and the quality seems very good to my ears. Don't have the details of the jitterbuffer parameters right now but hopefully this will give you some useful feedback. Good luck. On Thu, 2005-09-08 at 16:49 -0400, David Hajek wrote: Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voice over atlantic
Probably missing something here. Never heard of GSM commercial licence for asterisk. Do you have any URLs? Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, September 08, 2005 11:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic Pay the license fee and get the GSM codec would probably be best. The fee is nominal and the codec is a good one... $0.02 W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 1:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voice over atlantic Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? Thanks for any hints. -- David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voice over atlantic
Yep. Thats G729, not GSM. Btw, GSM codec implemented in Asterisk is EFR? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Friday, September 09, 2005 12:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic http://www.digium.com/index.php?menu=product_detailcategory=e xtrasprod uct=G729 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic Probably missing something here. Never heard of GSM commercial licence for asterisk. Do you have any URLs? Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler Sent: Thursday, September 08, 2005 11:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic Pay the license fee and get the GSM codec would probably be best. The fee is nominal and the codec is a good one... $0.02 W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 1:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voice over atlantic Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? Thanks for any hints. -- David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speex codec - Out of buffer space
Hi, I'm running Asterisk 1.0.7 and would like to add Speex support. I downloaded Speex 1.0.5, installed and recompile Asterisk again. Now trying from X-Lite to connect using Speex but getting lot of weird erros on Asterisk console: Sep 7 15:03:25 WARNING[28605]: codec_speex.c:166 speextolin_framein: Out of buffer space I was trying to setup Speex on my second Asterisk server and wanted to use this codec for IAX between these two boxes. But I'm getting unable to negotiate codecs. Other codecs works like a charm. Any ideas? Thank you. -- David ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghanns quadBRI on Dell PowerEdge [SOLVED]
We had two cards in the system - BRI card and analog TDM from Digium. The problem was caused by incorect modules.conf. There was a post-install directive for the TDM card which runs ztcfg. Additionaly I was running ztcfg from the asterisk startup scripts as well - which was the problem. So I can say Junghanns card works in Dell PowerEdge 2800. - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.com David Hajek wrote: Thanks. My cable is like 8-10m long. Hm, will try to make shorter one but it works in old system. Who knows. - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.com Michiel van Baak wrote: On 11:31, Wed 20 Jul 05, David Hajek wrote: Hi, we are trying to install Junghann's quadBRI into Dell PowerEdge 2800 system without success. I don't know if the issue can be that Junghann's card fits 32-bit slot and Dell PE 2800 has only 3 PCI-X 64-bit slots. Can this be an issue? We get CRC errors for HDLC frame when the card is initialized. Any idea what can be wrong? 1/ We use latest bristuff packages. 2/ We use TE mode 3/ Card is working on older 2.4 system, we use same cables and ISDN devices. 4/ On Dell we have a Centos 4.1 with 2.6.12 kernel. After loading the driver we got CRC errors like this: Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Hi, I had the same errors too when I started to test with the 4port card. After changing the 200M UTP cable that was all put in a corner for a 2 meter cable the problems went away. I read on some previous posts from Klaus-Peter that the CRC errors mean bad cables. In my case the way-too-long cable from the NT1 to my * box was the cause. Maybe this can be of any help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Junghanns quadBRI on Dell PowerEdge
