Re: [Asterisk-Users] sending fax
On Thu, Sep 08, 2005 at 06:42:49PM -0500, Chris Shipman wrote: I started working on a program using Ghostscript and Redmon to generate the tif in windows by a printer. So far I am using FTP to transfer the tiff and call file. At least until I figure something better out. Why don't you look at IPP (Internet Printing Protocol)? a protocol for submitting jobs over HTTP of some sort. Server is already implemented in e.g. cups. HTTP allows a nice header with some extra fields. I wonder if that can be abused to get the call information through. (and am I re-inventing some wheels in the process?) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Motherboard and processor recommendations
Hi I discovered that most onboard raid controllers are really software raid, and it uses the cpu to perform raid functions. I am not sure how much extra load this introduces, but anyway, its still not ideal when you need your cpu for transcoding voip stuff. my 2c. regards Clive On 8 Sep 2005 at 12:01, Soner Tari wrote: Thanks Tzafrir and canuck15 for your comments. Yes I don't think the NIC will be saturated, and I'll search the quality of the Onboard RAID. I guess I have to learn more about canuck15's comments though, because I am actually questioning what happens to the board when you're adding onboard peripherals and whether that would create problems with, say, Digium cards. I remember I've read comments on the list saying that some chipsets/motherboards cannot handle the interrupt frequency that Digium cards demand, thus miss some interrupts. So, even though a regular desktop user would not notice any problems, an Asterisk server would suffer a lot. But I'm afraid there is no rule of thumb on such matters (except Xeon motherboards?). The load on the computer will never be too high, but my purpose in asking about processor preference is that if there is any processor dependant dsp routines (such as G729 codec), then I thought that I might have problems. As another example, I don't know the details of the echocancelers on Asterisk (all 5 of them), but perhaps their performance is more satisfactory on, say, a P4 2.4 machine rather than, say, an AMD64, even though I'd expect AMD64 to be a more powerful processor. So I am questioning code compatibility/performance based on processor type rather than processor load. If that's irrelevant, please disregard this question (I need to learn more about dsp routines). Thanks again for your answers, Soner - Original Message - From: canuck15 [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, September 08, 2005 3:46 AM Subject: RE: [Asterisk-Users] Motherboard and processor recommendations Regarding Chipsets/Motherboards. I would stay FAR away from cheap ones. Any chipset/motherboard that electrically and logically separates some PCI slots (ie. interrupts) from onboard peripherals (network controller, VGA, USB etc.) makes compatibility issues with Digium cards much less likely. Many of the newer Intel chipsets do this. The Xeon chipsets/motherboards are the best IMHO because they usually have PCI-X slots connected directly to the memory controller hub, that you can put your Digium card(s) in, which are completely separate from the peripherals and PCI slots on the I/O controller hub. -Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 07, 2005 4:59 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Motherboard and processor recommendations On Wed, Sep 07, 2005 at 11:02:58PM +0300, Soner Tari wrote: Hi All, For sometime now I've been searching the wiki and googling, but I think I'm missing some of the very important answers. So I'll have to ask this to the list. I'm trying to decide on the right motherboard and processor. Here are my questions: 1. Would I have problems with all-onboard motherboards (Onboard VGA, LAN/GLAN, Sound, SATA, RAID) ? I've read the comment about an Onboard VGA on wiki. Considering the exceptional quality of graphics you'll need with Asterisk, and VGA-compatible adapter would suffice. The on-board one would be more than enough. Ditto for the sound card, at least in most cases. As for the network adapter: Are you going to get anything close to saturating the card? I figure that the efficiency of the network adapter and its driver will not be your bottleneck. Most of the WAN-oriented systems would have worked fine with an old 10Mbps card, probably without a noticable performance hit (right?). So their quality is not much of an issue. If you have the extra space, you can always add an extra one in an expansion slot. But it should not be required. An extra raid controller is something you may consider. But then-again, if it is a cheap software-based raid, it is practically the same as using linux for that (but with more problematic drivers). But it is for you to decide if it is worth the extra cost. 2. Which chipset should I prefer: Intel, SiS or VIA? I've read the old SiS chipset problem on wiki. There is much voodoo about this. There are good and bad boards made with each of those chipsets. In fact, for practically each model of board that has been sold for over a month or so, you'll probably find someone in this list who had bad experience with it. 3. Which processor has the least support problems: P4 (478 or LGA775, or even EMT64) or AMD64 ? For example, in G729 config file Athlon comment
Re: [Asterisk-Users] MAX PRI for single server
On Thu, 8 Sep 2005, Matthew Boehm wrote: Jason Becker wrote: Hmm, looks like someone in the know needs to update the wiki: http://www.voip-info.org/tiki-index.php?page=Digium+DS3000P Wow. Guess I'm not. I've got a 4 port PRI card in this brand new Dell 1850 3.0Ghz Xeon with 2GB RAM and I run an average of 50-60% CPU usage with just 47 calls. All G711. Yow and Huh? I have test 3.0GHz systems - Intel Desktop board. I've been testing with a TE405P with looped ports - 1 to 3, 2 to 4. My test is 20 second long calls with one side playing music on hold, the other playing gsm prompts. All channels full (60 calls out, 60 in). The system runs just under 40% CPU load - 1/2 system time, half user. This is with echo cancellation turned off. Incidentally, I successfully ran more than 3,000,000 calls through this hookup over a week and it was completely solid. I did try two TE405P with 8 looped spans - that did pretty much use up the CPU, but subjective call quality was still fine. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX PRI for single server
On Thu, 8 Sep 2005, Matthew Boehm wrote: Carlos Antunes wrote: Have you seen this? http://www.digium.com/index.php?menu=compatibility Yes, but I'm not using a Digium card. Ah - so the difference between your setup and mine is that you are using Sangoma (presumably) and I'm using Digium. Looks like the Digium is significantly more efficient then. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using E1 without power off simence pbx
hello everybody, i used asterisk with 2 ISDN BRI AVM cards in paralel with a panasonic ISDN pbx for testing putpose. is this also possible to use E1PRI to use in parallel with a simence PRI pbx for test purpose? |---asterisk public line| |---PBX best regards shaon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voice over atlantic
On Thu, Sep 08, 2005 at 04:49:49PM -0400, David Hajek wrote: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? Also consider iLBC . gsm consumes less CPU than either of those two. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Motherboard and processor recommendations
On Fri, Sep 09, 2005 at 08:05:09AM +0200, Clive wrote: Hi I discovered that most onboard raid controllers are really software raid, and it uses the cpu to perform raid functions. Also: in such a settings you can get comperable performance by using Linux's built-in software raid. And for that you won't depend on non-standard drivers from the vendor for that. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Motherboard and processor recommendations
Hi, I sucked the TE410 in a Siemens dual Xeon machine... lot of irq problems, digium support said: try the card in another machine. A cheap amd64 + via K8T800 and TE405 works perfectly... On Fri, 2005-09-09 at 08:05 +0200, Clive wrote: Hi I discovered that most onboard raid controllers are really software raid, and it uses the cpu to perform raid functions. I am not sure how much extra load this introduces, but anyway, its still not ideal when you need your cpu for transcoding voip stuff. my 2c. regards Clive On 8 Sep 2005 at 12:01, Soner Tari wrote: Thanks Tzafrir and canuck15 for your comments. Yes I don't think the NIC will be saturated, and I'll search the quality of the Onboard RAID. I guess I have to learn more about canuck15's comments though, because I am actually questioning what happens to the board when you're adding onboard peripherals and whether that would create problems with, say, Digium cards. I remember I've read comments on the list saying that some chipsets/motherboards cannot handle the interrupt frequency that Digium cards demand, thus miss some interrupts. So, even though a regular desktop user would not notice any problems, an Asterisk server would suffer a lot. But I'm afraid there is no rule of thumb on such matters (except Xeon motherboards?). The load on the computer will never be too high, but my purpose in asking about processor preference is that if there is any processor dependant dsp routines (such as G729 codec), then I thought that I might have problems. As another example, I don't know the details of the echocancelers on Asterisk (all 5 of them), but perhaps their performance is more satisfactory on, say, a P4 2.4 machine rather than, say, an AMD64, even though I'd expect AMD64 to be a more powerful processor. So I am questioning code compatibility/performance based on processor type rather than processor load. If that's irrelevant, please disregard this question (I need to learn more about dsp routines). Thanks again for your answers, Soner - Original Message - From: canuck15 [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, September 08, 2005 3:46 AM Subject: RE: [Asterisk-Users] Motherboard and processor recommendations Regarding Chipsets/Motherboards. I would stay FAR away from cheap ones. Any chipset/motherboard that electrically and logically separates some PCI slots (ie. interrupts) from onboard peripherals (network controller, VGA, USB etc.) makes compatibility issues with Digium cards much less likely. Many of the newer Intel chipsets do this. The Xeon chipsets/motherboards are the best IMHO because they usually have PCI-X slots connected directly to the memory controller hub, that you can put your Digium card(s) in, which are completely separate from the peripherals and PCI slots on the I/O controller hub. -Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 07, 2005 4:59 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Motherboard and processor recommendations On Wed, Sep 07, 2005 at 11:02:58PM +0300, Soner Tari wrote: Hi All, For sometime now I've been searching the wiki and googling, but I think I'm missing some of the very important answers. So I'll have to ask this to the list. I'm trying to decide on the right motherboard and processor. Here are my questions: 1. Would I have problems with all-onboard motherboards (Onboard VGA, LAN/GLAN, Sound, SATA, RAID) ? I've read the comment about an Onboard VGA on wiki. Considering the exceptional quality of graphics you'll need with Asterisk, and VGA-compatible adapter would suffice. The on-board one would be more than enough. Ditto for the sound card, at least in most cases. As for the network adapter: Are you going to get anything close to saturating the card? I figure that the efficiency of the network adapter and its driver will not be your bottleneck. Most of the WAN-oriented systems would have worked fine with an old 10Mbps card, probably without a noticable performance hit (right?). So their quality is not much of an issue. If you have the extra space, you can always add an extra one in an expansion slot. But it should not be required. An extra raid controller is something you may consider. But then-again, if it is a cheap software-based raid, it is practically the same as using linux for that (but with more problematic drivers). But it is for you to decide if it is worth the extra cost. 2. Which chipset should I prefer: Intel, SiS or VIA? I've read the old SiS chipset problem on wiki. There is much voodoo about this. There are good and bad
[Asterisk-Users] the number of incoming calls in queue
Hi, I had written a web application for queue report. In that I had calculate the incoming calls through parse the queue_log and the return info from management API. But for the realtime refresh about the current status, it is a little affectionto voice quality of our call center system, I want to move all the database and web application to another machine. But the only problem is the queue_log which can not be removed from local server. So I want to know, can I get the incoming call numbers in queue through other ways instead using queue_log? Any help and advice will be appreciated! Best Regards, Gary Li DO YOU YAHOO!? 雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P not detecting hangup and not hanging up
