Re: [Asterisk-Users] Grandstream Early Dial
On Thu, 2003-12-18 at 04:03, Brian West wrote: Stop using beta firmware... I honestly think that GrandStream needs to either fix the phones or stop making them.. THEY SUCKS! I think I would rather eat glass than work with a grandstream phone. bkw Brian, GS has people that works very hard on this BETA firmwares, and if you have any problem with there phones send them the tcpdump logs, asterisk logs, and explain the bugs your having. THEY are really really open mind to listen up their firmware problems. Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR strange problem with call files
Brian West wrote: Accually CDR will not be generated if the target is an appliction. exten = 1234,1,AGI,outbound.agi|19 then ref the exten not the appliction it will generate a cdr record. http://bugs.digium.com/bug_view_page.php?bug_id=240 Comment by John Todd in bugs added to please note-section on http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Thank you! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to detect process 256 frames
Hi folks, Does anybody have any idea what this is; WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames I see this all over when I make a call from SIP to H323 (chan_h323) in pass-through mode. Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] after hours
When setting include = daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours Any help appreciated Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX quesitons please.
Hello,everyone, I encoutered some difficult with IAX when I run the asterisk. internet -- asterisk + NAT -- DIAX my * box and NAT are at the same linux box which connecting to the internet using ADSL. The box has two network cards and two IP address,such as public IP:211.11.11.11private IP:192.168.1.10 the windows box running DIAX lies behind the NAT,it's IP is 192.168.1.12 the iax.conf like this: .. [marko]type=friendhost=dynamicusername=markosecret=moofoocontext=fromsipauth=plaintext .. the extensions.conf like this: .exten = 45678,1,Dial(IAX2/marko). Question1:I use the DIAX register to the *, if I set the server IP in the DIAX to 192.168.1.10,then it will OK, but if I set the server IP in the DIAX to 211.11.11.11, then I will fail to register. I have tried all version of DIAX and alway got the same result. Why? Question2:If I dial the IAX2 user registed to my * inside my NAT,it will success,but if I dial other IAX2 user registed to my * in the internet (not inside my NAT),I alway get the result: == Everyone is busy at this time Any help will be apprecated. Sorry for my poot English. Regards. franc
Re: [Asterisk-Users] IAX quesitons please.
Hi, - Original Message - From: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAX quesitons please. Hello,everyone, I encoutered some difficult with IAX when I run the asterisk. internet -- asterisk + NAT -- DIAX my * box and NAT are at the same linux box which connecting to the internet using ADSL. The box has two network cards and two IP address,such as public IP:211.11.11.11 private IP:192.168.1.10 the windows box running DIAX lies behind the NAT,it's IP is 192.168.1.12 Question1: I use the DIAX register to the *, if I set the server IP in the DIAX to 192.168.1.10,then it will OK, but if I set the server IP in the DIAX to 211.11.11.11, then I will fail to register. I have tried all version of DIAX and alway got the same result. Why? This is something normal. When you are inside your network use your internal server IP address. Usually the NAT router does not permit to connect to the external IP address from inside. Question2: If I dial the IAX2 user registed to my * inside my NAT,it will success,but if I dial other IAX2 user registed to my * in the internet (not inside my NAT),I alway get the result: == Everyone is busy at this time Take care that there is an issue with DIAX and IAX2... after some time (aprox.1min.) the other part does not ring anymore. I work on this now. Have you registered the external DIAX with the external IP address of the server? BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX quesitons please.
Question2: If I dial the IAX2 user registed to my * inside my NAT,it will success,but if I dial other IAX2 user registed to my * in the internet (not inside my NAT),I alway get the result: == Everyone is busy at this time Take care that there is an issue with DIAX and IAX2... after some time (aprox.1min.) the other part does not ring anymore. I work on this now. Have you registered the external DIAX with the external IP address of the server? Surely. The external DIAX registered with the external IP address of the * server. BR, Dan ___ 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] Re: transfer with threeway calling
But the issue is not: 'how does the alternative feature work', the issue is 'why is the original feature absent'. I haven't heard anyone giving any reason whatsoever why * does not allow a user to retrieve an on-hold call with old-fashioned flashing (or pressing #). I think that is what the debate should focus on, not on whether the customer is right... because is different new. Has new powerful features, and old functions has been abandoned for new ones. So if you wanna such system, you should change your minds (read you as the customer). If you wanna old habits, go for a old stlye pbx, pay more, and forget 'bout other features... or pay $ for them. why microshaft don't let me play old dos games under xp? same reason here. if you (customer) want new things, be prepared to change (a bit) your mind... -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): [EMAIL PROTECTED] - ext 201 Iaxtel: 1-700-56-62458 - ext 201 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 128 kbs satelite link
Mike M. Tkachuk wrote: Hello, I'm using satellite link (1024/256) Eutelsat. With Gnugk and Asterisk. The average roundtrip to my Gateway (DualTalk) is about 650 ms. I think that's fine for non business telephony, just for calling to friends. Hi, thanks for that. Could you give me a phone number to call so I can test it? Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trunk Groups and Multiple Asterisk Machines
Hi!
I have no problems setting up trunk groups in general, but is there a way to
set up a trunk group for outbound calls that includes channels on multiple
servers? I might have missed something somewhere, but I couldn't find any
reading about this topic. Thanks!
What exactly are you trying to do, paging multiple extensions at once
that are hosted in different servers?
If that's not it, why wouldn't this be sufficient for you:
exten = 1000,1,Dial(${Trunk1}/{$Exten})
exten = 1000,2,Dial(${Trunk2}/{$Exten})
exten = 1000,3,Dial(${Trunk3}/{$Exten})
Or how about this if you are trying to arrange some load balancing:
exten = 1000,1,AGI(generate-random-number)
exten = 1000,2,GotoIf($[${RANDOM} = 1]?4:3)
exten = 1000,3,GotoIf($[${RANDOM} = 2]?5:6)
exten = 1000,4,Dial(${Trunk1}/{$Exten})
exten = 1000,5,Dial(${Trunk2}/{$Exten})
exten = 1000,6,Dial(${Trunk3}/{$Exten})
exten = 1000,7,Congestion
Cheers, Philipp
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RE: [Asterisk-Users] 128 kbs satelite link
Hi, I have not incoming phone number to test, but I think I can call you. If I have termination to your country I'll call you (please give me your stationary phone, not mobile). -- Best regards, Mikemailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
Hi! How can I make * ring one phone then if no answer Go to a different extension ?? Read the handbook draft which is to be found on www.asterisk.org. Or read the Wiki and search for the description of the application DIAL. *sigh* Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Probably not hard but I'm just a no0b with *
matt wrote:
The problem is that I don't want to call an extension, I want to call
the number that was specified in the the connection i.e.
[EMAIL PROTECTED] number will be different every time it
is called so I don't want to have to put in an exten for every phone
number in the city I'm trying to call...
Ok, then do something like this:
exten = _9.,1,Dial(OH323/${EXTEN}...)
Michael.
Michael Manousos wrote:
matt wrote:
How do I get * to take an incoming oh323 call and let it dial a number?
I.E. if my boss sets up netmeeting with the gateway as
my.pabxbox.com, whenever he enters a number to dial it always just
dials into the pabx rather than calling that number
The context for the incoming H.323 calls must contain
entries for the extensions you want to handle. The initial extension
of the incoming H.323 calls is just the called number.
i.e. he wants to call 12345 he types it in and presses dial but it
just goes to the message
Setup the extension 12345:
[incoming-h323]
...
exten = 12345,1,Dial(...)
also I have developed an H323 client app which I thought I would dial
the numbers as [EMAIL PROTECTED] where XXX is the number and pabx.com is
the asterisk machine
You can do it. It works.
Peace out,
Matt Riddell
Michael.
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[Asterisk-Users] CAPI Calls Don't Bridge
I had a working configuration whereby an incoming call on an ISDN line would be sent out on the second ISDN line, but since I updated to the latest version of Asterisk I get this error message: WARNING[311315]: File res_parking.c, Line 226 (ast_bridge_call): Bridge failed on channels CAPI[contr1/s]/0 and CAPI[contr1/01624619052]/1 The message comes up as soon as the outgoing call is answered and the call is lost. I have switched to sending the outgoing call using an IAX connection and that does not have the problem. I am also getting this error message (just before the other one), which I don't recall seeing previously: WARNING[311315]: File channel.c, Line 1296 (do_senddigit): Unable to handle DTMF tone 'f' for 'CAPI[contr1/s]/0' Thanks, Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any Ideas
Hi! I need to come up with a solution that the user can place the caller on hold, the caller here MOH and the user hang the receiver up. Just as if they hit, the hold button on the phone. This can be done, using ADSI if need be. What you are trying to do doesn't seem to make much sense. First of all it sounds like you *really* want to do call parking. Secondly, if you hang up, then what are you going to do with the caller? Why not right away hang up on the caller - or do you want to collect phone fees from him while having him listen to MOH indefinitely? :- Here's one way to do it: Create an extension that looks like exten = 333,1,Answer exten = 333,2,MusicOnHold(default) and then use # to transfer the caller to that extension. Unless you use the manager interface (redirect) or some smart scripting/ dialplan layout you won't be able to get back to that caller though. But you didn't say that you need to do that. ;- Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: transfer with threeway calling
Hi! But the issue is not: 'how does the alternative feature work', the issue is 'why is the original feature absent'. I haven't heard anyone giving any reason whatsoever why * does not allow a user to retrieve an on-hold call with old-fashioned flashing (or pressing #). I think that is what the debate should focus on, not on whether the customer is right... If you wanna old habits, go for a old stlye pbx, pay more, and forget 'bout other features... or pay $ for them. Here I don't think that's the best approach - I basically agree with the original posting: It doesn't make much sense to try to disable the FLASH key on any hardware phone (Grandstream or others) only because if the standard user hits that key he'll get himself into trouble where there is no way out... or is there a way to teach the GS Flash key to do anything useful? Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: transfer with threeway calling
Matteo Brancaleoni [EMAIL PROTECTED] said: because is different new. Has new powerful features, and old functions has been abandoned for new ones. Yeah, so much is clear. However, because flash doesn't work at a certain moment *and*, AFAIK, has no other functions at that time, I'm simply wondering what the design constraint here is. Because if there is no design constraint, the old-style behavior could simply be added (should, even, IMO) and everyone would be happy... -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Expressions
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I'm having a problem with the following expression examples.
exten = s,1,NoOp($[$[${value} = 10] $[${value} 18]])
exten = s,1,GotoIf($[$[${value} = 10] $[${value} 18]]?3)
${value} is 13 in both examples above. First extension evaluates to 1 while
second evaluates to 0 even though it's the same expression used in both
examples. Am I doing something wrong?
