Re: [Asterisk-Users] Problem with SPA-2000 and Asterisk 1.0.5
Carlos Chavez wrote: I had everything working fine until today. Today the Sipura cannot dial anywhere. I just get the following: Feb 10 12:48:18 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10 12:48:19 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10 12:48:35 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Feb 10 12:48:35 NOTICE[1205]: chan_sip.c:7399 handle_request: Unable to create/find channel Every time you pick up the phone and dial anything the above appears in the console. The strange things is that I can dial to that extension and talk without problems. I did not change anything on both sides that may account for a problem, it just started today. I did a factory reset on the Sipura and re configured it and I still have the same problem. I don't know what to do next. Any ideas? I've recently encountered the same issue here. I've narrowed down the problem to be authentication failures, although it doesn't readily say it. See if this doesn't clear up if you remove the secret= line from the user/friend entry in sip.conf, and from the UA (sip client). If so, then could you please email me off list with your network setup so I can attempt to make a determination of wether or not there's something similar in the setup? -Josh (twisted) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Your Acerbic Tyrant will be off line for about 10 days
Next time you decide to move, have enough courtesy to *NOT* inform the other 8000+ subscribers of the list, as most of us could care less, and you're just wasting bandwidth.Yes, this post is too, but I feel it's okay since you also managed to CROSS POST THIS CRAP to the other lists. Thanks, have a great move. Don't drop a tv on your toe. Race Vanderdecken wrote: Greetings List, I know many of you are looking for advice from me but I am moving from the 28th until about the 4th of February. As moving does not always go as planned so I am letting you know that I may be out of internet touch for 10 days during the move depending on the closing and the Cable Modem guy. In case any cares to know, I am moving from South Florida to Asheville. I will try to check mail often but please do not think I am being rude if I do not answer for a while. Race Vanderdecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ResponseTimeout problem
Joseph wrote: [snip] I'm assuming you have snipped the priorities 1 and 2. exten = s,3,BackGround(welcome) exten = s,4,ResponseTimeout,15 exten = t,1,Goto(1,1) Description ResponseTimeout(seconds) Set the maximum amount of time permitted after falling through a series of priorities for a channel in which the user may begin typing an extension. If the user does not type an extension in this amount of time, control will pass to the 't' extension if it exists, and if not, the call would be terminated. If ResponseTimeout is not explicitly set in an extension, the default value of 15 seconds will be used. Thank you, it work! So it needs to be pass to: exten = t,1,Goto(1,1) I got confused by the last sentence ...and if not, the call would be terminated. This is correct. However, if you wish to have it pause between priorities for an extension change, try using the application WaitExten. Here's a show application waitexten: -= Info about application 'WaitExten' =- [Synopsis]: Waits for some time [Description]: Wait([seconds]): Waits for the user to enter a new extension for the specified number of seconds, then returns 0. Seconds can be passed with fractions of a seconds (eg: 1.5 = 1.5 seconds) or if unspecified the default extension timeout will be used. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ResponseTimeout problem
Joseph wrote: This is correct. However, if you wish to have it pause between priorities for an extension change, try using the application WaitExten. Here's a show application waitexten: -= Info about application 'WaitExten' =- Yes, I was looking at it already but it is available in ver. 1.0.0 and up; I'm on 0.9 on Gentoo. Gentoo is kind of slow when it comes to Asterisk. There is an unstable ver. 1.0.2 in unstable branch but it doesn't compile (there is an error when compiling). So I will have to learn how to upgrade using CVS or wait for Gentoo stable version. If I use CVS I'm not sure if startup scrip will be upgraded as well in /etc/init.d/ I don't see any reason the startup script would need to be updated. Apparently, however, gentoo has 1.0.1 is in the portage tree now, if you do an emerge sync, or possibly you need to look deeper. I'm not a gentoo user by any means, but this is being reported to me by a gentoo user at this very moment. -J ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SysMaster and GPL Violation (lets think before we
Okay.. this is geting out of hand. Please end this thread now, and take it off list. Personal attacks have NO place on a mailing list. Jay Milk wrote: I thought Kevin Walsh was Asterisk's bitch? -Original Message- From: Gary [mailto:[EMAIL PROTECTED] Sent: Sunday, November 14, 2004 8:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SysMaster and GPL Violation (lets think before we On Sun, 14 Nov 2004 17:30:23 -0800, Bruce Ferrell wrote: Only if we can move the top-post discussion there too Gary wrote: Hi folks, Might I propose a new mailing list ?? Asterisk-bitch Thus discussions such as the one with this topic could be moved to it rather than clutter up an already very busy list. All those in favour ? Only if we can move the top-post discussion there too Sure, any of those types of debate. In fact, i just wished people would NOT use the list for debates !! . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pause during dial
information of this nature is on the wiki at http://www.voip-info.org/wiki-Asterisk I will go ahead and give you a hint, however. the 'w' digit means wait, or pause, in dialing. -Josh Henry Devito wrote: Is there a way to put pauses in a dial string? I need * to dial a number then pause for 6 seconds and dial a second string of numbers. I have now Dial(ZAP/1/18005551212)But I need it to be Dial(ZAP/1/18005551212,pause for six seconds,454545) . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wcfxs module doesn't load
Tom Lahti wrote: Question on installation issue: I noticed that the product literature for the TDM400P states that it is PCI 2.2 compliant, but it doesn't say that PCI 2.2 is required. Does the TDM400P _require_ PCI 2.2 in order to function? I'm trying it in an older motherboard that is PCI 2.1 compliant. -- -- = Tom Lahti Tx3 Online Services (888)4-TX3-SVC (489-3782) http://www.tx3.net/ -- = First of all, please don't repost just because you haven't gotten an answer in 2 hours. Secondly, yes, it has to be 2.2, in my experience. I have not had any luck whatsoever getting it to work in a 2.1 slot. -Josh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] moh
Just an FYI: If you are *EVER* unsure that mpg123 is correctly installed (correct verison etc), you can enter the asterisk source tree, and type 'make mpg123' (without quotes), and mpg123 v0.59r will be download ed, unpacked, and built for you, and then a simple make install will install asterisk AND mpg123 in one smooth motion. -josh Richard wrote: Thanks Matthew, You are the MAN! It fixed the problem. Richard -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matthew Sent: Sunday, October 31, 2004 3:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] moh My solution to this (as the debian package appears to actually download mpg321 (instead of mpg123) when you install *, was to download mpg123 from the original website and compile/install it myself. http://www.mpg123.de/ mpg123 0.59r is the version im now running (just copied the executable over mpg123 and mpg321 and restarted asterisk (and killed dead looking mpg321 processes) started up astersik, caleld myself and shoved myself on hold, and VOILA, music on hold is working normally and not running 'really' slow Hope this helps! Richard wrote: Hi, I have * 1.0.0. Everything works well except moh. I followed the instruction in http://voip-info.org/wiki-Asterisk+config+musiconhold.conf. I use the default mp3 from *. The problem is that the music is really slow. Seems like it didn't get the right rate to play. Any one having this problem too? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-Dev] How to submit a patch?
Renato Mintz wrote: Josh, please help me as I'm really naive to this process. Is there a specific text you would like to see in the disclaimer? Where should I send this disclaimer file to? Tks, Renato On Mon, 25 Oct 2004 21:08:35 -0500, Josh Roberson [EMAIL PROTECTED] wrote: Just so everyone's clear on the patch submission etiquette here... (sorry for the crosspost, but I believe this is relevant to both users/dev lists) --SNIP-- If you read the etiquette, http://www.digium.com/bugguidelines.html , you will see that there are two versions of a disclaimer that need to be filled out, signed, and either faxed or mailed to digium for patch submission. I was just posting a few key problem points from this etiquette to clear things up a bit. -Josh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Type of T1 for T100P card
Cirelle Enterprises wrote: - Original Message - From: Pedro Aguayo [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 27, 2004 6:18 PM Subject: [Asterisk-Users] Type of T1 for T100P card | I'm currently setting up a PBX system using the T100P card, and was | wondering if it can handle the 2-way trunk type of T1s. Do 2-way trunk | T1s use RBS signaling? | Please excuse my ignorance, I have mostly dealt with PRI B and D channel | type of T1s. | | Thanks | | Pedro I found out by taking the long way around, the t100p requires the t1 pri hybrid isdn bchan, dchan A standard 24 timeslot t1 will not work. regards greg I apologize if i'm not understanding exactly wtf you're saying here, but if i'm reading this correctly, I COMPLETELY disagree. I have MANY instances of t100p's on a normal 24 channel t1 (rbs), and a few on PRI's using standard B and D channels. It will work fine as long as it's configured properly. -Josh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] How to submit a patch?