Hi, we are trying to install Junghann's quadBRI into Dell PowerEdge 2800 system without success. I don't know if the issue can be that Junghann's card fits 32-bit slot and Dell PE 2800 has only 3 PCI-X 64-bit slots. Can this be an issue? We get CRC errors for HDLC frame when the card is initialized. Any idea what can be wrong? 1/ We use latest bristuff packages. 2/ We use TE mode 3/ Card is working on older 2.4 system, we use same cables and ISDN devices. 4/ On Dell we have a Centos 4.1 with 2.6.12 kernel. After loading the driver we got CRC errors like this: Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Loading qozap driver: Jul 19 17:15:55 ustredna kernel: qozap: no version for zt_receive found: kernel tainted. Jul 19 17:15:55 ustredna kernel: qozap: Junghanns.NET quadBRI card configured at mem 0xf8836000 IRQ 77 HZ 1000 CardID 0 Jul 19 17:15:55 ustredna kernel: qozap: S/T ports: 4 [ TE TE TE TE ] Jul 19 17:15:55 ustredna kernel: qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Running ztcfg: Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: Channel map: Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: Channel 01: Individual Clear channel (Default) (Slaves: 01) Jul 19 17:15:56 ustredna ztcfg: Channel 02: Individual Clear channel (Default) (Slaves: 02) Jul 19 17:15:56 ustredna ztcfg: Channel 03: D-channel (Default) (Slaves: 03) Jul 19 17:15:56 ustredna ztcfg: Channel 04: Individual Clear channel (Default) (Slaves: 04) Jul 19 17:15:56 ustredna ztcfg: Channel 05: Individual Clear channel (Default) (Slaves: 05) Jul 19 17:15:56 ustredna ztcfg: Channel 06: D-channel (Default) (Slaves: 06) Jul 19 17:15:56 ustredna ztcfg: Channel 07: Individual Clear channel (Default) (Slaves: 07) Jul 19 17:15:56 ustredna ztcfg: Channel 08: Individual Clear channel (Default) (Slaves: 08) Jul 19 17:15:56 ustredna ztcfg: Channel 09: D-channel (Default) (Slaves: 09) Jul 19 17:15:56 ustredna ztcfg: Channel 10: Individual Clear channel (Default) (Slaves: 10) Jul 19 17:15:56 ustredna ztcfg: Channel 11: Individual Clear channel (Default) (Slaves: 11) Jul 19 17:15:56 ustredna ztcfg: Channel 12: D-channel (Default) (Slaves: 12) Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: 12 channels configured. Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna zaptel: Running ztcfg: succeeded Thank you, -- - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghanns quadBRI on Dell PowerEdge
Exactly. Junghanns works for me on old PIII system, but it does not work in newest Dell PE. - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.com Altus Snyman wrote: The 1ste pc I tried it on was on a expensive intel board and the second one that worked was on some cheap name board Ill say incompatibility ? Yes, I do use latest bri-stuff package (asterisk 1.0.9 incl) Any ideas? - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.com altus wrote: I had the same problems with a 4 port junghanns and a 4 por wcfxs I took the junghanns out and added it into a new box and all was ok So ether it was because the 2 cards was in together or it was the motherboard? U using the latest driver and asterisk? On Wed, 2005-07-20 at 11:31 +0200, David Hajek wrote: Hi, we are trying to install Junghann's quadBRI into Dell PowerEdge 2800 system without success. I don't know if the issue can be that Junghann's card fits 32-bit slot and Dell PE 2800 has only 3 PCI-X 64-bit slots. Can this be an issue? We get CRC errors for HDLC frame when the card is initialized. Any idea what can be wrong? 1/ We use latest bristuff packages. 2/ We use TE mode 3/ Card is working on older 2.4 system, we use same cables and ISDN devices. 4/ On Dell we have a Centos 4.1 with 2.6.12 kernel. After loading the driver we got CRC errors like this: Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 2 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 3 Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 4 Loading qozap driver: Jul 19 17:15:55 ustredna kernel: qozap: no version for zt_receive found: kernel tainted. Jul 19 17:15:55 ustredna kernel: qozap: Junghanns.NET quadBRI card configured at mem 0xf8836000 IRQ 77 HZ 1000 CardID 0 Jul 19 17:15:55 ustredna kernel: qozap: S/T ports: 4 [ TE TE TE TE ] Jul 19 17:15:55 ustredna kernel: qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Running ztcfg: Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: SPAN 4: CCS/ AMI Build-out: 399-533 feet (DSX-1) Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: Channel map: Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: Channel 01: Individual Clear channel (Default) (Slaves: 01) Jul 19 17:15:56 ustredna ztcfg: Channel 02: Individual Clear channel (Default) (Slaves: 02) Jul 19 17:15:56 ustredna ztcfg: Channel 03: D-channel (Default) (Slaves: 03) Jul 19 17:15:56 ustredna ztcfg: Channel 04: Individual Clear channel (Default) (Slaves: 04) Jul 19 17:15:56 ustredna ztcfg: Channel 05: Individual Clear channel (Default) (Slaves: 05) Jul 19 17:15:56 ustredna ztcfg: Channel 06: D-channel (Default) (Slaves: 06) Jul 19 17:15:56 ustredna ztcfg: Channel 07: Individual Clear channel (Default) (Slaves: 07) Jul 19 17:15:56 ustredna ztcfg: Channel 08: Individual Clear channel (Default) (Slaves: 08) Jul 19 17:15:56 ustredna ztcfg: Channel 09: D-channel (Default) (Slaves: 09) Jul 19 17:15:56 ustredna ztcfg: Channel 10: Individual Clear channel (Default) (Slaves: 10) Jul 19 17:15:56 ustredna ztcfg: Channel 11: Individual Clear channel (Default) (Slaves: 11) Jul 19 17:15:56 ustredna ztcfg: Channel 12: D-channel (Default) (Slaves: 12) Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: 12 channels configured. Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna zaptel: Running ztcfg: succeeded Thank you, -- - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http