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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iD8DBQE/4ZjJ2TEAILET3McRAnUIAJwJJTKPDX1iLxcWxdjcqa+b9LvMNACfX5Yj
Cs3L5GjGluSsuBqZhyGn7vs=
=VWAW
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[Asterisk-Users] Re: Expressions - solved
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 18 December 2003 13:08, Tais M. Hansen wrote:
I'm having a problem with the following expression examples.
exten = s,1,NoOp($[$[${value} = 10] $[${value} 18]])
exten = s,1,GotoIf($[$[${value} = 10] $[${value} 18]]?3)
${value} is 13 in both examples above. First extension evaluates to 1 while
second evaluates to 0 even though it's the same expression used in both
examples. Am I doing something wrong?
Answer() apparently changes channels and thus clears variables set prior to
the Answer() call. :(
... Setting variables in spooled calls becomes even more useless.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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uSfwYxsrN5N+UyGgFd18T2Q=
=uU4p
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Re: [Asterisk-Users] Re: transfer with threeway calling
Cees de Groot wrote: Matteo Brancaleoni [EMAIL PROTECTED] said: because is different new. Has new powerful features, and old functions has been abandoned for new ones. Yeah, so much is clear. However, because flash doesn't work at a certain moment *and*, AFAIK, has no other functions at that time, I'm simply wondering what the design constraint here is. Because if there is no design constraint, the old-style behavior could simply be added (should, even, IMO) and everyone would be happy... I agree with Cees, however, not wanting to throw away the 3 way conference feature, but giving the user a config choice might be best. Therefore I'm now testing a patch which will allow/disallow the 3 way conference. When disallowed it will fallback to normal old fashioned PBX behaviour namely FLASH puts caller on hold, FLASH again gets caller back. Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: transfer with threeway calling
Michiel Betel [EMAIL PROTECTED] said: I agree with Cees, however, not wanting to throw away the 3 way conference feature, but giving the user a config choice might be best. Therefore I'm now testing a patch which will allow/disallow the 3 way conference. When disallowed it will fallback to normal old fashioned PBX behaviour namely FLASH puts caller on hold, FLASH again gets caller back. Good. Choice is Good ;-) -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any Ideas
Ok... Let me give a better example. A caller calls in and a user picks up the phone. Then the user needs to put the caller on hold so he can go check on something. He would like to press the hold button on the phone and hang the receiver up. He can do this, but the caller never hears MOH. The user does what he needs to do and comes back and picks up the receiver and press hold to release the caller from hold. I would like this functionality - but for the caller to hear MOH. You mentioned I could do some redirects via the manager interface to get the call back if I just put in out in an extension playing MOH. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Posted At: Thursday, December 18, 2003 6:44 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Any Ideas Subject: Re: [Asterisk-Users] Any Ideas Hi! I need to come up with a solution that the user can place the caller on hold, the caller here MOH and the user hang the receiver up. Just as if they hit, the hold button on the phone. This can be done, using ADSI if need be. What you are trying to do doesn't seem to make much sense. First of all it sounds like you *really* want to do call parking. Secondly, if you hang up, then what are you going to do with the caller? Why not right away hang up on the caller - or do you want to collect phone fees from him while having him listen to MOH indefinitely? :- Here's one way to do it: Create an extension that looks like exten = 333,1,Answer exten = 333,2,MusicOnHold(default) and then use # to transfer the caller to that extension. Unless you use the manager interface (redirect) or some smart scripting/ dialplan layout you won't be able to get back to that caller though. But you didn't say that you need to do that. ;- Cheers, Philipp ___ 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] Asterisk Crash
Asterisk Crash I have an application that using the System() command. When ever I invoke the command my asterisk crashes. I have updated to the latest CVS and it crashes. Can someone offer some suggestions on how to diagnose and correct this problem? Thanks Kevin Extensions.conf exten = 2810,1,System(date) exten = 2810,2,Goodbye ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 128 kbs satelite link
Mike M. Tkachuk wrote: Hi, I have not incoming phone number to test, but I think I can call you. If I have termination to your country I'll call you (please give me your stationary phone, not mobile). Ok, thanks for that. (USA)1-212-400-7921 Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any Ideas
Hi! A caller calls in and a user picks up the phone. Then the user needs to put the caller on hold so he can go check on something. He would like to press the hold button on the phone and hang the receiver up. He can do this, but the caller never hears MOH. The user does what he needs to do and comes back and picks up the receiver and press hold to release the caller from hold. As for SIP and Grandstream (firmware 4.17): It just works the way that you describe (including MOH), however if you hang up then the caller is disconnected. So just don't hang up (but use for example the speakerphone). As for ZAP: Dunno, don't have ZAP hardware. As to the manger interface and redirect: Do some reading on the Wiki and see if that can help you. Or play with astman or gastman. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI and broken pipe
Hi All, I was able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this? Below is a patch that solves the problem. Index: asterisk/apps/app_agi.c === RCS file: /usr/cvsroot/asterisk/apps/app_agi.c,v retrieving revision 1.22 diff -u -r1.22 app_agi.c --- asterisk/apps/app_agi.c 5 Nov 2003 23:43:31 - 1.22 +++ asterisk/apps/app_agi.c 18 Dec 2003 13:48:38 - @@ -167,6 +167,10 @@ /* close what we're not using in the parent */ close(toast[1]); close(fromast[0]); + + // [PHM 12/18/03] + close(audio[0]) + *opid = pid; return 0; ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some questions concerning update
Hi! I'm somewhat unsynced with the features of todays CVS. What's the status of the SQL-support for voicemail? Can you everything (including messages) in the DB? Do you still(?) have to recompile when you change the emailbody-variable in voicemail.conf? Is there anything else I should be thinking of when updating from CVS-09/18/03-16:48:44? Is the .conf-syntax backwards-compatible with my previous build? Thanks! //Filip ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Expressions - solved
At 1:16 PM +0100 12/18/03, Tais M. Hansen wrote:
On Thursday 18 December 2003 13:08, Tais M. Hansen wrote:
I'm having a problem with the following expression examples.
exten = s,1,NoOp($[$[${value} = 10] $[${value} 18]])
exten = s,1,GotoIf($[$[${value} = 10] $[${value} 18]]?3)
${value} is 13 in both examples above. First extension evaluates to 1 while
second evaluates to 0 even though it's the same expression used in both
examples. Am I doing something wrong?
Answer() apparently changes channels and thus clears variables set prior to
the Answer() call. :(
... Setting variables in spooled calls becomes even more useless.
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
If Answer() clears variables, then this is a bug, where bug is
defined as behavior that occurs that a reasonable user or developer
would not expect given the inputs to the process.
If you do not use Answer() explicitly, are the values still cleared?
Can you please document and put in the bug tracker.
JT
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[Asterisk-Users] Zaprtc compile error - virtual device for conferencing
Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe(SIP/-08118800, 1234) in new stack == Parsing '/etc/asterisk/meetme.conf': Found WARNING[245776]: File app_meetme.c, Line 162 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' (language 'en') i read from the lists, that I need to install zaprtc to solve this problem. when i try to compile zaprtc, which i got from http://www.junghanns.net/asterisk/downloads/zaprtc.0.0.1.tar.gz It gives me the following error, zaprtc.c:1077: warning: implicit declaration of function `barrier' zaprtc.c:1078: warning: implicit declaration of function `cpu_relax' zaprtc.c: At top level: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Can anyone please guide me how to compile zaprtc. Thanks in advance. Best Regards, Kannaiyan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trunk Groups and Multiple Asterisk Machines
At 7:44 PM -0500 12/17/03, Sean Cheesman wrote: Hello all, I have no problems setting up trunk groups in general, but is there a way to set up a trunk group for outbound calls that includes channels on multiple servers? I might have missed something somewhere, but I couldn't find any reading about this topic. Thanks! Sean You should be able to do this with TDMoE (TDM over Ethernet) which is a little-used feature of the Zap drivers. There are two ways of doing this: individual trunk groups in a cascading failover situation, which is a well-known configuration with TDMoE (see below for howto) and then perhaps a more interesting way that creates one huge trunk group spanning multiple hosts, which is not documented in the howto but I can't think of why it wouldn't work. I have not actually tried this, but in theory it should work as long as your servers are all on the same ethernet. See Wasim's mini How-to on http://www.convergence.com.pk/TDMoE-HOWTO and then think about it for a while and modify it so that the group spans multiple hosts. Let us know how it goes; I think this might actually be a pretty interesting experiment if it functions as expected. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Telemarketer Torture
On Wed, 10 Dec 2003, Cees de Groot wrote: Chris Albertson [EMAIL PROTECTED] said: My brother has the BEST solution for sales people. He makes an appointment with them to come out and gives an address across the street. It really wastes a real estate salesman or house painter's time to drive out to a dead end. Keeps em off the phone too. I once got Reader's Digest direct mail department off my back by sending them a formal offer to check their mail service - every received mail piece would be reported by me (including a 'quality report' - folded, cracked, ...) and I would invoice only some 50 dollars per mail piece for that. Sending mail would constitute acceptance of the offer - never got a single piece of mail from them again (a pity, I could've been rich ;-)). Wonder whether one could build up a similar construction (the paper one was legally quite watertight, of course) for telemarketeers... Why not re-direct them with their ANI to a 900 number that you own? Announce that they have reached a pay per minute service, and that the first 2 minutes are free of charge, but that subsequent minutes would be charged at a rate of $20 / minute? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream Budgetone * Error
On Tue, 9 Dec 2003, John Breeden wrote: Just started putting my first * together with a tdm400p and x100p. Analog phones, xlite and diax I've got working. Just got Grandstream budgetone-100. The budgetone registers with * just fine. * accepts the dtmf and dials the number. The remote phone rings. From there things go south. The CLI reports this: -- Executing StripMSD(SIP/jrb-683a, 1) in new stack -- Executing Dial(SIP/jrb-683a, Zap/1/9384074) in new stack -- Called 1/9384074 -- Zap/1-1 answered SIP/jrb-683a WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 52852 (Response) -- Hungup 'Zap/1-1' == Spawn extension (home, 9384074, 2) exited non-zero on 'SIP/jrb-683a' The budgetone is using a fixed ip, dtmf signaling, firmware version is 1.0.4,17 My sip.conf for the budgetone is: [jrb] type=friend host=dynamic username=jrb secret=x dtmfmode=rfc2833 context=home reinvite=no canreinvite=no qualify=1000 I can't find a solution to in the archives and I've looked at all the documentation I can find setting up the budgetone on *. Any pointers would be appreciated. Thanx in advance. I had a similar experience and called Digium, who informed me that I have to specifically add a disallow all and then allow ulaw command to my sip.conf for that extension. Ever since then, it has worked like a champ. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: transfer with threeway calling
- Original Message - From: Cees de Groot [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 18, 2003 7:58 AM Subject: [Asterisk-Users] Re: transfer with threeway calling Michiel Betel [EMAIL PROTECTED] said: I agree with Cees, however, not wanting to throw away the 3 way conference feature, but giving the user a config choice might be best. Therefore I'm now testing a patch which will allow/disallow the 3 way conference. When disallowed it will fallback to normal old fashioned PBX behaviour namely FLASH puts caller on hold, FLASH again gets caller back. Good. Choice is Good ;-) -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Shouldn't this be an feature of the parking app? Something like: parkext =700 parkhold =701 700, Normal Park 701, Personal Park/EasyRetreive I transfer to 701. * says: Next Available hold is 1 I dial 1. * parks caller and hangs up on me. I hang up. time passes I dial 701. * says: Next Available hold is 2 (or * knows this isn't a transfer, and reads off the list of holds in use, possibly including callerid) I dial 1. * puts the call back together, or hangs up and rings that line back to me. Criticism and further discussion appreciated. No flames please, this is just an idea. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] after hours
On Thu, 2003-12-18 at 02:26, [EMAIL PROTECTED] wrote: When setting include = daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours This is where it is important for you to flex your mind while reading what was given to you and whatever documentation it leads you to. It will benefit you a great deal, and the community a bit. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Expressions
- Original Message -
From: Tais M. Hansen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 18, 2003 7:08 AM
Subject: [Asterisk-Users] Expressions
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I'm having a problem with the following expression examples.
exten = s,1,NoOp($[$[${value} = 10] $[${value} 18]])
exten = s,1,GotoIf($[$[${value} = 10] $[${value} 18]]?3)
${value} is 13 in both examples above. First extension evaluates to 1
while
second evaluates to 0 even though it's the same expression used in both
examples. Am I doing something wrong?
Your example sort of confuses me. How do you know what NoOp returns?
Even though NoOp could theoretically take arguments, the point (IMHO) of
NoOp is to do nothing. If it did return something, why wouldn't it always
return the same thing?
Also, doesn't the ?3 in thesecond expression change things? Or is that
outside the scope of what you're asking?
-
Andrew Thompson http://aktzero.com/
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.
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Re: [Asterisk-Users] Zaprtc compile error - virtual device for conferencing
- Original Message - From: Kannaiyan Natesan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 18, 2003 9:52 AM Subject: [Asterisk-Users] Zaprtc compile error - virtual device for conferencing Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe(SIP/-08118800, 1234) in new stack == Parsing '/etc/asterisk/meetme.conf': Found WARNING[245776]: File app_meetme.c, Line 162 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' (language 'en') i read from the lists, that I need to install zaprtc to solve this problem. when i try to compile zaprtc, which i got from http://www.junghanns.net/asterisk/downloads/zaprtc.0.0.1.tar.gz It gives me the following error, zaprtc.c:1077: warning: implicit declaration of function `barrier' zaprtc.c:1078: warning: implicit declaration of function `cpu_relax' zaprtc.c: At top level: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Can anyone please guide me how to compile zaprtc. Did you modprobe ztdummy? It should return nothing(successfully). Confirm that it is loaded with lsmod. You'll also need to put that line in an init script so that it gets loaded into memory again when you reboot. ztdummy and zaprtc serve the same purpose, but go about it different ways. They both provide a timing source. ztdummy uses info from the USB bus. zaprtc uses the realtime clock of your pc. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 128 kbs satelite link
At 10:46 AM -0800 12/17/03, Paul Mahler wrote: While this thread is already in the archives, I'll throw my opinion on the table, too. The latency is about .25 seconds to and .25 seconds from the satellite. There is additional propagation delay in the system. Also, TCP/IP relies on propagation delay to optimize traffic, but doesn't handle delays this long well. There is special purpose hardware available the sends the traffic to an from the satellite with a different protocol. This helps the satellite calls quite a bit. http://www.mentat.com/skyx/skyx-whitepaper.html John Todd reamed me out the last time I suggested that you need this hardware, he says he has voice working fine over a satellite link. And he knows what he's talking about. I thought I would save him some time, this time, by putting this in. ;-) Again, for the archives, I'll comment on why this hardware/software probably isn't useful for VoIP. :-) All of the satellite accelerators are based on TCP spoofing, which ACK's TCP packets before the other side actually sends the ACK. This keeps things flowing faster because it breaks the way TCP functions. Since VoIP is almost always UDP, this doesn't do any good. Plus, since VoIP is real-time human-perceived communication, you can't make it any faster than it currently is - the speed of radio waves in a vacuum (and a short duration in the slower atmosphere) is fixed. Geosync satellites are ~22,000 miles away - there is no speeding up that process or predicting what is being said before it's actually said. I have used SIP RTP over satellite, and while the delay is uncomfortable it is workable if that is the only method to get to the endpoint at a reasonable cost. As previously noted, discomfort is a function of cost; the more expensive the transport, the less discomfort one is willing to suffer through. Quality of sound in UDP-based VoIP services are 100% a function of available bandwidth, and not a function of delay, though with IAX2 I can't say if that's the case or not since I am unaware of how it handles unreasonable duration between frames of send/receive. The trunking features of IAX2 may help tremendously in saving transmission bandwidth, though - the removal of the IP overhead from subsequent sessions (after the first one) is a huge savings over SIP or H.323, so you should be able to get much more out of your very expensive bitstream than you would ordinarily be able to get with other VoIP protocols, so from that perspective (to the original poster or anyone else with IAX2 over satellite transport) please let us know how you make out with testing. PS: To answer the original question, the _theoretical_ maximum for channels through 128kbps is 18 calls with LPC10, based on my codec comparisons (see the archives) with IAX2. LPC10 sucks, and adding a satellite delay would make it infuriating for anyone except the most hardened cheapskate. A more reasonable codec would be G.729, at a theoretical maximum of 11 calls, but you'd probably really only get 8 or 9 without starting to get loss or buffer issues. JT There is, of course, also the problem of the upload speed. The consumer grade satellites don't offer much upstream bandwidth. You can get better upload speed from some of the commercial satellite carriers. http://www.tachyon.net/ Paul Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805) Sent: Wednesday, December 17, 2003 7:06 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 128 kbs satelite link Similar to online gaming, I would think that the propagation delay with the satelite connection would make calls unbearable. Half-duplex at its worst. my $0.02 -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Senad Jordanovic Sent: Wednesday, December 17, 2003 9:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] 128 kbs satelite link Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? In particular I am interested to hear how many calls could be put through 128Kbs satelite link simultaneously? Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI and broken pipe
On Thu, 18 Dec 2003 11:48:59 -0300 Paulo Mannheimer [EMAIL PROTECTED] wrote: Hi All, I was able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this? Below is a patch that solves the problem. Thanks Paulo, I've patched the app_agi.c and now asterisk with EAGI applications is not leaking pipes anymore :-) Angel Index: asterisk/apps/app_agi.c === RCS file: /usr/cvsroot/asterisk/apps/app_agi.c,v retrieving revision 1.22 diff -u -r1.22 app_agi.c --- asterisk/apps/app_agi.c 5 Nov 2003 23:43:31 - 1.22 +++ asterisk/apps/app_agi.c 18 Dec 2003 13:48:38 - @@ -167,6 +167,10 @@ /* close what we're not using in the parent */ close(toast[1]); close(fromast[0]); + + // [PHM 12/18/03] + close(audio[0]) + *opid = pid; return 0; ___ 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] Where is D channel in a PRI link?
We have contracted with Eschelon to provide voice and data over a T1 link. The plan is to terminate this link at a T100P card in the * system. The vendor has said that they will provide the D channel contiguous to the voice channels (voice on channels 1-8 and D channel on 9). The data channels would be 20-24. Will the T100P be able to accept this configuration? Does the PRI specification mandate where the D-channel should be? Thanks for your help. Michael Welter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Asterisk problem
I have then downloaded newer version of the Asterisk about 10 days ago,
but
then my Asterisk would start to crash on me, the module would just stop
running by itself and I had to restart the Asterisk. Sometimes, it would
just stop running, but 75% of the time, I would see this error message:
Connected to Asterisk CVS-12/06/03-03:06:28 currently running on localhost
(pid = 23720)
-- Remote UNIX connection
localhost*CLI /usr/sbin/safe_asterisk: line 6: 23720 Segmentation fault
asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
I have it core dump on me with latest cvs,
at chan_h323.c:1164-connection_made - if (!p-owner)
I beleive this is a race condition on the the channel, since the struc 'p'
is valid
but the channel appears to have vanished right after 'find_call'
do a back trace on the core file when it dumps, compare is to this
#0 connection_made (call_reference=1153301536) at chan_h323.c:1164
#1 0x44bd8c23 in
MyH323EndPoint::OnConnectionEstablished(H323Connection, PString const)
(this=0x8164878,
[EMAIL PROTECTED], [EMAIL PROTECTED]) at ast_h323.cpp:360
#2 0x4544bab1 in H323Connection::OnEstablished() () from
/usr/src/openh323/lib/libh323_linux_x86_r.so.1.12.2
#3 0x45453629 in H323Connection::InternalEstablishedConnectionCheck() ()
from /usr/src/openh323/lib/libh323_linux_x86_r.so.1.12.2
#4 0x4544af0a in H323Connection::HandleSignalPDU(H323SignalPDU) ()
from /usr/src/openh323/lib/libh323_linux_x86_r.so.1.12.2
#5 0x4544aba2 in H323Connection::HandleSignallingChannel() () from
/usr/src/openh323/lib/libh323_linux_x86_r.so.1.12.2
#6 0x45457f45 in H225CallThread::Main() () from
/usr/src/openh323/lib/libh323_linux_x86_r.so.1.12.2
#7 0x44dfb3c4 in PThread::PX_ThreadStart(void*) () from
/usr/src/pwlib/lib/libpt_linux_x86_r.so.1.5.2
#8 0x400279e1 in pthread_start_thread () from /lib/i686/libpthread.so.0
void connection_made(unsigned call_reference)
{
struct ast_channel *c = NULL;
struct oh323_pvt *p = NULL;
p = find_call(call_reference);
if (!p)
ast_log(LOG_ERROR, Something is wrong: connection\n);
if (!p-owner) {
printf(Channel has no owner\n);
return;
}
c = p-owner;
ast_setstate(c, AST_STATE_UP);
return;
}
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Re: [Asterisk-Users] Re: Telemarketer Torture
- Original Message - From: Greg Boehnlein [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 18, 2003 10:01 AM Subject: Re: [Asterisk-Users] Re: Telemarketer Torture On Wed, 10 Dec 2003, Cees de Groot wrote: Chris Albertson [EMAIL PROTECTED] said: My brother has the BEST solution for sales people. He makes an appointment with them to come out and gives an address across the street. It really wastes a real estate salesman or house painter's time to drive out to a dead end. Keeps em off the phone too. I once got Reader's Digest direct mail department off my back by sending them a formal offer to check their mail service - every received mail piece would be reported by me (including a 'quality report' - folded, cracked, ...) and I would invoice only some 50 dollars per mail piece for that. Sending mail would constitute acceptance of the offer - never got a single piece of mail from them again (a pity, I could've been rich ;-)). Wonder whether one could build up a similar construction (the paper one was legally quite watertight, of course) for telemarketeers... Why not re-direct them with their ANI to a 900 number that you own? Announce that they have reached a pay per minute service, and that the first 2 minutes are free of charge, but that subsequent minutes would be charged at a rate of $20 / minute? While an exceptionally devious concept, I don't think it'd work out like you planned. Wouldn't that mean you'd have to dial out the 900 number yourself, meaning You would be charged for the 900 call. Instead, I'd have someone record a similar message states they will be required to pay to complete the telephone call, but that the charges may be waived. Or you could just give them the Number has Changed to message with your previously mentioned 900 number. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Zaprtc compile error - virtual device for Conferencing
Thanks for the reply but it could not solve my problem. Did you modprobe ztdummy? modprobe ztdummy modprobe: Can't open dependencies file /lib/modules/2.4.20-6um/modules.dep (No such file or directory) Can you please guide me what should I do for this? It should return nothing(successfully). Confirm that it is loaded with lsmod. lsmod Module Size Used byNot tainted lsmod: QM_MODULES: Function not implemented You'll also need to put that line in an init script so that it gets loaded into memory again when you reboot. Can you please add a line how can I add that. ztdummy uses info from the USB bus. I'm not sure USB bus available on the system. Since I could not modprob ztdummy zaprtc uses the realtime clock of your pc. I want to setup my conference bridge running. Kannaiyan Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe(SIP/-08118800, 1234) in new stack == Parsing '/etc/asterisk/meetme.conf': Found WARNING[245776]: File app_meetme.c, Line 162 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' (language 'en') i read from the lists, that I need to install zaprtc to solve this problem. when i try to compile zaprtc, which i got from http://www.junghanns.net/asterisk/downloads/zaprtc.0.0.1.tar.gz It gives me the following error, zaprtc.c:1077: warning: implicit declaration of function `barrier' zaprtc.c:1078: warning: implicit declaration of function `cpu_relax' zaprtc.c: At top level: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Can anyone please guide me how to compile zaprtc. Did you modprobe ztdummy? It should return nothing(successfully). Confirm that it is loaded with lsmod. You'll also need to put that line in an init script so that it gets loaded into memory again when you reboot. ztdummy and zaprtc serve the same purpose, but go about it different ways. They both provide a timing source. ztdummy uses info from the USB bus. zaprtc uses the realtime clock of your pc. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring DG-104s
On Thu, 18 Dec 2003 14:10:08 +0200, Anton Yurchenko [EMAIL PROTECTED] wrote: Hello, I asked on the asterisk mailing list about dlink DG-104SH, some people wrote that they had DG-104S working, so I kicked that 104SH , and got an 104S. And now I`m having trouble configuring it( I`m kinda new to MGCP) What do I put in the Config Call agent IP section? I now have it like this: Notify Entity RGW Name DNS IP . . . DNS State SDP IP address for NAT . . . Make sure that Notify Entity is [EMAIL PROTECTED]:2427 (172.20.0.50 is your asterisk box, right? Make RGW Name DG104S and in the mgcp.conf i have: [general] port = 2427 bindaddr = 0.0.0.0 [172.20.0.98] Make this [DG104S] Let me know if this helps ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Expressions
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 18 December 2003 16:12, Andrew Thompson wrote: Your example sort of confuses me. How do you know what NoOp returns? It was two examples. I use NoOp() everytime I'm in doubt about the contents of a specific variable or expression. Even though NoOp could theoretically take arguments, the point (IMHO) of NoOp is to do nothing. If it did return something, why wouldn't it always return the same thing? It's not supposed to. It's for informative reasons only. Also, doesn't the ?3 in thesecond expression change things? Or is that outside the scope of what you're asking? It says, if true goto priority 3. Outside the scope. :) - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/4dDN2TEAILET3McRAv+lAKCLbgaXgKLMC1Kg4H0AFchRm1eg/ACgjaj6 ATfKbiJg+U5poBENGR9nk6o= =PfTJ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] International calling forbidden?
Please forgive me if the answer is obvious, but my new Asterisk server
gives back a forbidden message when I try to call my UK office. It
should go out simply via X100p and PTSN. Here's the relevant lines from
extensions.conf.
[outbound-analog-int'l]
; allowed to call interntional long distance numbers via PSTN
; dial 8 to signify overseas calling
exten = _8011,1,Dial(${PSTNOUTBOUND/${EXTEN},70)
exten = _8011,2,Macro(fastbusy)
The number I'm calling is 011 44 1223 721 000. What am I doing wrong?
Thanks,
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]
FWD 54245
Can't you see it all makes perfect sense, expressed in dollars and cents,
pounds, shillings and pence - Roger Waters
** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704
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[Asterisk-Users] /var/spool/asterisk/outgoing -- call joining ?/
I want to join two calls invoked from asterisk, Here is my 1.call in /var/spool/asterisk/outgoing, Channel: IAX2/[EMAIL PROTECTED]/847512,20,tr MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: 13732 Extension: s Priority: 1 it successfully rings at extension 847512 and I could answer the call. In my context, ( extensions.conf ) [13732] exten = s,1,Dial(SIP/[EMAIL PROTECTED]) I want to join both calls. But it hangs up when the first call is made. It is not dialling the s,1 extension. I even tried with Answer App without Answer App. Can anyone please guide me how can i join both the calls. Thanks, Kannaiyan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Residential router w/ QoS support?
I use a Linksys BEFSR81 which ia an 8 port model with QoS. I paid about $90 USD. I had to buy a QoS router when I first installed a Vonage line about a year ago. Without it using FTP to d/l loarge files would simply kill my calling. Michael On Wed, 17 Dec 2003 17:01:52 +0100, Thilo Salmon wrote: Did anybody ever come across an affordable, residential cable/dsl router with support for QoS? The ones I've seen so far (Netgear, D-Link and W-Linx) do not seem to support it. I noticed that even email can damage a G.711 stream on an 128kbit uplink, leave alone file-sharing applications. I understand this is strictly related to *, but nevertheless of interest to many of us. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 Philosophers and plowmen, each must know their part to sow a new mentality, closer to the heart. - Geddy Lee, Rush ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where is D channel in a PRI link?
It doesn't matter for the zaptel (since you can set dchan=any_channel) but in chan_zap.c in asterisk dchannel for t1 cards is hardcoded to by on 24th channel. You can change that though. regards Martin On Thu, 18 Dec 2003, Michael Welter wrote: We have contracted with Eschelon to provide voice and data over a T1 link. The plan is to terminate this link at a T100P card in the * system. The vendor has said that they will provide the D channel contiguous to the voice channels (voice on channels 1-8 and D channel on 9). The data channels would be 20-24. Will the T100P be able to accept this configuration? Does the PRI specification mandate where the D-channel should be? Thanks for your help. Michael Welter ___ 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] Re: Expressions - solved
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 18 December 2003 15:43, John Todd wrote:
Answer() apparently changes channels and thus clears variables set prior
to the Answer() call. :(
If Answer() clears variables, then this is a bug, where bug is
defined as behavior that occurs that a reasonable user or developer
would not expect given the inputs to the process.
If you do not use Answer() explicitly, are the values still cleared?
Can you please document and put in the bug tracker.
A first it seemed like the variable was set until Answer() was called but it
is not the case. More testing revealed the vars gets cleared when Asterisk
bridges the channel and chosen extension set in the call spool file.
-- Executing NoOp(Local/[EMAIL PROTECTED],1, id = 36) in new stack
-- Executing Answer(Local/[EMAIL PROTECTED],1, ) in new stack
-- Executing NoOp(Local/[EMAIL PROTECTED],1, id = 36) in new stack
-- Executing Wait(Local/[EMAIL PROTECTED],1, 1) in new stack
== Spawn extension (macro-dialprovider, s, 5) exited non-zero on
'Local/[EMAIL PROTECTED],2' in macro 'dialprovider'
== Spawn extension (default, 1234, 2) exited non-zero on
'Local/[EMAIL PROTECTED],2'
-- Executing NoOp(IAX2[x.x.x.x:4569]/1, id = ) in new stack
A matching segment of the dialplan:
exten = s, 2,NoOp(id = ${id})
exten = s, 3,Answer()
exten = s, 4,NoOp(id = ${id})
exten = s, 5,Wait(1)
exten = s, 6,NoOp(id = ${id})
- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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[Asterisk-Users] Polycom phones update
Hello, We have updated the Wiki page for Polycom phones: http://www.voip-info.org/tiki-index.php?page=Polycom+Phones We posted several configuration specs as well as a link to an admin guide for the phone. We also posted a link on there to two firmware versions for download. The official Asterisk-Polycom support website should be up and live sometime in January. If anyone has anything to add or would like to be on our list of experts for Asterisk-Polycom when it goes live, please let me know. Enjoy, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] International calling forbidden?
- Original Message -
From: Michael Graves [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 18, 2003 11:06 AM
Subject: [Asterisk-Users] International calling forbidden?
Please forgive me if the answer is obvious, but my new Asterisk server
gives back a forbidden message when I try to call my UK office. It
should go out simply via X100p and PTSN. Here's the relevant lines from
extensions.conf.
[outbound-analog-int'l]
; allowed to call interntional long distance numbers via PSTN
; dial 8 to signify overseas calling
exten = _8011,1,Dial(${PSTNOUTBOUND/${EXTEN},70)
exten = _8011,2,Macro(fastbusy)
The number I'm calling is 011 44 1223 721 000. What am I doing wrong?
Are you doing ignorepat somewhere?
If not, you'll need to chop off the 8 at the beginning.
Also, since I'm not sure...
Does * pick up the line and dial, then fail? Or does it fail before it picks
up the line?
Do you have X100 -- PSTN working for non-UK calls? If so, try your entry
into the other context and see if it fails there as well. (Maybe something
leading up to this particular choice is hosed?
-
Andrew Thompson http://aktzero.com/
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The opinions
stated above are yours. You cannot imagine why you ever felt otherwise.
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Re: [Asterisk-Users] International calling forbidden?
Duh,
I finally go it! Missing } afterter PSTNOUTBOUND.
Michael
On Thu, 18 Dec 2003 10:06:23 -0600, Michael Graves wrote:
Please forgive me if the answer is obvious, but my new Asterisk server
gives back a forbidden message when I try to call my UK office. It
should go out simply via X100p and PTSN. Here's the relevant lines from
extensions.conf.
[outbound-analog-int'l]
; allowed to call interntional long distance numbers via PSTN
; dial 8 to signify overseas calling
exten = _8011,1,Dial(${PSTNOUTBOUND/${EXTEN},70)
exten = _8011,2,Macro(fastbusy)
The number I'm calling is 011 44 1223 721 000. What am I doing wrong?
Thanks,
Michael
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]
FWD 54245
Can't you see it all makes perfect sense, expressed in dollars and cents,
pounds, shillings and pence - Roger Waters
** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704
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--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]
FWD 54245
I used to be snow white, but I drifted - Mae West
** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704
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RE: [Asterisk-Users] fedora core 1 install problem
I used root user but I do not understand what you mean running it by prompt or screen. These are the packages I installed. Do you know if some are missing? David [EMAIL PROTECTED] root]# rpm -q kernel-source readline readline-devel openssl openssl-devel bison cvs gcc newt-devel ncurses-devel libtermcap-devel zlib zlib-devel kernel-source-2.4.22-1.2115.nptl readline-4.3-7 readline-devel-4.3-7 openssl-0.9.7a-23 openssl-devel-0.9.7a-23 bison-1.875-5 cvs-1.11.5-3 gcc-3.3.2-1 newt-devel-0.51.6-1 ncurses-devel-5.3-9 libtermcap-devel-2.0.8-36 zlib-1.2.0.7-2 zlib-devel-1.2.0.7-2 [EMAIL PROTECTED] root]# -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens matt Verzonden: woensdag 17 december 2003 23:30 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] fedora core 1 install problem Strange... I just did one yesterday on Fedora Core 1 and eventually got everything sweet... what user did you use? did you run it from screen or a prompt? did you install the pre-requisites and/or check for them with rpm-q package name? i.e. rpm - q kernel-source readline readline-devel openssl openssl-devel (according to the piece of paper digium sent me with the kit - although some people on this list have said that readline is no longer neccesary with the latests cvs of asterisk) Kind regards, Matt Riddell David Luyens wrote: Hi, I am trying ti install an asterisk system on fedora core 1. During the make of asterisk I got the folowing problem: bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: *** [ast_expr.c] Broken pipe Does anybody know how to solve this? David \/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o tdd.o tdd.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o acl.o acl.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o rtp.o rtp.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o manager.o manager.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o asterisk.o asterisk.c bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: *** [ast_expr.c] Broken pipe [EMAIL PROTECTED] asterisk]# ___ 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
Re: [Asterisk-Users] International calling forbidden?
Hello,
On Thu, 2003-12-18 at 13:06, Michael Graves wrote:
[outbound-analog-int'l]
; allowed to call interntional long distance numbers via PSTN
; dial 8 to signify overseas calling
exten = _8011,1,Dial(${PSTNOUTBOUND/${EXTEN},70)
exten = _8011,2,Macro(fastbusy)
The number I'm calling is 011 44 1223 721 000. What am I doing wrong?
The number you are calling (011 44 xxx) does not match the dialplan. You
have to remove the 8:
exten = _011,1,Dial(${PSTNOUTBOUND/${EXTEN},70)
exten = _011,2,Macro(fastbusy)
If you want to 8 signify overseas, as the comentary line says,
you should dial 8 before 011, and remove one digit from the extension,
in order to not send that 8 to the PSTN.
exten = _8011,1,Dial(${PSTNOUTBOUND/${EXTEN:1},70)
exten = _8011,2,Macro(fastbusy)
All of this will work if you are including this context in
the proper place. Best regards,
--
Nicolas Gudino [EMAIL PROTECTED]
House Internet S.R.L.
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Re: [Asterisk-Users] Unable to detect process 256 frames
SW wrote: Hi folks, Does anybody have any idea what this is; WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames Do not try to do inband DTMF on G.729 Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Crash
On Thu, 2003-12-18 at 07:25, Kevin wrote: Asterisk Crash I have an application that using the System() command. When ever I invoke the command my asterisk crashes. I have updated to the latest CVS and it crashes. Can someone offer some suggestions on how to diagnose and correct this problem? Thanks Kevin Extensions.conf exten = 2810,1,System(date) exten = 2810,2,Goodbye repeating your question with no additional information will likely have it ignored. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI and broken pipe
Great ;-) Can someone else confirm this doesn't have any side effects besides solving the problem? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angel Carpintero Sent: quinta-feira, 18 de dezembro de 2003 12:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] AGI and broken pipe On Thu, 18 Dec 2003 11:48:59 -0300 Paulo Mannheimer [EMAIL PROTECTED] wrote: Hi All, I was able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this? Below is a patch that solves the problem. Thanks Paulo, I've patched the app_agi.c and now asterisk with EAGI applications is not leaking pipes anymore :-) Angel Index: asterisk/apps/app_agi.c === RCS file: /usr/cvsroot/asterisk/apps/app_agi.c,v retrieving revision 1.22 diff -u -r1.22 app_agi.c --- asterisk/apps/app_agi.c 5 Nov 2003 23:43:31 - 1.22 +++ asterisk/apps/app_agi.c 18 Dec 2003 13:48:38 - @@ -167,6 +167,10 @@ /* close what we're not using in the parent */ close(toast[1]); close(fromast[0]); + + // [PHM 12/18/03] + close(audio[0]) + *opid = pid; return 0; ___ 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
[Asterisk-Users] G729 question
I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this answer. I know that G729A is low complexity which seems to be what Cisco 7960's use but I have some others that support G729B which has comfort noise and reduced transmission during silence. If anyone knows how the different G729 codecs interoperate I would be eager to know. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Different Dial tones for internal and external.
On systems even key systems it is customary to have an internal
dial tone.
Since Asterisk simply ignores the 9 and keeps the tone going
it is hard to tell for some new users if they can make a call.
My first idea was to change the generated dial tone via
source. Then if the user presses 9 go to a different context where I would record
about 30 seconds of the normal dial tone and then let them enter the numbers to
dial. Something it this:
[internal]
exten = 123,1,macro-stdexten(blah,blah,blah)
exten = 124,1,macro-stdexten(blah,blah,blah)
exten = 125,1,macro-stdexten(blah,blah,blah)
exten = 9,1,Goto(trunkgroup,s,1)
[trunkgroup]
exten = s,1,DigitTimeout,5
exten = s,2,ResponseTimeout,10
exten = s,3,Playback(bell-dialtone)
exten = _X.,1,Dial(Zap/g2/${EXTEN})
It Works but there HAS to be a better way!!!
Maybe instead of ignorepat a changetonepat in the context.
How do others do this or am I the first
Re: [Asterisk-Users] G729 question
It doesn't matter ... A B are compatible On Thu, 18 Dec 2003, Clif Jones wrote: I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this answer. I know that G729A is low complexity which seems to be what Cisco 7960's use but I have some others that support G729B which has comfort noise and reduced transmission during silence. If anyone knows how the different G729 codecs interoperate I would be eager to know. Thanks. ___ 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] another
As far as I understand it, daytime is a context? so you just use like [daytime] s,1,blahblah etc [weekend] s,1,blahblahweekend etc [EMAIL PROTECTED] wrote: Matt I understand that bit but How do I express the sound file for after that time period ?? Here is what I need to do include = daytime|9:00-21:00|mo-fri|*|* include = weekend|10:00-19:00|sat-sun|*|* I think the above is correct ?? Bit how do I specify the after hours config ??? Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of matt Sent: Thursday, 18 December 2003 2:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] another [EMAIL PROTECTED] wrote: Hi again How do I change the message played on initial pickup for after hours ?? Thanks in advance Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users In the context put: include = daytime|9:00-17:00|mo-fri|*|* which will include the daytime context during these hours ___ 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] Different Dial tones for internal and external.
On Thursday 18 December 2003 13:31, Alex Lopez wrote:
On systems even key systems it is customary to have an 'internal'
dial tone.
Since Asterisk simply ignores the 9 and keeps the tone going it is
hard to tell for some 'new users' if they can make a call.
My first idea was to change the generated dial tone via source.
Then if the user presses 9 go to a different context where I would
record about 30 seconds of the normal dial tone and then let them
enter the numbers to dial. Something it this:
[internal]
exten = 123,1,macro-stdexten(blah,blah,blah)
exten = 124,1,macro-stdexten(blah,blah,blah)
exten = 125,1,macro-stdexten(blah,blah,blah)
exten = 9,1,Goto(trunkgroup,s,1)
[trunkgroup]
exten = s,1,DigitTimeout,5
exten = s,2,ResponseTimeout,10
exten = s,3,Playback(bell-dialtone)
exten = _X.,1,Dial(Zap/g2/${EXTEN})
It Works but there HAS to be a better way!!!
Maybe instead of ignorepat a changetonepat in the context.
How do others do this or am I the first
Here's how to do it:
In the zaptel driver, create a new tone in zonedata.c at the END of
a zone (so you don't throw off existing tone indexes). Then, in
asterisk, in the specific channel driver (e.g. chan_zap.c), locate
ast_ignore_pattern and change ZT_TONE_DIALTONE to the
index of your new tone. You can probably use ZT_TONE_CUST1
as your index.
Recompile, reinstall, restart (both the zaptel driver, as well as
asterisk).
-Tilghman
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Re: [Asterisk-Users] Re: Zaprtc compile error - virtual device for Conferencing
ztdummy isn't compiled by default.. you have to take the # from infront of it in the Makefile. But then again it only works with usb-uhci and not usb-ohci. Buy an x100p and call it a day. bkw On Thu, 18 Dec 2003, Kannaiyan Natesan wrote: Thanks for the reply but it could not solve my problem. Did you modprobe ztdummy? modprobe ztdummy modprobe: Can't open dependencies file /lib/modules/2.4.20-6um/modules.dep (No such file or directory) Can you please guide me what should I do for this? It should return nothing(successfully). Confirm that it is loaded with lsmod. lsmod Module Size Used byNot tainted lsmod: QM_MODULES: Function not implemented You'll also need to put that line in an init script so that it gets loaded into memory again when you reboot. Can you please add a line how can I add that. ztdummy uses info from the USB bus. I'm not sure USB bus available on the system. Since I could not modprob ztdummy zaprtc uses the realtime clock of your pc. I want to setup my conference bridge running. Kannaiyan Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe(SIP/-08118800, 1234) in new stack == Parsing '/etc/asterisk/meetme.conf': Found WARNING[245776]: File app_meetme.c, Line 162 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' (language 'en') i read from the lists, that I need to install zaprtc to solve this problem. when i try to compile zaprtc, which i got from http://www.junghanns.net/asterisk/downloads/zaprtc.0.0.1.tar.gz It gives me the following error, zaprtc.c:1077: warning: implicit declaration of function `barrier' zaprtc.c:1078: warning: implicit declaration of function `cpu_relax' zaprtc.c: At top level: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: *** [zaprtc.o] Error 1 Can anyone please guide me how to compile zaprtc. Did you modprobe ztdummy? It should return nothing(successfully). Confirm that it is loaded with lsmod. You'll also need to put that line in an init script so that it gets loaded into memory again when you reboot. ztdummy and zaprtc serve the same purpose, but go about it different ways. They both provide a timing source. ztdummy uses info from the USB bus. zaprtc uses the realtime clock of your pc. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest ___ 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] Interesting problem
I have three cisco 7910 phones connected to * through skinny protocol. When one of the phones is called, and the phone is ringing, you can hear what's going on in the room even though the caller hasn't answered. It's crazy and very hard to ignore when someone is calling :) God forbid you should cough while the phone is ringing. C. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom phones update
On Thu, 2003-12-18 at 13:30, mattf wrote: Hello, We have updated the Wiki page for Polycom phones: http://www.voip-info.org/tiki-index.php?page=Polycom+Phones We posted several configuration specs as well as a link to an admin guide for the phone. We also posted a link on there to two firmware versions for download. What I missed is the Release Notes to compare the changes between 2.3.0 and 2.4.1. At least in both firmwares if the prefered codec is G.729 and the prefered codec in sip.conf of * is GSM the Polycom doesnt take this in account, even if the second choice of both endpoint is G711u, Asterisk drops this logs since the Polycom starts to spit G729 audio: Dec 18 17:06:30 NOTICE[193554]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to GSM Dec 18 17:06:30 NOTICE[193554]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from GSM to G729A Dec 18 17:06:30 WARNING[193554]: File codec_gsm.c, Line 136 (gsmtolin_framein): Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP (20)? Dec 18 17:06:30 WARNING[193554]: File chan_sip.c, Line 1159 (sip_write): Asked to transmit frame type 2, while native formats is 256 (read/write = 2/2) Dec 18 17:06:30 WARNING[193554]: File app_dial.c, Line 282 (wait_for_answer): Unable to forward frame Does anybody know if Polycoms has a three finger salute as Cisco 79XX does ? I really hate to unplug ethernet cable since you have to release the stand first. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fedora core 1 install problem - CAN SOMEONE ELSE HELP HIM
Sorry I can't help you anymore than that...hopefully some guru can...I checked all of your versions against mine and they're EXACTLY the same... Was it a clean install? I.e. newish? Hopefully someone else will continue the thread from here... Sorry, Matt David Luyens wrote: I used root user but I do not understand what you mean running it by prompt or screen. These are the packages I installed. Do you know if some are missing? David [EMAIL PROTECTED] root]# rpm -q kernel-source readline readline-devel openssl openssl-devel bison cvs gcc newt-devel ncurses-devel libtermcap-devel zlib zlib-devel kernel-source-2.4.22-1.2115.nptl readline-4.3-7 readline-devel-4.3-7 openssl-0.9.7a-23 openssl-devel-0.9.7a-23 bison-1.875-5 cvs-1.11.5-3 gcc-3.3.2-1 newt-devel-0.51.6-1 ncurses-devel-5.3-9 libtermcap-devel-2.0.8-36 zlib-1.2.0.7-2 zlib-devel-1.2.0.7-2 [EMAIL PROTECTED] root]# -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens matt Verzonden: woensdag 17 december 2003 23:30 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] fedora core 1 install problem Strange... I just did one yesterday on Fedora Core 1 and eventually got everything sweet... what user did you use? did you run it from screen or a prompt? did you install the pre-requisites and/or check for them with rpm-q package name? i.e. rpm - q kernel-source readline readline-devel openssl openssl-devel (according to the piece of paper digium sent me with the kit - although some people on this list have said that readline is no longer neccesary with the latests cvs of asterisk) Kind regards, Matt Riddell David Luyens wrote: Hi, I am trying ti install an asterisk system on fedora core 1. During the make of asterisk I got the folowing problem: bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: *** [ast_expr.c] Broken pipe Does anybody know how to solve this? David \/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o tdd.o tdd.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o acl.o acl.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o rtp.o rtp.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o manager.o manager.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o asterisk.o asterisk.c bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: *** [ast_expr.c] Broken pipe [EMAIL PROTECTED] asterisk]# ___ Asterisk-Users mailing list [EMAIL PROTECTED]
RE: [Asterisk-Users] after hours
Stevie If you do not have any thing intelligent to say Why waste both your time and ours Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, 19 December 2003 1:40 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] after hours On Thu, 2003-12-18 at 02:26, [EMAIL PROTECTED] wrote: When setting include = daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours This is where it is important for you to flex your mind while reading what was given to you and whatever documentation it leads you to. It will benefit you a great deal, and the community a bit. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] another
Thanks Regards Mick [weekend] s,1,blahblahweekend etc [EMAIL PROTECTED] wrote: Matt I understand that bit but How do I express the sound file for after that time period ?? Here is what I need to do include = daytime|9:00-21:00|mo-fri|*|* include = weekend|10:00-19:00|sat-sun|*|* I think the above is correct ?? Bit how do I specify the after hours config ??? Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of matt Sent: Thursday, 18 December 2003 2:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] another [EMAIL PROTECTED] wrote: Hi again How do I change the message played on initial pickup for after hours ?? Thanks in advance Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users In the context put: include = daytime|9:00-17:00|mo-fri|*|* which will include the daytime context during these hours ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 question
Found it. Anyone interested can look in RFC3551 RTP Profile for Audio and Video Conferences with Minimal Control. You can piece together that G.729, G.729a G.729b will play together and the other annexes will not due to bandwidth differences. Clif Jones wrote: I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this answer. I know that G729A is low complexity which seems to be what Cisco 7960's use but I have some others that support G729B which has comfort noise and reduced transmission during silence. If anyone knows how the different G729 codecs interoperate I would be eager to know. Thanks. ___ 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] moh problems
Hi, I'm trying to setup Moh default config. When I dial the ext. I get this: WARNING[1200884528]: File res_musiconhold.c, Line 303 (moh0_exec): Unable to start music on hold (class 'default') on channel SIP/user1-f2d3 What could it be? Tx., ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] after hours
Are you sweet with it now? The other option is to go to the documentation link on digium's website where there are demo config files...that's probably the single thing that helped me the most... Also the people on the irc group can be nice from time to timealthough it helps to be demure and seem like you have absolutely no clue - they tend to get pissed and not help you if you know a bit about what you're doing. I've had conversations there with a few people and they helped me to get asterisk fully up and running within a day of receiving the hardware! And I hadn't used linux for like 5-7 years or something! It makes it somewhat harder being in a different timezone, but you must be closer to the US timezone than me, seeing as you're in Ozzie not NZ. Kind regards, Matt Riddell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Excessive VNAK's and jitter over IAX2
Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone --(ulaw over LAN)-- *1 IAX2 (ulaw over Internet) -*2(GSM over Internet) ---*3(ulaw over LAN)-- SIP phone Now what is shown below is the Asterisk in the middle, that is doing the conversion between the other two, one of which only speaks ulaw and the other only speaks GSM. The call basically seemed to work, except the audio quality was terrible, but it did seem to be basically connected. Asterisk started spewing out these VNAK messages, thousands of them as fast as it could. In the middle of it I did an IAX2 show channels to show what was in progress. The asterisk version shown here is a completely stock, CVS version from just a few days ago. The outboard Asterisks are somewhat modified but also re-synchronized with CVS within the last week. Also, all Asterisks have iax jitterbuffer=no. So, my questions are: 1. What do the excessive VNAKs indicate? Some type of communication error? NAT-related perhaps? 2. Does the 20,000+ jitter have something to do with the audio sounding terrible? 3. Why is there jitter at all if all Asterisks have their IAX2 jitter buffers turned off? 4. Is there any significance to the Username (none) for one of the peers? The Asterisk has both peer and user names for both machines. The caller name shows up, but the callee name is always (None) Ideas anyone? Thanks. DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK s Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter Format 24.9.xx.xxx i58 9/3 00015/6 0ms 0169ms ULAW 66.167.xx.xxx(None) 00010/4 8/00013 9ms 20743ms GSM 2 active IAX channel(s) *CLI DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fedora core 1 install problem - CAN SOMEONE ELSE HELP HIM
The name bru1voip kindof looks familarish to me ;) (hi btw), but may
just be concidental.
What are the specs on the box you are trying to compile it on? (disk/ram/etc)
If its a pitiful machine, compile it on another machine, and transfer it
over. Sometimes Makefiles have a PREFIX (or so) option, you might be able to do
like mkdir complete; make PREFIX=${PWD}/complete install; tar czvf install.tgz
install; and copy install.tgz to the machine you're trying to compile on
and extract the install.tgz in /
It helps if you are building on the same distro/patch level as your target.
- andrewg
Hi,
I am trying ti install an asterisk system on fedora core 1. During the
make of asterisk I got the folowing problem:
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
make: *** [ast_expr.c] Broken pipe
Does anybody know how to solve this?
David
\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\
-DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\
-DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o tdd.o tdd.c
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o acl.o acl.c
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o rtp.o rtp.c
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o manager.o manager.c
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o asterisk.o asterisk.c
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
make: *** [ast_expr.c] Broken pipe
[EMAIL PROTECTED] asterisk]#
___
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Re: [Asterisk-Users] after hours
reorganized - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 18, 2003 3:30 PM Subject: RE: [Asterisk-Users] after hours -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, 19 December 2003 1:40 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] after hours On Thu, 2003-12-18 at 02:26, [EMAIL PROTECTED] wrote: When setting include = daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours This is where it is important for you to flex your mind while reading what was given to you and whatever documentation it leads you to. It will benefit you a great deal, and the community a bit. -- Steven Critchfield [EMAIL PROTECTED] Stevie If you do not have any thing intelligent to say Why waste both your time and ours Regards Mick Because people sometimes need to be kicked before they will think for themself. Give a man a fish and he eats for a day. Teach him to fish and he eats for a lifetime. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Different Dial tones for internal and external.
On Thursday 18 December 2003 14:04, Tilghman Lesher wrote: In the zaptel driver, create a new tone in zonedata.c at the END of a zone (so you don't throw off existing tone indexes). Then, in asterisk, in the specific channel driver (e.g. chan_zap.c), locate ast_ignore_pattern and change ZT_TONE_DIALTONE to the index of your new tone. You can probably use ZT_TONE_CUST1 as your index. Before anybody else tries this, know that I just wedged the driver for the card by attempting to do this. NOT for the faint of heart. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3-way calling bug
Hi, I discovered a problem in asterisk with the following scenerio: 1) I make an outbound call 2) Called person answers phone 3) I hit the flashhook to initiate a 3-way call 4) I hear dial tone and called person is on hold 5) I hang up my phone 6) called person hangs up their phone 7) my phone starts ringing 8) I answer and no one is there, I hang up 9) endless loop between step 7 8 happens after this happens and this endless ringing loop begins asterisk cannot be stopped from within the console but must be killed with kill -9. Any help or insight into the matter would be greatly appreciated. Thanks, Derek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] after hours
Although it's hard to see the original proverb writer saying RTFM :-) Matt Andrew Thompson wrote: Give a man a fish and he eats for a day. Teach him to fish and he eats for a lifetime. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] after hours
Yeah And give him a gun And look at what happened at Port Arthur Regards Mick Although it's hard to see the original proverb writer saying RTFM :-) Matt Andrew Thompson wrote: Give a man a fish and he eats for a day. Teach him to fish and he eats for a lifetime. ___ 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] Headless Linux system for Asterisk
Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? TIA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Headless Linux system for Asterisk
You can see any thing Sorry I could not resist If you need to admin Linux without a monitor Try webmin Regards Mick Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? TIA ___ 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] Headless Linux system for Asterisk
well i never had a asterisk server with a monitor or keyword all my servers i do remote login with ssh its better more private. Miguel On Thu, 2003-12-18 at 22:02, Michael Welter wrote: Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? TIA ___ 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] Headless Linux system for Asterisk
You will probably find quite a few people do. I know I do. I have a monitor hooked up to a keyboard switch that is attached to my Asterisk server but I never click over to it. I use SecureCRT and monitor the console that way only. No need really for a monitor unless you want the graphical view which is not really needed anyway. I can do all I need to do with no monitor at all. I use RedHat 9 and just issue a up2date -u when I need to do updates via the RedHat network as well. I can't think of 1 reason you would HAVE to use a monitor. Good luck, Robert - Original Message - From: Michael Welter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 18, 2003 2:02 PM Subject: [Asterisk-Users] Headless Linux system for Asterisk Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? TIA ___ 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] Headless Linux system for Asterisk
There are no issues. There is no reason to have a K/B or monitor on the server. Just sucks up power and adds to global warming. You may also want to pull any CDROM or flopy drive from the box too for the same reason. Seriusly, you should be using ssh from a remote machine to access the server. I did install X11 and many X11 clients like but NEVER run X11 on the server just do ssh -X and work on a remote machine. --- Michael Welter [EMAIL PROTECTED] wrote: Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? TIA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Headless Linux system for Asterisk
On Thu, Dec 18, 2003 at 03:02:42PM -0700, Michael Welter wrote: Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? The main issue is that you need to think before you type something; lest you do something that requires you to plug a monitor in. Some things can include, rebuilding your kernel, changing network settings, playing with firewall rules, stopping/starting sshd's, etc. A good line of advice would be to get it working with a console attached, and use it remotely and ssh to it. Oh. and your bios settings might need to be changed so it will boot without a keyboard present. TIA ___ 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] Headless Linux system for Asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter Sent: Thursday, December 18, 2003 5:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Headless Linux system for Asterisk Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? I would doubt that many real installations have monitors attached. And whether it works or not has nothing to do with the OS or any applications running on the machine. It is strictly a hardware support issue. Most equipment should have no problems without a mouse or keyboard if properly configured. Most hardware can't even detect if a monitor is attached or not. Daryl G. Jurbala BMPC Network Operations Tel: +1 215 825 8401 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Headless Linux system for Asterisk
Michael Welter wrote: Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? no issues whatsoever use ssh screen :D TIA ___ 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] fedora core 1 install problem - CAN SOMEONE ELSE HELP HIM
Hi, the machine is a dual xeon, 1 G of memory and 160 G harddisk...
I am not such a linux guru and your suggestion kind sounds like chinees,
but thanks anyway.
I will try to do a new install and let you know...
David
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens
[EMAIL PROTECTED]
Verzonden: donderdag 18 december 2003 22:00
Aan: [EMAIL PROTECTED]
Onderwerp: Re: [Asterisk-Users] fedora core 1 install problem - CAN
SOMEONE ELSE HELP HIM
The name bru1voip kindof looks familarish to me ;) (hi btw), but may
just be concidental.
What are the specs on the box you are trying to compile it on?
(disk/ram/etc) If its a pitiful machine, compile it on another machine,
and transfer it over. Sometimes Makefiles have a PREFIX (or so) option,
you might be able to do
like mkdir complete; make PREFIX=${PWD}/complete install; tar czvf
install.tgz install; and copy install.tgz to the machine you're trying
to compile on
and extract the install.tgz in /
It helps if you are building on the same distro/patch level as your
target.
- andrewg
Hi,
I am trying ti install an asterisk system on fedora core 1. During
the
make of asterisk I got the folowing problem:
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
make: *** [ast_expr.c] Broken pipe
Does anybody know how to solve this?
David
\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\
-DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\
-DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o tdd.o tdd.c
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\
-DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o acl.o acl.c
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\
-DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o rtp.o rtp.c
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\
-DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o manager.o manager.c
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\
-DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o asterisk.o asterisk.c
bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c
make: *** [ast_expr.c] Broken pipe
[EMAIL PROTECTED] asterisk]#
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Re: [Asterisk-Users] Unable to detect process 256 frames
Thanks, Jeremy, that was indeed the problem. Message: 2 Date: Thu, 18 Dec 2003 12:56:48 -0500 From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Unable to detect process 256 frames Reply-To: [EMAIL PROTECTED] SW wrote: Hi folks, Does anybody have any idea what this is; WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 256 frames Do not try to do inband DTMF on G.729 Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Headless Linux system for Asterisk
Michael Welter wrote: Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? Most of my Linux boxes are sans-monitor. I highly recommend it. Just be sure SSH works well and you haven't firewalled your admin workstation from being able to communicate (not that I speak from experience, he he, blush). Some computers have to have a keyboard to boot. Others are a little more sane. YMMV Good luck, and I hope your server can breathe. David Gomillion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Headless Linux system for Asterisk
On Thu, 2003-12-18 at 17:02, Michael Welter wrote: Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? TIA Run using a serial console (http://www.tldp.org/HOWTO/Remote-Serial-Console-HOWTO/). No monitor, VGA adapter, keyboard etc needed. Use SSH to log into the asterisk box for any maintenance, etc. If the box gets hosed, connect the serial port to a working PC and fire up minicom and your all set. You'll find this type of setup quite often in data center environments. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Inuse Count Wrong
I am currently using a copy of Asterisk checked out as the code of 10 days ago from Asterisk and the: sip show inuse reports that I have 3 incoming connections to one of the Grandstream phones, even though that isn't the case. I believe I have tracked the problem down to the following error message, which also (conveniently) showed up 3 times: -- Got SIP response 481 back from 192.168.252.101 Incidentally I am using the code from 10 days ago because as explained earlier today the CAPI support with the current Asterisk is unable to do a native bridge between two CAPI calls (instead it just drops both calls). Incidentally am I sending this report to the right mailing list, or is there a better one? Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729 question
Hi Clif, My experience with G.729 and asterisk is not good. My first registration was good, it worked. Then I bought more license and tried to upgrade it, it blew everything off. Still waiting Digium support to give me a helping hand. If you use pass-through feature then I guess you are fine. I have SIP users going to h.323 g/w and I need g.729. So now I have it in pass-through mode, I think that requires less CPU overhead and I do not have to mess with licenses. Cheers SW Message: 5 Date: Thu, 18 Dec 2003 13:59:06 -0500 From: Clif Jones [EMAIL PROTECTED] To: asterisk users [EMAIL PROTECTED] Subject: [Asterisk-Users] G729 question Reply-To: [EMAIL PROTECTED] I am thinking about using the G729 codecs on my endpoint devices and purchasing some G729 licenses for Asterisk but I have several questions: 1. Which G729 codec is sold by Digium for Asterisk, G729, G729A, B...I? 2. If I have G729A on one end and G729B on the other, are they compatible? I have looked all over the place for question 2, but without buying the ITU docs I cannot seem to find this answer. I know that G729A is low complexity which seems to be what Cisco 7960's use but I have some others that support G729B which has comfort noise and reduced transmission during silence. If anyone knows how the different G729 codecs interoperate I would be eager to know. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Headless Linux system for Asterisk
Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? My home asterisk box is a headless machine. I generally keep my Linux machines headless. You can use SSH to connect to your server and do anything you would ever do from the console. One catch is you will want to verify that your computer is happy booting without a keyboard present. Most PCs can be set in the BIOS to ignore a missing keyboard, but some machines will halt during boot waiting for you to press F1 to continue. This is an annoying problem especially if you must reboot the system from remote. -- Tony Kava Network Administrator Pottawattamie County, Iowa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sphinx
Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2 decoder called with command line arguments. executing /usr/local/bin/sphinx2-server, please wait ioctl(SETDUPLEX) failed: Invalid argument Calibrating background noise level...done server.c(443): Bad or missing port# argument, using 7027 srvcore.c(382): Listening at port 7027 srvcore.c(409): Connected 192.168.1.99 at Thu Dec 18 15:24:19 2003 Hit CR to start listening, qCR to quit client connection -- Executing Answer(SIP/test-ff55, ) in new stack -- Executing EAGI(SIP/test-ff55, eagi-sphinx-test) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/eagi-sphinx-test Environment: 'agi_request' is 'eagi-sphinx-test' Environment: 'agi_channel' is 'SIP/test-ff55' Environment: 'agi_language' is 'en' Environment: 'agi_type' is 'sip' Environment: 'agi_uniqueid' is '1071786651.21' Environment: 'agi_callerid' is 'blah 1234' Environment: 'agi_dnid' is 'unknown' Environment: 'agi_rdnis' is 'unknown' Environment: 'agi_context' is 'default' Environment: 'agi_extension' is '911' Environment: 'agi_priority' is '2' Environment: 'agi_enhanced' is '1.0' Environment: 'agi_accountcode' is '' Ooh, got a response from Asterisk: '200 result=0 endpos=46560' 1. Result is '200 result=0 endpos=46560' Ooh, got a response from Asterisk: '200 result=0 endpos=30720' 2. Result is '200 result=0 endpos=30720' -- Playing 'digits/20' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/million' (language 'en') -- Playing 'digits/4' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/50' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/thousand' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/40' (language 'en') -- Playing 'digits/5' (language 'en') Ooh, got a response from Asterisk: '200 result=0' 3. Result is '200 result=0' -- Playing 'demo-enterkeywords' (language 'en') Ooh, got a response from Asterisk: '200 result= (timeout)' 4. Result is '200 result= (timeout)' Ooh, got a response from Asterisk: '200 result=0 endpos=9440' 5. Result is '200 result=0 endpos=9440' -- AGI Script eagi-sphinx-test completed, returning 0 Is the endpos number something significant? What is it referring to? Am I doing this right? Does anyone have any other EAGI Sphinx examples? Maybe something that asks for you to say a word and it puts it into text? Sorry, this is pretty neat but there's hardly any information on it. I've tried searching the lists and I see a couple people that have used it but that's all. Thanks, Kevin B. _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] after hours
Well I feel you are right there are a few people on this list That could use a good kick. Aren't there Andrew Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Thompson Sent: Friday, 19 December 2003 7:41 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] after hours reorganized - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 18, 2003 3:30 PM Subject: RE: [Asterisk-Users] after hours -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, 19 December 2003 1:40 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] after hours On Thu, 2003-12-18 at 02:26, [EMAIL PROTECTED] wrote: When setting include = daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours This is where it is important for you to flex your mind while reading what was given to you and whatever documentation it leads you to. It will benefit you a great deal, and the community a bit. -- Steven Critchfield [EMAIL PROTECTED] Stevie If you do not have any thing intelligent to say Why waste both your time and ours Regards Mick Because people sometimes need to be kicked before they will think for themself. Give a man a fish and he eats for a day. Teach him to fish and he eats for a lifetime. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ 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] Headless Linux system for Asterisk
Michael Welter wrote: Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? There is no fancy GUI, unlike that other semi-popular OS, so there is absolutely no need for a monitor and keyboard on a Linux box. Just make sure you make sure the BIOS is set not to halt the system on any errors. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Excessive VNAK's and jitter over IAX2
I'm also getting this issue, for some reason some calls yield mass VNAK's. I also get iseq problems, but that might be my code - eg DEBUG[98311]: File chan_iax2.c, Line 4368 (socket_read): Received iseqno 4 not within window 0-2 DEBUG[98311]: File chan_iax2.c, Line 4368 (socket_read): Received iseqno 4 not within window 0-2 DEBUG[98311]: File chan_iax2.c, Line 4368 (socket_read): Received iseqno 4 not within window 0-2 Both happen in 5% of the time. - Original Message - From: Matt Lawson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, December 19, 2003 7:59 AM Subject: [Asterisk-Users] Excessive VNAK's and jitter over IAX2 Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone --(ulaw over LAN)-- *1 IAX2 (ulaw over Internet) -*2(GSM over Internet) ---*3(ulaw over LAN)-- SIP phone Now what is shown below is the Asterisk in the middle, that is doing the conversion between the other two, one of which only speaks ulaw and the other only speaks GSM. The call basically seemed to work, except the audio quality was terrible, but it did seem to be basically connected. Asterisk started spewing out these VNAK messages, thousands of them as fast as it could. In the middle of it I did an IAX2 show channels to show what was in progress. The asterisk version shown here is a completely stock, CVS version from just a few days ago. The outboard Asterisks are somewhat modified but also re-synchronized with CVS within the last week. Also, all Asterisks have iax jitterbuffer=no. So, my questions are: 1. What do the excessive VNAKs indicate? Some type of communication error? NAT-related perhaps? 2. Does the 20,000+ jitter have something to do with the audio sounding terrible? 3. Why is there jitter at all if all Asterisks have their IAX2 jitter buffers turned off? 4. Is there any significance to the Username (none) for one of the peers? The Asterisk has both peer and user names for both machines. The caller name shows up, but the callee name is always (None) Ideas anyone? Thanks. DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK s Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter Format 24.9.xx.xxx i58 9/3 00015/6 0ms 0169ms ULAW 66.167.xx.xxx(None) 00010/4 8/00013 9ms 20743ms GSM 2 active IAX channel(s) *CLI DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK DEBUG[98311]: File chan_iax2.c, Line 4649 (socket_read): Sending VNAK ___ 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] Headless Linux system for Asterisk
On Thu, Dec 18, 2003 at 03:02:42PM -0700, Michael Welter wrote: Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? Not only can you run Linux without a monitor or keyboard, you can also just plug a keyboard and monitor in if you need to get at the console. When you're done, just unplug them. (You won't do that on an NT server). What I do is install linux with a keyboard and monitor connected. After I've got it on the network, I go back to my office and finishe the configuration. When I install the server at the clients site, I just make sure that my ssh connection can route to the server so I can go back to my office and maintain it. The end user can ssh to the server if they need to (are interested in) observe or manage Asterisk. It's great, two connections to the server: network and power. Walker -- DataCrest, Inc. -- Technically Superior ** Walker Haddock http://www.datacrest.com DataCrest, Inc.e-mail: [EMAIL PROTECTED] 1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589 Birmingham, AL 35216 fax: 1-205-823-7838 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fedora core 1 install problem - CAN SOMEONE ELSE HELP HIM
Reinstaling the OS is not going to do anything ut get you right back to where yu are now. Look at the last command the the Makefile tried to run. Did it choke while running bison? If so run the bison command by hand. What happens? You need to cd to the directory where Makewas at when it tried to run bison first. You might also try typing which bison just to verify that bison is installed and in your path. In general when a Makefie fails you try to get the failed command to run by hand and then you fix up the Makefile to do whenever you neede to do by hand. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Headless Linux system for Asterisk
Run using a serial console (http://www.tldp.org/HOWTO/Remote-Serial-Console-HOWTO/). No monitor, VGA adapter, keyboard etc needed. Use SSH to log into the asterisk box for any maintenance, etc. If the box gets hosed, connect the serial port to a working PC and fire up minicom and your all set. You'll find this type of setup quite often in data center environments. Except there is a known problem of dropping/missing interrupts with running serial consoles with certain Digium boards. You also have the same problem if you use a framebuffer console. If you truly want it headless with a serial console without that problem, stick a PC Weasel board in it (http://www.realweasel.com). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