Just so everyone's clear on the patch submission etiquette here... (sorry for the crosspost, but I believe this is relevant to both users/dev lists) To submit a patch, PLEASE PLEASE PLEASE help us out by doing the following: Search the bugtracker to see if someone has already submitted a patch/request/bug that may be relevant to your patch. If so, please use the existing bug, and follow the relevant sections below. If not, please follow the following guidelines: 1) make sure you save your patch in a unified diff, based on stable/head cvs if possible. (cvs diff -u path/to/file.c [within the asterisk source tree will keep you from having to have two copies of the source file, too]) and save it into a file ending in a .txt extension (for faster review by us bug marshals, and Mark. 2) make sure you add the header [patch] without quotes to the subject/summary field. This will help us quickly identify your bug as a patch, and will once again, help speed up the process. 3) please refrain from using whitespaces instead of tabs within your patch, also, please try not to add any unnecessary whitespace. and last but not least 4) if you don't already have a disclaimer on file, please file one. Once filed, PLEASE mention somewhere in your bug that you have one on file. This saves a lot of time, and has held back lots of good patches, because the powers that be have to ask, then wait on a response as to wether or not the patch has been properly disclaimed. Thanks for listening! Kevin Walsh wrote: I've found some problems in the implementation of say_number for portuguese and I have some corrections that I would like to submit to be incorporated to the CVS. How do I do that? Is there any place where this process is already written? Or can someone explain to me? My feeling is that I should first open a bug at Mantis describing the problem I found and attaching my patch. Then what? That's all. As you said, you just need to open a bug report and attach your patch. The report and patch will be evaluated at some point and might make it into CVS if it corrects an identifiable problem or provides a new/enhanced feature. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: phone line roaming
Benjamin on Asterisk Mailing Lists wrote:
On Wed, 15 Sep 2004 14:15:24 +0200, Pavel Jezek [EMAIL PROTECTED] wrote:
thanks for idea, but this is not exactly what I need,
It wasn't supposed to be *exactly* what you need ;-) That's not what a
mailing list is about. We can give each other clues and ideas for how
things can be done *in principle*, elaboration and implementation is
up to whoever wants the feature requested.
assume: one employee working in office (open-space cubes),
when this employee leave the work, on the same place come another employee
so that, I can't ring both lines and can't use bluetooth device :(
I thing to do some login to phone asterisk and download appropriate extension/phone line
according to e.g. username/login id,
Yes, that's a good thought to elaborate on.
but how to configure/implement?
The easiest way in terms of both implementation and use is probably to
use the dialplan and some extensions to trigger the login/logoff
For example, you could use *21 and *22 as a prefix to login and
logoff, then use DBput/DBget and the built in database to determine
the state of a virtual extension.
*212000 would tell Asterisk that extension 2000 is now on the device
from which this was dialled.
*222000 would tell Asterisk to cancel the previous state.
You'd maintain a database entry for each virtual extension (or user
depending on your) which would be your key. Then when somebody dials
*21 for that virtual extension, you check the phone it was coming
from and that is going to be your value for this key.
something along the lines of ...
exten = _*21,1,DBput(VIRTUAL${EXTEN:3}/physical={CALLERIDNUM})
but ideally you'd want to use something other than CALLERIDNUM to
identify the physical device.
All you have to do then is check for each incoming call whether there
is a value stored in the key for the virtual extension in question and
if there is send the call to the phone associated with that value.
Once you've got this basic functionality working, You can make it more
fancy with such things as PIN numbers, automatic cancelling of a value
if the user signs in again on another phone, timeouts taking into
account office hours etc etc etc.
And if you have done all this, don't forget to share and post the code
on the Wiki ;-)
rgds
benjk
benjk, pavel,
I have had something similar to this i made long ago. There are a
couple downfalls, as MWI doesn't work, but all in all, it works great.
http://www.indigent-networks.com/asterisk/roaming.txt
Also note, this is just an EXAMPLE. I claim no usability for it, even
though it does work ;)
-twisted
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Re: [Asterisk-Users] Dialing pstn-asterisk
I see your problem, unless you point out this is already the case:
Matthias Leeb wrote:
Hello list
When i'm trying to dial into our pstn the following errors occure:
-- Executing Dial(SIP/snomsip-dbd0, /2100) in new stack Sep 9 10:02:22
WARNING[59409]: channel.c:1901 ast_request: No channel type registered for
''
Sep 9 10:02:22 NOTICE[59409]: app_dial.c:715 dial_exec: Unable to create
channel of type ''
== Everyone is busy/congested at this time
-- Executing Congestion(SIP/snomsip-dbd0, ) in new stack
== Spawn extension (default, 02100, 2) exited non-zero on
'SIP/snomsip-dbd0'
- snip -
extensions.conf should be setup something like this:
Everything seems to be allright. Here is a part of my extensions.conf:
; all hard set variables need to be in global
[global]
CONSOLE=Console/dsp
TRUNK=Zap/g1
; sip phones set into context=default in sip.conf, for example.
[default]
ignorepat = 0
exten = _0.,1, Dial(${TRUNK}/${EXTEN:1})
exten = _0.,2,Congestion
Has anybody got some hints for me?
Beste regards
matthias
Try that, it should work.
-twisted
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Re: [Asterisk-Users] Cepstral
I wrote cepstral regarding this at the beginning of the week, thought it might be relevant to post the reply: Thanks for contacting us. Our Linux package is off the site right now because we are releasing a new version, 3.02, next week. This is an incremental release. The major update of this version is a new Linux SDK. Please check back with us in 6-7 days and we should have what you're looking for. We appreciate your patience. -Craig Now hopefully, they'll hold up to it and release the new Linux SDK in a week or so... -twisted Shane Young wrote: Quoting Jerry Geis [EMAIL PROTECTED]: Cepstral offers Linux versions. Just contact them. http://www.cepstral.com/cgi-bin/downloads?page=voices Note that you can not download any Linux versions from that page. They changed something a while back. Released a new TTS engine for Windows and Windows CE, but have not as of yet released it for Linux. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [patch] allow the transfer keys from app_dial's 't' and 'T' and hangup key 'H' to be configured via features.conf
Roger, at no point did I say I was finished with this patch. I did get a little frustrated early on in the development. Currently this patch is broken due to recent changes in cvs, and I'm about to tag it with a post-1.0 tag in the bugtracker since there seems to be lots of interest in it's existance, but it needs a little work. Please do not make assumptions as you have below, because I am the author of this patch, and I do *NOT* feel as though i'm finished, nor did i say so anywhere in the bugnotes. Thanks. twisted box100 wrote: Can anyone tell me how I can implement the features added in the following link for call transfer? The authors seem to feel they are finished but it doesn't appear to have been integrated into what everyone can download. It is referred to as a patch but I don't understand how it could be applied. Here is the link: http://bugs.digium.com/bug_view_page.php?bug_id=0002010 I guess I just don't understand how to apply patches Thanks in advance, Roger Easlick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [patch] allow the transfer keys from app_dial's't' and 'T' and hangup key 'H' to be configured via features.conf
Yeah, sorry i kinda jumped the gun on ya there..I plan to update it again pretty soon, but just for future reference, they normal method to apply a patch would be to use the patch command, as follows: patch -p0 patch.txt usually this is done from the top-level source directory for the package you are trying to patch, replacing patch.txt with the name of the diff (patch). Sometimes patch makers don't use paths in their patch, so you kinda have to know where the files are that you're trying to patch, but i try to make mine all work against the top-level source directory. twisted box100 wrote: Wow, you guys are fast. My apologies, twisted. I realize there must have been a reason why it wasn't marked resolved and included in the CVS HEAD, but I was under the impression that those who wanted to and have the knowhow could download and apply the patch. Didn't mean to imply you or anyone else *stated* that it was finished, it just seemed from the dialog in the bug report that work on it had been completed and just not marked as such. My mistake. It will be great when it is done whenever that happens since the added functionality really will increase the usefulness of Asterisk as a PBX. Thanks for the speedy reply. Roger From: [EMAIL PROTECTED] on behalf of Josh Roberson Sent: Tue 9/7/2004 00:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [patch] allow the transfer keys from app_dial's't' and 'T' and hangup key 'H' to be configured via features.conf Roger, at no point did I say I was finished with this patch. I did get a little frustrated early on in the development. Currently this patch is broken due to recent changes in cvs, and I'm about to tag it with a post-1.0 tag in the bugtracker since there seems to be lots of interest in it's existance, but it needs a little work. Please do not make assumptions as you have below, because I am the author of this patch, and I do *NOT* feel as though i'm finished, nor did i say so anywhere in the bugnotes. Thanks. twisted ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ex-girlfriend logic not working in latest CVS?
Maybe it's just me, but it looks as if you have one too many X's in your pattern matching.. 615NXX is all you need, i see 615NXXX. Same for 931. -twisted James Sizemore wrote: Ex-girlfriend logic not working in latest CVS? Incoming sip calls don't work. Anyone else seen this problem? Extension logic looks good: exten = 6153248305/_931NXXX,1,Queue(queue1); exten = 6153248305/_615NXXX,1,Queue(queue2); ;exten = 6153248305,1,Queue(queue3); show dialplan looks good: -- Added extension '6153248305' priority 1 (CID match '_931NXXX')to vantage -- Added extension '6153248305' priority 1 (CID match '_615NXXX')to vantage ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blocking the 'Do Not Call List
Chris, While you are thinking logically, This will just as un-effective as putting them all in the dialplan, as the DBGet() and DBPut() functionality deals with the internal astdb (db1 database). I would reccomend going the AGI route at this time, until we have better functionality for DB handling. -Josh Chris Shaw wrote: Why use AGI? Why not just use the builtin DBGet() and DBPut() functions in *? -Chris - Original Message - From: drodden [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 10, 2004 9:22 AM Subject: [Asterisk-Users] Blocking the 'Do Not Call List Anybody have any experience with blocking numbers in the U.S's Do Not Call list? We have a customer that will be getting their own Asterisk server from us, and they want it to be check outbound numbers against the do not call list; this is for a backup, in case there's a slip up and one of their people try to dial somebody on the do not call list. The list has millions of numbers, and I don't think the extensions.conf file could handle me listing all million+ phone numbers and making it play a sound like That number is on the do not call list, and then creating a _NXXNXXNXXX extension at the very bottom. The list would take up all it's memory. Anybody have a more elegant solution? Maybe an AGI script to match the outbound phone number against a column in a table in a MySQL database? Is there something similar already written that I can just modify? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AbsoluteTimeout Inside A Macro
Absolute timeout is 'T', and your standard timeout is 't'. If he's
looking for absolute timeout, he is, indeed, looking for the T extension.
