Re: [Asterisk-Users] T100P No D-channels
H I guess from a troubleshooting standpoint to try and pinpoint the problem what I would do is remove all cards from the system and then only replace the cards that are absolutely necessary like your SCSI card and your Video card and of course the T100P and then check /proc/interrupts to see if you're having any more MISses... Also are you getting interrupt ERRs as well? Is APIC enabled for your board? If it is, you'll see things like IO-APIC-edge or IO-APIC-level in your /proc/interrupts. If not, you'll see XT-PIC for all interrupts... If you can't get APIC turned on you might try upgrading your kernel, your motherboard/bios may be blacklisted in that particular kernel... You definitely DO NOT want to share interrupts on the T100P unless it's a low-interrupt device like a USB controller or your video card. You definitely don't want to share with say a NIC or a SCSI controller... If it is sharing with one of those, try shuffling the cards around in different slots and make sure that your T100P isn't in slot 5. Slot 5 is usually shared with Slot 1 if they're on the same bus... Of course you've probably already tried all of this but just in case you haven't... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Headset for Cisco 7960?
On Tue, 31 Aug 2004, Benjamin Johnson wrote: I found the same with lots of headsets and my 7940, but I've just plugged the headset from my Norstar system into the *handset* port on my and it works perfectly. It's not ideal but it'll do for now! Ah, yeah, didn't think of that - works fine. On Tue, 31 Aug 2004, Dan Austin wrote: GN-Netcom has a nice little headset for about US $120. As to the pin-out, I believe that the headset port uses pins 14 instead of 23. I'll have to try building an adapter - shouldn't be tough. Thanks all! | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P No D-channels
On Tue, 2004-08-31 at 15:58, Pliva, Josef wrote: Unfortunately, I am seeing great many missed IRQs continually...if in fact it is that which causes the loss of D-channel. Then you need to find out why interrupts are being locked for long enough to make the T100P miss interrupts. Common causes: frame buffer, graphics, RAID drivers, IDE DMA, and many, many other things. Doing an hdparm -u1 /dev/hda might help -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why is it called 'Comedian Mail?
Kris Boutilier [EMAIL PROTECTED] wrote: Inquiring (management) minds want to know. I'm assuming it's because 'it's funny how simple it really is to write a really decent voicemail system'? Perhaps it was written by someone with a red nose, oversized shoes and a custard pie. I don't know either. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why is it called 'Comedian Mail?
I've wondered that myself... obviously the writer has a sense of humor! :) I like the sound of Digium Mail, it sounds cool... - Original Message - From: Kevin Walsh [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 2:30 PM Subject: RE: [Asterisk-Users] Why is it called 'Comedian Mail? Kris Boutilier [EMAIL PROTECTED] wrote: Inquiring (management) minds want to know. I'm assuming it's because 'it's funny how simple it really is to write a really decent voicemail system'? Perhaps it was written by someone with a red nose, oversized shoes and a custard pie. I don't know either. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can asterisk detect BUSY signal?
suppose I have agents waiting on a queue and I configure asterisk to dial out and to forward the call to the first agent enqueued. Asterisk will do it even if the answer to the call is busy. Is it possible to configure asterisk to detect the busy signal and, in that case, dial another number, without wasting agent's time? Spam-dialling should be made illegal. I, for one, wouldn't spend two seconds adding features to support this sort of usage. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why is it called 'Comedian Mail?
hmm Meridian Voice Mail == Comedian Voice Mail:) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Client
On Tue, 31 Aug 2004 14:14:40 -0400, Jon Bebeau [EMAIL PROTECTED] wrote: Hello all, I'm working an a switchboard console for Asterisk and would like to investigate using IAX Client library to Asterisk. I don't seem to be able to find the source. I'm planning on a Win32 app. Guidance on where the source is or who to take to is requested. Jon iaxclient.sf.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why is it called 'Comedian Mail?
