Re: [Asterisk-Users] X101P minor nuisances..
At 20:22 21-3-2003 +, you wrote: Do you know for sure whether the PBX issues a call termination pulse (ie zero or reverse battery) on completion of a call? No, I don't. Anyone in .nl or with an MD110 got experience on this ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and NAT - more
On Fri, 21 Mar 2003, Mark Spencer wrote: have you tried nat=1 in your friend declaration? I notice in your dump it says non-NAT I´m in the same situation, trying to debug an ATA 186 behing a NAT. And i´m stuck with SIP/2.0 407 Proxy Authentication Required debug messages. Does anyone have any hints on thisone? It would also be nice if someone could post a working ATA186 config. This would help against stupid mistakes in that end. But back to the NAT/No-NAT issue. What exactly is the difference protocolwise inbetween the two of them? /Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] core dump in chan_h323 (again)
hi running debian now... installed openh323, pwlib and chan_h323 from cvs. Starting asterisk with -gvvvc, the following happens: it starts, writes some status info, and 'KaBoom' on the output, waits ~13 seconds and dies. any ideas? vmstat output: -- procs memoryswap io system cpu r b w swpd free buff cache si sobibo incs us sy id 0 0 0 1540 13732 60216 264724 0 0 0 0 10718 0 0 100 0 0 0 1540 10012 60216 264724 0 0 0 0 545 573 36 7 57 0 0 0 1540 10012 60216 264724 0 0 0 0 104 208 0 0 100 0 0 0 1540 10012 60216 264724 0 0 0 0 104 211 0 0 100 0 0 0 1540 10012 60216 264724 0 0 0 0 103 209 0 0 100 0 0 0 1540 10008 60220 264724 0 0 0 160 112 222 0 0 100 0 0 0 1540 10004 60220 264724 0 0 0 0 105 209 0 0 100 0 0 0 1540 10004 60220 264724 0 0 0 0 106 211 0 0 100 0 0 0 1540 10004 60220 264724 0 0 0 0 104 208 0 0 100 0 0 0 1540 10004 60220 264724 0 0 0 0 103 211 0 0 100 0 0 0 1540 10004 60220 264724 0 0 0 0 103 212 0 0 100 0 0 0 1540 6420 60240 268320 0 0 0 0 107 161 0 4 96 1 0 0 1540 6420 60240 268320 0 0 0 0 10310 0 0 100 0 0 0 1540 10112 60240 268320 0 0 0 0 11032 0 0 100 Console output: -- [chan_h323.so] = (The NuFone Network's Open H.323 Channel Driver) == Creating H.323 Endpoint == Parsing '/etc/asterisk/h323.conf': Found == Registered channel type 'H323' (The NuFone Network's Open H.323 Channel Driver) == H.323 listener started on ip$*:4 KaBoom? Segmentation fault (core dumped) Backtrace: -- #0 0x400fcc1b in free () from /lib/libc.so.6 (gdb) bt #0 0x400fcc1b in free () from /lib/libc.so.6 #1 0x400fcaa3 in free () from /lib/libc.so.6 #2 0x41246854 in __builtin_delete () from /usr/lib/libstdc++-libc6.2-2.so.3 #3 0x40c783e1 in H323TransportUDP::~H323TransportUDP () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #4 0x40ccbae6 in H323Transactor::~H323Transactor () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #5 0x40cc8628 in H225_RAS::~H225_RAS () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #6 0x40c8c85a in H323Gatekeeper::~H323Gatekeeper () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #7 0x40c35f61 in H323EndPoint::InternalRegisterGatekeeper () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #8 0x40c35d02 in H323EndPoint::SetGatekeeper () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #9 0x402746bf in h323_set_gk (gatekeeper_discover=0, gatekeeper=0x40283620 192.168.144.253, secret=0x402836a0 ) at ast_h323.cpp:908 #10 0x4026f268 in load_module () at chan_h323.c:1590 #11 0x08053736 in ast_load_resource (resource_name=0x80c2f80 chan_h323.so) at loader.c:272 #12 0x080538c2 in load_modules () at loader.c:318 #13 0x0807804b in main (argc=2, argv=0xbdc4) at asterisk.c:1293 (gdb) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