Re: [Asterisk-Users] Junghanns quadBRI on Dell PowerEdge
My zaptel.conf: [channels] switchtype = euroisdn ; BRI CARD nationalprefix = 0 internationalprefix = 00 signalling = bri_cpe pridialplan = local prilocaldialplan = local echocancel = yes rxgain=-2 txgain=4 echotraining=yes context=from-isdn usecallerid=yes hidecallerid=no group = 1 channel = 1,2,4,5,7,8,10,11 Yes, quadBRI does not share any interrupts. Do you use this card in Dell PE 2800? I suspect that this card can't work in 64-bit PCI slots? Thanks, David David Hajek wrote: We get CRC errors for HDLC frame when the card is initialized. Any idea what can be wrong? After loading the driver we got CRC errors like this: Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 What are your settings in /etc/zaptel.conf ? | I once had the same issue, make sure your signalling is: signalling = bri_cpe Also, check with # cat /proc/interrupts that the quadBRI is not sharing an irq with something else. Jul 19 17:15:55 ustredna kernel: qozap: S/T ports: 4 [ TE TE TE TE ] Jul 19 17:15:55 ustredna kernel: qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Your card is found, so hardware seems to be okay.. Best regards, Kristof. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghanns quadBRI on Dell PowerEdge
Yes, I tried signalling = bri_cpe_ptmp. When I put the card into older system and use same cables, same ISDN units, same Asterisk configs (but older bristuff!) it works fine. When I put the card into Dell, I got the CRC errors as I wrote before. Maybe someone from Junghanns is watching this thread and can give some help? === Here is my zaptel.conf loadzone=nl defaultzone=nl # qozap span definitions span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 Kristof Hardy wrote: David Hajek wrote: [channels] switchtype = euroisdn ; BRI CARD nationalprefix = 0 internationalprefix = 00 signalling = bri_cpe Ouch, we are talking about zapata.conf I guess :-) But, that all seems okay.. Have you tried signalling = bri_cpe_ptmp instead of bri_cpe? Do you use this card in Dell PE 2800? I suspect that this card can't work in 64-bit PCI slots? We do not use it in a Dell, but on SuperMicro with pci-x slots, so that should work I guess. Keep us posted, in my opinion it has to work :-) Cheers, Kristof. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghanns quadBRI on Dell PowerEdge
Yes, that was zapata.conf - sorry. OK, I can give it a try - but I don't think I have a mistake in confings. But who knows. ;-) Tzafrir Cohen wrote: On Wed, Jul 20, 2005 at 04:38:07PM +0200, David Hajek wrote: My zaptel.conf: /etc/asterisk/zapata.conf, you mean? And I'd like to use this opportunity to introduce a newer version of genzaptelconf. It should basically identify zaphfc and qozap as well as E1 and T1 (hopefully). http://tzafrir.org.il/genzaptelconf [channels] switchtype = euroisdn ; BRI CARD nationalprefix = 0 internationalprefix = 00 signalling = bri_cpe pridialplan = local prilocaldialplan = local echocancel = yes rxgain=-2 txgain=4 echotraining=yes context=from-isdn usecallerid=yes hidecallerid=no group = 1 channel = 1,2,4,5,7,8,10,11 Yes, quadBRI does not share any interrupts. Do you use this card in Dell PE 2800? I suspect that this card can't work in 64-bit PCI slots? Thanks, David David Hajek wrote: We get CRC errors for HDLC frame when the card is initialized. Any idea what can be wrong? After loading the driver we got CRC errors like this: Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 What are your settings in /etc/zaptel.conf ? | I once had the same issue, make sure your signalling is: signalling = bri_cpe Also, check with # cat /proc/interrupts that the quadBRI is not sharing an irq with something else. Jul 19 17:15:55 ustredna kernel: qozap: S/T ports: 4 [ TE TE TE TE ] Jul 19 17:15:55 ustredna kernel: qozap: 1 multiBRI card(s) in this box, 4 BRI ports total. Your card is found, so hardware seems to be okay.. Best regards, Kristof. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Junghanns quadBRI on Dell PowerEdge