They are case sensitive, and should work.
Mr. Wade: Have you tried using the T extension outside of the macro?
Although it *SHOULD* work within the macro, we may have stumbled upon a
bug..
-Josh
Chris Shaw wrote:
For one thing it's 't' not 'T', just like invalid is 'i' not 'I'
-Chris
- Original Message -
From: Christopher L. Wade [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 10, 2004 10:03 AM
Subject: Re: [Asterisk-Users] AbsoluteTimeout Inside A Macro
Christopher L. Wade wrote:
Hi all,
Is it just me and not reading the docs right, or has anybody else had
problems with the AbsoluteTimeout application and the 'T' extension when
used inside a macro?
[macro-attended]
; ARG1 is the device to dial out on, SIP or Zap, or whatever
; ARG2 is the extension to dial using 'attended' dialing
exten = s,1,AbsoluteTimeout(30)
exten = s,2,AGI(attended-extension,${ARG1},${ARG2})
; attended-extension takes a device string and an extension
; and builds a dial string according to some crazy internal logic
exten = s,3,Dial(${DIALSTRING},5,t)
exten = s,4,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Goto(s,1)
exten = T,1,NoOp(i got here here)
exten = T,2,Goto(s,1)
The purpose of this macro is to be able to say something like
exten = _8XX,1,Macro(attended,SIP,${EXTEN})
and have the the dialed extension rung, then, if no answer within 5
seconds, have the dialed extension plus an 'attendant' for that
extension rung, (etc. etc. etc.). If nobody answers after 30 seconds,
the caller is (read 'will be') offered the chance to leave a voicemail,
otherwise re-enter the loop, ringing the 'full' attendant list for the
requested extension.
When I test this, everything works according to plan, except when
AbsoluteTimeout expires, my T extension inside the macro is not
executed, the call is simply hungup. What am I doing wrong?
Thanks,
Chris
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Anybody?
Right now I'm considering doing this inside an AGI app, but I don't like
the way Dial is 'blocking' (AGI or not). I guess I could use chan_local
in my dial string inside the AGI to make it 'fork' but that just creates
a whole new ball of ear wax to deal with. :(
This 'bug' seams strange though, because I've seen examples that, at
least to my eyes, appear exactly the same as my above code.
Any help would be appreciated.
Thanks,
Chris
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Re: [Asterisk-Users] IAX2 'no authority found' problem
Simon, i was having the exact same problem, the only solution I found, was to remove the secret, then it worked great.. I thought I must have been missing something too, but apparently not. I'm not sure exactly what is causing this, as if i set the servers up to register with each other, they register fine, but the moment they try to pass a call to one another, they fail, unless there is no secret listed in iax.conf for the connections. -twisted Simon Ward wrote: Hi everyone, I'm having some problem trying to set up an IAX connection between two * servers. The scenario is : serverA has an X100p card and will direct all calls from the X100p over IAX to a specific extension on serverB which is at the other end of an unfirewalled VPN connection. At the moment serverA tries to redirect the call to serverB but recieves this message (it appears on both servers) : -- Executing Dial(Zap/1-1, IAX2/test:[EMAIL PROTECTED]/cardiff) in new stack -- Called test:[EMAIL PROTECTED]/cardiff Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 6ms SCall: 1 DCall: 0 [192.168.1.250:4569] VERSION : 2 CALLED NUMBER : cardiff LANGUAGE: en USERNAME: test FORMAT : 2 CAPABILITY : 65283 ADSICPE : 2 DATE TIME : 151287361 Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT Timestamp: 1ms SCall: 1 DCall: 1 [192.168.1.250:4569] CAUSE : No authority found Aug 4 14:50:02 WARNING[147465]: chan_iax2.c:5339 socket_read: Call rejected by 192.168.1.250: No authority found Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 1 DCall: 1 [192.168.1.250:4569] -- Hungup 'IAX2/192.168.1.250:4569/1' == No one is available to answer at this time Here are excerpts from the config files : ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback doesn't work whith h323
Steve Szmidt wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 04 August 2004 09:29 pm, Seth Remington wrote: On Wed, 2004-08-04 at 20:29, Jeremy McNamara wrote: M. Willigs wrote: Hi Jeremy My entry in the extensions.conf is like this: exten = 011001,1,Playback(tt-monkey) I didn't asociate the cmd Dial whit this entry, so, I can't answer the line You are not answering the line and that extension looks weird to me. Jeremy McNamara exten = 011001,1,Answer exten = 011001,2,Playback(tt-monkey) -Seth Crikey, what kind of extension is That!? Ten digits long?!! - -- Steve well... my first glance told me that tt-monkey doesn't exist... it's tt-monkeys. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf
Kevin Walsh wrote: Chris Shaw [EMAIL PROTECTED] wrote: Not in configs or /etc/asterisk/. Asterisk is still running, just curious why I am not seeing that file. Hmmm.. Maybe try checking out a fresh new copy in a different dir? It's been in there for over a week now, I just checked out a new copy and it's in there... Or simply rename musiconhold.conf as features.com and restart Asterisk. no.. WRONG. rename parking.conf, as parking.conf is what features.conf is derived from. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] features.conf
Kevin Walsh wrote: Josh Roberson [EMAIL PROTECTED] wrote: no.. WRONG. rename parking.conf, as parking.conf is what features.conf Oops. I knew it was one of them. At least I didn't say sip.conf :-) True that. This is another reminder that everyone needs to make sure that when they update, they check all of the files in the configs/ path in the src tree to see what's changed. Also, if you're confused about why something that's supposed to be in cvs isn't, a good method would be to make clean; make update; make install. If that still doesn't cure it, blow away the source tree and start with a new checkout. Just a friendly reminder to the list. twisted ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 to IAX2...i'm obviously an idiot!!
[EMAIL PROTECTED] wrote:
Hi All
I'm trying to get two Asterisk servers to talk to each other using IAX(2).
I've read the WiKi and the docs and tried the examples.
You sure?
I can't get it to work (I have 2 x 7960's registering on one server and 1 x 7960
registering on the other).
I've set them up as follows...
The two servers are set up as friends and have consecutive IP address's.
The setup is that the prefix 3 determines that the server dials the extension number
on the other servers local context:-
extensions.conf
exten = _3,1,Dial(IAX2/OtherServer:[EMAIL PROTECTED]:5036/${EXTEN:[EMAIL
PROTECTED])
5036 is not IAX2. That's abundantly clear in the wiki and the
examples. 4569 is the port you are looking for.
When I do a dial say 32221 this is what comes up in the console:-
Executing GoTo(SIP/2231-, intern-post|32221|1) in new stack
GoTo (intern-post,32221,1)
Executing Dial(SIP/2231-, IAX2/OtherServer:[EMAIL PROTECTED]:5036/[EMAIL
PROTECTED]) in new stack
Called OtherServer:[EMAIL PROTECTED]:5036/[EMAIL PROTECTED]
Warning: chan_iax2.c:1413 attempt_transmit: Max retries exceeded to host OtherServerIP
on IAX2/OtherServerIP:5036/3 (type = 6, subclass = 1, ts=2, seqno=0)
Hungup 'IAX2/OtherServerIP:5036/3'
then the regular cleanup commands
In IAX2 Show Peers I get:-
OtherServerOtherServerIP(S) 255.255.255.255 4569UnMonitored
it even tells you this right here...
I'm confused
why is the connection showing on port 4569 in show peers? Is this a default?
*nods*
Is there a way to test the validity of the IAX2 connection from the console?
Thanks in advance.