I remember at some point getting 488 media errors if I didn't enable gsm. As I mentioned, I'm getting 480 Temporarily not available, not 488 media errors. I tried the grotesque hack of making handle_response() ignore 480 errors, which *seems* to work. Hmm. -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP and rfc2833 dtmf + # transfer
Has anybody noticed that # transfers aren't working anymore when SIP is used with rfc2833 dtmfmode? They work as espected with inband dtmf. lele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] core dump in chan_h323 (again)
This looks a whole lot like the -rx 'reload' problem we've been having. Are you running latest CVS of everything? Jeremy McNamara Roy Sigurd Karlsbakk wrote: hi running debian now... installed openh323, pwlib and chan_h323 from cvs. Starting asterisk with -gvvvc, the following happens: it starts, writes some status info, and 'KaBoom' on the output, waits ~13 seconds and dies. any ideas? vmstat output: -- procs memoryswap io system cpu r b w swpd free buff cache si sobibo incs us sy id 0 0 0 1540 13732 60216 264724 0 0 0 0 10718 0 0 100 0 0 0 1540 10012 60216 264724 0 0 0 0 545 573 36 7 57 0 0 0 1540 10012 60216 264724 0 0 0 0 104 208 0 0 100 0 0 0 1540 10012 60216 264724 0 0 0 0 104 211 0 0 100 0 0 0 1540 10012 60216 264724 0 0 0 0 103 209 0 0 100 0 0 0 1540 10008 60220 264724 0 0 0 160 112 222 0 0 100 0 0 0 1540 10004 60220 264724 0 0 0 0 105 209 0 0 100 0 0 0 1540 10004 60220 264724 0 0 0 0 106 211 0 0 100 0 0 0 1540 10004 60220 264724 0 0 0 0 104 208 0 0 100 0 0 0 1540 10004 60220 264724 0 0 0 0 103 211 0 0 100 0 0 0 1540 10004 60220 264724 0 0 0 0 103 212 0 0 100 0 0 0 1540 6420 60240 268320 0 0 0 0 107 161 0 4 96 1 0 0 1540 6420 60240 268320 0 0 0 0 10310 0 0 100 0 0 0 1540 10112 60240 268320 0 0 0 0 11032 0 0 100 Console output: -- [chan_h323.so] = (The NuFone Network's Open H.323 Channel Driver) == Creating H.323 Endpoint == Parsing '/etc/asterisk/h323.conf': Found == Registered channel type 'H323' (The NuFone Network's Open H.323 Channel Driver) == H.323 listener started on ip$*:4 KaBoom? Segmentation fault (core dumped) Backtrace: -- #0 0x400fcc1b in free () from /lib/libc.so.6 (gdb) bt #0 0x400fcc1b in free () from /lib/libc.so.6 #1 0x400fcaa3 in free () from /lib/libc.so.6 #2 0x41246854 in __builtin_delete () from /usr/lib/libstdc++-libc6.2-2.so.3 #3 0x40c783e1 in H323TransportUDP::~H323TransportUDP () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #4 0x40ccbae6 in H323Transactor::~H323Transactor () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #5 0x40cc8628 in H225_RAS::~H225_RAS () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #6 0x40c8c85a in H323Gatekeeper::~H323Gatekeeper () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #7 0x40c35f61 in H323EndPoint::InternalRegisterGatekeeper () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #8 0x40c35d02 in H323EndPoint::SetGatekeeper () from /usr/src/openh323/lib/libh323_linux_x86_r.so.1 #9 0x402746bf in h323_set_gk (gatekeeper_discover=0, gatekeeper=0x40283620 192.168.144.253, secret=0x402836a0 ) at ast_h323.cpp:908 #10 0x4026f268 in load_module () at chan_h323.c:1590 #11 0x08053736 in ast_load_resource (resource_name=0x80c2f80 chan_h323.so) at loader.c:272 #12 0x080538c2 in load_modules () at loader.c:318 #13 0x0807804b in main (argc=2, argv=0xbdc4) at asterisk.c:1293 (gdb) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and rfc2833 dtmf + # transfer
Ciao lele. Yes, this is known. also, if you use inband dtmf you can't log into the voicemail (doesn't recon. the dtmf), but voicemail works with rfc2833 matteo Il sab, 2003-03-22 alle 14:45, Lele Forzani ha scritto: Has anybody noticed that # transfers aren't working anymore when SIP is used with rfc2833 dtmfmode? They work as espected with inband dtmf. lele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk w/ Netmeeting?