Chris Shaw [EMAIL PROTECTED] lazily top-posted: I've wondered that myself... obviously the writer has a sense of humor! :) I like the sound of Digium Mail, it sounds cool... I like the sound of, err, nothing. Mine just prompts for Mailbox? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can asterisk detect BUSY signal?
On Tuesday 31 August 2004 17:36, Kevin Walsh wrote: Spam-dialling should be made illegal. I, for one, wouldn't spend two seconds adding features to support this sort of usage. I can think of at least one legitimate use for this -- reverse spam dialling, or at least real person detection. I hate sitting in hold queues and my usual method is to put the phone on speaker and listen to Muzak while I wait. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can asterisk detect BUSY signal?
Lol reverse hold! I can't see that working ever though, I tried it once and the agent at the other end hung up on me... I had to wait another hour in the queue... - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 2:53 PM Subject: Re: [Asterisk-Users] Can asterisk detect BUSY signal? On Tuesday 31 August 2004 17:36, Kevin Walsh wrote: Spam-dialling should be made illegal. I, for one, wouldn't spend two seconds adding features to support this sort of usage. I can think of at least one legitimate use for this -- reverse spam dialling, or at least real person detection. I hate sitting in hold queues and my usual method is to put the phone on speaker and listen to Muzak while I wait. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why is it called 'Comedian Mail?
I had always thought it was because an early clone of 'meridian mail' was called 'chameleon mail' and 'comedian mail' is a really good take off on 'chameleon mail'. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - Voicemail Password Help
On Tue, 2004-08-31 at 10:53, Stephen Hon wrote:
Paul,
What you can do is modify the source code for the voicemail application.
Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the file
'vm-password' to 'pls-enter-vm-password'.
Recompile and install.
Then in your macro remove the line that plays the 'pls-enter-vm-password' file.
Steve
Why do that ? when you can simply replace the prompt file. Using your
method will need a recompile every time a different prompt needs to be
used.
From: [EMAIL PROTECTED] on behalf of Java Rockx
Sent: Mon 8/30/2004 8:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Newbie - Voicemail Password Help
Hello All.
I'm just beginning with Asterisk and I have it all working now. I'm using
Asterisk 1.0 RC1.
My only question is this; when I check my voice mail the PBX simply says
password. I wanted to make it say please enter your voice mail password so
I am using Background(pls-enter-vm-password).
However now I hear Please enter your voice mail password password when I
check my messages.
That's not a type-o. It says password twice.
Here is my extensions.conf file.
[macro-vmanswer]
exten = s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
exten = s,2,Background(pls-enter-vm-password)
exten = s,3,VoicemailMain(${ARG1})
exten = s,4,Hangup
exten = s,5,Voicemail(u${ARG1})
exten = s,6,Hangup
[default]
exten = 1002,1,Macro(vmanswer,1002)
The whole point of the vmanswer macro is to go to the voice mail main menu
automatically when calling from your own phone, otherwise it sends callers to
the voice mail system to leave a message. Perhaps there's a better way to do
this as well. If so, please let me know.
Regards,
Paul
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Re: [Asterisk-Users] Newbie - Voicemail Password Help
On Tuesday 31 August 2004 23:22, Umar Sear wrote: On Tue, 2004-08-31 at 10:53, Stephen Hon wrote: Paul, What you can do is modify the source code for the voicemail application. Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the file 'vm-password' to 'pls-enter-vm-password'. Recompile and install. Then in your macro remove the line that plays the 'pls-enter-vm-password' file. Steve Why do that ? when you can simply replace the prompt file. Using your method will need a recompile every time a different prompt needs to be used. [...] It should be neither of those ways. The system should ideally be reading a definition file as to what files should be played and when. This could also allow for different languages. B Why can't people on this list delete the signatures? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap ANSWER the Call
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Nobody knows about that strange behaviour of Zap channels or
at least if is that right?
Thanks in advance.