P
-twisted
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Re: [Asterisk-Users] Parking renamed to feature in 7/17/04 CVS
Seth Remington wrote: I just updated from CVS and noticed that Mark has renamed all of the parking related files (parking.conf, parking.h, res_parking.c) to features.conf, features.h, res_features.c respectively. The CVS log mentions that this is in preparation for some more (possibly post 1.0) feature additions. The header file still #define(s) _PARKING_H though so let the confusion ensue ;) Time to update the wiki. -Seth Actually, no, that was fixed also. -twisted ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy
Hi florian.. Florian Overkamp wrote: Hi, -Original Message- - Sometimes a call won't go through, dialtone stays on after keypresses. Strange. A powercycle of the iaxy usually helps Are you changing which phone is plugged into the iaxy before this happens? The changing of resistance levels I have noticed to cause this occasionally. Nope, no changes in the phone connection. By the way, what does this mean: -- Accepted AUTHENTICATED TBD call from ip that means that the iaxy has authenticated to * to make a call, but has not yet sent digits. This is normal activity. - No support for hookflash transfers (will this be possible ??) There is most definately support for flashhook transfers. Hit Flash, then #, then dial the number you're' transferring to. I've done this for awhile now, and it works fine. Can you show your IAXy provisioning config ? This doesn't do anything with my setup. The provisioning config has nothing to do with this. Make sure you're running revision 14 of the firmware. type iax2 show firmware from the cli to see which firmware you have on your asterisk system.If it's not the latest, i reccomend doing a cvs update to latest -HEAD . If you're not running HEAD this is futile. Also, to see which version of firmware you have on the iaxy, do an iax2 debug, and watch for the iaxy's talk, it will tell you what firmware revision it's running. I did, and I can confirm the IAXy is running version 14. Florian Okay.I'm not sure what to say then... I've been using an IAXy now since the first batch was done, and it works exactly as I described. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy
Okay.. here's the scoop I love the little blue boxes in the sense that it just works. Disadvantages as I have seen them these few days: - American tones only ? (or I just don't know how to change them) As far as I can tell this is true. - Sometimes a call won't go through, dialtone stays on after keypresses. Strange. A powercycle of the iaxy usually helps Are you changing which phone is plugged into the iaxy before this happens? The changing of resistance levels I have noticed to cause this occasionally. - No support for hookflash transfers (will this be possible ??) There is most definately support for flashhook transfers. Hit Flash, then #, then dial the number you're' transferring to. I've done this for awhile now, and it works fine. Florian Make sure you're running revision 14 of the firmware. type iax2 show firmware from the cli to see which firmware you have on your asterisk system.If it's not the latest, i reccomend doing a cvs update to latest -HEAD . If you're not running HEAD this is futile. Also, to see which version of firmware you have on the iaxy, do an iax2 debug, and watch for the iaxy's talk, it will tell you what firmware revision it's running. --Josh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BugTracker Information - REPOST
All, As a bug marshal, I have noticed quite a few bugs that seem to get overlooked/ignored due to the fact that they do not have the appropriate information in the bug information fields, or bugnotes. I am, therefore, Reposting an old post from July 26th, 2003, to help clarify the use of the Bugtracker. Sorry if you have read it before; this is just an attempt to re-familiarize everyone with the information that should be included in the posts, which should help speed things along. Thank you for your time, and attention. twisted ORIGIONAL POST From: Mark Spencer Subject: [Asterisk-Users] Bug Tracker Official Launch Date: Sat, 26 Jul 2003 13:48:14 -0500 (CDT) ANNOUNCEMENT: Bug Tracker/Feature Request System http://bugs.digium.com/ Digium has introduced a bug tracking and feature request system for Asterisk developers and users. Due to the increased traffic on the mailing list, and an inadequate number of hours in the day to parse it, it has been decided that a more meaningful method of tracking bugs, features, and patches had to be implemented. The Asterisk developers pledge to do their best update and keep the bug tracking system up to date so long as the users choose to utilize it adequately. We would encourage people from this point forward to log their bugs and features in this system. Simply sending things to the list is insufficient notification for bug repair and tracking. Unless submitted to the bug tracker, there are no guarantees that your bugs are even read, much less worked on. (No guarantees if they are in the bug tracker that they will be repaired, either, but they will be read and examined.) If you're a developer looking for a project, the bug tracker represents a good place to start looking. When you send patches to implement features and fix bugs, be sure to referenche what bugs they fix (or features they implement). BUGS: Before submitting a bug into the system, make sure you have the following information to submit: - your CVS date (show version) - your operating system and revision (uname -a) - your hardware configuration, if relevant (all cards and their configs) - your VoIP environment (SIP phones? H.323? MGCP?) - if a corefile has been produced, please have a backtrace gdb /usr/sbin/asterisk /path/to/corefile.1234 then type bt and include the output - include copies of relevant configuration files - full console error messages - debug traces (asterisk -vgcd) if applicable - WITHOUT ENOUGH DATA, YOUR BUG REPORT WILL BE REJECTED OR IGNORED What is a bug? A bug is something that causes unexpected adverse effects, contrary to what the stated or understood meaning of the program intended. A bug can be non-adherence to an RFC specification that causes conflict with other packages in a specific command set. A bug is a typo or syntax error in code. A bug can be an example in the documentation that does the opposite of what was intended. What is not a bug? Anything that adds functionality past what was intended in the code is a feature, and should be requested as such. Clarification of documentation or comments in code, extension of a protocol to include additional functionality, or support for a different model or card would all be features. FEATURES There are many features and requests that are made of the system. Please be as clear as you can as to what the feature is that you need, and why it should be given priority over other features currently in the queue. The developers will examine all feature requests, and at their discretion some may be implemented. Of course, if you supply code to implement the feature, it will be much more likely that it will be integrated into the codebase. See license section, below this text. LICENSE Please be aware that the Asterisk project, while Open Source under GPL, code and patches which are contributed for distribution with core Asterisk have additional requirements beyond the GPL. In order to prevent even the slightest possibility that a lawsuit could be brought against Digium (the primary sponsor, and holder of the copyright,) it is required that ALL patches and feature submitters have signed a waiver on the code that they submit. Before ANY patch is applied, you MUST sign and return either of the following document by fax, snail mail. Email only is unacceptable for legal reasons. http://www.digium.com/disclaimer.txt http://www.digium.com/disclaim.changes Personally, I'd like to thank everyone who has participated in fixing bugs and adding features. It's largly thanks to the feedback and assistance from the community that Asterisk has managed to become so powerful in such a short amount of time. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
RE: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
I agree in stopping the thread, but I do have one question... What would Qwest think of her posting to the list under a yahoo mail account representing her company, badmouthing this community, who, in the long run, could be VERY much worth their interest? Hmm Just my $.02 - no flames please. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: Saturday, January 03, 2004 8:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk. Hi All, Can we stop this thread pl. This lady has no intentions to learn asterisk. She is just a troll and wasting our time. With her corporate attitude, what she expects is support that available with paid commercial products. Her company has enough money to buy commercial products, let she go there. Hey lady, whoever u are, dont waste our time. this is not for u. Lets move on to something useful pl. -B - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 03, 2004 5:36 PM Subject: RE: [Asterisk-Users] New to asterisk? RUN... don't walk. On Sat, 2004-01-03 at 14:12, Me wrote: Mr. West, Sorry to burst your bubble, but that is not me. My name is Barbara Simpson. Either you are lying or someone is trying to remove any credibility from my original post. I now stand by my original post with more conviction than ever. You had little to no credibility when you show up acting like a troll from what most people would consider a throw away account. There were a lot of insightful replies. However, none of them were able to address the real problems of the asterisk community and come up with solutions. If you can't see your own faults, you are in for a bumpy ride. This is due to the problem residing in the general population, not the community. The problem resides in users who can't be bothered to either expend energy, or patience for the software to develop. Remember you came here, we didn't go recruiting you. So if you are disappointed in your experience, blame yourself for your expectations. As far as I can tell here, you haven't paid a single person for anything, so any help you have received has been at a cost to the other people of this community. So the solution is for you to grow up. You need to learn that the comment you have made in this thread are worthless as they don't advance anything here. If you want credibility in a technical forum, you will have to show some technical skills. Otherwise you will be cast aside and hopefully ignored. Barbara Simpson Qwest Voice Over Packet Services --- Brian West [EMAIL PROTECTED] wrote: You said it good Look what this person posted to my blog... Now thats what I call grown up. Date: Thu, 1 Jan 2004 10:10:24 -0600 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] IP Address: 24.10.200.168 Name: Jeff Sowery Email Address: [EMAIL PROTECTED] URL: Comments: You're a complete idiot. Grow a brain or at least some balls. -Jeff NEXT!!! bkw On Thu, 1 Jan 2004, JR Richardson wrote: Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness. This platform for learning is one of the best tools in existence to come to a finite understanding of VoIP and legacy telephony with the versatility to expand beyond and develop originality in the field of telecommunications excellence, product development. Learn it, understand it, appreciate it, then take it past where you found it and if you're capable contribute, if not, enjoy it. But always, always maintain respect for those who created it and continue to refine it. Learning is intrinsically human, and in this world of Industry (There is no substitution for knowledge. [Edward Deming]). Find your inner child, re-capture and embrace what God has given you, the ability to learn. It will require you to put down the remote control, get off the couch and decrease your apparently frequent visits to McDonalds. Search and find the knowledge which you seek to ultimately fulfill your destiny; build an Asterisk Server that works. Hell, we all did. JR Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL
RE: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk.