OpenPhone is a windows only GUI application. ohphone is the text based application. Gregg On Fri, 2003-03-21 at 18:54, Mike Diehl wrote: On Friday 21 March 2003 1:49 pm, Jeremy McNamara wrote: Mike Diehl wrote: Unfortunately, I'm kinda commited to NetMeeting. The problem is that I have friends with a mix of Windows and Linux machines. Between Netmeeting and Gnomemeeting, I should be able to get everyone connected. Good guy why? Why can't you use a decent H.323 client like OpenPhone? You may have to educate me here. What is wrong with NetMeeting/Gnomemeeting? I've gotten the impression that OpenPhone was a text-based application, true? I have friends/family who wouldn't be able/willing to use such a tool. I did notice that Gnomemeeting didn't support as much functionality as I would have expected, but it's free and easy to get. Looks like someone needs to write a really goood win32 endpoint application. Actually, someone needs to write a really good win32, period. grin Thanx for your thoughts. Mike Diehl. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] core dump in chan_h323 (again)
This looks a whole lot like the -rx 'reload' problem we've been having. Are you running latest CVS of everything? yes roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with linejack as a trunk?
On Friday 21 March 2003 22:55, Dave Packham wrote:
I have a linejack and a phone jack in my asterisk server
working well between the SIP phones and the phonejack. what I
cannot get to work is the outbound linejack Phone/phone0 trunk
line? how can I get a SIP or Phone/phone1 phonejack phone to
dial 9 then outside number and pickup Phone/phone0 and dial
it? right now it accepts a 95551212 but busy's on the last
digit 2. no outside dial. would the sip debug help? ill post
if you need
Thanks
Dave Packham
I have this in my extensions.conf
and have tried both of the below options
---snip---
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten = _9NXX,1,StripMSD,1
exten = _NXX,1,Dial,Phone/phone0/BYEXTENSION
Here's your problem: you've defined an initial priority to
stripMSD, but no secondary priority for the stripped number
(btw, BYEXTENSION is now deprecated):
exten = _9NXXNXXX,1,StripMSD,1
exten = _NXXNXXX,2,Dial,Phone/phone0/${EXTEN}
-Tilghman
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Re: [Asterisk-Users] SIP and rfc2833 dtmf + # transfer
Whoa, this is pretty weird. Somebody find me on IRC and lets get this fixed! Mark On 22 Mar 2003, Matteo Brancaleoni wrote: Ciao lele. Yes, this is known. also, if you use inband dtmf you can't log into the voicemail (doesn't recon. the dtmf), but voicemail works with rfc2833 matteo Il sab, 2003-03-22 alle 14:45, Lele Forzani ha scritto: Has anybody noticed that # transfers aren't working anymore when SIP is used with rfc2833 dtmfmode? They work as espected with inband dtmf. lele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Luke Howard wrote:I remember at some point getting 488 media errors if I didn't enable gsm. As I mentioned, I'm getting 480 Temporarily not available, not 488 media errors. I tried the grotesque hack of making handle_response() ignore 480 errors, which *seems* to work. Hmm. I tried that, and at least for me it has a number of subtle side effects: 1. Calls all cut off after just a few minutes 2. Subsequent calls after those messages have been ignored are bollixed up. I have complained repeatedly to iconnecthere, but they don't have much of a customer service model. Canned email responses is about the long and short of it. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with linejack as a trunk?
Does asterisk work now with the linejack in terms of both incoming and outgoing calls? That would be sweet, but I thought it was only one direction or the other, and so I gave up on the idea of using mine. ?? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with linejack as a trunk?
On Saturday 22 March 2003 12:11, Brian Capouch wrote: Does asterisk work now with the linejack in terms of both incoming and outgoing calls? That would be sweet, but I thought it was only one direction or the other, and so I gave up on the idea of using mine. AFAIK, it has always worked for both incoming and outgoing calls. What it canNOT do is to provide both an FXO (line) as well as an FXS (station) interface at the same time. In other words, it provides a single channel which can be provisioned as an FXO or an FXS at any single time, reprovisioning the interface would require a configuration file change and a restart of asterisk, and reprovisioning cannot happen dynamically within the call plan. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Standalone S100U: Is there some trick?