Thanks. My cable is like 8-10m long. Hm, will try to make shorter one but it works in old system. Who knows. - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.com Michiel van Baak wrote: On 11:31, Wed 20 Jul 05, David Hajek wrote: Hi, we are trying to install Junghann's quadBRI into Dell PowerEdge 2800 system without success. I don't know if the issue can be that Junghann's card fits 32-bit slot and Dell PE 2800 has only 3 PCI-X 64-bit slots. Can this be an issue? We get CRC errors for HDLC frame when the card is initialized. Any idea what can be wrong? 1/ We use latest bristuff packages. 2/ We use TE mode 3/ Card is working on older 2.4 system, we use same cables and ISDN devices. 4/ On Dell we have a Centos 4.1 with 2.6.12 kernel. After loading the driver we got CRC errors like this: Jul 19 17:02:30 ustredna kernel: qozap: CRC error for HDLC frame on card 1 (cardID 0) S/T port 1 Hi, I had the same errors too when I started to test with the 4port card. After changing the 200M UTP cable that was all put in a corner for a 2 meter cable the problems went away. I read on some previous posts from Klaus-Peter that the CRC errors mean bad cables. In my case the way-too-long cable from the NT1 to my * box was the cause. Maybe this can be of any help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_cornet status
Hi, what is the status of chan_cornet? Does someone here use it in production? I can't find enough info about it. Some URLs will be great. Thank you, -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dropcount
Hi, can someone here explain in detail what exactly the dropcount value represents and how it is calculated? From the help I can read the value of 3 represents of 1.5% frames dropped. What value of 5 means? Is it reasonable to set this to higher value, like 10? Thank you, -- - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 [EMAIL PROTECTED] http://www.systinet.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gsm gateway
Hi, can you recommend cheap GSM Gateway which works with Asterisk? VoiceBlue solution is quite expensive. Thanks, -- - David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dell PowerEdge + TDM
Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell PowerEdge + TDM
Thanks for the reply. I'm more interested in lower series then 2850, like PE SC1420, PE 800. I don't need so much power. ;-) -David Here we have PowerEdge 2850's doing the donky work with a Wildcard TE405P in each. I have seen no operational issues at all with the system or the cards. We are running CentOS 3 as the operating system and the stable version of asterisk The only niggle is that when the cards are modprobed on start up they sometimes 2 in a 100 give an NMI message, causing an error code on the servers little window, its not affected the stability at all, and its on my list of things to do to find out what causes it! The systems generally have around 400 - 500 SIP extensions comming off the back, running around a dual xeon 3Ghz and 3Gb of ram (no transcoding all G711.u) - we are very happy! David On 17/06/05, David Hajek [EMAIL PROTECTED] wrote: Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell PowerEdge + TDM
Do you have analog TDM in it? -David Oswaldo Arratia wrote: I bought a Dell SC1425 and installed a T1/E1 card from Digium and I tried to configure it using [EMAIL PROTECTED] scripts and did not work, so I went the long way and configure with zaptel's instructions and voila! It works like a charm. Oswaldo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Friday, June 17, 2005 8:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dell PowerEdge + TDM Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 behind NAT issue, Vonage is working
I have installed Vovida STUN server and point Sipura to use it. But no luck, I still can't hear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touching the Linksys. How come they can get through it? Any hints? Do you have the NAT Enable and NAT keepalive set to Yes on the Sipura? Yes, I do. I have find out that Sipura works when I set it as DMZ host on the Linksys firewall. Why Vonage can work without any special settings? -David http://hajek.net/blog ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 2000 behind NAT issue, Vonage is working
Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear the other party (to be more precise I can hear first two secs then nothing). So it must be the incoming RTP is blocked on Linksys. Here I think STUN server enters the game and give some help? I have installed Vovida STUN server and point Sipura to use it. But no luck, I still can't hear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touching the Linksys. How come they can get through it? Any hints? -- David Hajek http://hajek.net/blog ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 behind NAT issue, Vonage is working