Rodrigo P. Telles wrote:
| Hi,
|
| I'm using a TDM400 with one FXS and one FXO module (developer kit) and
| I've been testing termination from SIP phones to PSTN and it works fine,
| but
| asterisk accounting is doing something strange (for me).
| Scenario:
| 1 - extension 1009 (SIP phone - BT101)
| 2 - Zap/4-1 (TDM400 FXO module)
|
| extensions.conf:
| [dialout]
| exten = _9.,1,Dial(Zap/4/${EXTEN:1}|25|r)
| exten = _9.,2,Congestion
|
| [sip]
| include = dialout
| exten = 1009,1,Dial(SIP/1009,20,rt)
|
| So, when I dial 9something from 1009, something rings and then I
| hangup the phone.
| I realised that asterisk thought that something ANSWERED the call:
|
| ,1009,9something,sip,Tests,SIP/1009-42fb,Zap/4-1,Dial,
| Zap/4/something|25,2004-08-27 20:15:34,200
| 4-08-27 20:15:36,2004-08-27 20:15:43,9,7,ANSWERED,BILLING
|
| Is that right?
|
| Version: Asterisk 0.9.0
|
| Thanks in advance.
|
| --
|
| Rodrigo P. Telles [EMAIL PROTECTED]
| Project Manager
| Devel-IT - http://www.devel-it.com.br
| TDKOM Group
|
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- --
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Project Manager
Devel-IT - http://www.devel-it.com.br
TDKOM Group
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TGwaMVemCOPE1uOZQPFKdnc=
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[Asterisk-Users] Asterisk SIP between two networks
Hi all, I have next configuration: SIP Provider---ADSL router---localnet 192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones first localnet 192.168.20.0 second localnet 172.28.240.0 in second localnet we have softphone and the first localnet is connected to ADSL router to connect to our SIP provider. The problem is the next: if I put in [general] section in sip.conf, bindaddr=0.0.0.0, I can't register in my SIP provider. If I put 192.168.20.10 in bindaddr I can register in my SIP provider but softphones can't register into asterisk. I 'm using asterisk RC1. Any idea? srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie - Voicemail Password Help
The vm-password file is used else where, such as queues. If you changed
it, then you would change for all the other applications.
I guess, since we alter the source code often.. it's not that big of a
deal. We just create our own patch files and if we update from cvs, we
patch against the new source.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Umar Sear
Sent: Tuesday, August 31, 2004 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Newbie - Voicemail Password Help
On Tue, 2004-08-31 at 10:53, Stephen Hon wrote:
Paul,
What you can do is modify the source code for the voicemail
application.
Edit line 3579 in /usr/src/asterisk/apps/app_voicemail.c. Change the
file 'vm-password' to 'pls-enter-vm-password'.
Recompile and install.
Then in your macro remove the line that plays the
'pls-enter-vm-password' file.
Steve
Why do that ? when you can simply replace the prompt file. Using your
method will need a recompile every time a different prompt needs to be
used.
From: [EMAIL PROTECTED] on behalf of Java Rockx
Sent: Mon 8/30/2004 8:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Newbie - Voicemail Password Help
Hello All.
I'm just beginning with Asterisk and I have it all working now. I'm
using
Asterisk 1.0 RC1.
My only question is this; when I check my voice mail the PBX simply
says
password. I wanted to make it say please enter your voice mail
password so
I am using Background(pls-enter-vm-password).
However now I hear Please enter your voice mail password password
when I
check my messages.
That's not a type-o. It says password twice.
Here is my extensions.conf file.
[macro-vmanswer]
exten = s,1,GotoIf($[${ARG1} = ${CALLERIDNUM}]?2:5)
exten = s,2,Background(pls-enter-vm-password)
exten = s,3,VoicemailMain(${ARG1})
exten = s,4,Hangup
exten = s,5,Voicemail(u${ARG1})
exten = s,6,Hangup
[default]
exten = 1002,1,Macro(vmanswer,1002)
The whole point of the vmanswer macro is to go to the voice mail main
menu
automatically when calling from your own phone, otherwise it sends
callers to
the voice mail system to leave a message. Perhaps there's a better way
to do
this as well. If so, please let me know.