Canuck15, No, I hadn't played with the gains. But I've now done so and no difference unfortunately. Thanks for the suggestion though. I have discovered that after Asterisk has answered the call and the remote caller has hung up, if I lift the receiver on a phone connected to the line (in parallel with Asterisk), Asterisk then DOES instantly hang up. Would it be reasonable to assume the voltage drop caused by lifting the receiver causes this? It only happens when I set the BATT_(whatever it was) in wcfxs.c to 8. If I set it to a lower level, Asterisk won't even answer at all and so nothing works. Also possibly relevant: When I disconnect Asterisk completely from the equation and just answer the remote caller myself, when the remote caller hangs up the line does not actually drop: Instead I just get a disconnect (or number unobtainable) tone. Could this be the problem (i.e. there's no actual voltage drop happening to signal the call has ended)? Or is there some sort of other change in the line that I wouldn't detect audibly? Could it be that any inaudible voltage drop might be happening too quickly for zaptel to detect? What might I change in the source code to see if this is the case? Does nobody else in the UK use these cards? I'm sure that's not the case. So if you do use them, please stand up and be counted -- did you have to make any adjustments or did it just work out of the box? Incidentally, when callprogress=yes, Asterisk goes nuts and keeps detecting strange things happening: Essentially every time the CLI comes through (polarity reversal) between rings, asterisk picks up and hangs up (though not physically - the caller hears ringing). This may or may not be related but have you tried adjusting your RX and TX gains? I see both are at the default (0.0) which leads me to believe you have not. Search the Asterisk Wiki for the procedure. Stevanus, I think the hanguponpolarity switch is relevant to a patch to to Zaptel that may or may not have actually been added to the released version. I'm not sure. However, thanks for pointing this out -- I've tried it too and didn't get anywhere. I have similar problems like you. In the past, I did adjusted my RX and TX gain, but didn't know if it has been optimal yet. Fxotune is seemed do not working, perhaps caused of my asterisk's version ( I use stable v1.0).. Just curious, is rx and tx gain really a sole setting option here in order to make things the way it's meant to be? Or is there others? FYI, my tdm04b occasionally don't detect call-in as well as hangup signal. I've searched in the wiki and have activated hanguponpolarity swicth. But I don't notice any difference at all. Any help would be greatly appreciated. (I've asked this in another thread, but got no respon :( ) SUMMARY OF THREAD: hardware=TDM400P 2xFXS, 1xFXO. Location=UK. *ver=1.0.9. Zaptel 1.0.9.1. Problem: Asterisk does not detect that the remote caller has hung up and carries on as though nothing has happened. Disconnect Supervision/Hangup Detection has been discussed quite extensively on this list. But considering the information you provide and given that you tried to play with BATT_THRESH setting, I tend to think that there may be other problems. My concern would be the opermode setting of FXO modules. So I would get a fresh copy of asterisk and zaptel (and would not play with BATT_THRESH). I think the default opermode is FCC, so I would change it to UK. If it still does not work, I would try hanguponpolarityswitch (perhaps your telco provides disconnect supervision). If it still doesn't work, my only option would be busydetect. Hope this helps... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP/2.0 487 Request Terminated problem on Cisco 7960
Chris Stenton wrote: With todays CVS head I am getting the following being sent after a call has been terminated on my Cisco 7960. It eventually gives up with a critical error. chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) Any ideas I am sure it was working ok with cvs head a month ago. Chris Chris, one error message out of context won't say anything to me more than the phone is having a problem with it's mental state. Propably a cousin to Marwin, the depressed robot. Please give me a full SIP debug with verbose set to 4 and debug set to 4 so I can figure out what's going on!! /O ;-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sending fax
HI Chris I am interested, I would like to know how I can have the opportunity to test your program. On 9/9/05, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Sep 08, 2005 at 06:42:49PM -0500, Chris Shipman wrote: I started working on a program using Ghostscript and Redmon to generate the tif in windows by a printer. So far I am using FTP to transfer the tiff and call file.At least until I figure something better out.Why don't you look at IPP (Internet Printing Protocol)? a protocol forsubmitting jobs over HTTP of some sort. Server is already implemented ine.g. cups.HTTP allows a nice header with some extra fields. I wonder if that can be abused to get the call information through. (and am I re-inventingsome wheels in the process?)--Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's[EMAIL PROTECTED] | |bestICQ# 16849755 | | friend___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: SIP/2.0 487 Request Terminated problem onCisco 7960
This may also cause a hanging SIP channel. You can check it by issuing 'sip show channels' in CLI. CCF -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: Friday, September 09, 2005 16:52 To: Chris Stenton Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: SIP/2.0 487 Request Terminated problem onCisco 7960 Chris Stenton wrote: With todays CVS head I am getting the following being sent after a call has been terminated on my Cisco 7960. It eventually gives up with a critical error. chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) Any ideas I am sure it was working ok with cvs head a month ago. Chris Chris, one error message out of context won't say anything to me more than the phone is having a problem with it's mental state. Propably a cousin to Marwin, the depressed robot. Please give me a full SIP debug with verbose set to 4 and debug set to 4 so I can figure out what's going on!! /O ;-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: MAX PRI for single server
Yes, you missed something: 4 PRIs = 92 Lines per Card * 4 Cards = 368 Lines Isn't that just in North America? I believe most of the world uses E1 PRIs with 30 lines per PRI. right, we are in italy here, 1 PRI == 30 lines (calls) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups
i guess may be it's a 64bit variable. so you can only use 0-63. CCF -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of René Mayorga Sent: Wednesday, September 07, 2005 15:56 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups Hi, I'm working with this issue for a while, Now I already solve the dialplan issues, but I still have a question about the Callgroups, I read at www.voip-info.org that , there is a 63 limit of callgroups. And I'm wondering why?? and if the 1.2.0beta version supported more than 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any unoficial patch for that ? Thanks in advance. -- René Mayorga [EMAIL PROTECTED] El Salvador Telecom S.A. de C.V. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk pri heavy load testing (was MAX PRI for single server)
I have test 3.0GHz systems - Intel Desktop board. I've been testing with a TE405P with looped ports - 1 to 3, 2 to 4. My test is 20 second long calls with one side playing music on hold, the other playing gsm prompts. All channels full (60 calls out, 60 in). Niiice, can I ask what software/extension/script did you used to do such a test ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card
Hi all, I am new on this list an I hope the posting is correct. I am using Debian Sarge, kernel 2.6.13 from www.kernel.org and want to install the drivers for an AVM Fritz!PCI v2.0 ISDN card. I used the directions from AVM but Asterisk allways stops with CAPI not installed! I am new to Asterisk but this problem must already have been solved. I did not find a search function in the asterisk list. How can I find a thread to this and other problems ? Does anyone know of a helpful page ? Best regards Rainer Maier -- Lust, ein paar Euro nebenbei zu verdienen? Ohne Kosten, ohne Risiko! Satte Provisionen für GMX Partner: http://www.gmx.net/de/go/partner ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip log messages every few seconds
Andres wrote: 8.1.50;tag=as12a1c927 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY As you can see. This is just a NOTIFY message. Probably a Keep Alive. User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 No, it's a message waiting indication. You have requested this in sip.conf by entering a mailbox= entry in the peers configuration. /Olle --- Astricon 2005 - October 12-14 Anaheim, California Read the conference program on the web now htpp://www.astricon.net/2005/ - register today ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT Humo[u]r IVR Menu sample
Some one on another list I subscribe to had a session with an annoying IVR system at their doctor and posted this link. http://www.pendulum.org/humor/humor_psych_hotline.html -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card
In article [EMAIL PROTECTED], Rainer Maier [EMAIL PROTECTED] wrote: Hi all, I am new on this list an I hope the posting is correct. Welcome! I can't help with your ISDN problem, but I wanted to point out a common posting mistake that newcomers make, which you did also. When starting a new topic, please don't reply to an existing message to the list. That links your new topic into the middle of the existing thread in threaded mail clients, because it carries with it an In-Reply-To header pointing at the message you replied to. Always start a fresh topic with New Mail or similar (then enter the list address) and not Reply. Thanks! Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Huge Echo
Hello In the following setup: call coming from a pstn line - into FXO card - asterisk - SIP phone i get an incredible loud echo in the SIP phone (about 0,5-1s) (everything i speak into SIP phone microphone i hear in its speaker). The person calling from PSTN is not getting any echo. Which piece of the call could be causing the trouble so i can look into it? thanks, Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Debian Sarge, Kernel 2.6.13 and AVM Fritz!PCI v2.0 card
Hi all, I am new on this list an I hope the posting is correct. I am using Debian Sarge, kernel 2.6.13 from www.kernel.org and want to install the drivers for an AVM Fritz!PCI v2.0 ISDN card. I used the directions from AVM but Asterisk allways stops with CAPI not installed! I am new to Asterisk but this problem must already have been solved. I did not find a search function in the asterisk list. How can I find a thread to this and other problems ? Does anyone know of a helpful page ? Best regards Rainer Maier -- GMX DSL = Maximale Leistung zum minimalen Preis! 2000 MB nur 2,99, Flatrate ab 4,99 Euro/Monat: http://www.gmx.net/de/go/dsl ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Huge Echo
[EMAIL PROTECTED] wrote: In the following setup: call coming from a pstn line - into FXO card - asterisk - SIP phone i get an incredible loud echo in the SIP phone (about 0,5-1s) (everything i speak into SIP phone microphone i hear in its speaker). The person calling from PSTN is not getting any echo. Make sure you're not playing the recorded sound from your microphone back to your loudspeakers. -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siupra-2002 with astersik
Ahh wow.. that dial plan is seriously messed up... Try the default one... it will work alot better and give you less lag time between dialing a number and actually going through. On 9/8/05, Joseph [EMAIL PROTECTED] wrote: On Thu, 2005-09-08 at 23:29 +0200, Sander wrote: What is your problem with asterisk ans sipura ? Config files ?? Settings Give some more info on the problems Sipura-2002 CAN NOT dial out, incoming call works OK. I just got a new Sipura-2002 to my collection (I have few Sipura-3000 units that work OK). I setup the unit, Sipura-2002 to register with Asterisk and it registers OK. The unit will accept the call but I can not make a call out. My sip.conf entry: [SPA-2] ; incoming/outgoing calls on FXS Sipura-2002-Line1 ext.711 type=friend secret=711 username=711 mailbox=711 host=dynamic port=5068 ; port on FXS line dtmfmode=rfc2833 nat=no context=incoming callgroup=1 pickupgroup=1 Dial Plan on Sipura-2002: (xx.|*xx.|#xx.) (this dial plan works OK on Sipura-3000) I tried to compare the setup of 2002 unit to 3000 but I can not find anything that would be blocking outgoing calls. The firmware on Sipura-2002: Software Version:3.1.5 When I try to make a call out the asterisk is not registering anything on the command line from the unit. When I turn the SIP Debugging: SIP Debugging Enabled for IP: 10.0.0.155:5068 --- debug output --- Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-ceffa3af From: sip:[EMAIL PROTECTED];tag=e96b9f56902aab60o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5068 Expires: 240 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 420 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 1015871 1015871 IN IP4 10.0.0.155 s=- c=IN IP4 10.0.0.155 t=0 0 m=audio 16434 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv 14 headers, 19 lines Using latest request as basis request Sending to 10.0.0.155 : 5068 (non-NAT) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-ceffa3af From: sip:[EMAIL PROTECTED];tag=e96b9f56902aab60o0 To: sip:[EMAIL PROTECTED];tag=as3395f791 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=05664a87 Content-Length: 0 to 10.0.0.155:5068 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user 'SPA-2' syscon2*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-ceffa3af From: sip:[EMAIL PROTECTED];tag=e96b9f56902aab60o0 To: sip:[EMAIL PROTECTED];tag=as3395f791 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5068 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 0 10 headers, 0 lines syscon2*CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-93cd1d87 From: sip:[EMAIL PROTECTED];tag=e96b9f56902aab60o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username=SPA-2,realm=asterisk,nonce=05664a87,uri=sip:[EMAIL PROTECTED],algorithm=MD5,response=da6bd6dd8a890f2e37a88ff339ec0419 Contact: sip:[EMAIL PROTECTED]:5068 Expires: 240 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 420 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 1015871 1015871 IN IP4 10.0.0.155 s=- c=IN IP4 10.0.0.155 t=0 0 m=audio 16434 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv 15 headers, 19 lines Using latest request as basis request Sending to 10.0.0.155 : 5068 (non-NAT) Found user 'SPA-2' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 10.0.0.155:16434 Found description format PCMU Found description format G726-32 Found description format G723 Found description
[Asterisk-Users] BRI debug, national ISDN speech call problem