I have canreinvite=no already, below is my sip.conf entry. [1360] username=1360 callerid=Phone 1 1360 secret=mysec1 host=dynamic auth=md5 qualify=1000 dtmfmode=rfc2833 context=from-sip-unrestricted mailbox=1360 type=friend disallow=all allow=ilbc allow=g729 allow=gsm allow=g726 nat=yes canreinvite=no - David Hajek http://hajek.net/blog Rich Adamson wrote: I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear the other party (to be more precise I can hear first two secs then nothing). So it must be the incoming RTP is blocked on Linksys. Here I think STUN server enters the game and give some help? I have installed Vovida STUN server and point Sipura to use it. But no luck, I still can't hear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touching the Linksys. How come they can get through it? Any hints? Add canreinvite=no to the sipura def's in sip.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and CAS
Hi, is it possible to use Asterisk with T110P and CAS (channel associated signalling)? Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Many analog lines
Hi, how to use Asterisk where I need to have lets say 40 analog lines. Any ideas? Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk ChangeLog
Hello, is it possible Asterisk's ChangeLog will contain a reference to appropriate bug number in bugzilla? This can be very handy. Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registration issues with Sipura SPA-841
I have no such issues with Sipura/Asterisk. I don't think this is common to Sipura. Try Asterisk's SIP debug to see what happens when registration fails. -D GliTcH wrote: This is a common issue with all Sipura devices I've seen. I set the registration interval to 5 minutes, so that NAT doesn't interfere. I've done this with Cisco IP Phones, Cisco ATA Converters, and Sipura SPA-1001, SPA-2000, SPA-2100, and SPA-841's. All Sipura's have the same issue, 1 out of every 10 or so registrations will fail. I will suddenly see a Registration failed notice in the Asterisk console, 5 minutes later the phone will go to register, and it works fine. The right userid and password is programmed into the phone. What this leaves is a 5 minute gap when the phone can not be called, if somebody calls that Sipura device within that 5 minutes, they will go straight to voicemail. It's damned annoying. I've tried this on several different interenet connections (At home (where I have DSL and Cable), at work, at my friends home, etc.) and with multiple Asterisk servers and different providers. I've also confirmed this with a sister company we helped build, they've seen the same issues w/ Sipura's, they contacted Sipura about it and Sipura said it was a known issue they were working to resolve, but no ETA. Go figure. I was thinking about going to Linksys's PAP2-NA but I don't know much about it, except they're glorified Sipura's and may potentially have the same issue w/ Asterisk. I don't like how Linksys doesn't support them w/ end users, doesn't offer firmware, makes them a pain to obtain, etc. etc. I'm trying to investigate going to a different manufacturer, but I don't like the Cisco ATA-186's very much and they're too pricey, so I don't know where to go next. voipsupply has a pretty big collection, maybe I'll order 1 of each for testing. We're about to deploy 150 converters to residential and business users and I can't even find one I like, jeesh. Don't say the IAXY, I only use IAX for Asterisk-Asterisk communication, not for end users. On Fri, 18 Mar 2005 16:41:55 -0800, Wolfgang S. Rupprecht [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] (Iassen Hristov) writes: Anyone having problems with registration to * from a SPA-841? I have a spa-841 (firmware 3.1.1a) on my desk right next to me and it registers just fine w. ~current asterisk (CVS-HEAD-03/16/05-08:43:40). The one problem I do notice that the phone is very touchy about the cat-5 ethernet cable one uses. The one that came with the phone works just fine (of course). Any of my 5 longer store-bought cat5e cables don't work at all. Tcpdump shows the phone registering and asterisk answering, but the phone never hears the reply. Might you be seeing something like this? -wolfgang ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] international calls and NOANSWER
Hello, I'm doing lot of international calls via Sixtel and VoipJet. And there are some calls which do not go through - Asterisk immediatelly returns with NOANSWER. And it is not because the dialed party does not pickup the phone, it is because the call does not go through the provider. I've written a dial macro which route the call via second provider if the first returns CHANUNAVAIL, but I don't know how to handle NOANSWER when it is actually CHANUNAVAIL... Any ideas? Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoipJet issues?