I agree in stopping the thread, but I do have one question... What would Qwest think of her posting to the list under a yahoo mail account representing her company, badmouthing this community, who, in the long run, could be VERY much worth their interest? Hmm Just my $.02 - no flames please. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Balaji NJL Sent: Saturday, January 03, 2004 8:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] STOP THIS THREAD New to asterisk? RUN... don't walk. Hi All, Can we stop this thread pl. This lady has no intentions to learn asterisk. She is just a troll and wasting our time. With her corporate attitude, what she expects is support that available with paid commercial products. Her company has enough money to buy commercial products, let she go there. Hey lady, whoever u are, dont waste our time. this is not for u. Lets move on to something useful pl. -B - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 03, 2004 5:36 PM Subject: RE: [Asterisk-Users] New to asterisk? RUN... don't walk. On Sat, 2004-01-03 at 14:12, Me wrote: Mr. West, Sorry to burst your bubble, but that is not me. My name is Barbara Simpson. Either you are lying or someone is trying to remove any credibility from my original post. I now stand by my original post with more conviction than ever. You had little to no credibility when you show up acting like a troll from what most people would consider a throw away account. There were a lot of insightful replies. However, none of them were able to address the real problems of the asterisk community and come up with solutions. If you can't see your own faults, you are in for a bumpy ride. This is due to the problem residing in the general population, not the community. The problem resides in users who can't be bothered to either expend energy, or patience for the software to develop. Remember you came here, we didn't go recruiting you. So if you are disappointed in your experience, blame yourself for your expectations. As far as I can tell here, you haven't paid a single person for anything, so any help you have received has been at a cost to the other people of this community. So the solution is for you to grow up. You need to learn that the comment you have made in this thread are worthless as they don't advance anything here. If you want credibility in a technical forum, you will have to show some technical skills. Otherwise you will be cast aside and hopefully ignored. Barbara Simpson Qwest Voice Over Packet Services --- Brian West [EMAIL PROTECTED] wrote: You said it good Look what this person posted to my blog... Now thats what I call grown up. Date: Thu, 1 Jan 2004 10:10:24 -0600 From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] IP Address: 24.10.200.168 Name: Jeff Sowery Email Address: [EMAIL PROTECTED] URL: Comments: You're a complete idiot. Grow a brain or at least some balls. -Jeff NEXT!!! bkw On Thu, 1 Jan 2004, JR Richardson wrote: Piping in 2 cents, This is a great example of the Internet, Fast Food generation, showing their appreciation for all the magic that happens in the labs, hearts and minds of the courageous, hard working, dedicated and motivated group of people truly interested and guided to accomplish greatness. This platform for learning is one of the best tools in existence to come to a finite understanding of VoIP and legacy telephony with the versatility to expand beyond and develop originality in the field of telecommunications excellence, product development. Learn it, understand it, appreciate it, then take it past where you found it and if you're capable contribute, if not, enjoy it. But always, always maintain respect for those who created it and continue to refine it. Learning is intrinsically human, and in this world of Industry (There is no substitution for knowledge. [Edward Deming]). Find your inner child, re-capture and embrace what God has given you, the ability to learn. It will require you to put down the remote control, get off the couch and decrease your apparently frequent visits to McDonalds. Search and find the knowledge which you seek to ultimately fulfill your destiny; build an Asterisk Server that works. Hell, we all did. JR Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL
RE: [Asterisk-Users] Re: Grandstream Early Dial
I've never had early dial working, however, I resolved my multiple digit issue by simply putting both the GS phones and asterisk in INFO mode. This worked on both 10.0.3.81 firmware on the budgetone and the ATA286, as well as 10.0.4.30 firmware. I'm not saying I don't believe you, but doubelcheck your lines in asterisk to be dtmfmode=info and the gs devices are on SIP INFO method, and your DTMF Payload type is 101. Just my $.02 -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen R. Besch Sent: Wednesday, December 31, 2003 12:59 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Grandstream Early Dial I've just checked my voicemail with 1.0.4.30 and get the same multiple digits problem. sip.conf and GS config are both at info, for me this is a new problem voicemail has always worked perfectly with the GS. This has come up many times in this list, with no consensus for a solution. According to Grandstream, the multiple digit problem arises from a difference in the interpretation of the SIP standard. I'm not sure I really understand this, so no flames please, but, paraphrasing a conversation I had with GS, apparently they retransmit the digit as long as the key is pressed and expect asterisk to know that it is a re-transmission by examining other data in the packet. Asterisk does not handle the SIP packet in the way GS expects, resulting in multiple digit transmission. This flaw (?) is avoided by setting DTMF to INBAND. Why this behaviour is not repeatable on everyones installations escapes me. However, I have noticed one thing that may be a clue. I have one phone that is older hardware (redial button instead of send and an unused battery compartment on the bottom). This phone behaves differently than all the other, later, models. For example, it is the only phone on which the flash button actually works to answer the alternate line (eg when an incoming call waiting call arrives). All phones are using 3.81 firmware. Early dial has never worked for me, and I just upgraded to the 1.0.4.30 load yesterday. Now, I am having DTMF recognition issues, making it impossible to check my voice mail. This is an acknowleged bug on the GS. They have connected to my * server and acknowleged the problem. A fix has been promised but not yet delivered. Until then, the only solution is to turn early dial off and let the phone send the entire dial string in one packet. Since this does not affect later single digit transmission for IVR's, etc, the only consequence is the irritating delay between the last entered digit and the actual placing of the call. But, you can always hit the send key. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.554 / Virus Database: 346 - Release Date: 12/20/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.554 / Virus Database: 346 - Release Date: 12/20/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New to asterisk? RUN... don't walk.
Well, since everyone else is top-quoting on this message, so will I :P I'm no veteran either. As a matter of fact, I have had ZERO prior knowledge to the telcom industry or more than 'user level' experience with telecommunications in general. I decided that I wanted to expand my knowledge, and actually LEARN a few things, so I jumped into asterisk. I was, and quite frankly, IMO, still AM a 'n00b' to *. However, after playing around, and learning what things do, by reading the documentation that IS there, searching the archives, and just trolling the list and IRC, I have learned more in the last 4-5 months of having * than a lot of people I've noticed have learned in a lifetime of experience.I now have a fully functional (well, minus MOH, because mpg123 isn't yet compiled on my new box), * implementation, serving myself and my roommates strictly over VoIP, and a couple ata's and a Internet PhoneJack card. I love it. And I'm STILL learning to this date. Asterisk is not something you can expect everyone to just drop what their doing and help you with. Sure, it can be frustrating, but if you are so dense that you can't sit down an play with it and learn what happens when you type something in the cli, or change a few things in your dialplan, then get out, I agree. If you liked taking apart mom's hairdryer as a kid and seeing how it worked, and then later on, rewired up a few things to do what you wanted them to, or even took a hex editor to command.com in msdos to change what it says to suit your taste (mucho guilty on that one.. lol), then you will have no problem finding out what you can and can't change simply by editing files, and trying things out. Take off your training wheels, and just TRY IT. - Josh R. [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of SW Sent: Wednesday, December 31, 2003 4:13 PM To: [EMAIL PROTECTED] Digium. Com Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Hello, I am not a veteran here, but would like to share my thoughts on this subject. True, * is opensource and freely available, but it is not a computer program that you download and run. It is a very versatile telecommunication product you would otherwise pay at least 100 K to buy from a telecom vendor, if not more based on modules and usage, license hash-codes etc. Even to try * one would need some pre requisite knowledge in telecom, if not many years in the field. I work for a large telecom company and my specialty is voice over broadband (or xDSL). I worked with asterisk for couple of months now and I am amazed to see areas of telecom that * touch upon with. Starting from Linux, to SIP, H323, DSL technologies (PPP, PPPoE, PPPoA, DHCP, NAT), Call routing(Dial Plan), IVR, Transcoding, STUN are few areas that one would have to master even thinking about *. True one would know the syntax, and howtos etc, but also would have to have the ability to troubleshoot. For last two-three months in this list, I have not seen any newbi posting a sip trace (from a ethereal or a TCP dump) and asking a question about it. I have seen many question for instance, asking syntax of h.323 dial, but never seen a question asked on a h323 trace. I think, having * openly available is like keeping an airplane openly available in a airfield, so that anybody can try flying. Tell me how many of us would go try and fly that airplane if we do not know how to fly :) Point that I want to make here is simple, please try to understand what * is all about. If you like it's features and would like it to run in a production environment try to get some professional help. If you are learning these technologies for fun then get educated, use tools available to troubleshoot. Hooking up couple of phones and making a call is far from knowing *. Asterisk is a great product (thanks Mark and many others) and if you know what you are doing, you can do wonders with it. Don't put it down, because you do not have the background to understand it or work with it. Cheers SW Message: 4 Date: Wed, 31 Dec 2003 12:37:24 -0800 (PST) From: Me [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] New to asterisk? RUN... don't walk. Reply-To: [EMAIL PROTECTED] As a newcomer to Asterisk, you will not be welcomed with open arms. First, you will find almost no documentation on it's features. Second, if you try to ask questions, you will be flamed and pointed to worthless how-tos and 'the wiki'. These worthless documents can only be useful for explaining how things work to those already in-the-know. Lastly, Asterisk is so bug ridden, expect frequent segmentation faults. With a community so 'anti-n00b', don't expect your problems to be fixed anytime soon. RUN!!! Don't walk... away from Aterisk. ___ Asterisk-Users mailing
RE: [Asterisk-Users] XBOX as and * Dedicated Server
I have done this, but I haven't put the server in place yet... It appears to run absolutely fabulous, with the exception that OSS/dsp is noisy as all get-out. Alsa drivers tend to fix this problem, though. Other than that, I can pass at least 5-10 calls through with no problem whatsoever. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Miguel Cavazos Sent: Thursday, December 04, 2003 6:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] XBOX as and * Dedicated Server Hello guys, i have been on this mailing list for some weeks now, and i was wondering if someone here has installed linux on the XBOX and use it as a dedicated server. Its a 200 USD computer and could make it perfect to asterisk, its little and doesnt really take much space. My question is could this make it for a stable server??? here are some links i found for linux on XBOX http://xbox-linux.sourceforge.net/ some intresting screenshots found on that URL http://xbox-linux.sourceforge.net/docs/screenshots.html The only real thing that i dont know is where am i going to put the X100p. Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] XBOX as and * Dedicated Server
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Friday, December 05, 2003 5:18 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] XBOX as and * Dedicated Server Miguel Cavazos wrote: On Fri, 2003-12-05 at 07:13, Steven Critchfield wrote: During Phreaknic, Mark was showing off a Xbox running asterisk with 4 S100U interfaces connected to the game ports on the front. It was interesting. In the end, I don't think it is cost effective as a real PC since you can also build a PC of similar or better specs for that price now and you get PCI slots. the S100U is a good idea, and yes you can get a pc for what 30bucks a P200, but i was looking for something small and good looking, i dont have a big room and another CPU. Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users I like you idea. Very Cool :) Is RAM upgradable on xbox? Thanks Yes the ram is upgradable... *IF* you can do extremely small surface mount soldering. You can upgrade it to a whopping 128 megs. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cisco 7960 power suplies?
Also, I see them on eBay all the time for around $35 US. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lists Sent: Sunday, November 30, 2003 5:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cisco 7960 power suplies? Does anyone know where to get cisco 7960 power suplies? What should they cost? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.545 / Virus Database: 339 - Release Date: 11/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Van Donselaar
Sent: Tuesday, November 18, 2003 10:12 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Updated iaxComm binaries available for
WinXP,
Red Hat 9.0
On Tue, 18 Nov 2003 17:02:42 +0200, Dan [EMAIL PROTECTED] wrote:
Hi,
Tried on WinXP Pro and it loads, but in the background (no window).
There is something needed from the wxWindows package to just run the
executable?
Nothing needed from the wxWindows package. I think it's because it
can't
find
the rc directory.
I'm sorry that I didn't put this in the README. Bad coder. No donut.
You must run iaxComm from the installation directory beacuse it looks
for
rc
files in ${cwd}/rc.
Steve put an error dialog on failure in the CVS sources, but I'm
working
on a
better solution.
Please let me know if this solves it, or if the problem lies
elsewhere.
Nope still crashes on XP on load. Ran from directory extracted to, etc.
Below are crash details:
AppName: iaxcomm.exe AppVer: 0.0.0.0 ModName: iaxcomm.exe
ModVer: 0.0.0.0 Offset: 0008e98c
Don't know if that helps any at all, but the other details screen is WAY
too long to attach.
---
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RE: [Asterisk-Users] MeetMe problem
Also, unless something has changed, If you don't have any zap devices, you'll need to have the ztdummy module loaded to provide zap timing to the meetme app. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Saturday, November 15, 2003 8:59 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MeetMe problem On Saturday 15 November 2003 08:59, [EMAIL PROTECTED] wrote: Hi guys, Having a bit of a problem trying to get conference bridges working. In my meetme.conf file I have the following line [rooms] conf = 6000 In my extensions.conf file I have: exten = 1000,1,MeetMe,6000 My problem is that when I dial into extension 1000 it is telling me this is not a valid conference number. Can anybody telling me what I'm doing wrong here? You need to do a restart after defining new conference numbers, otherwise they won't work (i.e. not on a reload). -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.538 / Virus Database: 333 - Release Date: 11/10/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.538 / Virus Database: 333 - Release Date: 11/10/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Graphical Interface
I would like to propose the name astmaster control in all seriousness. I agree, this isnt a name for an actual possible business implementation, but I think it has a nice ring to it for a project name.. J -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Friday, November 14, 2003 8:57 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Graphical Interface Hello, I don't have a project name yet, any suggestions? What in your mind should a full client app have in it? This program pretty much has everything that my company needs from a client app in it. What other things (within the limitations of a Zap/Sip Asterisk system with unmodified source code) need to be added to it to make it complete? MATT--- -Original Message- From: marin blu [mailto:[EMAIL PROTECTED] Sent: Friday, November 14, 2003 9:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Graphical Interface Hi, What is the project name? Do you thing thatyour project could be a step to a full client appl ? Best Regards, Marin Blu mattf [EMAIL PROTECTED] wrote: Hello, I have developed a graphical interface using Perl/TK that has the following features: I'm still cleaning up the client code, but it will be released before the end of the month on Sourceforge. Here are some of the things I have added to the code: - Recording of any Zap channel by extension they are connected to at the click of a button - A refreshing list of active Zap channels - dialing a number by entering in a number or selecting from a list of recently dialed numbers and clicking a DIAL button - Asterisk based conference-calling of up to 6 external channels(even on single-line phone) - Admin section that allows you to Hangup any Zap channel at the click of a button - Call Parking and retrieval from specific extensions - Runs on Linux and Windows On the server side you will need a MySQL server, a couple AGI scripts and some custom dialplan extensions, but the Asterisk code itself is unaltered. On the Client side you just need to have perl and Tk/tcl modules installed on Linux and on windows you just need Activestate perl, you also need to make sure you have the Net:Telnet and Net::MySQL perl modules loaded on both(these are easy to get and have no prerequisites). MATT--- -Original Message- From: David Winkler [mailto:[EMAIL PROTECTED] Sent: Thursday, November 13, 2003 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Graphical Interface Hello. Was just curious to know if anyone is working on a graphical interface to Asterisk using X windows, or something else similar. Thanks! David Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003
[Asterisk-Users] I hate to do this but..
I hate to bring this thread back to life, but... it may be possible to get it supported, do you think the price point is remotely competitive with Digium hardware? Also as I am not about to divulge my information to them to look in the downloads section, what is the licensing of their SDK? What is the licensing of the driver? Steven On Tue, 2002-11-26 at 14:52, Jamin W. Collins wrote: Is there any current/planned support for Aculab hardware? http://www.aculab.com Looks like they have Linux drivers and an SDK. Has any advancement taken place in this? Has someone developed a working channel driver for this product? I have one, and would like to see if it would be a possibility to get working... Thanks -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] grandstream ntp
I've noticed the same problem on the BT-102. I would also like to know this... (cc'ed grandstream to get their opinion) -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Rodger Sent: Friday, November 07, 2003 7:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] grandstream ntp I am running ntpd on the same machine as asterisk in order for the grandstream phones to display the time. After a while the time display fails until the phone is re-booted. Has anyone run into this problem before? Is it simply a bug in the GS firmware? Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.532 / Virus Database: 326 - Release Date: 10/27/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Beta testers for visual configuration tool for asterisk
I would like to beta test this tool. :) Looks like it could be a good thing. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paulo Mannheimer Sent: Friday, October 17, 2003 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Beta testers for visual configuration tool for asterisk Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland doesn't crash because of a missing sound file. Beta users will have to download and install either a 16 Mb or a 4Mb Windows program, depending if you already have or not JRE 1.4.2 installed. This client works together with a tomcat-based application, which will be running on our servers during the trial. If you wish to participate, please let me know off-list. I'll get in touch with the first 5 answers to arrange how the test will be performed. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Ringing from PSTN
Well, the ATA uses SIP to communicate with the * box. SIP by default doesn't generate a ringing indicator when the far side is ringing, you indeed DO have to tell it to ring, using the r flag in the extension. **Note, this is just from my experience. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Dolloff Sent: Thursday, October 09, 2003 3:17 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringing from PSTN That does make a ringing sound, but any idea what's causing the problem? Stephen Subject: Re: [Asterisk-Users] No Ringing from PSTN You can send a fake ring by using something like: exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],20,r) Assuming the ATA is in the sip.conf as [1234] However, this does NOT solve the underlying problem. On Thu, 2003-10-09 at 15:29, Steve Dolloff wrote: Here is my Configuration PSTN - Cisco AS5350 - SIP - ASTERISK - SIP - ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] eBay Sip Phone Scam.
Ok, see, now you're confusing what I said. Nowhere did I say I had the 102D. I said he never mentioned that it was the 102, irregardless of the D. I *DO* have the 101, which is what he was talking about. No, it doesn't mention it's the 101. This argument has now proved silly, especially since you're confusing what I'm saying, with what he supposedly is. *I CLAIM END OF THREAD!* -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Thursday, October 02, 2003 7:04 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Ok, in addition you are confusing the 102 with the 102D. If you had done your homework you would have noticed that the 102D (see the big D?) is a different model. Than one has the 16x2 LCD and 3 way conferencing. I spent a lot of time studying these phones. So no, you don't have that phone. check http://www.chagres.net -- Original Message -- From: Josh Roberson [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Thu, 2 Oct 2003 07:31:43 -0500 My bad... It's a .net, not a .com :P Oops... Sorry JMB (sheepish grin) http://www.chagres.net -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Low, Adam Sent: Thursday, October 02, 2003 6:22 AM To: '[EMAIL PROTECTED] ' Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Josh, Pls can you confirm that URL, www.chagres.com doesn't seem to mention the sale of any Grandstream phones Adam -Original Message- From: Josh Roberson To: [EMAIL PROTECTED] Sent: 02/10/03 13:04 Subject: RE: [Asterisk-Users] eBay Sip Phone Scam. Actually, had you taken the time to READ the auction details, He says (direct copy/paste from auction) -Begin Copy/Paste- Flash Based OS Easy to install and manage, Cost effective, Easy to use - Friendly GUI for 1st time user, Easy to learn - User's guide and on-line tutorial Big information and management LCD blue back light User friendly keypad Universal AC/DC adapter Ergonomic design 25-button keypad 12-digit caller ID LCD Universal Switching Power Adaptor Input: 100-240VAC Output: +5VDC, 400mA, 1. Auto-sensing 10/100 Base-TX Ethernet Port 2.UL/CE/FCC 3.Power Supply : Universal 90 ~ 264V Support all major Network Operating Systems (Windows, MAC, Linux/Unix) Web-Based Management TCP/IP Configuration with DHCP support Free Flash Firmware update No User Licenses System Restart/ Shutdown Password Access control 1 x 10/100Mbps Ethernet Port (RJ-45 Interface) Support STUN and SIMPLE extension Interoperable with 3d parties Proxy, Registrar and gateway products DSP technology for the best voice quality Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (alaw and u-law), G.726 (40K/32K/24K/16K), as well as G.728 (Model 102D) In and out-off-band DTMF Support 3-way conferencing (Model 102D), full duplex hands-free speakerphone, redial, call log, volume control, voice mail with indicator Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Remote software upgrade capability via TFTP Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control) -End Copy/Paste- Nowhere does he make the claims you're stating. He DOES, have (Model 102D) in one of the descriptions, but that is a direct quote from Grandstream's product brochure. Also, this phone *IS* out on the market.. I own one, and I'm quite happy with it.. I will tell you this though: Go order one from Chagres (http://www.chagres.com). They are an asterisk supporter/user on this list, and the price is MUCH better. ;) /rant -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Thursday, October 02, 2003 4:50 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] eBay Sip Phone Scam. I did shop around. Nowhere does he say the phone is the 101. If you look at his ad he says the phone has 102D features and has 16x2 lines and 3 way conference. The starting price was $90. A reasonable opening price I thought. He also does not say the phone is not available until end of year. I only called Grandstream to find out some info on it after I placed the Bid. In a way Grandstream is also at fault. Nowhere do they say the phone is not available. I was suprised when they told me it wasnt even out. When I sent a message to this thief, he said its for the 101 and they are hard to get. Thats why he jacked up the price. He did cancel my bid after telling me what a bad
RE: [Asterisk-Users] IAX and IAXTEL
Well, that's odd.. Can you, then, with IAX, determine in which section (first, second, last, etc...) you read your configuration in iax.conf, rather than matching up with passwords? -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: Thursday, October 02, 2003 7:16 AM To: ASTERISK USERS Subject: Re: [Asterisk-Users] IAX and IAXTEL The location of the guest / iaxtel section having to be at the end is, as it turns out, a configuration error on iaxtel. I hope to have it straightened out shortly. Mark On Thu, 2 Oct 2003, Bartosz Jozwiak wrote: Sometime yes sometimes no :) But thats the life :) Ok but I fixed it. Just put the guest section in iax.conf all the way on the end. And right now it works for me. :) -- Bart - Original Message - From: bill black [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 02, 2003 12:47 AM Subject: Re: [Asterisk-Users] IAX and IAXTEL Hello Bart: Did anyone ever follow up to your question? I have the same issue. thanks, Bill On Wednesday 01 October 2003 07:27, Bartosz Jozwiak wrote: Hello, Could somebody tell me what I should change in iax.conf file to be able to receive calls from iaxtel. I am already registered and I can make calls to IAXtel users but what I should do in iax.conf to be able to receive call also. -- Bart ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.516 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR Web Search Frontend
Since * and MySQL have had a licensing scuffle, is there a way to set it up so that we can specify wether or not it's in the mysql database, or use the plaintext file that * generates with cdr_csv.so? Just a thought.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jamie Carl Sent: Saturday, September 27, 2003 1:42 AM To: Asterisk Users (E-mail); Asterisk Dev (E-mail) Subject: RE: [Asterisk-Users] CDR Web Search Frontend *This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, New versions available. Now written in PHP with totals for Billing Seconds and Duration. Help yourselves and please send me more suggestions!!! Thanx! J -Original Message- From: Dimitri Bellini [mailto:[EMAIL PROTECTED] Sent: Friday, 26 September 2003 10:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CDR Web Search Frontend *This message was transferred with a trial version of CommuniGate(tm) Pro* Hi Carl i see web frontend i action is very good!! The total time at end is good thing. Thanks for great work. Can you put the script in some place to download. Dimitri *This message was transferred with a trial version of CommuniGate(tm) Pro* Hey all, I've just done a quick (but functional) web front end for searching the CDRs in a MySQL database. Anyone interested in trying it out? I'm wondering what to add to it next. So far you can seach using source, destination, CLI, channel and date ranges. It also displays ALL fields in the database table. If interested, email me on [EMAIL PROTECTED] Do not reply directly to this email, it will bounce. Depending on the level of interest, I may post this somewhere for your free downloading pleasure. Regards, Jamie Carl Jazz Inc. http://www.jazz-inc.net Email: [EMAIL PROTECTED] JID: [EMAIL PROTECTED] Phone: +61-414-365466 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR Web Search Frontend
You wanted suggestions, didn't you? Well, you got them! :P Also another major search function... Allow to search via account code, for accounting purposes (obviously). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jamie Carl Sent: Saturday, September 27, 2003 8:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CDR Web Search Frontend *This message was transferred with a trial version of CommuniGate(tm) Pro* Gimmie a break, I only learnt PHP yesterday.. :) J - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 27, 2003 10:13 PM Subject: Re: [Asterisk-Users] CDR Web Search Frontend *This message was transferred with a trial version of CommuniGate(tm) Pro* Since * and MySQL have had a licensing scuffle, is there a way to set it up so that we can specify wether or not it's in the mysql database, or use the plaintext file that * generates with cdr_csv.so? Or do something really smart like the Perl guys and have a backend-mostly-independent DB infrastructure. Hell I think that PHP finally smartened up and went this way, too. Regards, Andrew ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxtel and iax.conf
Bkw, Yes, and we have documented a bug on it... seems to only happen with the iaxtel entry, so we're not sure if it's IAXTel that's at fault, or a bug with * causing iaxtel to read the wrong entry. http://bugs.digium.com/bug_view_page.php?bug_id=296 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Tuesday, September 23, 2003 12:34 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] iaxtel and iax.conf I have tried for over a month off and on to get iaxtel for inbound to work... and tonight after alot of troubleshooting we noticed this: iaxtel inbound will use the last entry in your iax.conf to auth against. So if [iaxtel] is at the top and say [voicepulse] at the bottom. An inbound call will try to auth against that [voicepulse] entry even with the [iaxtel] entry at the top of the file. Has anyone else seen this happen? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
FW: RE: [Asterisk-Users] iaxtel and iax.conf (HTML CONTENT, FYI)
I hate to post HTML to the list, but I refuse to respond to this, and I would like to say that whomever is using this service is kinda stupid for subscribing an email address to the list using this service. I hope they learn a lesson by us, the list users, NOT responding to this, and eventually, after not receiving any list mail, theyll wonder hmm why is this list so dead? Just my .02. Again, sorry for the html post. -Original Message- From: AntiSpam UOL [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 23, 2003 3:27 AM To: [EMAIL PROTECTED] Subject: RE:RE: [Asterisk-Users] iaxtel and iax.conf Olá, Você enviou uma mensagem para [EMAIL PROTECTED] Para que sua mensagem seja encaminhada, por favor, clique aqui Esta confirmação é necessária porque [EMAIL PROTECTED] usa o Antispam UOL, um programa que elimina mensagens enviadas por robôs, como pornografia, propaganda e correntes. As próximas mensagens enviadas para [EMAIL PROTECTED] não precisarão ser confirmadas*. *Caso você receba outro pedido de confirmação, por favor, peça para [EMAIL PROTECTED] incluí-lo em sua lista de autorizados. Atenção! Se você não conseguir clicar no atalho acima, acesse este endereço: http://tira-teima.as.uol.com.br/challengeSender.html?data=""> Hi, You´ve just sent a message to [EMAIL PROTECTED] In order to confirm the sent message, please click here This confirmation is necessary because [EMAIL PROTECTED] uses Antispam UOL, a service that avoids unwanted messages like advertising, pornography, viruses, and spams. Other messages sent to [EMAIL PROTECTED] won't need to be confirmed*. *If you receive another confirmation request, please ask [EMAIL PROTECTED] to include you in his/her authorized e-mail list. Warning! If the link doesn´t work, please copy the address below and paste it on your browser: http://tira-teima.as.uol.com.br/challengeSender.html?data=""> Use o AntiSpam UOL e proteja sua caixa postal --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003