Regards,
Paul
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[Asterisk-Users] Can only call asterisk once
I have Asterisk RC2 setup here with a Fritz ISDN card on Debian Woody. I'm using chan_capi-0.3.5 and fcpci-suse8.2-03.11.02. Settings are pretty much the default, except in order to get the fcpci module to compile, I had to follow the instructions here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20AVM%20Fritz%20CAPI%20Driver%20Install but the drivers would only compile if I left the CCFLAGS as they were. Now I've been able to successfully call Asterisk from a POTS phone. We have a block of 10 numbers that are on the ISDN line. Once I call one of those numbers, no more calls will go through after that. I have to restart Asterisk in order for it to answer another call. Has anyone seen this behaviour before? It's certainly not ideal ;) Thanks -- James Doherty Zeald.com Network Operations Ph: +64 9 415 7575, Fax: +64 9 443 9794 Web: http://www.zeald.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] good Dutch TTS ?
hi; anyone can recommend a good TTS for the dutch language compat in linux? -- Best regards, Danny mailto:[EMAIL PROTECTED] belGOnet.com a Euro-pictures division - internet solutions place princesse elisabeth 9/11 - 1030 Brussels - Belgium Tel : +32-(0)2-215.67.65 - Fax : +32-(0)2-215.66.65 domains - hosting - hardware - VoiP - consultancy - backuping CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL No legal consequences can be derived from the contents of the email neither is belGOnet.com committed to them. The content of this email is exclusively intended for adressee(s) and information purposes. belGOnet.com accepts no liability for any damage resulting from the use and/or acceptation of the content of this email. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk codecs and packet size
Does anybody knows if it's posible or if there is some develoment in course to be able to use longer transmit packet sizes (as long as I know this is fixed in 20ms now) with the compressed voip codecs in asterisk (g729, g726, gsm, etc). I need to use asterisk to connect remote sip clients with 24kb bandwidth lines and I'm using a licences g729 codec but because I can't increase the packet size to 40 or 60 ms in asterisk the connection is useless. Thanks very much Luis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap ANSWER the Call
The standard for loop start does not send answer supervision, so * and all
other telcom devices that do CDR records have to 'assume' that the call was
answered.
Lyle
- Original Message -
From: Rodrigo P. Telles [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, August 31, 2004 5:31 PM
Subject: Re: [Asterisk-Users] Zap ANSWER the Call
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Nobody knows about that strange behaviour of Zap channels or
at least if is that right?
Thanks in advance.
Rodrigo P. Telles wrote:
| Hi,
|
| I'm using a TDM400 with one FXS and one FXO module (developer kit) and
| I've been testing termination from SIP phones to PSTN and it works fine,
| but
| asterisk accounting is doing something strange (for me).
| Scenario:
| 1 - extension 1009 (SIP phone - BT101)
| 2 - Zap/4-1 (TDM400 FXO module)
|
| extensions.conf:
| [dialout]
| exten = _9.,1,Dial(Zap/4/${EXTEN:1}|25|r)
| exten = _9.,2,Congestion
|
| [sip]
| include = dialout
| exten = 1009,1,Dial(SIP/1009,20,rt)
|
| So, when I dial 9something from 1009, something rings and then I
| hangup the phone.
| I realised that asterisk thought that something ANSWERED the call:
|
|
,1009,9something,sip,Tests,SIP/1009-42fb,Zap/4-1,Dial,
| Zap/4/something|25,2004-08-27 20:15:34,200
| 4-08-27 20:15:36,2004-08-27 20:15:43,9,7,ANSWERED,BILLING
|
| Is that right?
|
| Version: Asterisk 0.9.0
|
| Thanks in advance.
|
| --
|
| Rodrigo P. Telles [EMAIL PROTECTED]
| Project Manager
| Devel-IT - http://www.devel-it.com.br
| TDKOM Group
|
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- --
Rodrigo P. Telles [EMAIL PROTECTED]
Project Manager
Devel-IT - http://www.devel-it.com.br
TDKOM Group
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TGwaMVemCOPE1uOZQPFKdnc=
=zq37
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[Asterisk-Users] Cisco 79XX SIP Ring Tones
Hi all, Has anyone gotten custom ring tones to work using ALERT_INFO with the Cisco 7940 SIP phone? I've read the wiki, but just can't get this to work. I'm currently using the 7.2 SIP image. Thanks, Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MP3Player strange error
Hi all! I downloaded right mpg123, chabged path to mpg123 binary in app_mp3.c, rebuilt app_mp3.so, and got MusicOnHold to work. But MP3Player refuses to do properly: -- Accepting AUTHENTICATED call from x.x.x.x, requested format = 1024, actual format = 1024 -- Executing Answer(IAX2/[EMAIL PROTECTED]/3, ) in new stack -- Executing MP3Player(IAX2/[EMAIL PROTECTED]/3, mohmp3) in new stack Aug 31 08:07:25 NOTICE[1135618864]: chan_iax2.c:2375 iax2_read: I should never be called! Aug 31 08:07:28 NOTICE[1135618864]: app_mp3.c:91 timed_read: Selected timed out/errored out with 0 -- Timeout on IAX2/[EMAIL PROTECTED]/3 == CDR updated on IAX2/[EMAIL PROTECTED]/3 -- Executing Hangup(IAX2/[EMAIL PROTECTED]/3, ) in new stack == Spawn extension (litnimax-in, t, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/3' -- Executing Hangup(IAX2/[EMAIL PROTECTED]/3, ) in new stack == Spawn extension (litnimax-in, h, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/3' -- Hungup 'IAX2/[EMAIL PROTECTED]/3' new*CLI ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk codecs and packet size
Luis Vazquez [EMAIL PROTECTED] wrote: Does anybody knows if it's posible or if there is some develoment in course to be able to use longer transmit packet sizes (as long as I know this is fixed in 20ms now) with the compressed voip codecs in asterisk (g729, g726, gsm, etc). I need to use asterisk to connect remote sip clients with 24kb bandwidth lines and I'm using a licences g729 codec but because I can't increase the packet size to 40 or 60 ms in asterisk the connection is useless. Thanks very much Luis It wouldn't help you if there was an easy Asterisk patch for this, as the G.729 code is closed source and is therefore un-patchable. You could try SpeeX or LPC10 - or a 56k modem. :-) -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: AW: [Asterisk-Users] SMS Asterisk
On Tue, 31 Aug 2004 15:34:51 +0200, Axel Eble [EMAIL PROTECTED] wrote: On Tue, 31 Aug 2004 15:22:26 +0200, Michael Labuschke [EMAIL PROTECTED] wrote: Pick up mobile phone.. enter sms .. send it to the * phone number. Done On the * side.. follow the sms howto (voip-info.org might have some infos) Done Ah. That requires SMS to be available on land lines. Axel -- Axel Eble, CISSP * Trienter Str. 6b * 87437 Kempten (Allgäu) * Germany VoIP: [EMAIL PROTECTED] * cell: +49.178.285-3265 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users calluk.com gives free 0870 DIDs. I registered my 0870 to FWD account, and FWD passes all to my * box. When I send SMS to my 0870 DID, * shows nothing and I get SMS error. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SMS Asterisk - an explanation
Maxim- This will not work through a FWD DID as you suggest. BT requires each telephone number to be registered in order to receive SMS messages. You need a either an analogue, BRI, or PRI line that terminates in your asterisk box directly. The way a line gets registered is that you must initiate a special SMS message on the asterisk box from the number you are registering (see the details on the asterisk Wiki page, under the SMS command) Once you have registered a number, you can send an SMS text message from a UK mobile to that number, and the asterisk box will receive it, assuming that you've defined SMS handling for that number in your dialplan. IMPORTANT NOTE: As of two weeks ago when I tested this, BT is only accepting SMS's from Vodafone mobiles - O2 and Orange do not work yet. (not sure about T-Mobile) You can send messages to all carriers however. This is expected to change in the next couple months when BT will accept SMS's from all carriers. Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxim Litnitsky Sent: Tuesday, August 31, 2004 5:19 PM calluk.com gives free 0870 DIDs. I registered my 0870 to FWD account, and FWD passes all to my * box. When I send SMS to my 0870 DID, * shows nothing and I get SMS error. ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration with public dynamic ip address
I set up my own STUN server and turned reinvite off. Lyle - Original Message - From: [EMAIL PROTECTED] To: '[EMAIL PROTECTED]' Sent: Tuesday, August 31, 2004 8:53 AM Subject: [Asterisk-Users] SIP registration with public dynamic ip address Hi, I'm trying to configure a natted budgetone phone to a asterisk server as described in wiki using port forwarding. I successfully make call from the client but it seems it does not register the client ip address and when I try to recall it is not reacheable. Asterisk can manage natted sip client with dynamic ip address ? Bye ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Monday, August 30, 2004 12:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration I just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up The volume reset is intentional on Polycom's partt, due to US ADA restrictions (Americans with Disabilities Act). It must reset after each call. This can be overridden through the config files, althrough I can't recall the exact setting right now. Email me off list if you can't find it and I'll look it up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Losing voice on Digium demo server - how to spotproblem ?
try steven sokol's iaxphone and see if you have the same problems dialing his box while taking * out of the equation. same problem=network, no problem = * http://www.sokol-associates.com/IaxPhoneDownload.htm - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 1:51 PM Subject: [Asterisk-Users] Losing voice on Digium demo server - how to spotproblem ? Hi, I'm trying to get Asterisk working on P4 2.8 server behind NAT and Firewall (all ports we're set according to instructions) on DSL line. When pbx connects to Digium demo server( I'm located in Slovenia, Europe), it gets first few words, then silence and then comes back when enumerating dial possibilities (4 for accounting ...). Same happens from SIP or IAX local extension. I guess this is network problem, but would kindly ask for guidance for what measures should I take and what seetings are first to try to avoid this problems. I have another server running at my home on dialup line (28.8kbps) and it connects to digium without problems, so I'm little suspicious being only network traffic problem. Thanks in advance for your effort, regards, Robert. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog lines and TDM card
Correct. TDM (time division multiplex) FXO is for analog ports coming from the telco. - Original Message - From: Marcello Lupo [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 31, 2004 1:52 PM Subject: [Asterisk-Users] Analog lines and TDM card Hi, sorry to bother you, but i need to connect 8 standard analog lines to 2 asterisk servers (one in Italy (4 lines) and one in USA (4 lines)) and after let this 2 systems to interact between them. I was thinking to use the TDM400 card equipped with 4 FXO modules on both sides. Is it correct to do this (use the TDM card to terminate analog lines) or i have to use 4 X100P PCI card in both servers? Thank you in advance. Bye, Marcello ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P Configuration for Mixed Voice Data
I need to know how to setup the data side of the T1 on my Linux Box. I have found information about configuring a PRI and HDLC but nothing about the Frame-Relay type setup for data. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15 is-slave = off DS0 Provisioning: analog-begin = 1 analog-end = 16 data-begin = 17 data-end = 24 alignment = same (The following is what our Vina T1 Integrator currently has in its settings. Our Linux box will replace the Vina) Synchronous Interface: encapsulation = Frame-Relay HDLC-inversion = off Encap-data: LMI-Type = T1.617-annex-D N391 = 6 N392 = 3 N393 = 3 T391 = 6 PVC1 : DLCI = 100 IP = ---.---.---.--- netmask = 255.255.255.252 RIP = disabled NAT = OUT Thank you for the help ahead of time! --Shawn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLC (Packet loss cancel) questions
On Aug 31, 2004, at 8:42 AM, Steve Underwood wrote: Chris Shaw wrote: - Channel Support: IAX2 in asterisk IAX2 in libiax2 Other IP channels in asterisk (RTP-based ones, I guess are all that is left). CNG/VAD and DTX in SIP is a must if * is to be taken seriously as a complete solution... As much as we all hate it's complexity and wish that everything would speak IAX (I know I do) a large number of devices support (and will be supporting) SIP, making it equally as important as IAX2 in using * as a complete telephony solution... This is nothing to do with SIP. It is an RTP issue, common to everything which uses RTP - SIP and H.323 included. Sending no packets is perfectly valid, and normal, in RTP. If the receiving end takes no packets (other than, perhaps, an extremely long silence) as a disconnect it does not comply with the RTP spec. DTX is much despised, and CNG only slightly better. They just sound good (pun intende) on paper. They make a big difference in conferences, actually; In a big conference, you generally have just zero to 2 speakers at any time, and many many calls that are not transmitting. In this case, it is wasteful of all kinds of resources for these clients to transmit anything. Not just bandwidth, but also processing in your mixer. We currently have deployed app_conference (sources with iaxclient), which depends on iaxclient's ability to do DTX transmission in order to be scalable. DTX Support: Sending a single CN packet (in IAX2, this should probably sent reliably) would probably be good. I second, third and fourth this one as does anyone who's tried to use BroadVoice with Voicemail... Currently when * is not making any noise (e.g. recording) absolutely NO packets are sent back to the proxy... A lot of proxies take this as a sign that the far end has disconnected... Including BroadWorks! But they do recognize small CN packets as a sign that the SIP device (Asterisk) is still there... A lot of CNG spec. call for only one transmission, and then silence. Continued CNG has real benefits, but it certainly not the norm. PLC I think is somewhat implemented already in codecs that support it, but I could be wrong, I remember seeing mention of it in the code... PLC is seldom included in the codecs. If you read the specs they often mention PLC, but only in terms of how the codec mitigates the awfulness of a lost packet. Few codecs actually include it. iLBC and speex both include it. Speex' PLC is triggered by simply passing in NULL instead of a pointer to compressed audio. It will then generate a predicted frame without any information: I imagine it basically continues the energy transmission of previous packets. app_conference currently emulates PLC for GSM or by sending the GSM decoder the previous packet (it only does this if it's sucessfully gotten a few packets in a row). This isn't just something that looks good on paper, but something that significantly improves the quality of conversations in practice. (although, in this practice, the lost packets are just as often artifacts of scheduling issues and the present jitter buffer as they are from actual network loss). PLC is also really important when using a dynamic jitter buffer, because whenever the jitter buffer wants to grow, PLC can hide the sparse frames. While the current jitter buffer will just give you a big gap, ideally, if you're growing, you'd be able to spread the gaps a bit, and use PLC to hide them. For example, if you have 100ms worth of frames already queued, and you need to grow the jitterbuffer to 180ms, you'd intersperse the 80ms gaps into 20ms (or whatever) size gaps, and use PLC to interpolate them. This would be much less noticable than an 80ms dropout in audio. -SteveK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()
Tobias Jönsson wrote: Sorry, I did not know these american specialities. I just noticed in Larry's PRI debug info that he received a STATUS message during the waiting, so I thought that the waiting could lead to some kind of timeout at the telco. In EuroISDN the callerid always come in first SETUP message and so it did in Larry's pri debug. The calling number _is_ delivered in the SETUP message; what is not delivered (in National ISDN-2) is the calling name. That comes later in a FACILITY message, and if you Dial() an extension before it has arrived, the destination phone won't see the calling name. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] All you polycom folks.....
Just out of curiosity, What version of CVS and Polycom SIP software are you running happily? Are you running SIP 2.3.0 yet? 2.2.0? 2.1.0? I tried upgrading the CVS yesterday, with a mixed mode of 2.2 and 2.1 with poor results. Transferring did not work as expected. Using the # key to do blind transfers after a call was on hold did not work. Just curious. Thanks, - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] install software version to mediatrix 1204 (how to)
i'm new here and i need help on how where can i get software version 4.0.x of the mediatrix and how can i install it... mediatrix unit im using has a software version of 2.4.9.57. i would like to use H.323 not SIP... please need help asap!... hope to hear from anyone of you soon.. thanks in advance! -- eder ___ Do you Yahoo!? Win 1 of 4,000 free domain names from Yahoo! Enter now. http://promotions.yahoo.com/goldrush ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does anyone have a working GR-303 config?
Chris Jensen wrote: I am hooking up to a DMS500 (100250 together) and wanted to see if anyone had any experience with this. We have the GR-303 span up, the IDT is built. I have not yet heard of anyone doing this, but would be _extremely_ interested in your experiences. Please keep in touch with me off-list, if you would. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration
Look up the word persist in the XML config file... - Brent On Tue, 31 Aug 2004, Reid A. Forrest wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Monday, August 30, 2004 12:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Polycom SoundPoint IP 300 Configuration I just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up The volume reset is intentional on Polycom's partt, due to US ADA restrictions (Americans with Disabilities Act). It must reset after each call. This can be overridden through the config files, althrough I can't recall the exact setting right now. Email me off list if you can't find it and I'll look it up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Headset for Cisco 7960?
On Tue, 31 Aug 2004 15:58:16 -0500, B. J. Bomar [EMAIL PROTECTED] wrote: I use a Plantronics Supra H51 plugged straight into the headset port, and it works great. B. J. Same here. They're wonderful headsets. -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line death not recognized on TDM400P?
A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the number via g1 and that it is connecting sip/bla with zap/1-1 (or some such)... On my X100P I get a red alarm if the phone cable is not plugged in. Is there any way to do this with the TDM400P? They would like to be able to unplug lines and use them for other purposes at times. Make sense? I kinda thought that asterisk would realise that nothing was connected to the TDM card and try the second socket, the third etc... Any help greatly appreciated. Cheers, Matt Riddell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial/Zap doesn't work
Hi, I'm trying to dial in from one phone and give it access to another line (ie incoming on zap/1 and outgoing on zap/2)... how can I transfer the call from channel 1 and give it the dial tone on channel 2? I can use dial but that takes a phone number, which I want the user to be able to select. thanks Imran ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line death not recognized on TDM400P?
On 1 Sep 2004 at 17:15, [EMAIL PROTECTED] wrote: A customer of mine has 3 TDM400P cards in a box running asterisk. On each card he has four FXO modules. I have set up the dialplan to dial via group 1 for an outgoing call. Channels 1-12 are in group 1. If he plugs a telephone cable into socket 2 or 3 etc, but not 1, when he dials out, it still tries to make the call via socket 1. Straight away the console says that it has dialed the number via g1 and that it is connecting sip/bla with zap/1-1 (or some such)... On my X100P I get a red alarm if the phone cable is not plugged in. Is there any way to do this with the TDM400P? They would like to be able to unplug lines and use them for other purposes at times. Make sense? I kinda thought that asterisk would realise that nothing was connected to the TDM card and try the second socket, the third etc... Any help greatly appreciated. The problem is, if lines 1,3,4,5,6,7,8,9,10,11 and 12 are plugged in, you would only be able to make 1 concurrent call...(because the next call would try to go out line two which would never work)...maybe if four people called at the same time only 1 wouldn't get through but then the next call wouldn't get through... Is this so? Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