Title: BRI debug, national ISDN speech call problem hello, I have a Junghanns QuadBRI card in my asterisk server. I'm able to dial connect to local numbers through the ZAP interfaces however when I try to dial national numbers with the according area code the connection fails, an intense BRI debug is shown below. The error being Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) about half way through the debug. Thanks, steve T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (21) [ 00 e7 01 2b ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 115 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 021 P/F: 1 0 bytes of data -- Restarting T203 counter [ 02 e7 01 1f ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 115 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 015 P/F: 1 0 bytes of data -- ACKing all packets from 14 to (but not including) 15 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 counter [ 00 e7 01 1f ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 115 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 015 P/F: 1 0 bytes of data -- ACKing all packets from 14 to (but not including) 15 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Unsolicited RR with P/F bit, responding Sending Receiver Ready (21) [ 02 e7 01 2b ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 115 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 021 P/F: 1 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter -- Accepting AUTHENTICATED call from 213.107.182.203, requested format = 2, actual format = 2 -- Executing Dial(IAX2/[EMAIL PROTECTED]/3, ZAP/r1/01462647120) in new stack -- Requested transfer capability: 0x00 - SPEECH PBX*CLI [ 00 e7 1e 2a 08 01 09 05 04 03 80 90 a3 18 01 82 6c 02 00 c3 70 0c c1 30 31 34 36 32 36 34 37 31 32 30 a1 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 115 EA: 1 N(S): 015 0: 0 N(R): 021 P: 0 31 bytes of data -- Restarting T203 counter Stopping T_203 timer Starting T_200 timer Protocol Discriminator: Q.931 (8) len=31 Call Ref: len= 1 (reference 9/0x9) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 01 82] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Preferred Dchan: 0 ChanSel: B2 channel ] [6c 02 00 c3] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Number not available (67) '' ] [70 0c c1 30 31 34 36 32 36 34 37 31 32 30] Called Number (len=14) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '01462647120' ] [a1] Sending Complete (len= 1) -- Called r1/01462647120 PBX*CLI [ 00 e7 01 20 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 115 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 016 P/F: 0 0 bytes of data -- ACKing all packets from 14 to (but not including) 16 -- ACKing packet 15, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Restarting T203 counter [ 02 e7 2a 20 08 01 89 02 18 01 8a ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 115 EA: 1 N(S): 021 0: 0 N(R): 016 P: 0 7 bytes of data -- ACKing all packets from 15 to (but not including) 16 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=7 Call Ref: len= 1 (reference 137/0x89) (Terminator) Message type: CALL PROCEEDING (2) [18 01 8a] Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 ChanSel: B2 channel ] Sending Receiver Ready (22) [ 02 e7 01 2c ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 115 EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 022 P/F: 0 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter [ 02 e7 2c 20 08 01 89 45 08 02 82 81 1e 02 82 88 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 115 EA: 1 N(S): 022 0: 0 N(R): 016 P: 0 12 bytes of data -- ACKing all packets from 15 to (but not including) 16 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=12 Call Ref: len= 1 (reference 137/0x89) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1
Re: [Asterisk-Users] T400P vs TE405P
T400P(tormenta2) is based entirely off of the public zaptel spec, Digium doesn't make them anymore, You can still get almost an exact copy of the T400P from Varion a clone card maker. The TE405P has several design and firmware optimizations over the T400P and the TE405P switches lines faster. The T400P is T1-only whereas the TE405P can take T1s or E1s The TE405P is 5v only, the TE410P is identical to the TE405P but it is 3v only. Hope that helps, MATT---On 9/9/05, Darren Wright [EMAIL PROTECTED] wrote: Anyone care to elaborate on the differences between the T400P and the TE405P? -Darren ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] remote SIP phones
Hi, I am looking for some suggestion on getting remote SIP phone connected to an asterisk server where their is a NAT on the remote home users network/remote site and whether I need the asterisk box on a public IP. I know their is some problem with SIP and NAT (Although certainly not an expert) and some routers are coming out that have a built in SIP proxy. I would like to have 2 or more SIP phones on each remote site with using an asterisk server on each. Has anyone implemented these types of scenarios. These will be connected to DSL lines most with Static IP addresses but some do not. If we need to replace DSL routers then thats not a problem. Just looking for a heads up really. BTW: Would it be easier using H323 connections? Thanks -- Gary ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge Echo
On Friday 09 of September 2005 13:14, Andreas Sikkema wrote: [EMAIL PROTECTED] wrote: In the following setup: call coming from a pstn line - into FXO card - asterisk - SIP phone i get an incredible loud echo in the SIP phone (about 0,5-1s) (everything i speak into SIP phone microphone i hear in its speaker). The person calling from PSTN is not getting any echo. Make sure you're not playing the recorded sound from your microphone back to your loudspeakers. How could I have done that? I'm not recording any sound (at least nothing i'm aware of). The echo doesn't happen when the call is incoming from SIP provider (instead of PSTN) - so i assume the problem is related to the analog line. The SIP phone is stand-alone AT-320 Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM11B pinout
Thanks Steve most helpful, Yes I am in the UK and we have master and secondary sockets for implementing ring. Master sockets have the ring Capacitor and secondary sockets do not, where it is designed that the ring voltage is already supplied on the 3rd pin. I am attempting to get my first asterisk server up and running so this is a learning curve. Thanks Gary [EMAIL PROTECTED] wrote: On Mon, 5 Sep 2005, Gary Smith wrote: I have a development with a TDM11B in it. I am trying to connect this to a exchange line as well as a UK telephone and are looking for some pinout information for the FXS port. Is it the centre 2 pins that at the tip and ring. I have been digging around the Digium site but cannot seem to pick up this info. On the RJ45 socket its is pins 4 and 5 (aka the middle 2 pins). An rj11 also can be used and fits the socket fine by design. In that case its the middle 2 pins (2 and 3). If you are in the UK you will know about the UK three-wire system for premises wiring. Some UK phones may not ring when connected to a TDM board; if that happens you need to go to Maplin and buy the RJ11 plug to BT socket adapter that INCLUDES THE RINGING CAPACITOR. Steve -- Gary Smith - Director Phoenix Broadband Ltd 116 Henderson Street, Bridge of Allan, STIRLING, FK9 4HF, UK Tel : 0870 0553152 Fax : 0870 0553154 Mob: +44 (0)7971 504798 Sales / Accounts Tel 0870 2200573. Support Contact : [EMAIL PROTECTED] -- This e-mail transmission is intended exclusively for the individual(s) to whom it is addressed and may contain information that is privileged, or confidential. If you receive this e-mail in error, please advise the sender immediately and then delete the e-mail. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Huge Echo
Your config files?? Zaptel zapata ? Maybe you have a setting wrong i had the same problem but then with pri lines now it's gone. You can hear yourself as loud as the other person that is calling you? And what sipphone do you use -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Marek Zachara Verzonden: vrijdag 9 september 2005 13:27 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Huge Echo On Friday 09 of September 2005 13:14, Andreas Sikkema wrote: [EMAIL PROTECTED] wrote: In the following setup: call coming from a pstn line - into FXO card - asterisk - SIP phone i get an incredible loud echo in the SIP phone (about 0,5-1s) (everything i speak into SIP phone microphone i hear in its speaker). The person calling from PSTN is not getting any echo. Make sure you're not playing the recorded sound from your microphone back to your loudspeakers. How could I have done that? I'm not recording any sound (at least nothing i'm aware of). The echo doesn't happen when the call is incoming from SIP provider (instead of PSTN) - so i assume the problem is related to the analog line. The SIP phone is stand-alone AT-320 Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge Echo
Did you search the maillist archives for hybrid echo cancellation? Hello In the following setup: call coming from a pstn line - into FXO card - asterisk - SIP phone i get an incredible loud echo in the SIP phone (about 0,5-1s) (everything i speak into SIP phone microphone i hear in its speaker). The person calling from PSTN is not getting any echo. Which piece of the call could be causing the trouble so i can look into it? thanks, Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] pri gateway
Not all providers use crc4 you can try to remove the entry -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens altus Verzonden: vrijdag 9 september 2005 7:24 Aan: Baris Simsek CC: asterisk Onderwerp: Re: [Asterisk-Users] pri gateway These are my configs for a sangoma 4 port connected to E1's in the UK loadzone = us loadzone=uk defaultzone=uk span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 span=3,1,0,ccs,hdb3,crc4 span=4,1,0,ccs,hdb3,crc4 # card 0 - span 1 bchan=1-15,17-31 dchan=16 # card 0 - span 2 bchan=32-46,48-62 dchan=47 # card 0 - span 3 bchan=63-77,79-93 dchan=78 # card 0 - span 4 bchan=94-108,110-124 dchan=109 and zapata.conf [channels] context=default switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown priindication = outofband usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes callwaitingcallerid=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes callgroup=1 pickupgroup=1 ; card 0 - span 1 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 1-15,17-31 ; card 0 - span 2 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 32-46,48-62 ; card 0 - span 3 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 63-77,79-93 ; card 0 - span 4 switchtype = euroisdn signalling = pri_cpe group = 1 context = incoming channel = 94-108,110-124 Maybe its your telco?? On Thu, 2005-09-08 at 15:23 +0300, Baris Simsek wrote: hi, my asterisk version is 1.0.9 /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 it is comfortable with Turkish Telecom. i tried before and it works. /etc/asterisk/zapata.conf [channels] switchtype=euroisdn signalling=pri_cpe context=incoming group=1 channel=1-15,17-31 Leds are lighting at start. When i run /etc/init.d/zaptel they go out. And i can see the modules are installed. and i see that, layer 1 is going up after zaptel. So i am sure there is no problem with drivers. I think it is connected to asterisk. any idea? thanks... altus wrote: what about a copy of your zapata.conf and zaptel.conf,what color is the leds On Thu, 2005-09-08 at 12:42 +0300, Baris Simsek wrote: hello, i installed an asterisk as a pri gateway. Everything is okay. /etc/init.d/zaptel starts successfull with wct4xxp module. /etc/init.d/asterisk starts also successfully. I tested my pri cable and it works. But still my span isn't up. I don't see any error. Do you have any idea? What else i should check? Thanks. My card is 4 span Wildcard TE410P http://www.digium.com/index.php?menu=product_detailcategory=hardwa reproduct=TE410P # lsmod wct4xxp 106688 62 zaptel226820 129 wct4xxp # asterisk -r gw*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, In Alarm, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T400P vs TE405P
On Friday 09 September 2005 00:25, Darren Wright wrote: Anyone care to elaborate on the differences between the T400P and the TE405P? This is described on Digium's site. In a nutshell: newer, more efficient design, utilizes PCI burst mode and can reduce load on your server. They even have pretty pictures. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge Echo
On Friday 09 of September 2005 13:38, Sander wrote: Your config files?? Zaptel zapata ? Maybe you have a setting wrong i had here they are: zapata.conf: [channels] context=incoming signalling=fxs_ks usecallerid=yes cidsignalling=v23 cidstart=ring callerid=asreceived busydetect=yes busycount=6 echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 rxgain=9.0 txgain=4.0 channel = 1 i tried various rxgain/txgain settings, also commenting out echotraining, but havn't noticed any difference zaptel.conf: fxsks=1 loadzone=pl defaultzone=pl the same problem but then with pri lines now it's gone. You can hear yourself as loud as the other person that is calling you? Actually, i can hear myself much louder than the person calling... :) And what sipphone do you use as i wrote, its stand-alone AT-320 Marek -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Marek Zachara Verzonden: vrijdag 9 september 2005 13:27 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Huge Echo On Friday 09 of September 2005 13:14, Andreas Sikkema wrote: [EMAIL PROTECTED] wrote: In the following setup: call coming from a pstn line - into FXO card - asterisk - SIP phone i get an incredible loud echo in the SIP phone (about 0,5-1s) (everything i speak into SIP phone microphone i hear in its speaker). The person calling from PSTN is not getting any echo. Make sure you're not playing the recorded sound from your microphone back to your loudspeakers. How could I have done that? I'm not recording any sound (at least nothing i'm aware of). The echo doesn't happen when the call is incoming from SIP provider (instead of PSTN) - so i assume the problem is related to the analog line. The SIP phone is stand-alone AT-320 Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
On Thu, Sep 08, 2005 at 07:28:34PM +, Mike Hemstock wrote: On Tuesday 06 September 2005 15:27, Mike M wrote: Imagine what a network of systems composed of Asterisk, ham radio, wifi, generators, batteries, and a reserve of fuel could have done for the Gulf coast. I have all of the components above except the ham radio. That's a very interesting idea. I've initiated a request to join my local amateur radio yahoo group. I'm going to see if I can enlist help to demonstrate this idea. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge Echo
On Friday 09 of September 2005 13:39, Soner Tari wrote: Did you search the maillist archives for hybrid echo cancellation? well, yes i googled a lot beforehand, came across the hybrid issue, but from what i unerstand, the hybrid is a piece of hardware that sits on the X100P card. I'm not sure what can be done about it - the card doesn't seem to have any serviceable parts and i found no 'programmatic' way to change the hybrid parameters (but maybe there is some)? Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp txfax multi page problem
Dear All, I'm having problem with spandsp/txfax, I'm not able to send a multi paged tiff file, the fax machine receives the first page of the document and complains about communication problem. The file what I'm trying to send has 2 pages and is received generated by spandsp/rxfax. Searched in the archives, but found no solution. I'm using: Dell Optiplex PC Inoteska Quad E1 card Fedora Core 3 asterisk 1.0.9 (tried 1.0.7) spandsp 0.0.2pre20 (tried 0.0.1k, 0.0.2pre19) libtiff 3.6.1 from distro (tried 3.5.7, 3.6.0 3.7.1 from source) heard the same problem on a Debian/Sangoma machine thanks in advance, Bertalan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Doesn't finishes callerid spill
Hi,
I am a beginner in asterisk. Implementing it in my dept in India
using TDM400b card with asterisk, zaptel, libpri version latest of CVS
HEAD
Callerid on my system is coming tough.
Asterisk doesnot finishes the callerid spill and Cancells it.
After going through code in Callerid.c and chan_zap.c I found that my
line is providing caller id of length 8867.
Flow enters in zt_call and generates callerid id of length 8867 from
callerid generate in callerid.c
*snip** zt_call** chan_zap.c**
if (p-cidspill)
p-cidlen = ast_callerid_generate(p-cidspill, ast-cid.cid_name,
ast-cid.cid_num, AST_LAW(p));
p-cidpos = 0;
send_callerid(p);
//Flow enters in send callerid in a while loop which checks
cidposcidlen; Initial cidpos=0 and cidlen =8867
***snip** send_callerid*chan_zap.c
//
while(p-cidpos p-cidlen) {
if(!p-cidpos)
{
res = write(p-subs[SUB_REAL].zfd, p-cidspill + p-cidpos,
p-cidlen - p-cidpos);
//res here comes out to be 160
}
if (res 0) {
if (errno == EAGAIN)
return 0;
else {
ast_log(LOG_WARNING, write failed: %s\n, strerror(errno));
return -1;
}
}
if (!res)
return 0;
// res increments pos by 160
p-cidpos += res;
}
*
The strange thing happens here when loop is executed 35-37 times
cidpos is inreased to near about 5700 8867 and suddenly control gets
in zt_handle_event function in a switch case statement and cancells
the callerid spill and continues.
***snip***zt_handle_event***chan_zap.c*
case ZT_EVENT_RINGEROFF:
if (p-inalarm) break;
if (p-radio) break;
ast-rings++;
if ((ast-rings p-cidrings) (p-cidspill)) {
ast_log(LOG_WARNING, Didn't finish Caller-ID spill. Cancelling.\n);
free(p-cidspill);
p-cidspill = NULL;
p-callwaitcas = 0;
}
p-subs[index].f.frametype = AST_FRAME_CONTROL;
p-subs[index].f.subclass = AST_CONTROL_RINGING;
break;
***
I am seaching Why loop exits before reaching limit of 8867 or what
makes zt_handle_event to control the flow.
Please help me with any idea you have. Also tell if I am on wrong path
for right problem
PS: I have tried best to explain it but if ny doubt prevails pls tell me.
Regards
Gurminder
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Re: [Asterisk-Users] Huge Echo
Did you search the maillist archives for hybrid echo cancellation? well, yes i googled a lot beforehand, came across the hybrid issue, but from what i unerstand, the hybrid is a piece of hardware that sits on the X100P card. I'm not sure what can be done about it - the card doesn't seem to have any serviceable parts and i found no 'programmatic' way to change the hybrid parameters (but maybe there is some)? Marek I'd recommend the following link for the start: http://www.voip-info.org/tiki-index.php?page=Causes+of+Echo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge Echo
I'd recommend the following link for the start: http://www.voip-info.org/tiki-index.php?page=Causes+of+Echo I have read the echo related info on voip-info. But this didn't help me much. thats why i send my initial post to this list. I know the problem is related to the FXO card, but none of the hints there helped. I'm wondering however why the echo cancellation doesn't work as expected in asterisk. Any way to debug it? Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Want to use a remotely location POTS phone
spa3k is really an spa3000 (k = 000). Try: www.sipura.com www.voxilla.com www.voipsupply.com or any number of other suppliers of voip equipment. Oh Ok I guess I was taking it too literally!!! With a pair of SPA3000's, would I not even need *? Depends on what you are trying to do. If you only want remote dialtone (eg, toll bypass), then no. The spa3k's can be configured to play together to accomplish that. Example config's are on www.voxilla.com site. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
On 08/09/05, Mark Phillips [EMAIL PROTECTED] wrote: Bloddy 2E's; always wrong. Mark G7LTT/KC2ENI I know some G7s who are occasionally wrong, too :-) Peter G4MJS / 9M6BAA -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
On 09/09/05, Mike M [EMAIL PROTECTED] wrote: On Thu, Sep 08, 2005 at 07:28:34PM +, Mike Hemstock wrote: On Tuesday 06 September 2005 15:27, Mike M wrote: Imagine what a network of systems composed of Asterisk, ham radio, wifi, generators, batteries, and a reserve of fuel could have done for the Gulf coast. I have all of the components above except the ham radio. That's a very interesting idea. I've initiated a request to join my local amateur radio yahoo group. I'm going to see if I can enlist help to demonstrate this idea. The concept of combining VoIP and ham radio is by no means new - there are many skype-a-like systems around which are used as links or user access to the existing ham repeater network. I don't know of any using Asterisk, though. Peter G4MJS -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: spandsp txfax multi page problem
Hi ! I have a same problem here, tried even more different versions of *, libtiff, spandsp, and lot of different hardware (X100P,TE110P,mISDN ISDN BRI, and TDM400) and a lot of different faxes (but mostly Panasonic ones) on a various landlines (over couple of POTS lines, over ISDN BRI where receiving FAX is connected to the NT1 device with POTS ports, over Panasonic PBX that is connected to Telco over ISDN PRI, and the fax is PBXs extension). Result is more or less same: First page OK and without visible problem on paper, but after that, receiving fax machine just hangs up, and spandsp reports a bad quality signal. I have actualy managed to send 2 or 3 different multipaged faxes (tiffs generated by some Windows TIFF printer driver that has FAX resolutions and image coding support), but ONLY to a one model of Panasonic FAX machine (which is obviously not good enough :( ). On the other hand, all those fax machines I have tried are receiving multupage faxes from other fax machines without any problems. Judging from what I have read from Steve (author of spandsp) posts here and replies to my emails, the problem is somewhere in proces of retraining between pages (i.e. receiving fax machine say that signal is not good enough, but it will continue to receive after retraining, and spandsp doesn't retrain for some reason in that case). Anyway, that is as far as I have got with app_txfax. On the other hand, app_RXfax is doing great job on X100P and on TE110P in my case. Regards, Nenad Radosavljevic Dear All, I'm having problem with spandsp/txfax, I'm not able to send a multi paged tiff file, the fax machine receives the first page of the document and complains about communication problem. The file what I'm trying to send has 2 pages and is received generated by spandsp/rxfax. Searched in the archives, but found no solution. I'm using: Dell Optiplex PC Inoteska Quad E1 card Fedora Core 3 asterisk 1.0.9 (tried 1.0.7) spandsp 0.0.2pre20 (tried 0.0.1k, 0.0.2pre19) libtiff 3.6.1 from distro (tried 3.5.7, 3.6.0 3.7.1 from source) heard the same problem on a Debian/Sangoma machine thanks in advance, Bertalan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
The operative word here being occasionally. Of course, bad spelling doesn't count. flameprooftrousers And as for those half baked M3's ... /flameprooftrousers Peter Bowyer wrote: On 08/09/05, Mark Phillips [EMAIL PROTECTED] wrote: Bloddy 2E's; always wrong. Mark G7LTT/KC2ENI I know some G7s who are occasionally wrong, too :-) Peter G4MJS / 9M6BAA -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX PRI for single server
[EMAIL PROTECTED] wrote: Ah - so the difference between your setup and mine is that you are using Sangoma (presumably) and I'm using Digium. Looks like the Digium is significantly more efficient then. It could also be that I'm using Net-SNMP to query my cpu usage and even when the machine is idle, SNMP reports about 20% CPU usage which is incorrect. -Matthew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New CUT()
I store my speed dial numbers in the astdb key speeddial with the number and
then name separated by a -.
This dial plan works fine:
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,Cut(number=temp,,1)
exten = _*0XX,3,Goto(house-phones,${number},1)
The log informs me that cut is replaced with CUT.
I rewrote the dial plan using CUT (as best I can figure out) The plan below
returns the entire string number-name and fails?
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,CUT(temp,,1)
exten = _*0XX,3,Goto(house-phones,${temp},1)
What am I missing.
Thanks
--john
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Re: [Asterisk-Users] Huge Echo
echotraining=yes echotraining=800 This looks odd to me, I would use just: echotraining=800 Gain setting are important of course. You could use ztmonitor for that. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sending fax
I'm not writing a printer driver so I probably couldn't use the idea. I've always disabled CUPS. Regards, Chris - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, September 09, 2005 1:04 AM Subject: Re: [Asterisk-Users] sending fax On Thu, Sep 08, 2005 at 06:42:49PM -0500, Chris Shipman wrote: I started working on a program using Ghostscript and Redmon to generate the tif in windows by a printer. So far I am using FTP to transfer the tiff and call file. At least until I figure something better out. Why don't you look at IPP (Internet Printing Protocol)? a protocol for submitting jobs over HTTP of some sort. Server is already implemented in e.g. cups. HTTP allows a nice header with some extra fields. I wonder if that can be abused to get the call information through. (and am I re-inventing some wheels in the process?) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Montreal usergroup
Hi Montrealers ! I would like to create a usergroup for Montreal's asterisk users. If you are interested, contact me and we'll schedule a beer/coffee meeting downtown next week. Sincerely, Adrien -- Adrien Laurent - CIO 514-284-2020 ext 202 [EMAIL PROTECTED] www.modulis.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
Hold on here folks, I'm guessing that the original poster of this thread isn't a member of his local RAyNet team. Whilst I don't profess to be an expert at this I have been doing emergency radio for quite some time and have seen service at the Lockerbie bombing, Docklands bomb, Ground Zero (I'm sure I'm a terrorist target y'know - they seem to follow me everywhere) and soon I'll be in Louisiana. In all of these events the KISS principle must and does prevail. We need a system that is a simple and energy efficient as possible. Building a network of * servers and Wi-Fi links is all very well but how are you going to power them? Generators require fuel which is always in short supply and batteries die out quickly. Adding Ham Radio to the picture doesn't really add much when you are trying to do something like a * network. The radio gear just isn't designed to integrate with the * server. Ham radio is being used down in the Katrina affected area with great results for both emergency and heath/welfare related traffic. They are using both phone (that's when one talks in to the radio) and data modes and can be heard all over the 75 and 40 meter bands here in the US. Power for most of these stations comes from batteries they loot (with Police approval) from abandoned cars or a combo of solar and batteries. Many stations are only hear on the air after dark so that they can put as much sunlight into their batteries as possible. Yes, electricity is available in some places either all day or across the peak hours (allowing the workmen to restore power to other areas). Yes, there are radio to phone interconnects but these really are a single phone to a single radio. Think of it as a cordless phone in that the radio user can be anywhere within reach of the base station. Such technologies, whilst legal here in the US, may not be legal elsewhere. When last at home (UK) I was not able to connect my radio to the phone system by law (this may have changed recently - not been home for 8 years). Many countries have such restrictions and as we saw during the Tsunami, rules don't get relaxed just because there's a panic on. Without question a phone system would be much better than a radio station. As such I'll be taking a portable * server I've built, all the IP hard phones I can find and 5 DirectTV style Internet systems. My (approved by the Red Cross) plan is to install the * server and 2 phones in the HQ at Montgomery, AL. And then the other 4 systems in shelters where they have electricity thus relieving the Radio Hams for duty at other places. As hams are in short supply (they need over 700 every day) The best I could think of was to replace hams with phones rather than augment hams with phones. I guess after all this waffle I'm trying to say that ham radio is not a replacement for the telephone and cannot handle the kinds of load that is required by a phone system. Mark Mike M wrote: On Thu, Sep 08, 2005 at 07:28:34PM +, Mike Hemstock wrote: On Tuesday 06 September 2005 15:27, Mike M wrote: Imagine what a network of systems composed of Asterisk, ham radio, wifi, generators, batteries, and a reserve of fuel could have done for the Gulf coast. I have all of the components above except the ham radio. That's a very interesting idea. I've initiated a request to join my local amateur radio yahoo group. I'm going to see if I can enlist help to demonstrate this idea. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX PRI for single server
On Friday 09 September 2005 09:05, Matthew Boehm wrote: It could also be that I'm using Net-SNMP to query my cpu usage and even when the machine is idle, SNMP reports about 20% CPU usage which is incorrect. I'm sorry but if your Dell Xeon 3.0GHz is topping out at 50% CPU for 40 ulaw calls you've perhaps got other issues with the system, I think. It's hard to quantify with the information you've given, though. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] siemens pbx what i ask techinician?
im really newbie, and i have a siemens digital pbx work in my work. i have 4 outside lines and the pbx has a E1/PRI card. what i need to ask my siemens provider(techinicians) to do in the pbx? i only have in my pbx the 9 to get a line to go outside is very simple. but i dont know what i need to ask them to programming. please help me. -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Montreal usergroup
Hi, You might want to join MLUG which has a lot of VOIP users/experts. http://www.mlug.ca Andre Courchesne - Consultant http://www.net-forces.com Home of the RockHopper Firewall/Server Adrien Laurent wrote: Hi Montrealers ! I would like to create a usergroup for Montreal's asterisk users. If you are interested, contact me and we'll schedule a beer/coffee meeting downtown next week. Sincerely, Adrien -- Adrien Laurent - CIO 514-284-2020 ext 202 [EMAIL PROTECTED] www.modulis.ca ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Txfax
Chris Shipman schrieb: What build of SpanDSP did you use? spandsp-0.0.2pre18 I'm working on a windows program so users can print to a local printer which will be forwarded to the asterisk server to be faxed. So far the program FTPs a Tiff to the Asterisk server to be faxed with a Sample.Call file.(For lack of a better method thus far) I don't understand, what you are telling or asking us with this information. Has it something to do with your question? If not, please avoid confusing with additional infos which are not relevant! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge Echo
On Friday 09 of September 2005 15:23, Soner Tari wrote: echotraining=yes echotraining=800 This looks odd to me, I would use just: echotraining=800 I have commented the first echotraining. Not that it changed anything ;) I have also just compiled 1.2.0-beta1 asterisk. As far as my perception can be accurate, the echo delay seem to be much shorter now (like 0,1s) - but still its incredibly loud :( Gain setting are important of course. You could use ztmonitor for that. the asterisk server is a racked machine with no sound card. so can't use the ztmonitor. If everything fails i'll dig it out and try this Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: New CUT()
In article [EMAIL PROTECTED],
John Hill [EMAIL PROTECTED] wrote:
I store my speed dial numbers in the astdb key speeddial with the number and
then name separated by a -.
This dial plan works fine:
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,Cut(number=temp,,1)
exten = _*0XX,3,Goto(house-phones,${number},1)
The log informs me that cut is replaced with CUT.
I rewrote the dial plan using CUT (as best I can figure out) The plan below
returns the entire string number-name and fails?
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,CUT(temp,,1)
exten = _*0XX,3,Goto(house-phones,${temp},1)
What am I missing.
The new CUT is a function, and should be used within a Set command.
Something approximating (please check the detail):
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,Set(number=CUT(temp,,1))
exten = _*0XX,3,Goto(house-phones,${number},1)
Your second example is calling the same Cut command as the first.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] sending fax
It seems that using AstFax would mean that you would have to have a dedicated email server for faxing. AstFax expects the number in the email address.So all emails would have to be piped to the program. Which maybe fine in some circumstances. Am I wrong? regards, Chris___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge Echo
Gain setting are important of course. You could use ztmonitor for that. the asterisk server is a racked machine with no sound card. so can't use the ztmonitor. If everything fails i'll dig it out and try this You don't need a soundcard to use ztmonitor, what do you mean by that? Marek, you are making me suspicious about whether you've really read wiki in detail. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma
Echo can is just a few weeks away , and if I recall correctly, the Sangoma echo can will effectively monitor for, and handle, echo up to 128MS, whereas on a Quad Span Digium card I think you only get echo can up to 16MS. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory Nathan C. Smith wrote: Did they tell you anything about the timeframe for the echo can-on-a-card they have mentioned in the past? -Nate -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Thursday, September 08, 2005 9:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sangoma Had the pleasure of a visit from Doug Vilim of Sangoma today. Keep an eye on these guys, if you are a current Sangoma user, or have heard the name, they have some extremely innovative stuff coming down the pipe that will benefit the Asterisk community tremendously. Great company, great products, not knocking Digium but these guys will soon emerge as a major player in the industry. Cory Andrews Partner / Purchasing VOIPSupply.com ++ 454 Sonwil Drive Buffalo, NY 14225 ++ v - 800.398.VOIP Ext 22 f - 716.630.1548 e - [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VIP-050
Hi, I want to extend my asterisk stuff and buy some Planet devices, to be certain I'm going to buy PLANET VIP-050 with FXO and FXS modules. Has anyone heard about it. Is it compatible with Asterisk, or it would cause a lot of problems. Dose anyone have some experience with it?? All the best Andrutto -- Oferty sprzedazy samochodow... http://link.interia.pl/f18b1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Motherboard and processor recommendations
Or shitcan the onboard raid and get a real hardware raid controller like a 3ware card (if you are stuck on IDE / SATA.) Reduces complexity. On Fri, Sep 09, 2005 at 09:23:44AM +0300, Tzafrir Cohen said: On Fri, Sep 09, 2005 at 08:05:09AM +0200, Clive wrote: Hi I discovered that most onboard raid controllers are really software raid, and it uses the cpu to perform raid functions. Also: in such a settings you can get comperable performance by using Linux's built-in software raid. And for that you won't depend on non-standard drivers from the vendor for that. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
On Fri, Sep 09, 2005 at 01:46:57PM +0100, Peter Bowyer wrote: On 09/09/05, Mike M [EMAIL PROTECTED] wrote: On Thu, Sep 08, 2005 at 07:28:34PM +, Mike Hemstock wrote: On Tuesday 06 September 2005 15:27, Mike M wrote: Imagine what a network of systems composed of Asterisk, ham radio, wifi, generators, batteries, and a reserve of fuel could have done for the Gulf coast. I have all of the components above except the ham radio. That's a very interesting idea. I've initiated a request to join my local amateur radio yahoo group. I'm going to see if I can enlist help to demonstrate this idea. The concept of combining VoIP and ham radio is by no means new - there are many skype-a-like systems around which are used as links or user access to the existing ham repeater network. I don't know of any using Asterisk, though. I think this architecture has value: PSTN---asterisk---voip---radio===+==radio--voip--asterisk---POTS +==radio--voip--asterisk---POTS +==radio--voip--asterisk---POTS and this too: voip svc prvdrvoip---radio===+==radio--voip--asterisk---POTS +==radio--voip--asterisk---POTS +==radio--voip--asterisk---POTS POTS at the emergency end is good because it's familiar, simple, cheap, and runs on a central power source. I don't know radio equipment so I don't know if the upstream radio can multiplex streams onto different frequencies. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma
Cory, As I understand it the echo detection will only run in the first 100? Milliseconds. The statement that it will 'monitor for' echo is misleading. It will detect echo at the start of a call like all other current echo cancellation, correct? On another note, Sangoma tech support is good. I have used an a104 and they did a great job supporting it. Used it in a dell server with the only PCI slot sharing an interrupt with the onboard SATA controller! Crappy dell design - sc1425, the pci slot always shares with the sata regardless of bios settings). Echo can is just a few weeks away , and if I recall correctly, the Sangoma echo can will effectively monitor for, and handle, echo up to 128MS, whereas on a Quad Span Digium card I think you only get echo can up to 16MS. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory Nathan C. Smith wrote: Did they tell you anything about the timeframe for the echo can-on-a-card they have mentioned in the past? -Nate -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Thursday, September 08, 2005 9:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sangoma Had the pleasure of a visit from Doug Vilim of Sangoma today. Keep an eye on these guys, if you are a current Sangoma user, or have heard the name, they have some extremely innovative stuff coming down the pipe that will benefit the Asterisk community tremendously. Great company, great products, not knocking Digium but these guys will soon emerge as a major player in the industry. Cory Andrews Partner / Purchasing VOIPSupply.com ++ 454 Sonwil Drive Buffalo, NY 14225 ++ v - 800.398.VOIP Ext 22 f - 716.630.1548 e - [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Motherboard and processor recommendations
I agree, use either SCSI with hardware raid with a battery backed cache or use sata/ide with linux software raid. Linux raid SCSI also works well, but if you go for the scsi drives might as well get the controller too. The firmware raid on the cheap sata/ide cards have left me stranded several times, I have had experiences were both an HP and Promise IDE raid controller have SCRAMBLED both drives during a rebuild of a failed drive. What is the point of RAID if you have to restore tapes anyways? Or shitcan the onboard raid and get a real hardware raid controller like a 3ware card (if you are stuck on IDE / SATA.) Reduces complexity. On Fri, Sep 09, 2005 at 09:23:44AM +0300, Tzafrir Cohen said: On Fri, Sep 09, 2005 at 08:05:09AM +0200, Clive wrote: Hi I discovered that most onboard raid controllers are really software raid, and it uses the cpu to perform raid functions. Also: in such a settings you can get comperable performance by using Linux's built-in software raid. And for that you won't depend on non-standard drivers from the vendor for that. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge Echo
On Friday 09 of September 2005 16:08, Soner Tari wrote: Gain setting are important of course. You could use ztmonitor for that. the asterisk server is a racked machine with no sound card. so can't use the ztmonitor. If everything fails i'll dig it out and try this You don't need a soundcard to use ztmonitor, what do you mean by that? Marek, you are making me suspicious about whether you've really read wiki in detail. Well, i did read it. And as per soundcard - have you tried to run ztmonitor without it? When i tried i just got: arnor:~# /usr/src/zaptel-1.2.0-beta1/ztmonitor 1 Unable to open /dev/dsp: No such file or directory Cannot open audio ... so i guess it needs a soundcard after all... Anyway, i installed a soundcard and run the ztmonitor. I went with the rxgain/txgain down to -6.0 ... the echo is not that loud anymore, but still is quite annoying. i'm at loss... no other bright ideas ... Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Adtran TA 616
I know that the MGCP stack will work over the 10/100 port as I've been deploying them for the past few months and using them to terminate voice traffic for various customers using a Class5 VoIP capable softswitch. It almost seems like the adtran TA6xx doesn't reply back to the asterisk messages. If I try to audit the endpoint it shows: localhost*CLI mgcp audit endpoint aaln/[EMAIL PROTECTED] Posting Request: AUEP 85 aaln/[EMAIL PROTECTED] MGCP 1.0 F: A to 10.189.189.31:2427 Retransmitting #1 transaction 85 on [10.189.189.31] Retransmitting #2 transaction 85 on [10.189.189.31] Retransmitting #3 transaction 85 on [10.189.189.31] Retransmitting #4 transaction 85 on [10.189.189.31] Retransmitting #5 transaction 85 on [10.189.189.31] Sep 9 08:39:50 WARNING[8525]: chan_mgcp.c:610 retrans_pkt: Maximum retries exceeded for transaction 85 on [10.189.189.31] Sep 9 08:39:50 NOTICE[8525]: chan_mgcp.c:2274 handle_response: Transaction 85 timed out Nick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis Sent: Thursday, September 08, 2005 9:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Adtran TA 616 On Thu, 8 Sep 2005, Nick Colton wrote: Has anybody had any luck getting an Adtran Total Access 616 working via the Ethernet port/MGCP to an * box? The voice lines don't seem to be coming up and I wasn't sure if I had something missing. I had tried to get it working a few months ago but didn't get anywhere either. It wasn't clear if the mgcp stack could work over either interface or only over the wan side, so I set it up back to back with a cisco running to my asterisk server, but never got anything going. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Huge Echo
No that just means you are not calling ztmonitor properly. Try running ~# ztmonitor 1 -v Jared Armstrong OmniSpear, Inc. Web Network Solutions -Original Message- From: Marek Zachara [mailto:[EMAIL PROTECTED] Sent: Friday, September 09, 2005 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Huge Echo On Friday 09 of September 2005 16:08, Soner Tari wrote: Gain setting are important of course. You could use ztmonitor for that. the asterisk server is a racked machine with no sound card. so can't use the ztmonitor. If everything fails i'll dig it out and try this You don't need a soundcard to use ztmonitor, what do you mean by that? Marek, you are making me suspicious about whether you've really read wiki in detail. Well, i did read it. And as per soundcard - have you tried to run ztmonitor without it? When i tried i just got: arnor:~# /usr/src/zaptel-1.2.0-beta1/ztmonitor 1 Unable to open /dev/dsp: No such file or directory Cannot open audio ... so i guess it needs a soundcard after all... Anyway, i installed a soundcard and run the ztmonitor. I went with the rxgain/txgain down to -6.0 ... the echo is not that loud anymore, but still is quite annoying. i'm at loss... no other bright ideas ... Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sangoma
On Fri, 2005-09-09 at 08:28 -0600, Damon Estep wrote: On another note, Sangoma tech support is good. I can second that, 3/4 years ago I installed one of their cards and couldn't get it running. I phoned them and they talked me through for around 2 hours until it was really working. I had the tech guys and the boss helping me. It was only at the end when they realised it was 10 at night for me in France. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma
I'll have to have Doug refresh me on it, I don't want to mislead anyone. He was using the term Dynamic Echo Cancellation. Yes, their support is very good given the majority of the company are hardware/software engineers. They offer pretty much an unconditional guarantee that their hardware will work with any server platform manufactured in the last 2 years. If you have an IRQ or other issue getting their board installed, they will fix the problem within 24 hours. If they can't fix the problem, they will RMA the board and give you $1000.00 out of pocket for the inconvenience. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory Damon Estep wrote: Cory, As I understand it the echo detection will only run in the first 100? Milliseconds. The statement that it will 'monitor for' echo is misleading. It will detect echo at the start of a call like all other current echo cancellation, correct? On another note, Sangoma tech support is good. I have used an a104 and they did a great job supporting it. Used it in a dell server with the only PCI slot sharing an interrupt with the onboard SATA controller! Crappy dell design - sc1425, the pci slot always shares with the sata regardless of bios settings). Echo can is just a few weeks away , and if I recall correctly, the Sangoma echo can will effectively monitor for, and handle, echo up to 128MS, whereas on a Quad Span Digium card I think you only get echo can up to 16MS. Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory Nathan C. Smith wrote: Did they tell you anything about the timeframe for the echo can-on-a-card they have mentioned in the past? -Nate -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Thursday, September 08, 2005 9:22 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sangoma Had the pleasure of a visit from Doug Vilim of Sangoma today. Keep an eye on these guys, if you are a current Sangoma user, or have heard the name, they have some extremely innovative stuff coming down the pipe that will benefit the Asterisk community tremendously. Great company, great products, not knocking Digium but these guys will soon emerge as a major player in the industry. Cory Andrews Partner / Purchasing VOIPSupply.com ++ 454 Sonwil Drive Buffalo, NY 14225 ++ v - 800.398.VOIP Ext 22 f - 716.630.1548 e - [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registered SIP '202' ... expires 1800. Why does it expire
Hi, When a SIP client registers on Asterisk server, why it expires after certain amount of time? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Motherboard and processor recommendations
I also agree. You want a raid controller that has it's own CPU. You want hot spare, hot swapping, status lights, etc. to be handled by that controller. If you have a hot spare you want automatic cutover to that spare drive. You are not limited to SCSI with these controllers. Some manufactures offer ide and sata versions. If you want hot swap capability be sure to do your homework. Some drive hardware advertised as hot-swap capable might not work properly with the controller you select. Damon Estep wrote: I agree, use either SCSI with hardware raid with a battery backed cache or use sata/ide with linux software raid. Linux raid SCSI also works well, but if you go for the scsi drives might as well get the controller too. The firmware raid on the cheap sata/ide cards have left me stranded several times, I have had experiences were both an HP and Promise IDE raid controller have SCRAMBLED both drives during a rebuild of a failed drive. What is the point of RAID if you have to restore tapes anyways? Or shitcan the onboard raid and get a real hardware raid controller like a 3ware card (if you are stuck on IDE / SATA.) Reduces complexity. On Fri, Sep 09, 2005 at 09:23:44AM +0300, Tzafrir Cohen said: On Fri, Sep 09, 2005 at 08:05:09AM +0200, Clive wrote: Hi I discovered that most onboard raid controllers are really software raid, and it uses the cpu to perform raid functions. Also: in such a settings you can get comperable performance by using Linux's built-in software raid. And for that you won't depend on non-standard drivers from the vendor for that. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting retries in call files
Can anybody see a way of detecting the current number of retries remaining to a call file in the extension context that it is calling? E.g. If I want to schedule a fax and I want to feed an email back to the sender stating that the number is busy 2/5 retries remaining? Steve The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection
Ben,I assume from your posts that you are in an area serviced by a small independent phone company. We have this situation as well, and you might very well have to pay much higher rates than other areas. You might try contacting Long Distance carriers (we use Paetec) and find out if they can work a deal with the local provider and help you get it for cheaper. The savings on Long Distance can also make up the additional cost for the T1.TomOn Sep 8, 2005, at 11:49 PM, Ben Brown wrote: Heck...our CO still had analog switches until about 10 years ago! PRI is not an option going with the local telco. Luckily, there are a couple of non-local telco's that can apparently service me for a more reasonable rate. Damon Estep wrote: 500-600 is more typical these days depending on where you are, but insist on PRI unless the service central office can not provide it. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Ben Brown Sent: Thursday, September 08, 2005 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection Is this standard in the industry? My local telco in the US wants $1050/month for T1 (not PRI!) I can buy 24 POTS lines for $840/month. Gotta love small towns! BEN Sean Cook wrote: Not sure about where you are but 16 pots lines generally run about $25-$30 / month = $480/month. For about $400/month I can get a PRI (23+1) and go straight into a TE100. Just a thought. Sean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Erick Perez Sent: Thursday, September 08, 2005 11:04 PM To: Damon Estep Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection cant find a cisco one so far. So what i need is a voip gateway? i found the octtel sp1632 for about USD.1500 http://www.octtel.com.tw/eng/product_1632_2.php so i configure the equipment to register with the asterisk? as sip? or some other thing? it will only be used to make outgoing calls, the config is like this nortel (12 port pstn card)-(device with 16fxo)-asterisk-voip_provider no chance to use an e1/t1 on the nortel. On 9/8/05, Damon Estep [EMAIL PROTECTED] wrote: How about a used cisco iad2400 with a 16fxo module? Check ebay - no t1card required. Probably $1000 -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Erick Perez Sent: Thursday, September 08, 2005 8:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Solution for 12 to 16 FXO to asterisk connection Hi, today a customer asked how to use asterisk with 12 to 16 FXO ports. I can use a channel bank with 16 FXO ports and connect the channel bank with a T1 cable to a T1 card in the Asterisk Server. Asterisk will then send the calls to the Voip provider over the internet. However a 16 fxo port channel bank is about USD 1500 + a t1 card USD 500 + a USD 1000 computer = 3 thousand us dollars + my installation fees (life isn't free). Sounds expensive for such a small install. Suggestions? -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
On Fri, Sep 09, 2005 at 09:31:06AM -0400, Mark Phillips wrote: Generators require fuel which is always in short supply and batteries die out quickly. Fuel and batteries and power efficient systems need planning and management. Don't overlook solar panels as an energy source. They need to be in place all over the country and tested frequently. Adding Ham Radio to the picture doesn't really add much when you are trying to do something like a * network. The radio gear just isn't designed to integrate with the * server. It's software. It can be changed and added to. These things evolve from ideas in discussions like these. Such technologies, whilst legal here in the US, may not be legal elsewhere. What about authorized looting you mentioned? Sometimes you have to take a risk. Develop and demo where it's legal first. If it's not legal than we should ask why and work for change if we don't like the answer. Without question a phone system would be much better than a radio station. Well said. I guess after all this waffle I'm trying to say that ham radio is not a replacement for the telephone and cannot handle the kinds of load that is required by a phone system. What is the bandwidth potential? There are compression techniques from VoIP that might improve radio bandwidth utilization. New protocols can evolve to conserve bandwidth. Load control is a manageable problem. Radio telephony is not new. Telephony over ham might be new only because Asterisk puts telephonyi/voip into the same price range as ham radio gear. Maybe HAM is not the best technology. Maybe wi-fi is what we need. http://www.oreillynet.com/cs/weblog/view/wlg/448 Grassroots engineering can create an emergency civil communications system thereby creating some stored luck. Lucille Ball said, Luck? I don't know anything about luck. I've never banked on it, and I'm afraid of people who do. Luck to me is something else: Hard work -- and realizing what is opportunity and what isn't. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 for HEAD? 0.7.1 doesn't compile.
I have successfully been using OH323 v0.6.5 with Asterisk 1.0.x. I now need to move to CVS HEAD in order to use some features that are not in v1.0.x, and am trying to compile OH323 to use with it. On the InaccessNetworks site, it ways that OH323 v0.7.1 is for HEAD. However, when I compile it, it appears that it hasn't been updated since the channel structures were revamped. I get many errors, starting with the following: chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory chan_oh323.c: In function `oh323_exception': chan_oh323.c:1145: error: structure has no member named `pvt' Has anyone updated chan_oh323 to work with the latest HEAD? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] siemens pbx what i ask techinician?
It's not that easy then everytime you want to change someting for testing you have to ask them to change something i can give you the software for programming siemens pbx if you want -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 16:09 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] siemens pbx what i ask techinician? im really newbie, and i have a siemens digital pbx work in my work. i have 4 outside lines and the pbx has a E1/PRI card. what i need to ask my siemens provider(techinicians) to do in the pbx? i only have in my pbx the 9 to get a line to go outside is very simple. but i dont know what i need to ask them to programming. please help me. -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Storing extension prefs. in MySQL
Hello, In the examples on voip-info.org DBSet and DBGet are used to store configuration variables such as immediate call forwarding settings etc. I would like to store these seetings in a mysql database, so that they are more easily accessible from a user configuration page on a webserver. Since these settings need to be checked in the dialplan for each call to the extension, it seems a bit to much to have to connect, query and disconnect from mysql every time. Is there any way to keep a persistent connection to mysql that can be queried from the dialplan? Arnar ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge Echo
yes you are right. It does run this way. However this still does not solve the echo issue. I can see that RX is following TX with about 70% of the signal strength - but what to do about it? :( Marek On Friday 09 of September 2005 16:46, Jared Armstrong wrote: No that just means you are not calling ztmonitor properly. Try running ~# ztmonitor 1 -v Jared Armstrong OmniSpear, Inc. Web Network Solutions -Original Message- From: Marek Zachara [mailto:[EMAIL PROTECTED] Sent: Friday, September 09, 2005 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Huge Echo On Friday 09 of September 2005 16:08, Soner Tari wrote: Gain setting are important of course. You could use ztmonitor for that. the asterisk server is a racked machine with no sound card. so can't use the ztmonitor. If everything fails i'll dig it out and try this You don't need a soundcard to use ztmonitor, what do you mean by that? Marek, you are making me suspicious about whether you've really read wiki in detail. Well, i did read it. And as per soundcard - have you tried to run ztmonitor without it? When i tried i just got: arnor:~# /usr/src/zaptel-1.2.0-beta1/ztmonitor 1 Unable to open /dev/dsp: No such file or directory Cannot open audio ... so i guess it needs a soundcard after all... Anyway, i installed a soundcard and run the ztmonitor. I went with the rxgain/txgain down to -6.0 ... the echo is not that loud anymore, but still is quite annoying. i'm at loss... no other bright ideas ... Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Changing User-Agent: Asterisk PBX
Hello Folks! in my sip-logs i see that asterisk uses the User-Agent ID Asterisk PBX: SipClient: Received: 16:34:03.023 - BYE sip:[EMAIL PROTECTED]:44343;transport=udp SIP/2.0 Max-Forwards: 10 Record-Route: sip:213.2XX.XXX.XX8;ftag=as2eb3c466;lr=on Via: SIP/2.0/UDP 213.2XX.XXX.XX8;branch=z9hG4bK539a.47e6e8a7.0 #this is SER Via: SIP/2.0/UDP 213.2XX.XXX.XX7:5060;branch=z9hG4bK300d4e2b;rport=5060 # this is Asterisk From: 012341234sip:[EMAIL PROTECTED];tag=as2eb3c466 To: sip:[EMAIL PROTECTED];tag=7E24716A Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 Route: sip:[EMAIL PROTECTED]:44343;transport=udp this is an example from an incoming sip-call to a geographic number with a matching extension dialing the sip-account 1234. I'd like to customize this and thought activating the useragent=-setting in the sip.conf would change the display, but it doesn't seem to have any effect. is this the right place to change this distinct entry or do i have to edit it in the source? Asterisk-Version is 1.0.9 Thanks Christian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Storing extension prefs. in MySQL
[EMAIL PROTECTED] wrote: I would like to store these seetings in a mysql database, so that they are more easily accessible from a user configuration page on a webserver. Since these settings need to be checked in the dialplan for each call to the extension, it seems a bit to much to have to connect, query and disconnect from mysql every time. Is there any way to keep a persistent connection to mysql that can be queried from the dialplan? Well, if you do this before answering, nobody is going to notice. Even querying during an answered call will have hardly any outside consequences... -- Andreas Sikkema bbned NV Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voice over atlantic
Hi David, I just looked at my iax.conf on one of my boxes in Argentina and actually there are no jitterbuffer settings indicated so I'm assuming it is using Asterisk defaults. We are experimenting with G.729 on these IAX trunks also and I just realized I have no accurate means of measuring bandwidth consumption vis-a-vis GSM/G.729. I think I'll pose that question to the group in another message to see what recommendations and best practices are out there. Or, do some research. Best of luck. On Thu, 2005-09-08 at 17:49 -0400, David Hajek wrote: Nice. Thanks. What Asterisk version? Can you lookup jitterbuffer settings? Thanks a lot. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] reload
Hi, have looked around for some documentation what effect a reload has on a running system but I can't find any relevant information. What I would like to know is what type of configuration changes (if any) that will interfere with already established calls if I do a reload. I'm only using SIP. Any information or links are appreciated. /urban ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voice over atlantic
Forgot the version: Asterisk 1.0.7 On Thu, 2005-09-08 at 17:49 -0400, David Hajek wrote: Nice. Thanks. What Asterisk version? Can you lookup jitterbuffer settings? Thanks a lot. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Motherboard and processor recommendations
On Fri, Sep 09, 2005 at 10:55:52AM -0400, Paul wrote: I also agree. You want a raid controller that has it's own CPU. You want hot spare, hot swapping, status lights, etc. to be handled by that controller. If you have a hot spare you want automatic cutover to that spare drive. You are not limited to SCSI with these controllers. Some manufactures offer ide and sata versions. If you want hot swap capability be sure to do your homework. Some drive hardware advertised as hot-swap capable might not work properly with the controller you select. SATA is fast enough. In fact, ATAPI is also fast enough in most scenarios. It is just that SCSI disks/arrays tend to be of better quality (but usually much more expensive). IIRC Linux's raid support will support hot-swapping disks, but I'm not sure which disks are are supported. An external array with its own CPU doesn't necessarily mean better performance than one using the host CPU, BTW. Though it will take some load off of Asterisk. And if this is just about redundnacy and not about performance, consider not buying an expensive array at all, and using two cheap systems. The cost will be roughly the same, I believe. (RAID= Redundant Array of Inexpensive Disks). Any simple way to achive redundancy here? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siupra-2002 with astersik
The original dial plan was: (*xx|[3469]11|0|00|[2-9]xx|1xxx[2-9]xxS0|.) So I change it to: (xx.|*xx.|#xx.) I don't think it is complicated, beside it works with Sipura-3000, and I don't see a reason why shouldn't it work with Sipura-2002. I contact Sipura technical support but they didn't solve my problem yet. -- #Joseph On Fri, 2005-09-09 at 07:16 -0400, Matt wrote: Ahh wow.. that dial plan is seriously messed up... Try the default one... it will work alot better and give you less lag time between dialing a number and actually going through. On 9/8/05, Joseph [EMAIL PROTECTED] wrote: On Thu, 2005-09-08 at 23:29 +0200, Sander wrote: What is your problem with asterisk ans sipura ? Config files ?? Settings Give some more info on the problems Sipura-2002 CAN NOT dial out, incoming call works OK. I just got a new Sipura-2002 to my collection (I have few Sipura-3000 units that work OK). I setup the unit, Sipura-2002 to register with Asterisk and it registers OK. The unit will accept the call but I can not make a call out. My sip.conf entry: [SPA-2] ; incoming/outgoing calls on FXS Sipura-2002-Line1 ext.711 type=friend secret=711 username=711 mailbox=711 host=dynamic port=5068 ; port on FXS line dtmfmode=rfc2833 nat=no context=incoming callgroup=1 pickupgroup=1 Dial Plan on Sipura-2002: (xx.|*xx.|#xx.) (this dial plan works OK on Sipura-3000) I tried to compare the setup of 2002 unit to 3000 but I can not find anything that would be blocking outgoing calls. The firmware on Sipura-2002: Software Version:3.1.5 When I try to make a call out the asterisk is not registering anything on the command line from the unit. When I turn the SIP Debugging: SIP Debugging Enabled for IP: 10.0.0.155:5068 --- debug output --- Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-ceffa3af From: sip:[EMAIL PROTECTED];tag=e96b9f56902aab60o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5068 Expires: 240 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 420 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 1015871 1015871 IN IP4 10.0.0.155 s=- c=IN IP4 10.0.0.155 t=0 0 m=audio 16434 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv 14 headers, 19 lines Using latest request as basis request Sending to 10.0.0.155 : 5068 (non-NAT) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-ceffa3af From: sip:[EMAIL PROTECTED];tag=e96b9f56902aab60o0 To: sip:[EMAIL PROTECTED];tag=as3395f791 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=05664a87 Content-Length: 0 to 10.0.0.155:5068 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user 'SPA-2' syscon2*CLI Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-ceffa3af From: sip:[EMAIL PROTECTED];tag=e96b9f56902aab60o0 To: sip:[EMAIL PROTECTED];tag=as3395f791 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5068 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 0 10 headers, 0 lines syscon2*CLI Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.155:5068;branch=z9hG4bK-93cd1d87 From: sip:[EMAIL PROTECTED];tag=e96b9f56902aab60o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username=SPA-2,realm=asterisk,nonce=05664a87,uri=sip:[EMAIL PROTECTED],algorithm=MD5,response=da6bd6dd8a890f2e37a88ff339ec0419 Contact: sip:[EMAIL PROTECTED]:5068 Expires: 240 User-Agent: Sipura/SPA2002-3.1.5 Content-Length: 420 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 1015871 1015871 IN IP4 10.0.0.155 s=- c=IN IP4 10.0.0.155 t=0 0 m=audio 16434 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv
Re: [Asterisk-Users] Registered SIP '202' ... expires 1800. Why does it expire
Zeeshan Zakaria wrote: Hi, When a SIP client registers on Asterisk server, why it expires after certain amount of time? Because it is the way SIP registrations work. For more information, find a SIP book or read the SIP RFC 3261. SIP phones need to re-register every once in a while to tell the server where it can be reached. If you have a soft phone on a laptop that you move from network to network - home, office, airport, Barnes Noble etc - you want to be reached on the IP address you use there. SIP registration keeps the server up-to-date with your hectic life :-) /Olle --- Astricon - where you meet Asterisk friends and re-register with them by exchanging updated business cards! http://www.astricon.net/2005 - register today! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] woomera doesn't work (same OpenH323 problem as with chan_h323)
Banging my head against a brick wall trying to get a working H.323 implementation for CVS-HEAD. (The ONLY H.323 I have had working is OH323 v0.6.5 with CVS-STABLE - see my other post regarding compile problems on OH323 for HEAD) So, I thought, lets try this wonderful chan_woomera (dubbed H.323 for Asterisk that works!). I get exactly the same kind of problem as I have previously had with the in-tree chan_h323. OpenH323 can't open an audio codec, and I have a feeling it is because it is trying to mess around with devices. See my previous post about this with chan_h323 at http://lists.digium.com/pipermail/asterisk-users/2005-July/115331.html Now trying woomera it is doing the same kind of thing. Here is what is in my /var/log/woomera.console: Waiting for incoming connections on port 42420 Incoming connection from 127.0.0.1 0:02.840WoomeraThread:8432b58 woomera.cxx(763) Woomera Sending command EVENT HELLO 0 1.0 Post Increment 0:02.841WoomeraThread:8432b58 woomera.cxx(736) Woomera Sending response 200 Listener enabled 0:23.893 H225 Answer:b6f00ae0 woomera.cxx(763) Woomera Sending command EVENT INCOMING 1 Incoming connection from 127.0.0.1 0:23.912 WoomeraThread:b6f16740 woomera.cxx(763) Woomera Sending command EVENT HELLO 0 1.0 Post Increment 0:23.913 WoomeraThread:b6f16740 woomera.cxx(736) Woomera Sending response 200 Listener enabled 0:23.913 WoomeraThread:b6f16740 woomera.cxx(763) Woomera Sending command EVENT INCOMING 1 0:23.916 WoomeraThread:b6f16740 woomera.cxx(736) Woomera Sending response 200 Call answered 0:23.966 H225 Answer:b6f00ae0 woomera.cxx(763) Woomera Sending command EVENT MEDIA 1 AUDIO 0:23.968 LogChanRx:b6f09d98 codecs.cxx(482) Codec Write failed: Bad file descriptor 0:23.968 H225 Answer:b6f00ae0 channels.cxx(1147) LogChan Transmit thread aborted (open fail) for G.711-ALaw-64k 1 0:34.833 H225 Answer:b6f00ae0 h323pdu.cxx(1285) H225Read error (4): Interrupted system call 0:34.833 H323 Cleaner h323.cxx(1750) H323 Connection ip$194.54.172.1:33950/17506 terminated. 0:34.833 H323 Cleaner woomera.cxx(3239) Woomera OnCleared received 0:34.833 H323 Cleaner woomera.cxx(2180) Shutting down non-transferred call 0:34.834 H323 Cleaner woomera.cxx(763) Woomera Sending command EVENT HANGUP 1 And although the call control protocol seems to work ok, no audio is passed, no doubt due to the lack of a Transmit thread. Any suggestions would be greatly appreciated. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: siemens pbx what i ask techinician?
On Fri, Sep 09, 2005 at 05:39:22PM +0200, Sander wrote: thanks Sander but i have the soft, and i can enter to the pbx conf and modify all settings, but i dont know how settings i need to change. It's not that easy then everytime you want to change someting for testing you have to ask them to change something i can give you the software for programming siemens pbx if you want -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 16:09 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] siemens pbx what i ask techinician? im really newbie, and i have a siemens digital pbx work in my work. i have 4 outside lines and the pbx has a E1/PRI card. what i need to ask my siemens provider(techinicians) to do in the pbx? i only have in my pbx the 9 to get a line to go outside is very simple. but i dont know what i need to ask them to programming. please help me. -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Huge Echo
Did you try to get a milliWatt test phone number from your telco? It was really easy for me. I called the business office and told them that my new digital pbx was having some awful echo trying to deal with their lines; all I needed was a milliWatt test line to balance my receive and transmit gains properly. I then had to tell her what a milliWatt test line did and why I thought it would help me; she didn't know what it was, but she was more than happy to have repair call me to see if they could help me. Less than a half an hour later, somebody called up, asked for me, and said lets see.. milliWatt test line for Sitka... 747-1100 as easy as that. I spent a half an hour making calls to this number out of my (phew, only three!) zap lines and haven't had echo troubles since. BTW, the area code on that one is 907 if you want to listen to what it should sound like. If the telcos were to route amongst themselves fully digitally, wouldn't you be able to use my mW test line no matter where you are? As long as the only analog link between it and you was your local copper pair? Just in case, I wouldn't recommend anybody tuning themselves to this mW source. Mojo Jared Armstrong wrote: No that just means you are not calling ztmonitor properly. Try running ~# ztmonitor 1 -v Jared Armstrong OmniSpear, Inc. Web Network Solutions -Original Message- From: Marek Zachara [mailto:[EMAIL PROTECTED] Sent: Friday, September 09, 2005 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Huge Echo On Friday 09 of September 2005 16:08, Soner Tari wrote: Gain setting are important of course. You could use ztmonitor for that. the asterisk server is a racked machine with no sound card. so can't use the ztmonitor. If everything fails i'll dig it out and try this You don't need a soundcard to use ztmonitor, what do you mean by that? Marek, you are making me suspicious about whether you've really read wiki in detail. Well, i did read it. And as per soundcard - have you tried to run ztmonitor without it? When i tried i just got: arnor:~# /usr/src/zaptel-1.2.0-beta1/ztmonitor 1 Unable to open /dev/dsp: No such file or directory Cannot open audio ... so i guess it needs a soundcard after all... Anyway, i installed a soundcard and run the ztmonitor. I went with the rxgain/txgain down to -6.0 ... the echo is not that loud anymore, but still is quite annoying. i'm at loss... no other bright ideas ... Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registered SIP '202' ... expires 1800. Why does it expire
Olle E. Johansson wrote: SIP phones need to re-register every once in a while to tell the server where it can be reached. If you have a soft phone on a laptop that you move from network to network - home, office, airport, Barnes Noble etc - you want to be reached on the IP address you use there. SIP registration keeps the server up-to-date with your hectic life :-) Which is kinda annoying because grandstreams (at least 1.0.5.23 firmware anyway) don't do that... I have to powercycle the one on my desk once an hour if I want it to ring on incoming calls otherwise asterisk 'forgets' about it (the cisco in the other room seems to be fine). I'd love an option to work around that... Tony ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk connected to Concept XI520
Hi Asterisk 1.0.9-BRIstuffed-0.2.0-RC8m is connected to a T-Concept XI520 System. Phone calls on both directions do work, but transfers are not possible. Asterisk recognizes that some sip phone requests a transfer. Is it possible to forward this transfer request to the XI520? Users of analog and ISDN phones have to use the R key. Any hints? -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: siemens pbx what i ask techinician?
Do you want to connect the asterisk with pri or with internal isdn? And what model pbx do you have? then i can tell you how to configure? Maybe some screenshots with it -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 19:35 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: siemens pbx what i ask techinician? On Fri, Sep 09, 2005 at 05:39:22PM +0200, Sander wrote: thanks Sander but i have the soft, and i can enter to the pbx conf and modify all settings, but i dont know how settings i need to change. It's not that easy then everytime you want to change someting for testing you have to ask them to change something i can give you the software for programming siemens pbx if you want -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 16:09 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] siemens pbx what i ask techinician? im really newbie, and i have a siemens digital pbx work in my work. i have 4 outside lines and the pbx has a E1/PRI card. what i need to ask my siemens provider(techinicians) to do in the pbx? i only have in my pbx the 9 to get a line to go outside is very simple. but i dont know what i need to ask them to programming. please help me. -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Huge Echo
On Friday 09 of September 2005 19:04, Mojo with Horan Company, LLC wrote: Did you try to get a milliWatt test phone number from your telco? It was really easy for me. I called the business office and told them that Well, unfortunately not everyone has similiarly helpful telco providers ;) To just give you a hint, ten years ago here when you applied for a phone line you would have to wait about 3-7 YEARS to actually have it installed. Its improving now, but even now if the line is dead it takes them up to a few days to repair it ;) try not to laugh :) I'm pretty sure if i'd call them to ask for a balancing line they would probably assumed it an abuse ... Seriously however, even if i could get some reference signal, how can i tune the card apart from changing the rx/tx gain? even with these two down to -6.0 dB i'm still getting awful lot of echo ... The card is a simple X100P clone. The only piece i have not yet tampered with is the load the card places towards the PSTN line. The card's input circuits are all SMD - no old-fashioned (but good) separation with a transformer :( I'm thinking about playing around with increasing/decreasing resistance by placing additional resistors in the circut. Messy, but if it could help... What do you think? Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registered SIP '202' ... expires 1800. Why does it expire
Which is kinda annoying because grandstreams (at least 1.0.5.23 firmware anyway) don't do that... I have to powercycle the one on my desk once an hour if I want it to ring on incoming calls otherwise asterisk 'forgets' about it (the cisco in the other room seems to be fine). I'd love an option to work around that... This is very strange, as its usually the phones which tell the server how long their registration is to be valid. At least thats the case with all the sip phones i've seen so far. I assume this is a bug in the phone software, so upgrading the firmware could help. Marek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE:NewCUT()
In article 200509091317.j89DGtY3019393 at commserver.noach.com,
John Hill jhill at noach.com wrote:
I store my speed dial numbers in the astdb key speeddial with the number
and
then name separated by a -.
This dial plan works fine:
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,Cut(number=temp,,1)
exten = _*0XX,3,Goto(house-phones,${number},1)
The log informs me that cut is replaced with CUT.
I rewrote the dial plan using CUT (as best I can figure out) The plan
below
returns the entire string number-name and fails?
[speed-dial]
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,CUT(temp,,1)
exten = _*0XX,3,Goto(house-phones,${temp},1)
What am I missing.
The new CUT is a function, and should be used within a Set command.
Something approximating (please check the detail):
exten = _*0XX,1,Set(temp=${DB(speeddial/${EXTEN:2})})
exten = _*0XX,2,Set(number=CUT(temp,,1))
exten = _*0XX,3,Goto(house-phones,${number},1)
Your second example is calling the same Cut command as the first.
Cheers
Tony
exten = _*0XX,2,Set(number=${CUT(temp,,1)})
That fixed it. I only had to add the ${} to get the string returned.
Thanks
--john
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Re: [Asterisk-Users] Huge Echo
Seriously however, even if i could get some reference signal, how can i tune the card apart from changing the rx/tx gain? even with these two down to -6.0 dB i'm still getting awful lot of echo ... The card is a simple X100P clone. In my situation, before I found the reference signal, I found that dropping one of the gains to around -8 did indeed make the echo closer to inaudible. But, contrary to what I was expecting, both gains had to go _up_ when I did the proper testing. My rx to 2.53 and my tx to 0.25, to be exact, and the results were better than the -8 I had before. I am using a tdm card with fxo ports, though. My x100p clone at home I haven't had the chance to tune yet, but echo has never been a problem. I don't know anything about the resistance issues, good luck! Let the list know if you have any eurekas. Mojo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX PRI for single server
On Fri, 9 Sep 2005, Matthew Boehm wrote: [EMAIL PROTECTED] wrote: Ah - so the difference between your setup and mine is that you are using Sangoma (presumably) and I'm using Digium. Looks like the Digium is significantly more efficient then. It could also be that I'm using Net-SNMP to query my cpu usage and even when the machine is idle, SNMP reports about 20% CPU usage which is incorrect. Actually my figures also come out of net-snmp (via cricket). You're right, it reads higher than top. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: siemens pbx what i ask techinician?
On Fri, Sep 09, 2005 at 07:20:58PM +0200, Sander wrote: uuauuu that will great! i cant undertand too much about internal connection because. i have a PC with a te110p and my siemens is a hipath 3500 with a s2m/pri card is a E1 card. but i dont know how to connect between them. i have always the red alarm in the te110p. my conf files are both of this files i copy and paste from internet. /etc/zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 loadzone= us defaultzone = us and the /etc/asterisk/zapata.conf [channels] context=zap-in ;switchtype=qsig pridialplan=national signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no callerid=asreceived group=1 signalling=pri_net channel = 1-15,17-31 please help me!!! thanks a lot for your time Do you want to connect the asterisk with pri or with internal isdn? And what model pbx do you have? then i can tell you how to configure? Maybe some screenshots with it -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 19:35 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: siemens pbx what i ask techinician? On Fri, Sep 09, 2005 at 05:39:22PM +0200, Sander wrote: thanks Sander but i have the soft, and i can enter to the pbx conf and modify all settings, but i dont know how settings i need to change. It's not that easy then everytime you want to change someting for testing you have to ask them to change something i can give you the software for programming siemens pbx if you want -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Pablo Allietti Verzonden: vrijdag 9 september 2005 16:09 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] siemens pbx what i ask techinician? im really newbie, and i have a siemens digital pbx work in my work. i have 4 outside lines and the pbx has a E1/PRI card. what i need to ask my siemens provider(techinicians) to do in the pbx? i only have in my pbx the 9 to get a line to go outside is very simple. but i dont know what i need to ask them to programming. please help me. -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP registration issues
Hello. Is there any know issue with Asterisk 1.0.9 concerning intermittent SIP registration issues. My SIP hard phone (aastra 9133i) and soft phone (xlite) keep losing registration so calls to them go direct to VM although calling to other phones from them works fine. The logs show 'Transmitting (no NAT): SIP/2.0 403 Forbidden' which doesn't occur when they miraculously start working/registering. Asterisk seems to lose the user. Sep 9 11:47:36 VERBOSE[2444]: 12 headers, 0 lines Sep 9 11:47:36 VERBOSE[2444]: Using latest request as basis request Sep 9 11:47:36 VERBOSE[2444]: Sending to 192.168.1.100 : 5060 (non-NAT) Sep 9 11:47:36 VERBOSE[2444]: Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK289a5fe76 From: Martin sip:[EMAIL PROTECTED]:5060;tag=d6d383eca9b6910 To: Martin sip:[EMAIL PROTECTED]:5060;tag=as3c7c47f1 Call-ID: [EMAIL PROTECTED] CSeq: 54943697 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.1.100:5060 Sep 9 11:47:36 NOTICE[2444]: Registration from 'Martin sip:[EMAIL PROTECTED]:5060' failed for '192.168.1.100' Sep 9 11:47:36 VERBOSE[2444]: Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Sep 9 11:47:36 VERBOSE[2444]: Sip read: REGISTER sip:192.168.1.50:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKd88070866 Max-Forwards: 70 Content-Length: 0 To: No User sip:[EMAIL PROTECTED]:5060 From: No User sip:[EMAIL PROTECTED]:5060;tag=0e8bc4f3c760bc2 Call-ID: [EMAIL PROTECTED] CSeq: 535959059 REGISTER Contact: No User sip:[EMAIL PROTECTED] Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO User-Agent: Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 But then, some period of time later, they will start working at random times with no changes. Regards...Martin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