I am using an ATA 186 as my primary station interface, so I thought, Why not move that S100U over to another box and gain myself another remote client? So on the main machine I removed the line fxoks=2 and then in /etc/asterisk/zapata.conf I removed the references to channel 2. Then on the new machine, I did a full install of asterisk, and reversed what I had been doing, removing the fxsks=1 line, and then the references to channel 1 in /etc/asterisk/zapata.conf The main server restarted and appears to work just fine; the S100U server won't start. I get the following messages at the tail of the console: WARNING[16384]: File chan_zap.c, Line 580 (zt_open): Unable to specify channel 2: Device or resource busy ERROR[16384]: File chan_zap.c, Line 4591 (mkintf): Unable to open channel 2: Device or resource busy here = 0, tmp-channel = 0, channel = 2 ERROR[16384]: File chan_zap.c, Line 6189 (load_module): Unable to register channel '2' WARNING[16384]: File loader.c, Line 273 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[16384]: File loader.c, Line 368 (load_modules): Loading module chan_zap.so failed! Is there a trick to this? Thanks in advance for any help that might be out there. . . B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Standalone S100U: Is there some trick?
On Saturday 22 March 2003 13:26, Brian Capouch wrote: I am using an ATA 186 as my primary station interface, so I thought, Why not move that S100U over to another box and gain myself another remote client? So on the main machine I removed the line fxoks=2 and then in /etc/asterisk/zapata.conf I removed the references to channel 2. Then on the new machine, I did a full install of asterisk, and reversed what I had been doing, removing the fxsks=1 line, and then the references to channel 1 in /etc/asterisk/zapata.conf The main server restarted and appears to work just fine; the S100U server won't start. I get the following messages at the tail of the console: WARNING[16384]: File chan_zap.c, Line 580 (zt_open): Unable to specify channel 2: Device or resource busy ERROR[16384]: File chan_zap.c, Line 4591 (mkintf): Unable to open channel 2: Device or resource busy here = 0, tmp-channel = 0, channel = 2 ERROR[16384]: File chan_zap.c, Line 6189 (load_module): Unable to register channel '2' WARNING[16384]: File loader.c, Line 273 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[16384]: File loader.c, Line 368 (load_modules): Loading module chan_zap.so failed! Is there a trick to this? Channels are unique to the host machine. So, on your second machine, you should have 'fxoks=1'. It's erroring out because it has a channel 1, but no channel 2. Of course, you'll also need to set up iax.conf such that the two machines see each other, and set up extensions.conf such that when you communicate with the other machine, you Dial the IAX resource. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] will the digium cards
work with Bayonne? will i need special drivers to use them with Bayonne? thanks 4 your help dwayne ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
I tried the grotesque hack of making handle_response() ignore 480 errors, which *seems* to work. Hmm. I tried that, and at least for me it has a number of subtle side effects: Or maybe we should send an ACK to them -- I need to read the SIP RFC... -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Standalone S100U: Is there some trick?
Tilghman Lesher wrote: Is there a trick to this? Channels are unique to the host machine. So, on your second machine, you should have 'fxoks=1'. It's erroring out because it has a channel 1, but no channel 2. Of course, you'll also need to set up iax.conf such that the two machines see each other, and set up extensions.conf such that when you communicate with the other machine, you Dial the IAX resource. OK, now I have that much going. I can use the S100U to make calls to my other server. Now there is a new twist. The quality is horrible, and I notice when I run top that the asterisk process is eating up 96% of the CPU. It pegs right as I start it and stays that way without any sort of letup. On the other machine, asterisk when idle doesn't even show up on the first screen of top's processes. How could I figure out what asterisk could be doing with the CPU? The second machine is a laptop not a desktop, but the processors are comparable and this same laptop has been in long service w/o any hardware problems, including quite a few USB-based things. Not sure how I can look inside to see what asterisk could be doing. . Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX weirdness
I loaded the latest CVS this morning on the machine we use as our main IAX gateway. I often use this gateway to connect to an IVR machine that doesn't have a T-span, so I connect via IAX. However, after installing the new * release, calls to the IVR machine acted strangely. The main problem was that various voice prompts were missing or would cut off after a brief instant. For example, instead of playing Please record your message at the tone, beep, it just played the beep. There was no dead time when it should have been playing the prompt; it just skipped it. (Our prompts are in µ-law format if that makes a difference.) When I pressed the option for billing to a credit card, instead of Enter your credit card number, it would just play Ent This was not completely consistent. On some calls the prompts would play; on others we would get this behavior. Reverting to our prior * version on our IAX gateway made everything work again. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with linejack as a trunk?
What is the maximum capacity of simultaneous users an asterisk box can handle at 1 Ghz? 2Ghz? 3 Ghz? Shido - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, March 22, 2003 1:11 PM Subject: Re: [Asterisk-Users] Help with linejack as a trunk? Does asterisk work now with the linejack in terms of both incoming and outgoing calls? That would be sweet, but I thought it was only one direction or the other, and so I gave up on the idea of using mine. ?? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Moreover, if anyone has a packet trace of iConnectHere's SIP client making a call (which presumably does work), then please send it along... it would be interesting to see whether Asterisk, at fault or not, can be made to work around this properly. -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Luke Howard wrote:I tried the grotesque hack of making handle_response() ignore 480 errors, which *seems* to work. Hmm. I tried that, and at least for me it has a number of subtle side effects: Or maybe we should send an ACK to them -- I need to read the SIP RFC... Tried that, doesn't work. I should add that in my config I'm totally behind NAT, both asterisk and an ATA186 that talks to it. So that may be confounding me in terms of what I'm seeing. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Standalone S100U: Is there some trick?
On Saturday 22 March 2003 15:26, Brian Capouch wrote: Tilghman Lesher wrote: Is there a trick to this? Channels are unique to the host machine. So, on your second machine, you should have 'fxoks=1'. It's erroring out because it has a channel 1, but no channel 2. Of course, you'll also need to set up iax.conf such that the two machines see each other, and set up extensions.conf such that when you communicate with the other machine, you Dial the IAX resource. OK, now I have that much going. I can use the S100U to make calls to my other server. Now there is a new twist. The quality is horrible, and I notice when I run top that the asterisk process is eating up 96% of the CPU. It pegs right as I start it and stays that way without any sort of letup. On the other machine, asterisk when idle doesn't even show up on the first screen of top's processes. How could I figure out what asterisk could be doing with the CPU? The second machine is a laptop not a desktop, but the processors are comparable and this same laptop has been in long service w/o any hardware problems, including quite a few USB-based things. Not sure how I can look inside to see what asterisk could be doing. . What all extra are you running on the laptop? For example, are you running the FrameBuffer console or X? Either of these might take up enough interrupts to cause the bad sound quality. In terms of CPU usage, what is the load average on the system? I'd worry less about total CPU usage on laptops, as some of the more recent systems put the CPU in low power mode when there isn't much load to save battery life (thus CPU percentage is higher per process). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank FXS Questions
Here is my question. We have a Asterisk PBX with an T400P Card. We have 2 T1's comming into this card from a Channel bank. So this gives us 48 Channels and 48 Extensions. Here's my issue: How and where do my extensions get punched down? Is there a 24 extension FXS Card with an Amphenol Interface on it? Or ? This is where I am getting lost. I am used to seeing the Extension Cards be handled by the PBX System I am not seeing how this is done with the Asterisk System. I hope this is making sence. Could someone please shed some lite on this subject. Thank you, Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with linejack as a trunk?
Tilghman Lesher wrote: On Saturday 22 March 2003 12:11, Brian Capouch wrote: Does asterisk work now with the linejack in terms of both incoming and outgoing calls? That would be sweet, but I thought it was only one direction or the other, and so I gave up on the idea of using mine. AFAIK, it has always worked for both incoming and outgoing calls. What it canNOT do is to provide both an FXO (line) as well as an FXS (station) interface at the same time. In other words, it provides a single channel which can be provisioned as an FXO or an FXS at any single time, reprovisioning the interface would require a configuration file change and a restart of asterisk, and reprovisioning cannot happen dynamically within the call plan. I wonder if that's the case. . . I am working with a known-good (i.e. works with H.323 ohphone) Linejack. I have it conf'ed, basically, like this: mode=fxo device = /dev/phone0 But show channels doesn't show anything, and when I try to dial I get these errors: -- Executing Dial([EMAIL PROTECTED]:5036]/6,Phone/phone0/5551212) in new stack NOTICE[24591]: File app_dial.c, Line 443 (dial_exec): Unable to create channel of type 'Phone' == Everyone is busy at this time I thought I had read a couple of threads on this list that the Linejack only does inbound calling. . . . It does seem to work OK when I configure it that way. ?? B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channel Bank FXS Questions
The channel bank should have your amphenol connector(s). You should consult your channel bank doc's for wiring requirements, as some manufacturers deviate from the standard 50-pin amphenol wiring. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, March 22, 2003 6:04 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Channel Bank FXS Questions Here is my question. We have a Asterisk PBX with an T400P Card. We have 2 T1's comming into this card from a Channel bank. So this gives us 48 Channels and 48 Extensions. Here's my issue: How and where do my extensions get punched down? Is there a 24 extension FXS Card with an Amphenol Interface on it? Or ? This is where I am getting lost. I am used to seeing the Extension Cards be handled by the PBX System I am not seeing how this is done with the Asterisk System. I hope this is making sence. Could someone please shed some lite on this subject. Thank you, Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
I should add that in my config I'm totally behind NAT, both asterisk and an ATA186 that talks to it. Hmm, both our SIP phones and Asterisk are on visible IPs. -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Other IP Phone Possabilities
Has anyone looked into or used these? http://www.pingtel.com/pr_xpressa.jsp They look and sound compatible... Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank FXS Questions
So the Channel Bank that the ISP provides or the Telco company is sufficent? Or is this an additional piece of hardware I need to purcahse? Or another example lets say I have two T100P cards... how would this get utilized. Let's say I am using both T1 Cards for voice... Geoff On Sat, 22 Mar 2003, Richard Lyman wrote: if i'm understanding, you can simply use an 24 port FXS channel bank and feed if from one of the left over ports on the t400p [EMAIL PROTECTED] wrote: Here is my question. We have a Asterisk PBX with an T400P Card. We have 2 T1's comming into this card from a Channel bank. So this gives us 48 Channels and 48 Extensions. Here's my issue: How and where do my extensions get punched down? Is there a 24 extension FXS Card with an Amphenol Interface on it? Or ? This is where I am getting lost. I am used to seeing the Extension Cards be handled by the PBX System I am not seeing how this is done with the Asterisk System. I hope this is making sence. Could someone please shed some lite on this subject. Thank you, Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound card and other warning messages
Greetings Asterisk users. When I launch Asterisk, I get the following Asterisk CVS-03/20/03-16:56:24, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer [EMAIL PROTECTED] = [ Booting..-- Registered indication country 'us' -- Registered indication country 'au' -- Registered indication country 'fr' -- Registered indication country 'de' -- Registered indication country 'nl' -- Registered indication country 'uk' -- Setting default indication country to 'us' ...WARNING[16384]: File chan_oss.c, Line 342 (setformat): Requested 8000 Hz, got 8018 Hz -- sound may be choppy It means only that you have some cheap sound card. If it works for you that's fine. WARNING[98311]: File chan_oss.c, Line 228 (sound_thread): Read error on sound device: Resource temporarily unavailable If you don't want this messages put noload = chan_oss.so in /etc/asterisk/modules.conf .WARNING[16384]: File chan_iax2.c, Line 4927 (set_config): Ignoring port for now This channel driver is under developement. Also it reads the config files of chan_zap.c so you shouldn't bother about that. /var/spool/asterisk/outgoing .WARNING[16384]: File chan_zap.c, Line 6416 (load_module): Ignoring rxwink It's connected with Atlas ... just comment out in zapata.conf ] Asterisk Ready. *CLI 1. The warning messages listed above, should I be concerned? I tested the sound card, by entering the command dial, it plays the Asterisk greeting message on the speakers. I woudln't worry about yours. 2. Should I replace the sound card with a different one. Currently, it's a ES1370. (Ensoniq AudioPCI) If it works for you = you don't have to. 3. After reading multiple docs, FAQ and mailing list archives, I am unable to find out purpose of the sound card? Is it used to play music, while on hold? Telephony PBX's usually don't have a sound card in them. You don't need sound card in your PBX. It may be used as another extension (read: another phone) 4. The WARNING - chan_iax2.c, I think, this is a configuration problem (IAX), from the default files. That I have not gotten to yet. THis channel driver does not use port keyword. regards Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Developer's Kit (LITE) or 2 Wildcard X100P FXO's
I am setting up a simple information boxes and voicemail boxes solution with a round robin call forwarding extension. Would the Dev Kit LITE be the ticket or should I go with two (2) Wildcard X100P FXO's with two POTS lines? One for incoming calls... and if the caller chooses a specific extension the call is forwarded through the other device and out to the outside number. Recommendations appreciated. -- Mark Street, D.C. Red Hat Certified Engineer Cert# 807302251406074 -- Key fingerprint = 3949 39E4 6317 7C3C 023E 2B1F 6FB3 06E7 D109 56C0 GPG key http://www.streetchiro.com/pubkey.asc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with linejack as a trunk?
On Sat, 2003-03-22 at 15:35, shido wrote: What is the maximum capacity of simultaneous users an asterisk box can handle at 1 Ghz? 2Ghz? 3 Ghz? Depends on how what your users are doing, what else you are running on the machine, and what kind of interfaces you are using. VoIP users use the ethernet interface, and as long as you are not compressing, they shouldn't take much in processor load. Some PSTN interfaces such as the Zapata cards require that you are able to service the card quickly so you can't load it as heavy. If your users are talking to each other and there isn't compression or conversion in the way, it doesn't take much power. If they are interacting with the pbx in a IVR or application manner, then you use some more power. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank FXS Questions
It depends on the make of channel bank. The ones I'm most familier with (tellabs/NT D3/D4), have amphenol connectors on the chassis and those connectors feed to the cards/modules through the channel bank backplane connectors. Cable from the bank to some kind of distribution point... Punchdown, wirewrap (yeah, wirewrap!) whatever and you're done. Ahh for the days of butterfly tools. :) Bruce Ferrell [EMAIL PROTECTED] wrote: Here is my question. We have a Asterisk PBX with an T400P Card. We have 2 T1's comming into this card from a Channel bank. So this gives us 48 Channels and 48 Extensions. Here's my issue: How and where do my extensions get punched down? Is there a 24 extension FXS Card with an Amphenol Interface on it? Or ? This is where I am getting lost. I am used to seeing the Extension Cards be handled by the PBX System I am not seeing how this is done with the Asterisk System. I hope this is making sence. Could someone please shed some lite on this subject. Thank you, Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iconnecthere 480 error: is there a workaround?
Or maybe we should send an ACK to them -- I need to read the SIP RFC... Tried that, doesn't work. I should add that in my config I'm totally behind NAT, both asterisk and an ATA186 that talks to it. So that may be confounding me in terms of what I'm seeing. I do see the same problem: after a few minutes, the call is dropped (this is using Asterisk patched to ignore 480 Temporarily not available errors). From the log below it _seems_ like iConnectHere is waiting for an acknowledgment to the 480, but you noted that you tried this? It seems to be purely a signalling problem as the call is setup fine between the SIP phone and the gateway (which in this case appeared to be somewhere in Austria...) -- Executing Macro(SIP/515-Office-b8b1, iconnecthere|33145207135|60) in new stack -- Executing Dial(SIP/515-Office-b8b1, SIP/[EMAIL PROTECTED]|60|r) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnecthere-ec91 is ringing -- SIP/iconnecthere-ec91 answered SIP/515-Office-b8b1 -- Attempting native bridge of SIP/515-Office-b8b1 and SIP/iconnecthere-ec91 -- Got SIP response 408 Request Timeout back from 213.137.73.178 == Spawn extension (macro-iconnecthere, s, 1) exited non-zero on 'SIP/515-Office-b8b1' in macro 'iconnecthere' == Spawn extension (local, 933145207135, 1) exited non-zero on 'SIP/515-Office-b8b1' -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