Whats up to VoipJet.com? Their DNS servers are not reachable. Both primary and secondary are on the same subnet - weird setup. Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoipJet issues?
Anyway, they're not reachable since yesterday evening. -D Joe Greco wrote: Whats up to VoipJet.com? Their DNS servers are not reachable. Looks like their provider is maybe having problems. AS3728, onr.com, Onramp. Both primary and secondary are on the same subnet - weird setup. While that might be true, it also might not be. 206.55.64.64 and 206.55.64.65 are not on the same network or even the same city, for example (they used to actually be in different states). We use OSPF internally and those addresses are not on any Ethernet network. They're loopback interfaces. They can be moved around. In the case you're talking about, it's *likely* they're on the same network, and that's not good, of course. Those pesky rules about diversity of nameservers exist for a reason. ... JG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voice delay after call setup, outgoing calls
Hi, I'm experiencing some voice delay (2-3 sec) after outgoing call is setup. It means during the first 2-3 secs, audio is very choppy or nothing. So usually I can't hear the 'Hello. I use IAX2 for my ougoing calls with Grandstream phone as a client. Any hints to prevent this? Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] grandstream MWI?
Hello, it is possible to get MWI working with Grandstream and Asterisk? Thanks. -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] grandstream MWI?
Actually, I got the display flashing when I have a new message. But it is possible to get the Grandstream's Message button working? My goal is to pickup earphone and press Message button to retrieve my messages. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Monday, December 20, 2004 3:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] grandstream MWI? Hello, it is possible to get MWI working with Grandstream and Asterisk? Thanks. -David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] grandstream MWI?
Doug, thanks for the info. I'm curious where you get the BETA from? ;-) I sent a notice to Grandstream support anyway. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Monday, December 20, 2004 4:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] grandstream MWI? David Hajek wrote: Actually, I got the display flashing when I have a new message. But it is possible to get the Grandstream's Message button working? My goal is to pickup earphone and press Message button to retrieve my messages. David, I have both the message button and the MWI working under BETA 1.0.5.18 firmware. Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] first 2-3 secs choopy sound
Hello, I'm experiencing choppy sound first 2-3 secs of every call. Looks like there is some more handshaking between Asterisk and other party. After that voice quality is pretty good. Using IAX2. Any hints? Thanks. David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip trunking works?
Hello, I'm about to connect asterisk with Alcatel Enterprise PBX using SIP trunking, I can't find if Asterisk has this capability. Can you please advice? Thank you. -David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Max retries exceeded with voiceconnect
Hello,
currently I'm unable to make outgoing calls via voiceconect using IAX. I
have setup as following:
[outgoing]
exten =
_1NXXNXX,1,Dial(IAX2/MY_DEVICE_LOGIN:[EMAIL PROTECTED]
lse.com/${EXTEN})
exten =
_1NXXNXX,102,Dial(IAX2/MY_DEVICE_LOGIN:[EMAIL PROTECTED]
pulse.com/${EXTEN})
but getting error Max retries exceeded to host 66.234.228.160
(gwiaxt01.voicepulse.com). After that asterisk hangups, without checking
second voicepulse gateway.
I'm curious if it is possible to setup outgoing dialplan so it prevents
against Max retries.. error and follow the second voicepulse gateway
(gwiaxt02.voicepulse.com).
Any ideas?
Thanks,
David
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RE: [Asterisk-Users] Max retries exceeded with voiceconnect
What are you using now?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Totaro
Sent: Friday, November 05, 2004 9:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Max retries exceeded with voiceconnect
I canceled my account with voicepulse and ate the cost of
printed materials with the DID they supplied. I gave up on them.
- Original Message -
From: David Hajek [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Friday, November 05, 2004 2:59 PM
Subject: [Asterisk-Users] Max retries exceeded with voiceconnect
Hello,
currently I'm unable to make outgoing calls via voiceconect
using IAX. I
have setup as following:
[outgoing]
exten =
_1NXXNXX,1,Dial(IAX2/MY_DEVICE_LOGIN:[EMAIL PROTECTED]
iaxt01.voicepu
lse.com/${EXTEN})
exten =
_1NXXNXX,102,Dial(IAX2/MY_DEVICE_LOGIN:MY_DEVICE_PASSWORD@
gwiaxt02.voice
pulse.com/${EXTEN})
but getting error Max retries exceeded to host 66.234.228.160
(gwiaxt01.voicepulse.com). After that asterisk hangups,
without checking
second voicepulse gateway.
I'm curious if it is possible to setup outgoing dialplan so
it prevents
against Max retries.. error and follow the second
voicepulse gateway
(gwiaxt02.voicepulse.com).
Any ideas?
Thanks,
David
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[Asterisk-Users] Alcatel Enterprise
Hello, does someone have experience with connecting asterisk to Alcatel OmniPCX Enteprise phone switch? Alcatel can do SIP, so it sounds like it should work with Asterisk, but maybe some of you use it in production? Thanks, David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Nortel BCM
Hello, does anyone has an experience with connecting Asterisk to Nortel's BCM (http://www.nortelnetworks.com/products/01/eedge/bcm.html)? I would like to make this working using some voip protocol IAX, SIP, but it looks like Nortel's can't do that? My scenario is Nortel's BCM in central office and asterisk installations in satellites offices. Thanks, David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Nortel BCM
Thanks for your answer. We don't have to use Nortel's BCM, it is one of the option we're considering (not sure if it is still in the game now). I will ask this way, what commerical fullvoip PBX you will recommend? Unfortunatelly I can't use asterisk for this central point, but I can (and will) use asterisk on satellites offices. Can you please give some hints what vendors/makers I should not forget? 3com looks promising Thanks. -David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Van Meggelen Sent: Friday, October 29, 2004 4:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk with Nortel BCM Use H.323 and in the BCM set the protocol to Other. Do you HAVE to use the BCM? It's a really horrible system. I worked for many years in tech support, and I've been involved in BCMs since the beta trials of version 1.0, four years ago. I know BCM, and I can tell you that it is one of the worst telephone systems ever produced. Check out the spec sheet: - The operating system is Windows NT 4.0 -- no really, an EIGHT YEAR OLD OPERATING SYSTEM. - The MSC card is a Norstar KSU that they put on a PCI card. That's FIFTEEN YEAR OLD technology - and it shows. - The platform is an Intel Pentium III 700Mhz, with 256megs of RAM, and a 20meg hard drive. How much do they want you to pay for it? - Many of the critical scripts in the system are DOS batch files (I am NOT kidding!). The BCM is famous for it's instability (go figure), and mind-numbingly stupid interface. Unless you have a lot of money to waste on obsolescence, I'd remove the BCM completely from the equation. If you have to go Nortel, go with a Succession (even a Norstar would be a more stable choice, and you can tie it into a VoIP gateway with PRI trunks). You might want to consider not using Nortel's VoIP technology at all -- I don't think they fully understand VoIP yet. Better would be to tie any Nortel gear into your VoIP network using legacy trunking through, say, an Asterisk gateway, like this: [NT PBX/KSU]---PRI---[Asterisk]=(WAN cloud)=[Asterisk] I wouldn't use the BCM as a boat anchor, but for sure it should NEVER be used as the core of a VoIP network - it's just a key system, and not a very good one at that! Good luck! [EMAIL PROTECTED] wrote: Hello, does anyone has an experience with connecting Asterisk to Nortel's BCM (http://www.nortelnetworks.com/products/0 1/eedge/bcm.html)? I would like to make this working using some voip protocol IAX, SIP, but it looks like Nortel's can't do that? My scenario is Nortel's BCM in central office and asterisk installations in satellites offices. Thanks, David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/ asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] contexts based on time and date
Hello, I need to include many contexts based on time and date. But I have a so called midnight issue. I want to include context when time is between 10pm,Oct 21 - 3am,Oct 22. Is it possible to write it using one line? include = context1|22:00-00:00|*|21|Oct include = context1|00:00-03:00|*|22|Oct Thanks. -David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LDAP synchronization script
Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some share one with me/us? Thanks, -D ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LDAP synchronization script
I think I'll use something from this article - http://www.marko.net/asterisk/archives/0205/0006.html -David -Original Message- From: Stefan de Konink [mailto:[EMAIL PROTECTED] Sent: Thursday, June 17, 2004 1:12 PM To: David Hajek Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LDAP synchronization script I'm planning to incorporate this (native and dynamic) LDAP for my own system on short term. Do you have any LDAP design in mind? Stefan On Thu, 17 Jun 2004, Jeremy Jones wrote: David Hajek Sent: Thursday, June 17, 2004 2:41 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] LDAP synchronization script Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some share one with me/us? Not aware of any scripts like that, but... you could use the odbc support in asterisk in conjunction with some slick odbc-ldap connectivity. Jeremy Jones ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] where can I get toll-free number?
Hello, I'm running Asterisk and using VoicePulse for IAX termination. I would like to have toll-free number assigned to my asterisk, any hints where I can get this number? VoicePulse does not offer toll-free numbers. Thanks, David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
Is there english version of their sipgate.de website?
-D
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Birk Bremer
Sent: Friday, February 27, 2004 7:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anybody managed to call a phone
through sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi David,
no the number after the slash is necessary (and yes this is
my number) Without that slash/number I'm not able to get a
call anymore.
But thanks
Birk
David J Carter wrote:
| Hi,
|
| I would be tempted to get rid of the slash and number on
the register
line,
| unless your asterisk extension is 02115800.
|
| dave
|
| -Original Message-
| From: [EMAIL PROTECTED]
| [mailto:[EMAIL PROTECTED] Behalf Of
Birk Bremer
| Sent: 27 February 2004 16:47
| To: [EMAIL PROTECTED]
| Subject: [Asterisk-Users] Anybody managed to call a phone through
| sipgate.de
|
|
| Hello everybody,
|
| has anybody managed to call a (old fashioned) phone using
Sipgate.de
| and asterisk? (yes I have money on my account :-) )
|
|
| The configuration I got from the sipgate.de people is at
the botton of
| the mail
|
|
| Here is mine:
|
| sip.conf:
|
| register = 800:[EMAIL PROTECTED]/02115800
|
| [sipgate]
| type=friend
| username=800
| secret=SECRET
| host=sipgate.de
| fromuser=800
| fromdomain=sipgate.net
| nat=no
| ;dtmfband=3Dinband
| context=sipin
| canreinvite=no
|
|
| extension.conf:
| exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
|
| To be called on my sipgate number - no problem
|
| If I want to call somebody I get the following error:
|
| When I call a number directly out of the softphone:
| Executing Dial([EMAIL PROTECTED]/2,
SIP/[EMAIL PROTECTED]|30|tr)
| in new stack
| ~-- Called [EMAIL PROTECTED]
| ~-- Got SIP response 403 Forbidden back from 217.10.79.9
| ~ == No one is available to answer at this time
| ~-- Hungup '[EMAIL PROTECTED]/2
|
|
|
| when I use the webinterface at sipgate.de I get a ring at my
| softphone, when I pick the call I get the message (in the appearing
| box) Teilnehmer nicht gefunden - User/Number not found
|
| sometimes (while tried different config. I also got (at *
console) to
| many hops...
|
|
| Has anybody managed this - can you please send me your
configuration
| (sip, extensions) or can anybody help
|
| Thanks in advance
|
| Birk Bremer
|
|
|
|
|
| The configuration the sipgate people suggest:
|
| ~ register = 800:[EMAIL PROTECTED]/800
| ^ can't be correct
| |
| |
| |
| | [sipgate]
| |
| | type=friend
| |
| | username=800
| |
| | secret=sipgatepasswort
| |
| | host=sipgate.de
| |
| | fromuser=800
| |
| | fromdomain=sipgate.net
| |
| | nat=yes
| |
| | ;dtmfband=inband
| |
| | context=incomingsipgate
| |
| | canreinvite=no
| |
| |
| |
| | Aus der extensions.conf :
| |
| |
| |
| | [incomingsipgate]
| |
| | exten = h,1,Hangup
| |
| | exten = 800,1,Dial(SIP/internestelefon,20,tr)
| |
| |
| |
| | [sipgate]
| |
| | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| |
| | exten = _9.,2,Playback(invalid)
| |
| | exten = _9.,3,Hangup
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