RE: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
This may just be me, but When replying to a message from a digest, it would be proper to remove all the context except that to which you are replying so as not to have to scroll an entire mile to see your reply. I know if I was the person you were replying to, I probably wouldn't scroll all the way through the other 15 messages just to see a reply. Just my .02, Sorry if I seem a bit irrational, just irritated. -Josh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shimul Kanti Barua Sent: Wednesday, September 17, 2003 4:21 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs BIG OLE SNIP Message: 16 Date: Sat, 13 Sep 2003 16:32:32 +0300 From: Michael Manousos [EMAIL PROTECTED] Organization: inAccess Networks To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk using a h323 gateway Reply-To: [EMAIL PROTECTED] Cerrajetto wrote: Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 - PSTN gw)? - Asterisk ip: 192.168.1.10 - h323-PSTN gw: 192.168.1.20 I've tried: exten = _9,1,Dial(OH323/192.1.1.20) or exten = _9,1,Dial(OH323/[EMAIL PROTECTED]) I guess that 192.1.1.20 is a typo, right? You will have to give more info in order to be able to find the problem. Try to set these params in oh323.conf file: wrapLibTraceLevel=3 libTraceLevel=3 libTraceFile=/tmp/trace.txt Rerun and send me the /tmp/trace.txt file, oh323.conf and the screen log (off-list). but it does not work at all. If my h323 client directly uses 192.168.1.20 as h323 gateway, the calls are routed to the PSTN perfectly. What is the correct way to route some calls from Asterisk to another h323 gateway? Thank you, Mark Michael. Hi Mark, Yes, it is possible. I have test it with Asterisk and oh323. We have routed some calls thru a second h323 gateway (like Vegastream and Cirilium). Following is the configuration: ; Vegastream exten = _01XX,1,Dial(OH323/[EMAIL PROTECTED]) ; Crilium - exten = _9XX,1,Dial(OH323/[EMAIL PROTECTED]) Shimul --__--__-- ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer
Strange.. I had a symbolic link, and it wouldn't work. After I finally got it working properly, i even tried to remove it from /usr/bin and symlink it, and it wouldn't work again... couldn't for the life of me figure out why. - Original Message - From: Joseph Finley [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 4:20 PM Subject: RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer I used a symbolic link and it works just fine for me. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson Sent: Wednesday, September 03, 2003 4:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hello Mickey, I had a similar problem with the mp3 functions a while back, but I handled it off list, but since you're having the same issue, here's how I noted to fix it: 1. Make sure you have mpg123 in /usr/bin. Symbolic links will NOT work, and it has to be the REAL mpg123. 2. Make sure that the system has already passed the Answer call for the extension. For example: exten = 69,1,Wait(5) exten = 69,2,Answer exten = 69,3,MP3Player,/path/to/music.mp3 This example is the only way I found to make the mp3 player work. I haven't been able to test fully the music on hold functionality, as my system is'nt fully functional yet, and I don't have other clients to test with. -Josh - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist (E-mail) [EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 11:13 AM Subject: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback(file_which_dont_exist) just before the Moh or MP3Player I can hear the music. Actually I observed the same behavior internally when I used H323 for my Welltech Wellgates (which I have now changed to SIP). What can cause this kind of problem? Its not a huge issue since I can use the Playback to trigger the call, but it would be nice to find the source of the problem. regards Mickey Binder ___ 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] MusicOnHold and MP3Player not triggering answer
Along the same lines, though, I do agree that MP3Player app should cause the system to trigger the answer without having to do it manually, but I could see where you might not want it to, as well. In theory, the Playback app (pardon, this is what i've gathered by toying with it) triggers the 'answer' function if the call is not already answered. Couldn't we get the MP3Player app to do the same? I'm not that skilled of a programmer, otherwise, I'd hack it up and do it myself. -Josh - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 04, 2003 4:34 AM Subject: RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer -Original Message- From: Joseph Finley [mailto:[EMAIL PROTECTED] Sent: 3. september 2003 23:21 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer I used a symbolic link and it works just fine for me. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson Sent: Wednesday, September 03, 2003 4:30 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hello Mickey, I had a similar problem with the mp3 functions a while back, but I handled it off list, but since you're having the same issue, here's how I noted to fix it: 1. Make sure you have mpg123 in /usr/bin. Symbolic links will NOT work, and it has to be the REAL mpg123. 2. Make sure that the system has already passed the Answer call for the extension. For example: exten = 69,1,Wait(5) exten = 69,2,Answer exten = 69,3,MP3Player,/path/to/music.mp3 This example is the only way I found to make the mp3 player work. I haven't been able to test fully the music on hold functionality, as my system is'nt fully functional yet, and I don't have other clients to test with. -Josh Ok I get same results when using Answer, so I'll just stick with that thx Mickey - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist (E-mail) [EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 11:13 AM Subject: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback(file_which_dont_exist) just before the Moh or MP3Player I can hear the music. Actually I observed the same behavior internally when I used H323 for my Welltech Wellgates (which I have now changed to SIP). What can cause this kind of problem? Its not a huge issue since I can use the Playback to trigger the call, but it would be nice to find the source of the problem. regards Mickey Binder ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer
Hello Mickey, I had a similar problem with the mp3 functions a while back, but I handled it off list, but since you're having the same issue, here's how I noted to fix it: 1. Make sure you have mpg123 in /usr/bin. Symbolic links will NOT work, and it has to be the REAL mpg123. 2. Make sure that the system has already passed the Answer call for the extension. For example: exten = 69,1,Wait(5) exten = 69,2,Answer exten = 69,3,MP3Player,/path/to/music.mp3 This example is the only way I found to make the mp3 player work. I haven't been able to test fully the music on hold functionality, as my system is'nt fully functional yet, and I don't have other clients to test with. -Josh - Original Message - From: Mickey Binder [EMAIL PROTECTED] To: Asterisk maillist (E-mail) [EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 11:13 AM Subject: [Asterisk-Users] MusicOnHold and MP3Player not triggering answer Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback(file_which_dont_exist) just before the Moh or MP3Player I can hear the music. Actually I observed the same behavior internally when I used H323 for my Welltech Wellgates (which I have now changed to SIP). What can cause this kind of problem? Its not a huge issue since I can use the Playback to trigger the call, but it would be nice to find the source of the problem. regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
I downlaoded it and tried it, SIPPS. Nice featureful sip client, however, I haven't been able to get it to pass dtmf to *. I don't know if this is a software restriction or not, but I have emailed nero asking them for their opinion of this, as it is, in my case, a LARGE restriction when trying to deal with IVR's, and esp. * voicemail. - Original Message - From: Dave Packham [EMAIL PROTECTED] To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Monday, September 01, 2003 10:23 PM Subject: [Asterisk-Users] Sip Software from Nero Folk? http://www.nero.com/us/631911127302064.html Have you all seen this? Its a SIP softphone put out by the people that do the CD burning software Nero... Check it out it works with * Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Software from Nero Folk?
Actually, I do have that. i've tried inband, as well as rfc2883. Neither work. I'm going back and forth with ahead software on the issue, and they're doing a little bit of looking into it. Doesn't even work when clicking on the numbers, as required by the software, as someone else pointed out, that was an obvious feature i noticed right off the bat. -Josh - Original Message - From: Gavin Hollinger [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 02, 2003 1:58 AM Subject: Re: [Asterisk-Users] Sip Software from Nero Folk? haven't been able to get it to pass dtmf to *. I don't know if this Do you have dtmfmode=inband in sip.conf? http://www.sippstar.com/en/631927444894185.html Q.: DTMF generated by SIPPS is not recognized by other applications. SIPPS generates DTMF based on the standard set-op for DTMF for PSTN telephones. SIPPS transmits DTMF as tones and not as events. Hence, any application awaiting an event instead of a tone will not be able to work with SIPPS ___ 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] OT: My congestion music.
Wanna cheap laugh? IAXTel: 17005334094. -Josh
[Asterisk-Users] Dialogic cards...
Are the dialogic DTI series cards supported in asterisk? I know there's standard API, but I don't know if it supports only the cards listed on the digium site, or if it will support ALL dialogic cards.. Sorry, I *AM* a newbie to this stuff, just trying to get my hands on a good card. Thanks.
[Asterisk-Users] Brooktrout PRI-ISA48 card... info..
I have the option to purchase an Brooktrout PRI-ISA48 dual-span T1 card, which, upon checking with brooktrout, is supported for linux 2.x, but before I do this, I want to check and see what the opinions of your, the list members, and Mark, of course, as far as asterisk being able to use this card. ANY information would be helpful, as this offer will expire to me very soon. Thank you. :)
Re: [Asterisk-Users] Brooktrout PRI-ISA48 card... info..
It did I think, however I do still have an ISA slot to use My question was, will it work with asterisk? - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, August 19, 2003 10:44 AM Subject: Re: [Asterisk-Users] Brooktrout PRI-ISA48 card... info.. On Tue, 2003-08-19 at 04:08, Josh Roberson wrote: I have the option to purchase an Brooktrout PRI-ISA48 dual-span T1 card, which, upon checking with brooktrout, is supported for linux 2.x, but before I do this, I want to check and see what the opinions of your, the list members, and Mark, of course, as far as asterisk being able to use this card. ANY information would be helpful, as this offer will expire to me very soon. It is ISA. