Re: [Asterisk-Users] Asterisk goes international :-)
Hi, I'm from Spain and I have developed in Perl 2 scripts to say_number and say_digits in 3 diferents language (spanish, german and english). The problem is that I don't know how to adapt it to C in order to complement say.c If somebody can help me, I will be very pleasured. Thanks a lot. - Mensaje Original - De: Olle E. Johansson [EMAIL PROTECTED] Fecha: Jueves, Abril 29, 2004 8:49 am Asunto: Re: [Asterisk-Users] Asterisk goes international :-) Altus Snyman wrote: So what do I have to do to add South-Africa to this list? If you are saying numbers in a different way than english or if you are thinking about another language, check the latest version of say.c in CVS head and see if you can construct the syntax needed for your language. When you have a patch, open a bug report under Internationalization in http://bugs.digium.com and add the patch there. I don't know how Asian languages work, but it would certainly be interestingto see patches for those. Also, we need a larger group that works with the next generation architecture. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinf To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinf ___ 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] * Newbie installation advice
Hello, I'm about to install asterisk as the PBX at a location that my company has just moved into and I would like to get some comments and advice on the installation. I am new to * and don't want to make any big mistakes so I would love to hear whatever anyone has to say. Here is what I have so far Server: * 2.8Ghz P4 - 1G ram * T400P Tormenta II (is this as good as the wildcard?) Chanel Bank: * Adit 600 3FXS, 1FXO * We will start with 6 PSTN lines Phones: * Aastra PowerTouch 480 (Management, Customer Service etc) * Aastra Meridian 8004 (break room, warehouse floor etc) * Reception? Any comments or suggestions would be appreciated as I have no idea what type of phone to give reception. Reception typically has a multi-line phone to answer incoming calls. * Polycom SoundStation 200 EX (Conference room speaker phone) Am I missing anything? I see from the archives that a lot of people have used the PowerTouch phones. What do people think of them? Are there other ADSI phones that are better or just as good for less money? Thanks -Jon -- Jon J. Brandon [EMAIL PROTECTED] http://www.monsoonretail.com ___ 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] Best echo-free and trouble-free system?
On Fri, Apr 30, 2004 at 10:28:02AM -0500, Barton Hodges wrote: [EMAIL PROTECTED] wrote: The real problem arises when : - you have some echo induced somewhere (your call goes through a 2 wire line) - you have some delay induced somewhere (you use VoIP for instance) Following the 2-wire to 4-wire causes echo thought, the following should not result in noticable echo, true? - Analog phone - TDM10B-FXS - Asterisk - TDM01B-FXO - PSTN Yes, because although echo will exist, delay should be short enough so that you don't notice. Never tried such a setup myself, though. Please furthermore note that Asterisk uses pseudo TDM. In real telco world, PCM highways that interconnect trunks and devices switch one byte every 8000th/sec. OTOH, Zaptel devices switch eight bytes every 1000th/sec. This is due the to PC bus architecture (it would cause way too much overhead otherwise). So the delay is actually 8 times longuer (at least) than in the PSTN. - VOIP Phone - Asterisk - VOIP Phone - VOIP Phone - Asterisk - T100P - PRI Nobody's supposed to generate echo on VoIP phones. However, the PRI side will probably connect to a 2-wire PSTN set at the remote end, so you will get echo from there. However, the following could result in noticable echo (as I am experiencing): - Analog phone - ATA - Asterisk - TDM01B-FXO - PSTN - VOIP Phone - Asterisk - PSTN Definetly. Although Asterisk (zaptel, actually) make a fairly good job at cancelling it. What about the following as described in Raymond McKay's setup (Thank you Raymond) Does the channel bank provide the needed, and adequate echo cancellation? I don't have any experience with channels banks. Not very common stuff in Europe. Since you state that echo cancellation needs to be performed closest to the source, could the Grandstream HandyTone-286 be doing an inadequate job of echo cancellation? If this is the case, does anyone have experience with another ATA (Sipura SPA-1000, Cisco ATA-186, etc.) that does such a great job of echo cancellation, that the 2-wire to 4-wire situation is not an issue? Does Grandstream have improved echo cancellation scheduled for a future firmware upgrade? The Sipuras are definetly better than the HandyTones. I've heard that the forthcoming GS firmwares will enhance echo cancellation performance, though. -- Nicolas Bougues Axialys Interactive ___ 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] GrandStream 1.0.4.55 Firmware
On Fri, Apr 30, 2004 at 06:13:49PM +0100, Senad Jordanovic wrote: [EMAIL PROTECTED] wrote: Go to 1.04.54. This is pretty stable. Find it at www.telappliant.com/grandstream Does this version supports TFTP auto configuration? If it does, please contact me off the list for volume purchase discussion! Err, all (1.0.4.x at least) GS firmwares support TFTP autoconfiguration ! -- Nicolas Bougues Axialys Interactive ___ 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] * Newbie installation advice
On Mon, 2004-05-03 at 01:06, Jon Brandon wrote: Hello, I'm about to install asterisk as the PBX at a location that my company has just moved into and I would like to get some comments and advice on the installation. I am new to * and don't want to make any big mistakes so I would love to hear whatever anyone has to say. Your first mistake _may_ be the rush to learn. Good experiences normally require you to have a time to get used to the application and PSTN problems before you attempt to go through a roll out. Here is what I have so far Server: * 2.8Ghz P4 - 1G ram * T400P Tormenta II (is this as good as the wildcard?) Chanel Bank: * Adit 600 3FXS, 1FXO * We will start with 6 PSTN lines If you are going to start with 6 lines, you should decide how soon you might upgrade. You then should look into the cost difference to get either channelized T1 or PRI. You will be much happier with a T1 than analog lines. Specifically look at how many people here fight with echo, a T1 makes the risks of echo lower. It also becomes cheaper as the number of lines go up than analog lines. At some point in your growth, if you continue with analog lines, the telco will drop a similar Adit right next to yours to break the T1 they bring in out to the analog lines you order. Phones: * Aastra PowerTouch 480 (Management, Customer Service etc) * Aastra Meridian 8004 (break room, warehouse floor etc) * Reception? Any comments or suggestions would be appreciated as I have no idea what type of phone to give reception. Reception typically has a multi-line phone to answer incoming calls. Multiline isn't necessary. For that matter, a receptionist isn't overly necessary. You can help direct callers to a extension pretty easy with a menu system you script. Then you just need to designate a couple of people/phones that are used in the case the caller refuses to follow the menu or doesn't find the person they are looking for. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] South-Africa
My advice is just sell them. no-one I know is bothered with Icasa approval, as long as it works, its fine. That card has FCC approval, as far as I know. ALles van die beste! Regards Clive On Fri, 30 Apr 2004 15:17:13 +0100 WipeOut [EMAIL PROTECTED] wrote: Altus Snyman wrote: Good day all I'm in South-Africa,currently we are using openline4 cards for our pbx systems.Now we first need approval on the cards form icasa(a government standards) before we can use the card.The market here is very big for a system like asterisk.The only problem is to get a card approved(for a small company like us) its just about impossible. Now what I'm looking for is a company that will import an approve a card or if someone out of South-Africa now of such a card? The market is very big here Let me Know Thanks Altus Just don't tell anyone.. ;) We tried getting Modems approved in SA about 8 years ago and in the end it just wasn't worth it.. The regulators were a joke and their costs were rediculous.. It may have improved now.. Good luck.. ___ 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 _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ 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] * Newbie installation advice
On Mon, 3 May 2004, Steven Critchfield wrote: On Mon, 2004-05-03 at 01:06, Jon Brandon wrote: Hello, I'm about to install asterisk as the PBX at a location that my company has just moved into and I would like to get some comments and advice on the installation. I am new to * and don't want to make any big mistakes so I would love to hear whatever anyone has to say. Your first mistake _may_ be the rush to learn. Good experiences normally require you to have a time to get used to the application and PSTN problems before you attempt to go through a roll out. Yes... I have thought about this a lot. I do have experience with traditional PBX's, Tadiran and Panasonic to be specific, so that would be the safe route. * however is very exciting and there seems to be lots of help available. Here is what I have so far Server: * 2.8Ghz P4 - 1G ram * T400P Tormenta II (is this as good as the wildcard?) Chanel Bank: * Adit 600 3FXS, 1FXO * We will start with 6 PSTN lines If you are going to start with 6 lines, you should decide how soon you might upgrade. You then should look into the cost difference to get either channelized T1 or PRI. You will be much happier with a T1 than analog lines. Specifically look at how many people here fight with echo, a T1 makes the risks of echo lower. It also becomes cheaper as the number of lines go up than analog lines. At some point in your growth, if you continue with analog lines, the telco will drop a similar Adit right next to yours to break the T1 they bring in out to the analog lines you order. Okay this is a great suggestion. Echo is not something I have not had to deal with before. Is this a problem that would steer a person away from this type of PBX. Phones: * Aastra PowerTouch 480 (Management, Customer Service etc) * Aastra Meridian 8004 (break room, warehouse floor etc) * Reception? Any comments or suggestions would be appreciated as I have no idea what type of phone to give reception. Reception typically has a multi-line phone to answer incoming calls. Multiline isn't necessary. For that matter, a receptionist isn't overly necessary. You can help direct callers to a extension pretty easy with a menu system you script. Then you just need to designate a couple of people/phones that are used in the case the caller refuses to follow the menu or doesn't find the person they are looking for. Excellent :) this is exactly why I asked. Thanks Steven -- Jon J. Brandon [EMAIL PROTECTED] http://www.monsoonretail.com ___ 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 MGCP / NCS
Hi everybody, I have a MTA from Terayon that I try to make run with Asterisk using MGCP channel. The device is running with MGCP 1.0 NCS 1.0 Each time Asterisk try to send a Request (Request Notify, Audit Endpoint) the device returns error 510 Protocol Error Does anybody have already meet this problem and provide me support to make run it ?! (I have already try to change header into the chan_mgcp.c from MGCP 1.0 to 'MGCP 1.0 NCS 1.0 but always the same result) Thanks in advance Ignace CARIA ___ 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] * Newbie installation advice
On Mon, 2004-05-03 at 02:30, Jon Brandon wrote: On Mon, 3 May 2004, Steven Critchfield wrote: On Mon, 2004-05-03 at 01:06, Jon Brandon wrote: * We will start with 6 PSTN lines If you are going to start with 6 lines, you should decide how soon you might upgrade. You then should look into the cost difference to get either channelized T1 or PRI. You will be much happier with a T1 than analog lines. Specifically look at how many people here fight with echo, a T1 makes the risks of echo lower. It also becomes cheaper as the number of lines go up than analog lines. At some point in your growth, if you continue with analog lines, the telco will drop a similar Adit right next to yours to break the T1 they bring in out to the analog lines you order. Okay this is a great suggestion. Echo is not something I have not had to deal with before. Is this a problem that would steer a person away from this type of PBX. Echo seems to be a problem with VoIP situations mostly. It also creeps into analog PSTN links. It isn't a reason to not choose asterisk, but it does require some thought. If you take a poll of us that have deployed already, those who have analog on the PSTN side of asterisk seem to have a worse time with echo than those with digital on the PSTN side. It seems to be even worse when you put VoIP in the mix. My office has been using asterisk with T1 for almost 2 years now with little to no echo problems. You will also have better chances of making sure your fax machines work if there is only one analog to digital conversion on your end. This is all reasons to see if you can justify the added costs of the T1 line, and just think, the time to turn up additional lines is fairly minimal once you have the T1 line running. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] problems with bri-stuff.0.0.2rc20a
hi klaus-peter, yepp... with overlapdial=yes (almost) everything works great, again. one problem is left... touchtones are not working anymore so I can't use voicemail-system, parking and stuff. thank you for your help. ...bye thorsten - Original Message - From: Klaus-Peter Junghanns [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 03, 2004 12:35 AM Subject: Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a Hola, if you have overlapdial=no in zapata.conf then * will jump into the s extension on a NT span (this way you can use DigitTimeOut and ResponseTimeOut to make patterns like _X. work as expected.). So, either you create an s extension, e.g.: exten = s,1,DigitTimeOut(3) or you set overlapdial=yes in zapata.conf. best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ ___ 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] problems with bri-stuff.0.0.2rc20a
forget it... seems to work - no idea what was/is wrong. - Original Message - From: FastJack [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 03, 2004 11:38 AM Subject: Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a hi klaus-peter, yepp... with overlapdial=yes (almost) everything works great, again. one problem is left... touchtones are not working anymore so I can't use voicemail-system, parking and stuff. thank you for your help. ...bye thorsten ___ 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] Talking SIP to Vocal
Andres wrote: I think the username/secret items in sip.conf are busted. A quick ethereal trace shows that when placing an outbound call to another provider via SIP, * is not using the username defined during the authentication challenge, instead it uses the username of the phone placing the call. A rollback to CVS of a week ago fixes the issue. I did another CVS update and rebuild last night... and outgoing SIP authentication appears to work correctly now. Did someone fix the problem? Cheers, Mark. ___ 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] (no subject) MGCP
Il 21:38, venerdì 30 aprile 2004, Philipp von Klitzing ha scritto: Hi! I try to connect an MGCP device(Terayon) to asterisk. I have found many example BUT the Terayon always return error 510 ! Verb:'510' Identifiers :'2' Endpoint: 'Error' Version'(null)' 1. Which version of Asterisk exactly (!) are you using? 2. Try CVS-03/05/04-00:50:56 instead and see if that solves your problem. For me recent CVS has made using MGCP completely impossible (with Swissvoice ip10 having been upgraded to newer firmware) 3. Look at the MGCP bugs on bugs.digium.com to find out if you find a related issue. Add your comment plus debugging info there, or create a new bug. Cheers, Philipp So do I, I've got cisco ata 186 with MGCP. Last versions of asterisk are very unusable with MGCP! ___ 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] Digital Line Distortion
Firstly, the problem... Ever since I installed and setup asterisk, I have had various problems, initially it was echo caused by the ISDN (isdn4linux) card I was using. So, I upgraded to the X101P from digium. I still had echo, so I figured it was also caused by the ATA186 (cisco) I was using. So, I upgraded again to the TDM40B quad FXS card. This solved pretty much all my problems, except eventually, I needed more incoming lines. So, again, I upgraded to a digital line (10 channel PRI/E1) and purchased the brand new TE405p from digium... Now, eventually I got this working properly, for incoming and outbound calls, I have incoming callerid working, etc... However, ever since I did this, I continually get complaints from people about how terrible my phone lines are. Not *everyone* complains, but most people do Also, I hear the same problem when calls are 'diverted' to my mobile phone. ie, call arrives, and is then connected to a second channel back to my mobile. I still have my X100p installed, and don't have this problem with calls between the x100p and the tdm40b, only with calls between the te405p and the tdm40b and calls with both legs on the te405p. Here are some dumps from my configs/system/etc asterisk*CLI show version Asterisk CVS-04/13/04-18:00:23 built by [EMAIL PROTECTED] on a i686 running Linux [EMAIL PROTECTED]: ~# cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 6 model : 8 model name : AMD Athlon(tm) stepping: 1 cpu MHz : 1161.462 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 mmx fxsr sse syscall mmxext 3dnowext 3dnow bogomips: 2313.42 [EMAIL PROTECTED]: ~# lspci 00:00.0 Host bridge: VIA Technologies, Inc.: Unknown device 3116 00:01.0 PCI bridge: VIA Technologies, Inc. VT8633 [Apollo Pro266 AGP] 00:05.0 Communication controller: Xilinx, Inc.: Unknown device 0314 (rev 01) 00:06.0 Communication controller: Tiger Jet Network Inc. Model 300 128k 00:07.0 Network controller: Tiger Jet Network Inc. Model 300 128k 00:0e.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C (rev 10) 00:11.0 ISA bridge: VIA Technologies, Inc. VT8233A ISA Bridge 00:11.1 IDE interface: VIA Technologies, Inc. Bus Master IDE (rev 06) 01:00.0 VGA compatible controller: S3 Inc.: Unknown device 8d04 [EMAIL PROTECTED]: ~# cat /proc/interrupts CPU0 0: 683565744IO-APIC-edge timer 1:874IO-APIC-edge keyboard 2: 0 XT-PIC cascade 8: 1IO-APIC-edge rtc 14:4972250IO-APIC-edge ide0 15: 15IO-APIC-edge ide1 16: 2539525986 IO-APIC-level t4xxp 17: 2543170795 IO-APIC-level eth0, wcfxo 18: 2539516334 IO-APIC-level wctdm NMI: 0 LOC: 683553571 ERR: 0 MIS: 0 Now, some asterisk config files: [EMAIL PROTECTED]: ~# cat /etc/zaptel.conf |grep -v ^\# span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17-31 dchan=16 fxsls=125 fxoks=126 fxoks=127 fxoks=128 fxoks=129 loadzone = au defaultzone=au [EMAIL PROTECTED]: ~# cat /etc/asterisk/zapata.conf |egrep -v ^$\|^\; [channels] context=default signalling=fxo_ls usecallerid=no hidecallerid=no callwaiting=no callwaitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=no rxgain=0.0 txgain=0.0 busydetect=no pridialplan=local nationalprefix=0 internationalprefix=0011 usecallerid=yes hidecallerid=no callwaiting=no callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes echotraining=yes callerid=asreceived adsi=no busydetect=no callprogress=no switchtype = euroisdn signalling = pri_cpe callgroup = 1 group = 2 immediate = no context = remote channel = 1-10 immediate = yes usecallerid=no callerid=no group = 10 signalling = fxs_ls channel = 125 threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes usecallerid=yes hidecallerid=no callwaitingcallerid=yes context = inside immediate = no signalling = fxo_ks pickupgroup = 1 callgroup = 1 group = 3 callerid=Adam 601 mailbox=601 channel = 126 callerid=Doris 600 mailbox=600 channel = 127 callerid=Sales 603 mailbox=603 channel = 128 callerid=Technical 602 mailbox=602 channel = 129 So, you can see the te405p, followed by the x101p and finally the tdm40b If anyone can help me resolve this terribly annoying problem, I'd be most appreciative... I can't think of any other information which is helpful, but please let me know if you think something else will clearly show the problem (like a pri debug etc...) Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED]
[Asterisk-Users] AGI question
Hello, I'm using an AGI program written in C to manage incoming calls to some extensions. Its being used for a small call center (20 people). When the call comes in, the caller can listen the directory menu and then dial the extension. The AGI program is called and get one of the available extension to dial. After dialed, people start conversation up to a moment where the call hangs up and the caller goes to the start extension (s). It happens just sometimes and not for the same person. Sometimes happen a lot and sometimes happen once. What you guys think about this? I'm currently using the Asterisk version (Asterisk CVS-09/10/03-18:47:18). And I also use cdr_mysql for billing.. thank you Oz ___ 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] AGI question
Hello, Can we see your dialplan related to that ? On Mon, 2004-05-03 at 13:40, Osvaldo Mundim wrote: Hello, I'm using an AGI program written in C to manage incoming calls to some extensions. Its being used for a small call center (20 people). When the call comes in, the caller can listen the directory menu and then dial the extension. The AGI program is called and get one of the available extension to dial. After dialed, people start conversation up to a moment where the call hangs up and the caller goes to the start extension (s). It happens just sometimes and not for the same person. Sometimes happen a lot and sometimes happen once. What you guys think about this? I'm currently using the Asterisk version (Asterisk CVS-09/10/03-18:47:18). And I also use cdr_mysql for billing.. thank you Oz ___ 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
Re: [Asterisk-Users] Digital Line Distortion
Damn, I forgot to describe the actual problem. Basically as someone I spoke to today described it, it sounds like you have one of those new digital pbx systems... In more detail, when he spoke, he heard his voice come back, but distorted. The louder the sound he made, the louder he heard himself (distorted). At all times, if I am on the tdm40b side, I hear 100% perfect audio quality in both directions. (Which is bad, because now the customer gets the bad sound, before it was just the staff...) Regards, Adam On Mon, 2004-05-03 at 21:28, Adam Goryachev wrote: Firstly, the problem... Ever since I installed and setup asterisk, I have had various problems, initially it was echo caused by the ISDN (isdn4linux) card I was using. Now I'll do the thing most people forget about... [SNIP] the rest of the quoted text! Regards, Adam ___ 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] If Then Else Statements - Outbound Dialling on ISDN using CAPI -Individual Dial out Plans using msns
Hi All, Many thanks to Marc who helped me with a previous Capi Dialout plan - however. What I now would like to be able to do is: - We have 8 msn's 383590, 383591 383592 etc. What I would like to do is set up an If Then Else type statement along the following lines: - If extension 7957 Then Dialout on Capi msn 383590 ElseIf extension 7958 Then Dialout on Capi msn 383591 ElseIf extension 7959 Then Dialout on Capi msn 383592 Etc Etc If you could give me a simplistic example (as always!!!), including which files I put the coding in (i.e. extensions, capi etc.) I would be most grateful. Thanks as always. Nick ___ 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] Digital Line Distortion
Hi Adam, what is your echocancel setting in zapata.conf for the PRI spans? I once noticed this distorted sound by using echocancel=256 (using mec2.h for echo cancelation). How about echocancelwhenbridged and echotraining? best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-05-03 um 13.43 schrieb Adam Goryachev: Damn, I forgot to describe the actual problem. Basically as someone I spoke to today described it, it sounds like you have one of those new digital pbx systems... In more detail, when he spoke, he heard his voice come back, but distorted. The louder the sound he made, the louder he heard himself (distorted). At all times, if I am on the tdm40b side, I hear 100% perfect audio quality in both directions. (Which is bad, because now the customer gets the bad sound, before it was just the staff...) Regards, Adam On Mon, 2004-05-03 at 21:28, Adam Goryachev wrote: Firstly, the problem... Ever since I installed and setup asterisk, I have had various problems, initially it was echo caused by the ISDN (isdn4linux) card I was using. Now I'll do the thing most people forget about... [SNIP] the rest of the quoted text! Regards, Adam ___ 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
Re: [Asterisk-Users] Digital Line Distortion
On Mon, May 03, 2004 at 09:28:56PM +1000, Adam Goryachev wrote: Firstly, the problem... Ever since I installed and setup asterisk, I have had various problems, initially it was echo caused by the ISDN (isdn4linux) card I was using. So, I upgraded to the X101P from digium. I still had echo, so I figured it was also caused by the ATA186 (cisco) I was using. So, I upgraded again to the TDM40B quad FXS card. This solved pretty much all my problems, except eventually, I needed more incoming lines. So, again, I upgraded to a digital line (10 channel PRI/E1) and purchased the brand new TE405p from digium... Now, eventually I got this working properly, for incoming and outbound calls, I have incoming callerid working, etc... However, ever since I did this, I continually get complaints from people about how terrible my phone lines are. Not *everyone* complains, but most people do We did face what may be the same problem here. The problem came from the fact that on some motherboards (well, *most* motherboards, as far as I tested), the TE405P has a problem which makes it send every one in 8 (or was it 16?) bytes as 0xFF (instead of whatever the U/A-law value may have been). On the RX side of things, it was always perfect, thus when connecting to a local IP phone we heard a nice sound, but the remote party always had a quite garbled output. You can check it quite easily : - plug a crossover cable between two ports - do not start Asterisk (but load and ztcfg everything) - cat /dev/zap/span1/1 on one terminal - ls /dev/zap/span2/1 on another terminal (provided that spans 1 and 2 are connected together) - if everything works well, you should have a perfect output for your ls on the cat terminal. Otherwise, try hexdump and watch the columns with FF. It was solved by using a PCI 2.2 compliant motherboard (i865 based). It's quite an odd behaviour, and it's still not clear to me why it happens. I initialiy thought it could be solved by fixing the FPGA VHDL, but I'm not an expert in that field. -- Nicolas Bougues Axialys Interactive ___ 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] module help?
I've been running * for eight months in production mode without the init.d/zaptel script in place. Didn't know 'make config' from within the zaptel src directory even existed, and have never seen/heard anyone even mention that before. Its been running fine with a pair of x100p's, however the system is seldom rebooted. Does that imply that * loads the necessary zaptel modules automatically when its started? (Guess I would have expected to run into problems way before adding the TDM04B card this past week.) Thoughts? Why copy...use the make command(same with asterisk)... make config Will do all that for you. -- On Sun, 2004-05-02 at 22:32, Scott Weis wrote: Simple solution on redhat machines In the zaptel source tree (At least the CVS one) there is a file called zaptel.init. This is a script that will allow you to install all needed modules. To use it do this: cd /usr/src/zaptel cp zaptel.init /etc/init.d/zaptel chkconfig --add zaptel chkconfig --level 2345 zaptel on Now every time you reboot all the zaptel modules will be install automatically. PS Why this is not done in the make install script is beyond me. Scott 700-297-0469 - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 02, 2004 6:59 PM Subject: Re: [Asterisk-Users] module help? I've installed the new TDM04B 4-port FXO card and its working. After a reboot, when I do lsmod I see the wcfxo module but not the wcfxs even though both are listed modules.conf. If I modprobe wcfxs, then lsmod has both modules showing. why you need wcfxs on a quad-fxo ? Because the support folks at digium said on Friday the supporting routines for the new fxo card are actually in wcfxs. The wcfxs module is the last one in the modules.conf. Is the order of entries sensitive in modules.conf? modules.conf != loaded modules. as the name suggest, it contains only configuration params for modules Do I need to be concerned with wcfxs not showing before starting asterisk? Any suggestions? sure. learn something more about kernel, modules and what is modules.conf bug us with asterisk related questions, not with what-are-kernel-modules? questions. Okay, then let me reword this just for you. Is there a problem with the asterisk make install process that might be considered the root-cause for wcfxs not showing up in lsmod? ___ 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 ---End of Original Message- ___ 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] problems with bri-stuff.0.0.2rc20a
Hi klaus-peter, I thought I fixed this error... but when ever I pickup the phone before I dial the number (the sitution I got the former descibed problem fixed with overlapdial=yes) I can dial an extension but I cannot send any furhter digits so voicemail and early b3-connects with chan_capi do now work. I hope you can help me again. ... bye thorsten - Original Message - From: FastJack [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 03, 2004 11:38 AM Subject: Re: [Asterisk-Users] problems with bri-stuff.0.0.2rc20a hi klaus-peter, yepp... with overlapdial=yes (almost) everything works great, again. one problem is left... touchtones are not working anymore so I can't use voicemail-system, parking and stuff. thank you for your help. ...bye thorsten ___ 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] dialing a remote phone system and then entering an extension
I am trying to get a way to have * forward calls that are dialed to an extension, to end up at an extension on my old analog phone system. I will have 7 lines coming into * using the new Digium cards via PSTN, and then lines coming from * into the PSTN lines on the analog system. So that if for example someone dials extension 110: The system will call the analog system, the system will assume that a call is coming from the telco as always, pick up right away, and then listen for an extension to be entered. This should then connect the incoming call to the extension on the analog system. My question is, does my logic work, and also if I use the dial command, and I set the analog system to pick up immediately, will wait long enough before it dials? If that wouldn't work is there a way that I can tell * to dial then wait and then send digits? Thanks Joel Duffield Near North Business Machines www.NearNorthBusiness.com ___ 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] New ENUM service, what do you think?
Joe Baptista wrote: agreed - you see alot of business fluff - but the technicals are very important and on many of these ventures they fail to include them. As far as I'm aware they are providing an internet exchange peering point for voip providers, and to get access to their enum zone you need to sign NDA's and other agreements and buy rack/IP/port space from them and the whole point is to buy and sell minutes between providers. These NDA's prevent them from releasing any information on number ranges or URLs to anyone not signed up. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers ___ 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: Caller ID Re: [Asterisk-Users] Re: Support Digium
Someone wrote: The BT CD50 and soldering iron plan is looking more and more like the one I'll be going with for now If you don't fancy using a soldering iron to read UK CLI there's a mod to * that my colleague, Robb Boardman, uses. By placing a certain model of Hayes or Pace modem in parallel with * on the incoming PSTN line the CLI is collected (before the first ring) via a serial TTY port. I'm sure it was posted in here some while ago so if you're interested have a look in the archives or reply to this note and see if we can encourage him to re-post the details. From memory the new ProSlic chip used by Digium supports UK CLI at a physical interface level but appropriate drivers have not yet been coded. Mark Spencer is very aware of the community's demand for international CLI; I suspect that it's a case of ever growing demand for new functionality verses finite implementation/support resources (both financial and human). If we can obtain the ProSlic technical interface details does someone fancy a spot of coding in return for a bounty...? On the subject of line reversal detection I know of a major manufacturer whose LLU products were recently rejected by a UK Telco for failure to support this feature on V5 Access Network muxes. There were a number of problems with automatic telephony equipment (E.g.. subscriber's own (CPE) telephone answering machines) that could not detect the end of the call. One of the strengths of the PSTN is the backward compatibility that has been maintained (including physical standards like voltages as well as higher protocols) for more than 100 years. I would like to echo an earlier poster's comments about the necessity to maintain compatibility with the earlier electro-mechanical standards for as long as we can. Just my 2ds... Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Elkins Sent: 02 May 2004 23:21 To: [EMAIL PROTECTED] Subject: RE: Caller ID Re: [Asterisk-Users] Re: Support Digium On Mon, 2004-05-03 at 00:11, David J Carter wrote: Mark J Elkins wrote Um - Digium wants you to buy their hardware - but there is a CLID issue.. would it not make more financial sense to insert a dumb ISDN card (or two), and upgrade your PSTN to ISDN??? Would this not assist Digium in making sure CLID worked in the UK??? Isn't this a bit like cutting of the nose to spite the face. UK PSTN lines costs £30 /Qtr UK ISDN costs £65 /qtr, you could buy two X100P's every year and still be in pocket by staying with PSTN. ISDN BRI is two lines - so that makes it £2.50 more per line - or £10 a year..?? no need to purchase the BT50 (a caller-ID unit? - at what cost? you need one per line? and an RS232 interface per unit?) There was a post on the list in the not to distant past where someone had written two small scripts for getting the information from a BT50 and a serial modification and passing it to asterisk. Still seems the best way in the interim. As has been said many times in the list Digium have given us this software, we don't have to give them a hard time in return. Not a fair payback. True - the software is excellent. If they sold an ISDN BRI 4-port card (like Fritz) - I'd buy it from them. No intentions of bad mouthing Digium... but USA != World -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] IAX2
we are getting these errors too which cvs was it fixed in ? we just upgraded to cvs-stable from friday to see if that would help. On Sun, 2004-05-02 at 21:45, brian k. west wrote: I think this was fixed in CVS-HEAD because I do not see that message in the src at all while looking to see if t was fixed. bkw - Original Message - From: Serge Oleinikov To: [EMAIL PROTECTED] Sent: Sunday, May 02, 2004 2:40 PM Subject: [Asterisk-Users] IAX2 What does it mean ? May 2 20:37:21 WARNING[1205250992]: chan_iax2.c:2515 iax2_send: Out of trunk data space on call number 16386, dropping Asterisk CVS-05/02/04-23:04:14 built by [EMAIL PROTECTED] on a i686 running Linux from cvs checkout -r v1-0_stable asterisk ___ 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] Error building asterisk-0.9.0
Title: Message I am trying to build asterisk-0.9.0 on 533MHz 160MB Redhat Linux 9.0 machine. I have followed the instructions to build asterisk. Building zaptel and libpri seemed to go well (lots of messages but nothing that indicated an error) However, when I do the make clean ; make install for asterisk-0.9.0 after running for sometime I get the following: gcc -shared -Xlinker -x -o app_senddtmf.so app_senddtmf.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"0.9.0\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -fPIC -c -o app_parkandannounce.o app_parkandannounce.c /usr/include/bits/string2.h:992: internal error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. make[1]: *** [app_parkandannounce.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-0.9.0/apps' make: *** [subdirs] Error 1 Any thoughts? Regards, Jim O'Brien
RE: [Asterisk-Users] Error building asterisk-0.9.0
Title: Message Looks like you might have a hardware issue. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim O'Brien Sent: Monday, May 03, 2004 8:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Error building asterisk-0.9.0 I am trying to build asterisk-0.9.0 on 533MHz 160MB Redhat Linux 9.0 machine. I have followed the instructions to build asterisk. Building zaptel and libpri seemed to go well (lots of messages but nothing that indicated an error) However, when I do the make clean ; make install for asterisk-0.9.0 after running for sometime I get the following: gcc -shared -Xlinker -x -o app_senddtmf.so app_senddtmf.o gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\0.9.0\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP -fPIC -c -o app_parkandannounce.o app_parkandannounce.c /usr/include/bits/string2.h:992: internal error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See URL:http://bugzilla.redhat.com/bugzilla/ for instructions. make[1]: *** [app_parkandannounce.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-0.9.0/apps' make: *** [subdirs] Error 1 Any thoughts? Regards, Jim O'Brien
RE: [Asterisk-Users] IAX2
1. Its not an error. 2. It's a warning. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Justin Carlson Sent: Monday, May 03, 2004 3:21 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 we are getting these errors too which cvs was it fixed in ? we just upgraded to cvs-stable from friday to see if that would help. On Sun, 2004-05-02 at 21:45, brian k. west wrote: I think this was fixed in CVS-HEAD because I do not see that message in the src at all while looking to see if t was fixed. bkw - Original Message - From: Serge Oleinikov To: [EMAIL PROTECTED] Sent: Sunday, May 02, 2004 2:40 PM Subject: [Asterisk-Users] IAX2 What does it mean ? May 2 20:37:21 WARNING[1205250992]: chan_iax2.c:2515 iax2_send: Out of trunk data space on call number 16386, dropping Asterisk CVS-05/02/04-23:04:14 built by [EMAIL PROTECTED] on a i686 running Linux from cvs checkout -r v1-0_stable asterisk ___ 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
RE: [Asterisk-Users] Start recording during call by pressing button sequence
Does anyone know how these laws apply in interstate calls? For example, I am in a One-party consent state. This means I can legally record any telephone I am a part of, without notifying any other party. Say someone from Florida or another all-party state calls me, or I call someone in Florida. Which set of laws apply? Or is there a set of Federal laws that override what the state laws say? Next question is regarding how caller-ID plays a part of it. Say I have a system set up to record all calls. I have no idea where the call could be coming from. Would I be required to have caller-ID, or automatically stop recording of calls (or play warning messages) that came from area codes within the all-party states? I don't currently have a need to record any calls, but I just wanted to play devil's advocate and see if anyone knew the answers. Thanks and have a good one, Jeremy -Original Message- From: C. Maj [mailto:[EMAIL PROTECTED] Sent: Saturday, May 01, 2004 4:21 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Start recording during call by pressing button sequence On Fri, 30 Apr 2004, Dean Collins waxed: Ian, I'd love to see an example of this. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Friday, 30 April 2004 1:47 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Start recording during call by pressing button sequence --On Thursday, April 29, 2004 3:21 pm +0300 Vladyslav [EMAIL PROTECTED] wrote: On Thu, 2004-04-29 at 15:06, Andrew Kohlsmith wrote: Thank U for your reply, however I was asking about recording during call (for example I don't need record all calls, but only some of them and I want start recording during actual call process). You can activate call recording with a php script from a web page too. You can turn recording on and off without the called party knowing and at any time in the call. Iain Those in the US might want to check on what sort of laws affect recording of telephone conversations: http://archive.aclu.org/issues/cyber/phonelaw.html I recall it being mentioned on this list that people wished to spy on their kids with *, and that's specifically forbidden in most states, as it would be zero-party consent. You can log their IM all you want, tho. Then wonder why they hate you. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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
RE: [Asterisk-Users] IAX2
no I usually have 2 to 3 calls going down a full data T1(only voice data) and I get this message and 2 sec later calls are dropped. we look at our bandwidth for that time and we were no where near full utilization. On Mon, 2004-05-03 at 13:58, brian wrote: 1. Its not an error. 2. It's a warning. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Justin Carlson Sent: Monday, May 03, 2004 3:21 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX2 we are getting these errors too which cvs was it fixed in ? we just upgraded to cvs-stable from friday to see if that would help. On Sun, 2004-05-02 at 21:45, brian k. west wrote: I think this was fixed in CVS-HEAD because I do not see that message in the src at all while looking to see if t was fixed. bkw - Original Message - From: Serge Oleinikov To: [EMAIL PROTECTED] Sent: Sunday, May 02, 2004 2:40 PM Subject: [Asterisk-Users] IAX2 What does it mean ? May 2 20:37:21 WARNING[1205250992]: chan_iax2.c:2515 iax2_send: Out of trunk data space on call number 16386, dropping Asterisk CVS-05/02/04-23:04:14 built by [EMAIL PROTECTED] on a i686 running Linux from cvs checkout -r v1-0_stable asterisk ___ 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
RE: [Asterisk-Users] Start recording during call by pressing button sequence
Why don't you have a This call may be recorded for quality assurance when someone calls in That provides notification. Zac On Mon, 2004-05-03 at 09:01, Jeremy Hall wrote: Does anyone know how these laws apply in interstate calls? For example, I am in a One-party consent state. This means I can legally record any telephone I am a part of, without notifying any other party. Say someone from Florida or another all-party state calls me, or I call someone in Florida. Which set of laws apply? Or is there a set of Federal laws that override what the state laws say? Next question is regarding how caller-ID plays a part of it. Say I have a system set up to record all calls. I have no idea where the call could be coming from. Would I be required to have caller-ID, or automatically stop recording of calls (or play warning messages) that came from area codes within the all-party states? I don't currently have a need to record any calls, but I just wanted to play devil's advocate and see if anyone knew the answers. Thanks and have a good one, Jeremy -Original Message- From: C. Maj [mailto:[EMAIL PROTECTED] Sent: Saturday, May 01, 2004 4:21 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Start recording during call by pressing button sequence On Fri, 30 Apr 2004, Dean Collins waxed: Ian, I'd love to see an example of this. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Friday, 30 April 2004 1:47 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Start recording during call by pressing button sequence --On Thursday, April 29, 2004 3:21 pm +0300 Vladyslav [EMAIL PROTECTED] wrote: On Thu, 2004-04-29 at 15:06, Andrew Kohlsmith wrote: Thank U for your reply, however I was asking about recording during call (for example I don't need record all calls, but only some of them and I want start recording during actual call process). You can activate call recording with a php script from a web page too. You can turn recording on and off without the called party knowing and at any time in the call. Iain Those in the US might want to check on what sort of laws affect recording of telephone conversations: http://archive.aclu.org/issues/cyber/phonelaw.html I recall it being mentioned on this list that people wished to spy on their kids with *, and that's specifically forbidden in most states, as it would be zero-party consent. You can log their IM all you want, tho. Then wonder why they hate you. --Chris ___ 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] Start recording during call by pressingbutton sequence
Zac, Thanks for the input. This would cover it, however it is not stealth. In some cases, you may want it to be stealth. Again, my state allows me to do this, but some states do not. I've done a little searching and could not find an answer. Basically, to simplify the question: When is it legal to record interstate calls without giving notice, when the recording party is in a one-party consent state? As long as notice is given to all parties, it is always legal from my understanding. Jeremy -Original Message- From: Zac Amsler [mailto:[EMAIL PROTECTED] Sent: Monday, May 03, 2004 8:18 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Start recording during call by pressingbutton sequence Why don't you have a This call may be recorded for quality assurance when someone calls in That provides notification. Zac On Mon, 2004-05-03 at 09:01, Jeremy Hall wrote: Does anyone know how these laws apply in interstate calls? For example, I am in a One-party consent state. This means I can legally record any telephone I am a part of, without notifying any other party. Say someone from Florida or another all-party state calls me, or I call someone in Florida. Which set of laws apply? Or is there a set of Federal laws that override what the state laws say? SNIP ___ 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 -- Cisco router
Check the duplex on your ethernet conection on both the Cisco and the Asterisk box. Make sure neither are half duplex. Joseph wrote: What codec should be used to connect a * box to a cisco router which has a t1 with 24 trunks coming in? My router voip dial plan looks like this: dial-peer voice 2 voip destination-pattern [1,2,,3,5,8].. session protocol sipv2 session target ipv4:10.x.x.x dtmf-relay cisco-rtp codec g711ulaw no vad ! The problem I have is when more than one call is on it, sometimes the quality gets very bad. If more than one access the conference room it starts to blip real badly. Thots, ideas greatly appreciated. ___ 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] Start recording during call by pressingbutton sequence
I don;t know if stealth really that important.. How many times have u hung up on a company when they told you that your call may be recorded for quality and training purposes?? I hear that from every company I call. Most people brush off the message forget about it befor they even talk to someone. I have the theory of CYOA(Cover Your Own Ass) I hope that I have helped. Zac On Mon, 2004-05-03 at 09:45, Jeremy Hall wrote: Zac, Thanks for the input. This would cover it, however it is not stealth. In some cases, you may want it to be stealth. Again, my state allows me to do this, but some states do not. I've done a little searching and could not find an answer. Basically, to simplify the question: When is it legal to record interstate calls without giving notice, when the recording party is in a one-party consent state? As long as notice is given to all parties, it is always legal from my understanding. Jeremy -Original Message- From: Zac Amsler [mailto:[EMAIL PROTECTED] Sent: Monday, May 03, 2004 8:18 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Start recording during call by pressingbutton sequence Why don't you have a This call may be recorded for quality assurance when someone calls in That provides notification. Zac On Mon, 2004-05-03 at 09:01, Jeremy Hall wrote: Does anyone know how these laws apply in interstate calls? For example, I am in a One-party consent state. This means I can legally record any telephone I am a part of, without notifying any other party. Say someone from Florida or another all-party state calls me, or I call someone in Florida. Which set of laws apply? Or is there a set of Federal laws that override what the state laws say? SNIP ___ 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] Help with busydetect (no hangups)
I´m using * 0.9.0 and have a X100P connected to my analog PBX. I can´t detect hangups on this line, so I turned on busydetect=yes in zapata.conf. I also have busycount=6. While the line is connected to the PBX, I can never detect busy and the line hangs at the end of every call. If I connect the same X100P to the telco line, without the PBX, then it can detect busy and hangs up the line after 6 busy tones, as expected. I have recorded the busy tones and found that telco uses a standar one (250ms tone, 250 ms silence). My PBX, however, is using a 120ms tone, 80ms silence sequence. How can I adjust the detection routines to the tone I´m getting? I have tried to mess with busy_min, busy_max etc on dsp.c with no luck. I´m sure I doesnt really understand the meaning of those parameters. A also tryed to compile using TONE_ONLY but it gives a compilations error. Can someone suggest what times should I been using? Thanks a lot. Gelson ___ 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] zap x100p
I have 2 X100P cards that I am using to handle voicemail, but I have problem. It takes about 3 to 4 rings before they pick up. Anyone have any idea why it takes so long to pickup and answer? Or is there a way to control this? -- respectfully, Joseph ___ 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 -- Cisco router
I will double check. How much cpu does the MeetMe feature need per user? Or does it depend on how they connect? On Mon, 2004-05-03 at 11:54, James Sizemore wrote: Check the duplex on your ethernet conection on both the Cisco and the Asterisk box. Make sure neither are half duplex. Joseph wrote: What codec should be used to connect a * box to a cisco router which has a t1 with 24 trunks coming in? My router voip dial plan looks like this: dial-peer voice 2 voip destination-pattern [1,2,,3,5,8].. session protocol sipv2 session target ipv4:10.x.x.x dtmf-relay cisco-rtp codec g711ulaw no vad ! The problem I have is when more than one call is on it, sometimes the quality gets very bad. If more than one access the conference room it starts to blip real badly. Thots, ideas greatly appreciated. ___ 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 -- respectfully, Joseph ___ 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] iconnecthere behind NAT, strange deal
Have you found a solution yet? I am having the same issue. My account works fine with the IConnectHere soft phone client but not with Asterisk. Inbound lines work fine, outbound returns the same message. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Capouch Sent: Sunday, May 02, 2004 4:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] iconnecthere behind NAT, strange deal I've been to the WIKI and I've searched the archives. Is anyone on the list successfully using iconnecthere behind NAT? I was, for over a year, and then I changed my plan with them. Now all my calls get intercepted immediately, We're sorry, but your account is temporarily unavailable. Incoming calls work just fine. I contacted their so-called customer care, which has sent me repeated replies asking me to give them the version of my PC phone. When I say I don't have one, they say, Sorry, we only help those who do. I like to play with their GSM stuff, so I hate to let the account go, but if no one here knows what might be going on, they certainly don't. FWIW I used to prepend to the dialed number, and it worked fine until last week. Thx. B. ___ 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
Re: [Asterisk-Users] zap x100p
On Mon, 2004-05-03 at 10:04, Joseph wrote: I have 2 X100P cards that I am using to handle voicemail, but I have problem. It takes about 3 to 4 rings before they pick up. Anyone have any idea why it takes so long to pickup and answer? Or is there a way to control this? Do a minor amount of research, please. This smacks of being too lazy to do your own work since we have covered this at least 2 times in the last 2 weeks. You must have 1 ring to know there was a ring. Bellcore Callerid comes after the 1st ring, but can be out into the gap after the 2nd ring. If you are using callerid, asterisk waits for this. Then if you have a wait in your dialplan like the examples do, then you are pushing into the next ring. All explained in normal signaling patterns that we have gone over now more than 3 times in the last month. No one else should have an excuse to be this lazy again. -- Steven Critchfield [EMAIL PROTECTED] ___ 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 remains in the media path
Hi all, Just a quick question: I have an H323 terminal and some MGCP phones connected to *, and when they call each other * remains in the media path no matter what (while I'd like to have the RTP stream directly between the phones). - mgcp.conf has canreinvite=yes - extension.conf doesn't contain any Dial() instance with t or T Did I forget something? Thanks, Paul ___ 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] zap x100p
Thanks for the answer. I only joined the list 4 days and did not mean to ask a reduntant question. I will see if I can get it fixed. I assume turning callerid would not make it wait for that, so the shortest amount of time might be 1 to 2 rings then? Any tips on debugging callerid problems? On Mon, 2004-05-03 at 11:15, Steven Critchfield wrote: On Mon, 2004-05-03 at 10:04, Joseph wrote: I have 2 X100P cards that I am using to handle voicemail, but I have problem. It takes about 3 to 4 rings before they pick up. Anyone have any idea why it takes so long to pickup and answer? Or is there a way to control this? Do a minor amount of research, please. This smacks of being too lazy to do your own work since we have covered this at least 2 times in the last 2 weeks. You must have 1 ring to know there was a ring. Bellcore Callerid comes after the 1st ring, but can be out into the gap after the 2nd ring. If you are using callerid, asterisk waits for this. Then if you have a wait in your dialplan like the examples do, then you are pushing into the next ring. All explained in normal signaling patterns that we have gone over now more than 3 times in the last month. No one else should have an excuse to be this lazy again. -- respectfully, Joseph ___ 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 remains in the media path
Can't do it because you are changing from one technology to another. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul Berger Sent: Monday, May 03, 2004 10:29 AM To: Liste Asterisk Subject: [Asterisk-Users] Asterisk remains in the media path Hi all, Just a quick question: I have an H323 terminal and some MGCP phones connected to *, and when they call each other * remains in the media path no matter what (while I'd like to have the RTP stream directly between the phones). - mgcp.conf has canreinvite=yes - extension.conf doesn't contain any Dial() instance with t or T Did I forget something? Thanks, Paul ___ 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] txfax: Trainability test failed
Hi all,
I'm using txfax and rxfax to send and receive faxes. I find that txfax performs very
well when sending to an old fax machine. However, when trying to send to a Mac (OS X)
that accepts faxes, I get a Trainability test failed message on the * console (full
console log below).
I'm OK with being unable to fax to Mac, as I'm just using it to test. For future
reference, I'd like to know what causes this problem, and what steps (if any) I can
take to correct it.
Thanks
Ryan
exten = out_fax,1,txfax(${TXFAX_NAME}|caller)
---SNIP---
-- Attempting call on Zap/g1/2442790 for [EMAIL PROTECTED]:1 (Retry 1)
Channel Zap/1-1 was answered.
-- Executing Wait(Zap/1-1, 7) in new stack
-- Executing TxFAX(Zap/1-1, /var/spool/asterisk/fax/test.tif|caller) in new
stack
File name is '/var/spool/asterisk/fax/test.tif'
Changed from phase 0 to 2
Slow carrier up
Slow carrier down
Slow carrier up
CSI: 40 30 39 37 32 20 34 34 32 20 33 30 34 20 20 20 20 20 20 20 20
CSI without final frame tag
Remote fax gave CSI as: 403 244 2790
DIS: 80 00 6e 78
DIS with final frame tag
In state 10
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter, V.29 and V.17
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
Scan line length: 215mm
Recording length: A4 (297mm) and B4 (364mm)
Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85
DCS:
Selected data signalling rate: V.29, 9600bps
Scan line length: 215mm
Recording length: A4 (297mm)
Minimum scan line time: 0ms
Start sending document
Start tx document - compression 1
Fine mode
Changed from phase 2 to 4
Sending ident
TSI: 43 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20
DCS: 83 00 44 70
HDLC underflow in state 3
Changed from phase 4 to 6
Changed from phase 6 to 3
T4 timeout in state 4
Slow carrier up
FTT: 44
FTT with final frame tag
In state 4
Trainability test failed
Changed from phase 3 to 6
Changed from phase 6 to 3
T4 timeout in state 4
---SNIP---
___
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RE: [Asterisk-Users] Start recording during call by pressing button sequence
Jeremy, In the realm of US law, there are no definates. Before going any farther, I AM NOT LICENSED to practice law or provide legal advice in any way, form, or manner. That being said, talk to someone who is licensed to do so. And in this case, make sure that person is or has access to a telecommunications expert in the federal area as well. If you determine that you are safe under criminal law, you will also want to ask about any issues you may hit in civil law (this will probably mostly pertain to use of the recordings and results of those uses instead of the act of recording itself) -Chris On 07:01 AM 5/3/2004, Jeremy Hall wrote: Does anyone know how these laws apply in interstate calls? For example, I am in a One-party consent state. This means I can legally record any telephone I am a part of, without notifying any other party. Say someone from Florida or another all-party state calls me, or I call someone in Florida. Which set of laws apply? Or is there a set of Federal laws that override what the state laws say? ___ 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 remains in the media path
Le lun 03/05/2004 à 17:34, brian a écrit : Can't do it because you are changing from one technology to another. Thanks for your answer. H323 and MGCP are supposed to stay on the call control level, why isn't it possible to open RTP channels between the terminals then? Again, thanks, Paul ___ 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] zap x100p
On Mon, 2004-05-03 at 10:32, Joseph wrote: Thanks for the answer. I only joined the list 4 days and did not mean to ask a reduntant question. This is why the mailing list is archived by digium and indexed by google. Redundant questions should be easily covered. I'm not sure but the wiki may even cover the answer at voip-info.org. I will see if I can get it fixed. I assume turning callerid would not make it wait for that, so the shortest amount of time might be 1 to 2 rings then? Any tips on debugging callerid problems? usecallerid=no or some such item in zapata.conf, then make sure there are no wait() lines before the answer() line that your call would take. On Mon, 2004-05-03 at 11:15, Steven Critchfield wrote: On Mon, 2004-05-03 at 10:04, Joseph wrote: I have 2 X100P cards that I am using to handle voicemail, but I have problem. It takes about 3 to 4 rings before they pick up. Anyone have any idea why it takes so long to pickup and answer? Or is there a way to control this? Do a minor amount of research, please. This smacks of being too lazy to do your own work since we have covered this at least 2 times in the last 2 weeks. You must have 1 ring to know there was a ring. Bellcore Callerid comes after the 1st ring, but can be out into the gap after the 2nd ring. If you are using callerid, asterisk waits for this. Then if you have a wait in your dialplan like the examples do, then you are pushing into the next ring. All explained in normal signaling patterns that we have gone over now more than 3 times in the last month. No one else should have an excuse to be this lazy again. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Start recording during call by pressing button sequence
If you live in a one-party state you can record ANY call in or out .. doesn't matter if the call comes from/to out of state or not. You are within the rights of your state. I live in Oklahoma and I record EVERY call in our out of my house with asterisk. bkw ___ 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] zap x100p
Thanks a million. On Mon, 2004-05-03 at 12:04, Steven Critchfield wrote: On Mon, 2004-05-03 at 10:32, Joseph wrote: Thanks for the answer. I only joined the list 4 days and did not mean to ask a reduntant question. This is why the mailing list is archived by digium and indexed by google. Redundant questions should be easily covered. I'm not sure but the wiki may even cover the answer at voip-info.org. I will see if I can get it fixed. I assume turning callerid would not make it wait for that, so the shortest amount of time might be 1 to 2 rings then? Any tips on debugging callerid problems? usecallerid=no or some such item in zapata.conf, then make sure there are no wait() lines before the answer() line that your call would take. On Mon, 2004-05-03 at 11:15, Steven Critchfield wrote: On Mon, 2004-05-03 at 10:04, Joseph wrote: I have 2 X100P cards that I am using to handle voicemail, but I have problem. It takes about 3 to 4 rings before they pick up. Anyone have any idea why it takes so long to pickup and answer? Or is there a way to control this? Do a minor amount of research, please. This smacks of being too lazy to do your own work since we have covered this at least 2 times in the last 2 weeks. You must have 1 ring to know there was a ring. Bellcore Callerid comes after the 1st ring, but can be out into the gap after the 2nd ring. If you are using callerid, asterisk waits for this. Then if you have a wait in your dialplan like the examples do, then you are pushing into the next ring. All explained in normal signaling patterns that we have gone over now more than 3 times in the last month. No one else should have an excuse to be this lazy again. -- respectfully, Joseph ___ 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 remains in the media path
brian wrote: Can't do it because you are changing from one technology to another. Actually its cuz chan_h323 sucks like that. Jeremy McNamara -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul Berger Sent: Monday, May 03, 2004 10:29 AM To: Liste Asterisk Subject: [Asterisk-Users] Asterisk remains in the media path Hi all, Just a quick question: I have an H323 terminal and some MGCP phones connected to *, and when they call each other * remains in the media path no matter what (while I'd like to have the RTP stream directly between the phones). - mgcp.conf has canreinvite=yes - extension.conf doesn't contain any Dial() instance with t or T Did I forget something? Thanks, Paul ___ 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] Open Source SCGP
Hey, Someone told me an open source SGCP gateway was created for the Asterisk project. I'm looking for a little more information. I have two VG248s that I'd like to attach to my VoIP network; however, Cisco's documentation seems to indicate that Cisco CallManager is required for these things to operate. Cisco CallManager being a 10,000 dollar application, I would like to find any open source alternatives. Regards, Daniel -- Daniel Corbe, CCNP Senior Network Engineer Results Technologies, Inc. 952-921-2400 x104 ___ 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] Problem with new sipura firmware 1.0.35a
I just tried to upgrade my sipura to firmware 1.0.35a and now I can't connect to it. It still works but any connection to ports 23 and 80 makes it reboot. Even the flash tool makes it to crash when trying to connect. Anybody else experiencing this problem? Regards, Victor Perez ___ 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] iconnecthere behind NAT, strange deal
Hi, I had the same issue with iConnectHere as well as FWD (authenticated). If you are running CVS -head (4/26 - 5/1) i would suggest getting the latest CVS. I worked with Mark over the weekend to resolve this bug. http://bugs.digium.com/bug_view_page.php?bug_id=0001533 Hopefully, this will resolve the problem you are seeing. Bill Doll Jr Lists [EMAIL PROTECTED] Lists [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/03/2004 08:09 AM Please respond to [EMAIL PROTECTED] To [EMAIL PROTECTED] cc Subject RE: [Asterisk-Users] iconnecthere behind NAT, strange deal Have you found a solution yet? I am having the same issue. My account works fine with the IConnectHere soft phone client but not with Asterisk. Inbound lines work fine, outbound returns the same message. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Brian Capouch Sent: Sunday, May 02, 2004 4:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] iconnecthere behind NAT, strange deal I've been to the WIKI and I've searched the archives. Is anyone on the list successfully using iconnecthere behind NAT? I was, for over a year, and then I changed my plan with them. Now all my calls get intercepted immediately, We're sorry, but your account is temporarily unavailable. Incoming calls work just fine. I contacted their so-called customer care, which has sent me repeated replies asking me to give them the version of my PC phone. When I say I don't have one, they say, Sorry, we only help those who do. I like to play with their GSM stuff, so I hate to let the account go, but if no one here knows what might be going on, they certainly don't. FWIW I used to prepend to the dialed number, and it worked fine until last week. Thx. B. ___ 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 inline: graycol.gifinline: pic00041.gifinline: ecblank.gif
Re: [Asterisk-Users] Asterisk remains in the media path
On Mon, 2004-05-03 at 12:05, jimfl wrote: So does this mean you could get direct RTP steams between a SIP client and a IAX2 client? What about inband/out of band DTMF issues? IAX doesn't use rtp and therefore it couldn't do it either. All DTMF should be OOB to be reliable. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] If Then Else Statements - Outbound Dialling on ISDN using CAPI -Individual Dial out Plans using msns
Hi!
What I would like to do is set up an If Then Else type statement along the
following lines: -
If extension 7957 Then
Dialout on Capi msn 383590
Create a macro in extensions.conf:
exten = s,1,AbsoluteTimeout(${TIMEOUTABS})
exten = s,2,NoOp
exten = s,3,GotoIf($[$[${CALLERIDNUM} = 103] | $[${CALLERIDNUM} =
302]]?10:4)
exten = s,4,GotoIf($[$[${CALLERIDNUM} = 104] | $[${CALLERIDNUM} =
106]]?12:5)
exten = s,5,GotoIf($[${CALLERIDNUM} = 108]?8:6)
exten = s,6,Dial(CAPI/${MSN1}:b${MACRO_EXTEN:1},120,T) ; we are 102 or
have a CALLERIDNUM that was not checked for above
exten = s,7,Goto(20) ; unavailable
exten = s,8,Dial(CAPI/${MSN2}:b${MACRO_EXTEN:1},120,T) ; we are 108
exten = s,9,Goto(20) ; unavailable
...
Cheers, Philipp
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Re: [Asterisk-Users] Problem with new sipura firmware 1.0.35a
I have two units running 1.0.35a working just fine. On Mon, 2004-05-03 at 10:08, Victor Perez wrote: I just tried to upgrade my sipura to firmware 1.0.35a and now I can't connect to it. It still works but any connection to ports 23 and 80 makes it reboot. Even the flash tool makes it to crash when trying to connect. Anybody else experiencing this problem? ___ 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] Talking SIP to Vocal
Hi, Try latest CVS. There was an auth sip bug fixed on Saturday. http://bugs.digium.com/bug_view_page.php?bug_id=0001533 Hope this helps. Bill Doll Jr Mark Turner [EMAIL PROTECTED] Mark Turner [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/02/2004 05:24 AM Please respond to [EMAIL PROTECTED] To [EMAIL PROTECTED] cc Subject [Asterisk-Users] Talking SIP to Vocal I'm trying to get Asterisk to talk SIP to Vocal and so far have only managed to get it partially working. Calls in from Vocal are working fine but outbound calls aren't. In sip.conf I have: [ivv] secret=SECRET username=08452416761 host=sip.intervivo.net fromuser=08452416761 externip=mt104.dyndns.org nat=yes canreinvite=no reinvite=no notransfer=yes In extensions.conf I have: exten = 150,1,Dial(SIP/[EMAIL PROTECTED]) When I call 150 Asterisk sends an invite to Vocal which then asks for authentication. Asterisk sends another invite with auth details *but* the digest username is 0800800150 when it should (I think) be 08452416761. I'm using source from CVS, checked out yesterday. Calls out via IAX work fine. Calls out via SIP to Free World Dialup work fine, but then FWD doesn't ask for authentication. Is this a bug in the SIP auth code or am I misconfiged? Any ideas please? Thanks, Mark. ___ 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 inline: graycol.gifinline: pic29358.gifinline: ecblank.gif
[Asterisk-Users] quad fxo
Folks, I'm trying to install one of the new quad fxo cards remotely. I know the existing machine was too old to have a PCI 2.2 bus, so I had my helper at the other end try a few boxes that were sitting on a shelf with the new card and a Knoppix cd. He found one that reported the card as the Tiger Jet. Good. Now, we moved the HD from the existing machine, loaded with Debian, and the card is just seen as the generic communications device, like the bus is wrong. Any pointers on this? The machine is ~500 miles away at the moment, and off the network, so most of this is done by phone. :( Thanks, Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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] Start recording during call by pressing button sequence
On Mon, May 03, 2004 at 11:13:17AM -0500, brian said: If you live in a one-party state you can record ANY call in or out .. doesn't matter if the call comes from/to out of state or not. You are within the rights of your state. I live in Oklahoma and I record EVERY call in our out of my house with asterisk. The site: http://www.rcfp.org/taping/interstate.html has something to say about this. Apparently it depends on the state, and how the states handle conflict of law, and in which state someone files suit. The recommendation is to follow the laws that are more strict in cases where there is a conflict. I suppose you could do something with a database of area codes, and if no caller ID is present, assume you can't record. ___ 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] ISDN WAN ISDN bridge possible?
Hi list,is it possible to create something like a ISDN-WAN-WAN-ISDN bridge? We have to change our location, but our number and the telephone system should shoulb stay the same. kind regards,Patrick Stuckenberger ___ 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] ISDN WAN ISDN bridge possible?
sure. use * with: IAX2 for sending voice via WAN (I suppose is internet) and then for ISDN you can: if is PRI , get 2 digium cards if is BRI , get zapbri cards matteo Il lun, 2004-05-03 alle 19:41, Patrick Stuckenberger ha scritto: Hi list, is it possible to create something like a ISDN-WAN-WAN-ISDN bridge? We have to change our location, but our number and the telephone system should shoulb stay the same. kind regards, Patrick Stuckenberger ___ 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 -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ 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] quad fxo
On Mon, 2004-05-03 at 12:26, Tim Sailer wrote: Folks, I'm trying to install one of the new quad fxo cards remotely. I know the existing machine was too old to have a PCI 2.2 bus, so I had my helper at the other end try a few boxes that were sitting on a shelf with the new card and a Knoppix cd. He found one that reported the card as the Tiger Jet. Good. Now, we moved the HD from the existing machine, loaded with Debian, and the card is just seen as the generic communications device, like the bus is wrong. Any pointers on this? The machine is ~500 miles away at the moment, and off the network, so most of this is done by phone. :( Maybe you should start by getting your helper to recompile the kernel so it can be on the network and you can then do some real debugging of the bus. I'm guessing that the kernel you have on that drive isn't sufficiently smart enough to handle the newer hardware. Knoppix CDs seem pretty decent at running the hardware fast since they are at such a disadvantage booting and running from the CD. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] If Then Else Statements - Outbound Dialling on ISDN using CAPI -Individual Dial out Plans using msns
On Tue, 2004-05-04 at 06:46, Nick Grindley wrote: Hi All, Many thanks to Marc who helped me with a previous Capi Dialout plan - however. What I now would like to be able to do is: - We have 8 msn's 383590, 383591 383592 etc. What I would like to do is set up an If Then Else type statement along the following lines: - If extension 7957 Then Dialout on Capi msn 383590 ElseIf extension 7958 Then Dialout on Capi msn 383591 ElseIf extension 7959 Then Dialout on Capi msn 383592 Etc Etc Sounds like you are basically trying to set outbound callerid. Don't think of this as by extension, the extension number is an arbitrary number you placed on the physical port. Think of it as ports, or as a property of the line. You could do a lookup in a dbput/dbget manner to store the MSN. You could set each port to start in it's own context that defines the MSN. OR there is probably a few other easy to define ways also. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] iconnecthere behind NAT, strange deal
Thanks for the update! Michael From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, May 03, 2004 1:18 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] iconnecthere behind NAT, strange deal Hi, I had the same issue with iConnectHere as well as FWD (authenticated). If you are running CVS -head (4/26 - 5/1) i would suggest getting the latest CVS. I worked with Mark over the weekend to resolve this bug. http://bugs.digium.com/bug_view_page.php?bug_id=0001533 Hopefully, this will resolve the problem you are seeing. Bill Doll Jr Lists [EMAIL PROTECTED] Lists [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/03/2004 08:09 AM Please respond to [EMAIL PROTECTED] To [EMAIL PROTECTED] cc Subject RE: [Asterisk-Users] iconnecthere behind NAT, strange deal Have you found a solution yet? I am having the same issue. My account works fine with the IConnectHere soft phone client but not with Asterisk. Inbound lines work fine, outbound returns the same message. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Brian Capouch Sent: Sunday, May 02, 2004 4:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] iconnecthere behind NAT, strange deal I've been to the WIKI and I've searched the archives. Is anyone on the list successfully using iconnecthere behind NAT? I was, for over a year, and then I changed my plan with them. Now all my calls get intercepted immediately, We're sorry, but your account is temporarily unavailable. Incoming calls work just fine. I contacted their so-called customer care, which has sent me repeated replies asking me to give them the version of my PC phone. When I say I don't have one, they say, Sorry, we only help those who do. I like to play with their GSM stuff, so I hate to let the account go, but if no one here knows what might be going on, they certainly don't. FWIW I used to prepend to the dialed number, and it worked fine until last week. Thx. B. ___ 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 image001.gifimage002.gifimage003.gif
Re: [Asterisk-Users] quad fxo
The bus isn't wrong... debian is wrong. Like everything in debian... it ships with an old pci.ids (No flames intended... but still :P ) replace yours with one from: http://pciids.sourceforge.net/ And It *should* report it better... (Didn't verify) Not that any of this matters... Just load the driver and get on with it. Steven Critchfield wrote: On Mon, 2004-05-03 at 12:26, Tim Sailer wrote: Folks, I'm trying to install one of the new quad fxo cards remotely. I know the existing machine was too old to have a PCI 2.2 bus, so I had my helper at the other end try a few boxes that were sitting on a shelf with the new card and a Knoppix cd. He found one that reported the card as the Tiger Jet. Good. Now, we moved the HD from the existing machine, loaded with Debian, and the card is just seen as the generic communications device, like the bus is wrong. Any pointers on this? The machine is ~500 miles away at the moment, and off the network, so most of this is done by phone. :( Maybe you should start by getting your helper to recompile the kernel so it can be on the network and you can then do some real debugging of the bus. I'm guessing that the kernel you have on that drive isn't sufficiently smart enough to handle the newer hardware. Knoppix CDs seem pretty decent at running the hardware fast since they are at such a disadvantage booting and running from the CD. ___ 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] iconnecthere behind NAT, strange deal
Lists wrote: Have you found a solution yet? I am having the same issue. My account works fine with the IConnectHere soft phone client but not with Asterisk. Inbound lines work fine, outbound returns the same message. CVS update took care of my problem wrt outbound calls; it must have been the mis-authentication stuff referenced in other mails to the list. Ironically, it now seems to break *incoming* calls, which were working just fine before!! Here's a snippet from my CLI screen: May 3 13:43:14 NOTICE[114696]: chan_sip.c:3346 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again -- Got SIP response 482 Loop Detected back from 213.137.73.140 May 3 13:43:34 NOTICE[114696]: chan_sip.c:3346 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again -- Got SIP response 482 Loop Detected back from 213.137.73.140 May 3 13:43:54 NOTICE[114696]: chan_sip.c:3346 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again I changed nothing but for the CVS upgrade. Anyone know wtf is going on? Thx. B. ___ 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] Timeout Gives T in cdr.
Hans-Henrik Andresen wrote:
Hi,
If I do this in extensions.conf
exten = 411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],40,rS(10))
the line is cut of in 10 sec., thats fine, but in CDR I got dst as T,
and not 411.
How can I handle this so I still get kicked of after 10 sec., but
get 411 as dst in my cdr ?
I have worked around this issue by storing the extension in a variable,
then restoring it using a Goto in the 'T' processing. For example:
exten = 411,1,SetVar(ORIG_EXTEN=${EXTEN})
exten = 411,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],40,rS(10))
...
exten = 411,200,Playback(call-timed-out)
exten = 411,201,Hangup
exten = T,1,Goto(${ORIG_EXTEN},200)
I wonder if would make sense to add an additional column to the CDR
record to include the number that was originally dialed?
../fam
--
Frank A. Mandarino [EMAIL PROTECTED]
Spindrift Management, Toronto
416 642-3404
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Re: [Asterisk-Users] quad fxo
On Mon, 2004-05-03 at 12:39, Michael Sandee wrote: The bus isn't wrong... debian is wrong. Like everything in debian... it ships with an old pci.ids (No flames intended... but still :P ) If that was all, then it would have still showed up in the PCI bus just like you mention below. It also means he is probably running the STABLE version. Debian named it stable only because they don't change it very often. If you want something as new as the other distros, you have to go with Testing, or Unstable, possibly even Experimental. Just be glad they properly mark their releases unlike others whose x.0 releases shouldn't ever be trusted in a production environment. (Not flaming either, just filling in the picture) replace yours with one from: http://pciids.sourceforge.net/ And It *should* report it better... (Didn't verify) Same should work with an upgrade in debian. Not that any of this matters... Just load the driver and get on with it. While this may be true, the kernel should probably be recompiled for best performance on the new hardware. Steven Critchfield wrote: On Mon, 2004-05-03 at 12:26, Tim Sailer wrote: Folks, I'm trying to install one of the new quad fxo cards remotely. I know the existing machine was too old to have a PCI 2.2 bus, so I had my helper at the other end try a few boxes that were sitting on a shelf with the new card and a Knoppix cd. He found one that reported the card as the Tiger Jet. Good. Now, we moved the HD from the existing machine, loaded with Debian, and the card is just seen as the generic communications device, like the bus is wrong. Any pointers on this? The machine is ~500 miles away at the moment, and off the network, so most of this is done by phone. :( Maybe you should start by getting your helper to recompile the kernel so it can be on the network and you can then do some real debugging of the bus. I'm guessing that the kernel you have on that drive isn't sufficiently smart enough to handle the newer hardware. Knoppix CDs seem pretty decent at running the hardware fast since they are at such a disadvantage booting and running from the CD. ___ 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 -- Steven Critchfield [EMAIL PROTECTED] ___ 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] RE: FXS card dial digit wrong
Well, I just figured out that 4 digit dialing plan is used on the other
end so if I just 4 digit extension, i.e. 95222, the call works out fine
for both the sip phone and the pots phone.
However, I still don't understand that when I dial full 7 digit number,
sip phones work but pots does not. I'll ignore it for now:)
-Original Message-
From: Lisa Xie
Sent: Friday, April 30, 2004 9:52 AM
To: '[EMAIL PROTECTED]'
Subject: FXS card dial digit wrong
Hello, everyone,
I am currently trying to get the asterisk server to talk with a 3COM NBX
with T1 connection. My asterisk server has a T100p, TDM20B, a couple of
sip phones. Now the sip phones are calling 3COM NBX phones fine,
however, the analog phone has problem when dialing the NBX phones. The
connection is established and the NBX auto-attendant picks up the call
however the NBX end says that incorrect extension number is dialed,
From 3com NBX end to Asterisk is fine, i.e., 3com NBX phones call both
the sip phone and the analog phone with no problem.
Below is the console output from Asterisk when I tried to call the same
extension using both the sip phone and the analog phone.
Also my configuration files are attached: zapata.conf, zaptel.conf,
extensions.conf.
Thanks for your help.
Lisa
~~~Console output from Asterisk~~~
*CLI
-- Executing Dial(SIP/2001-2445, Zap/g1/222) in new stack
-- Called g1/222
-- Zap/1-1 answered SIP/2001-2445
-- Hungup 'Zap/1-1'
== Spawn extension (internal, 9222, 1) exited non-zero on
'SIP/2001-2445'
-- Starting simple switch on 'Zap/26-1'
-- Executing Dial(Zap/26-1, Zap/g1/222) in new stack
-- Called g1/222
-- Zap/1-1 answered Zap/26-1
-- Attempting native bridge of Zap/26-1 and Zap/1-1
-- Hungup 'Zap/1-1'
== Spawn extension (internal, 9222, 1) exited non-zero on
'Zap/26-1'
-- Hungup 'Zap/26-1'
~~~My configuration files are here~~~
---Zaptel.conf---
#add t100 card
span=1,0,0,esf,b8zs
em=1-24
loadzone = us
defaultzone=us
#add tdm20b card
fxoks=25-26
---Zapata.conf
;add for t100 card
signalling=em_w
context=incoming
group=1
immediate=yes
channel = 1-24
;add for tdm20b card
signalling=fxo_ks
context=internal
channel=25-26
---Extensions.conf---
[incoming]
exten = _XXX2001.,1,Dial(SIP/2001,20)
exten = _XXX2101.,1,Dial,Zap/26
include = internal
[internal]
exten = s,1,Playback(demo-congrats)
exten = 2001,1,Dial(SIP/2001,20)
exten = 2100,1,Dial,Zap/25
exten = 2101,1,Dial,Zap/26
;outbound calls
exten =_9.,1,Dial(Zap/g1/${EXTEN:1})
-Original Message-
From: Lisa Xie
Sent: Wednesday, April 28, 2004 5:51 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Extra digit needed for outbound call
Here is part of the files: extensions.conf for both of the servers
Asterisk server 1
[incoming]
include = internal
[internal]
exten = s,1,Playback(demo-congrats)
exten = 2000,1,Dial(SIP/2000,20)
exten = 2100,1,Dial,Zap/25
;outbound calls
ignorepat = 9
exten =_9.,1,Dial(Zap/g1/${EXTEN:1})
Asterisk server 2
[incoming]
include = internal
[internal]
exten = s,1,Playback(demo-congrats)
exten = 2000,1,Dial(SIP/2000,20)
;outbound calls
ignorepat = 9
exten =_9.,1,Dial(Zap/g1/${EXTEN:1})
Thanks!
Lisa
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, April 28, 2004 4:39 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Extra digit needed for outbound call
On Wed, 2004-04-28 at 14:49, Lisa Xie wrote:
Hi,
I've been working on starting a lab of end to end asterisk system and
now most of pieces seem to be working. The two asterisk servers are
connected by T1. Both servers have a couple of SIP phones connected
and
one of the servers has a FXS card with an analog phone hanging.
I can make calls across the T1 link however there is one thing that I
don't understand. I need to append one extra digit to get the correct
extension number at the other end. For example, when I tried to call
extension 2000 at the other end, I need to dial 92000x, where x can
be
anything between 0-9. Otherwise, if I dial 92000, the console says
something like extension 200 is not found.
Also internal calls are normal.
This looks very bizarre for me... How can I fix it?
Examine your dialplan. Post it here and maybe someone will point out
what you are doing wrong.
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] dialing a remote phone system and then entering an extension
On Mon, 3 May 2004, Joel Duffield waxed: I am trying to get a way to have * forward calls that are dialed to an extension, to end up at an extension on my old analog phone system. I will have 7 lines coming into * using the new Digium cards via PSTN, and then lines coming from * into the PSTN lines on the analog system. So that if for example someone dials extension 110: The system will call the analog system, the system will assume that a call is coming from the telco as always, pick up right away, and then listen for an extension to be entered. This should then connect the incoming call to the extension on the analog system. My question is, does my logic work, and also if I use the dial command, and I set the analog system to pick up immediately, will wait long enough before it dials? If that wouldn't work is there a way that I can tell * to dial then wait and then send digits? So, the legacy PBX already provides an analogue to *'s Background app, ie., you dial in and it sits to wait for the entered extension ? Then you might as well just bridge it right through * exten = s,1,Dial(Zap/g1/legacy_background_extension) Since the Dial app isn't eating the DTMF, it should just pass thru to the legacy PBX. If, however, you want to use * to do the Background, then dial that extension on the legacy PBX: exten = 110,1,Dial(Zap/g1/110) Probably a better option because it gets you migrating to * quicker. And I assume you are upgrading. I don't think you need to worry about the wait, * handles analog interfaces and this is a requirement of such an interface. Meaning it won't send audio while on hook, but wait for an answer. There's even support for pulse dialing in *, if it is that much of a legacy PBX. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] iconnecthere behind NAT, strange deal
Check 0001436 in the bugtracker. This was the original bug fix which broke outbound calls. Additional work was done on this bug to fix a problem with incoming calls (see marks comments at the end). Maybe you got a CVS while this was being worked on? Maybe there is still a problem? Bill Doll Jr Brian Capouch [EMAIL PROTECTED] Brian Capouch [EMAIL PROTECTED] Sent by: [EMAIL PROTECTED] 05/03/2004 11:47 AM Please respond to [EMAIL PROTECTED] To [EMAIL PROTECTED] cc Subject Re: [Asterisk-Users] iconnecthere behind NAT, strange deal Lists wrote: Have you found a solution yet? I am having the same issue. My account works fine with the IConnectHere soft phone client but not with Asterisk. Inbound lines work fine, outbound returns the same message. CVS update took care of my problem wrt outbound calls; it must have been the mis-authentication stuff referenced in other mails to the list. Ironically, it now seems to break *incoming* calls, which were working just fine before!! Here's a snippet from my CLI screen: May 3 13:43:14 NOTICE[114696]: chan_sip.c:3346 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again -- Got SIP response 482 Loop Detected back from 213.137.73.140 May 3 13:43:34 NOTICE[114696]: chan_sip.c:3346 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again -- Got SIP response 482 Loop Detected back from 213.137.73.140 May 3 13:43:54 NOTICE[114696]: chan_sip.c:3346 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again I changed nothing but for the CVS upgrade. Anyone know wtf is going on? Thx. B. ___ 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 inline: graycol.gifinline: pic00041.gifinline: ecblank.gif
Re: [Asterisk-Users] quad fxo
On Mon, May 03, 2004 at 07:39:06PM +0200, Michael Sandee wrote: The bus isn't wrong... debian is wrong. Like everything in debian... it ships with an old pci.ids (No flames intended... but still :P ) replace yours with one from: http://pciids.sourceforge.net/ And It *should* report it better... (Didn't verify) Not that any of this matters... Just load the driver and get on with it. Driver didn't load either. So I'm supposing that it's the bus. Hmm. MAybe that was on another machine we tried. I'll give it a whirl. Thanks, Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ 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] Start recording during call by pressing button sequence
brian wrote: If you live in a one-party state you can record ANY call in or out .. doesn't matter if the call comes from/to out of state or not. You are within the rights of your state. I live in Oklahoma and I record EVERY call in our out of my house with asterisk. Is there a list somewhere of one party versus (I assume) two party states? Thx. B. ___ 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] How does Novergence do it ?
I had just about about sold a new asterisk phone system to a local company when they called back asking if I could match a proposal from Novergence.com. I haven't seen anything on paper but was told their proposal was to provide a new phone system that would replace the existing 8 line 12 extension system, provide an internet T-1, unlimited local and long distance, voice mail, and two cellular phones with unlimited nationwide minutes all for the same $500 per month the business is spending now. The internet T-1 would be at least $500 so I'm a bit confused as to how they go about doing this. Does anyone have any details about Novergence and their phone systems and service ??? Thanks, Lance ___ 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] quad fxo
On Mon, May 03, 2004 at 03:13:56PM -0400, Tim Sailer wrote: On Mon, May 03, 2004 at 07:39:06PM +0200, Michael Sandee wrote: The bus isn't wrong... debian is wrong. Like everything in debian... it ships with an old pci.ids (No flames intended... but still :P ) replace yours with one from: http://pciids.sourceforge.net/ And It *should* report it better... (Didn't verify) Not that any of this matters... Just load the driver and get on with it. Driver didn't load either. So I'm supposing that it's the bus. Hmm. MAybe that was on another machine we tried. I'll give it a whirl. Getting the latest CVS zapatel seems to have done the trick. Now to test this live. :( Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ 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] Start recording during call by pressing button sequence
On Mon, May 03, 2004 at 02:15:09PM -0500, Brian Capouch said: Is there a list somewhere of one party versus (I assume) two party states? Google for telephone recording law ___ 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] Start recording during call by pressing button sequence
On Mon, 2004-05-03 at 14:15, Brian Capouch wrote: brian wrote: If you live in a one-party state you can record ANY call in or out .. doesn't matter if the call comes from/to out of state or not. You are within the rights of your state. I live in Oklahoma and I record EVERY call in our out of my house with asterisk. Is there a list somewhere of one party versus (I assume) two party states? 1, 2, 3, 4, 5, 6, 7, 8, 9, 10 Nope, didn't reduce frustration level. Whats wrong with your google interface? You posted about just short of 2 hours after Walt Reed posted a link to a site with the relevant information. Even if you didn't get his information in time, goole provides for a wonderful list of such resources. The terms I used are as follows, call recording law state. Since you seem to use english well, you should have been able to figure out those keywords. The first link on the list google provided to me was to RCFP, the same site Walt Reed mentioned a couple of hours ago. Then there are probably a few dozen more examples after that. Please, use a little effort, don't spam the list with easily answered questions that a child would be expected to look up and find the answer to. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] quad fxo
On Mon, 2004-05-03 at 13:18, Michael Sandee wrote: Steven Critchfield wrote: On Mon, 2004-05-03 at 12:39, Michael Sandee wrote: It also means he is probably running the STABLE version. Debian named it stable only because they don't change it very often. If you want something as new as the other distros, you have to go with Testing, or Unstable, possibly even Experimental. Just be glad they properly mark their releases unlike others whose x.0 releases shouldn't ever be trusted in a production environment. Well... tell me what is unstable about putting a new pci chip identification database into your distro? I run debian stable on my workstation... Some things are ok to be stable... but things like this are...well not so nice. Remember stable refers to change levels, not stability of software. Think of it like this, you could have a piece of software that fell over dead every time the wind blew, but it would be considered a stable version if it didn't change very often over time. Specifically, Todays version of stable shouldn't really change unless there is a REALLY good reason. Even then, the changes are usually part of a security add-on and not part of the main stable release. Changing the PDI ID database could potentially break something else that expected that card to slightly misrepresent. All of it is erring on the side of super caution. If you want to ride the cutting edge, you can choose the other versions. The choice is how much blood are you willing to lose as you ride the cutting edge. The closer you go, the more likely a upgrade will break something you hold critical. replace yours with one from: http://pciids.sourceforge.net/ And It *should* report it better... (Didn't verify) Same should work with an upgrade in debian. I seriously advice against that... vividly remebering the NPTL debacle in unstable... and loads of other glibc problems you can read about in the bugtracker. I haven't seen any NPTL stuff in debian. My laptop is fairly regularly synced with unstable, and the same goes for my home machine. My laptop is very stable while my home machine, well I'm still sorting out a hardware problem that makes it crash(hard lock, no console messages) with heavy network or disk activity. While we are at the subject anyway I can put some on-topic info here aswell. Recent benchmarks with 6 QuadBRI's on a Pentium-4 2.8Ghz resulted in a almost 100% improvement in load under Linux 2.6(.4) over Linux 2.4(.25). This ofcourse resulted in less (no) quirks in the sound under 2.6 than under 2.4. The loadtest was done looping the BRI's to each other and in such way that we used all 24 BRI's (2 channels) resulting in 48 active channels. http://voidptr.astmaster.org/loadtest1.jpg http://voidptr.astmaster.org/loadtest2.jpg The audio quality was easily monitored in contrary to often proposed test suites... Two phones at each end... Now when will they be able to work in the US, and what is the pricing for the cards? Truly impressed. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] How does Norvergence do it ?
ooops, this should have been norvergence.com -- Lance ___ 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] How do you close a VoicePulse Connect! account?
As best I can tell you remove your credit card info, cancel any phone numbers you have, and turn off the automatic billing stuff and when your account hits 0 your canceled. Scott - Original Message - From: Brian Cuthie [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 03, 2004 8:46 PM Subject: [Asterisk-Users] How do you close a VoicePulse Connect! account? Anybody figured out how to close a VoicePulse Connect! account? As bad as their web site is at most other things, the notion of actually closing an account doesn't appear to have even been contemplated. -brian ___ 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
Re: [Asterisk-Users] Start recording during call by pressing button sequence
I record ALL calls in and out of my house. If someone calls me too bad. Its not illegal in my state. bkw - Original Message - From: C. Maj [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 03, 2004 4:48 PM Subject: Re: [Asterisk-Users] Start recording during call by pressing button sequence On Mon, 3 May 2004, Steven Critchfield waxed: On Mon, 2004-05-03 at 14:15, Brian Capouch wrote: brian wrote: If you live in a one-party state you can record ANY call in or out .. doesn't matter if the call comes from/to out of state or not. You are within the rights of your state. I live in Oklahoma and I record EVERY call in our out of my house with asterisk. Is there a list somewhere of one party versus (I assume) two party states? 1, 2, 3, 4, 5, 6, 7, 8, 9, 10 Nope, didn't reduce frustration level. Whats wrong with your google interface? You posted about just short of 2 hours after Walt Reed posted a link to a site with the relevant information. I thought even better was my post from 2 *days* ago with just such a link. Granted, it's the ACLU, and I sent it on May Day so y'all were prolly out marching, but... a href=previous post in this thread http://archive.aclu.org/issues/cyber/phonelaw.html /a And I'm no legal expert either, but if you cross state lines, you are dealing with 2 sets of state laws. To think that only your state's laws matter is naive at best. Worse yet, you are under federal jurisdiction, too. You simply can't record everything going in and out of your * box, just because the feature is there doesn't make it legal. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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
Re: [Asterisk-Users] grandstream transfer, park and conference
Your English is just fine. :) What's your extensions.conf and sip.conf for your Grandstreams look like? What are your options in the GS config webpage for: 1) NAT transversal (and are you behind a NAT firewall) 2) Send Flash event 3) Send DTMF Best regards, Ryan Thrash On May 3, 2004, at 8:51 PM, Ing Isianto Istiadi wrote: I have 2 grandstream budgetone 100 series. I can call allright, but I cant do call transfer, park and call conference. (all features works with tdm devices) the The budgetone using 1.0.4.55. 1. If I called using sip to sip (from phone a to phone b), I cant transfer it at all or parking it or dial to conference. 2. if the call come from pstn, then the first phone who answer can park the call, and be picked up by the second phone, but after that the parking stuff wont work anymore. (it seems asterisk doesnt recognize #) 3. Ive already set dtmf to info 4. It seems on case 2 above, that even the # works for the first call from pstn to sip, but asterisk only recognize at most 2 digit after # being pressed (for example, I have ext 700 to park the call, when I look at * console, it only receive 70) Thanks and forgive my English ___ 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] Number of Digium cards in one box...
- Original Message - From: Alex Lopez [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 03, 2004 5:37 PM Subject: [Asterisk-Users] Number of Digium cards in one box... How much overhead would the 4 port cards put on a system?? At what point would the breakeven point be?? -- I have a deployed system with a 405, 2x40b, and a 31b. 3 spans on the 405 are in use, 1 to a provider, 1 in a p2p with another building, and 1 to a 750 with 6 fxs modules. model name : Intel(R) Pentium(R) 4 CPU 2.40GHz stepping: 5 cpu MHz : 2394.206 cache size : 512 KB [EMAIL PROTECTED]:~# uptime 19:10:42 up 6 days, 12:07, 2 users, load average: 0.78, 0.87, 0.88 0: 56208454 XT-PIC timer 1: 10999 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 562032946 XT-PIC t4xxp 5: 699933781 XT-PIC Intel ICH5, wctdm 8: 1 XT-PIC rtc 9: 562017411 XT-PIC usb-uhci, wctdm 10: 562030564 XT-PIC usb-uhci, wctdm 11:7923921 XT-PIC 3ware Storage Controller, usb-uhci, usb-uhci, eth0 12: 3015 XT-PIC PS/2 Mouse 15: 7 XT-PIC ide1 NMI: 0 ERR: 0 ___ 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] Number of Digium cards in one box...
I know, I know, check the archives but I can't find an answer since the new cards are well, NEW! I understand the whole issue of expandability and flexibility of using a T1 card and an Adtran 750. FXO or FXS, you mix and match. With the new card offerings from Digium I can easily put a 4CO by 8 Station system together. Barring the extra interrupts, extra CPU Cycles of extra interfaces, is there a reason why I would not do this?? How much overhead would the 4 port cards put on a system?? At what point would the breakeven point be?? The cost of the T1 card is not the problem, the cost of a channel bank, New, no-ebay, or used stuff here. I want to compare new apples to new apples, not used. The new TDM04B card (old card with new daughter boards) technical specs seem to be somewhat of an unpublished thing. System with two x100p's and one TDM04B shows: [EMAIL PROTECTED] cat /proc/interrupts CPU0 0:9953447 XT-PIC timer 1:253 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 0 XT-PIC usb-uhci 8: 1 XT-PIC rtc 9: 204875095 XT-PIC ehci-hcd, eth0, Intel ICH4, wcfxo, wctdm 10: 0 XT-PIC usb-uhci 11: 99034988 XT-PIC usb-uhci, wcfxo 12: 3451 XT-PIC PS/2 Mouse 14: 156855 XT-PIC ide0 15: 787553 XT-PIC ide1 NMI: 0 ERR: 0 [EMAIL PROTECTED] Don't see any interrupts associated with the wcfxs card. (Oh, I was told by digium support the software routines for the FXO daughter boards are in the wcfxs module. Looking at the file dates, that appears to be correct. But, no interrupts. Cool!) Don't have a clue what the limitations might be at this time, however it would appear: a. echo issues with the FXO card are identical (if not worse) then the x100p's. Looks/feels like the x100p prior to about Nov 2003. Thirty seconds of decreasing echo after anwser. b. the FXO card spontanously thinks the pstn line is ringing, and executes the dialplan entry assoicated with that. Rings twice and then disappears. (Note: callprogress=no/yes has absolutely no impact, as though it was not implemented on the FXO card.) c. CallerID seems to be a less reliable then the x100p. Nothing to back that up other then gut feeling. d. transmission levels are good and adjustable via rxgain/txgain, but has little impact on the stock echo problem from what I've seen thus far. Based on informal comments, I'd have to guess the software necessary to support the real FXO hardware needs are lagging by a fair amount. Still testing though... Rich ___ 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] How do you close a VoicePulse Connect! account?
Anybody figured out how to close a VoicePulse Connect! account? As bad as their web site is at most other things, the notion of actually closing an account doesn't appear to have even been contemplated. -brian ___ 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] A-B ok; B-C ok; A-C Crap
Here's a puzzlers for you expert Asteriskians: Person A in Australia with Xten-Lite connects to: Person B on an Asterisk server in Vancouver, Canada (B is on analog phone to FXS port) quality is fine. Person B on an Asterisk server in Vancouver IAX trunks to our Vancouver PSTN gateway (running Asterisk and connected to a PRI) and calls (placing a local landline call) to Person C on a regular PSTN phone line quality is fine. Person A links to B, trunks to C, connects to local Vancouver number. quality in both directions is crappy -- choppy to the point of unintelligible. So each leg tested separately works fine. The two put together do not. Where should we look? What parts of which configurations do you need to see to diagnose the problem? Thanks for any and all suggestions. g. ___ 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] RFD: With echo and other distortion, can ulaw/alaw line quality ever be good enough for faxing?
Folks, I've been following recent discussions regarding echo and echo training with much interest, since it's a problem we've never been able to eliminate here. We're facing two challenges presently, and they may be related (or not): a) Cisco 7960s in the office here echo back to our staff, but the customer hears decent sound. b) We've yet to be able to pass faxes through asterisk, despite trying a number of approaches A bit about our setup: Two analog lines come into the office, into the FXO card of an CAC ADIT 600. These are 'connected' to the TDM interface of the ADIT, and go out to Asterisk via 2 channels of a a 24-channel fxs_ks line into a port of a TE405P. We have two other ports on the TE405P occupied by digital fax boards, connected to asterisk via ATT 5ESS PRI. And of course there's the 7960 handsets, connected directly to a switch at 100MBps (fdx). Challenge a) = In terms of looking into a), we started with: http://asterisk.sohoskyway.net/Asterisk_Doc/current/docs-html/x939.html Clearly some of the details are outdated, since the echo canceller options live in zconfig.h now, not a Makefile, but it's a decent starting point. Which gives rise to my first question - which echo canceler should we be using? We have Steve, Steve 2, and Mark 1-3, with an aggressive option with 3 that could make calls scratchy ;-) Is there any conventional wisdom here? Next question ... we have control over gain at the zaptel level, and also at the ADIT level apparently. Which should we use when trying to adjust things with ztmonitor? Next question ... it's not clear to me what the target is with ztmonitor. Are we shooting for tx and rx levels that are balanced. and about 50% of the scale at their max energy? I found that when I called a voicemail service and listened to the auto-attendant that the RX meter of ztmonitor was about 50%, but I found the TX meter fluctuated wildly when I spoke. I could reduce my voice level slightly and it would be very low energy, but with a slight increase in volume it was pegged. The settings I made seemed to do nothing to change this (I stop asterisk, unload and reload all zaptel modules, and restart astertisk between each test). Is this really supposed to work? I've managed to get the connection extremely distorted with some settings, but have yet to make a change that improved the quality or removed the echo. Challenge b) In some ways, this is the most worrying problem for us, since we would love to be able to pass faxes through asterisk reliably, and from the traffic on this list and the fine efforts of Mr. Underwood, it seems many others would too. People have spoken here of the nearly immediate echo cancelation on pure TDM circuits, and so I would have thought we'd escape the echo problems above in either of the following setups: i) Plain ole' HP fax machine plugged into TDM400 FXS card ii) Brooktrout/Eicon fax board connected to TE405P In practice we're rarely able to train with the remote fax machine when dialing to an outside line through asterisk and the ADIT 600. If we do manage to get connected and the connection supports ECM error correction, we can see lots of retransmitted frames which again points to very poor line quality. In contrast, by connecting directly to the analog lines we can send faxes all day with the trusty little HP ... ie: the lines aren't inherently bad. I guess what I'm looking for here is a sanity check ... are we trying to push the limits here, or should this stuff be working much better than this? We're willing to invest some time getting faxing to work, but I'd hate to ask people here to dedicate their time and energy to something that's never really going to work well due to limitations in Asterisk/zaptel technology. -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 +1.215.243.8335 (fax ___ 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] grandstream transfer, park and conference
What's your extensions.conf and sip.conf for your Grandstreams look like? I'm not in my machine right now, but here's the relevant configs Extensions.conf [ext] Ignorepat=9 Exten=_9XX,1,Dial(zap/1,tTr,20) Exten=_9XX,2,hangup [sip] Include=ext Include=parkedcalls Sip.conf Posrt = 5060 Bindaddr = 0.0.0.0 [Isianto] Type=friend Username=Isianto Secret=xxx Host=dynamic Qualify=50 Context=sip Mailbox=22 Disallow=all Allow=gsm Allow=ulaw Dtmfmode=info [Istiadi] Type=friend Username=istiadi Secret=xxx Host=dynamic Qualify=50 Context=sip Mailbox=23 Disallow=all Allow=gsm Allow=ulaw Dtmfmode=info What are your options in the GS config webpage for: 1) NAT transversal (and are you behind a NAT firewall) Set to no 2) Send Flash event Set to no 3) Send DTMF Dtmf = info (I tried rfcxxx, I forgot) Thanks Isianto ___ 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
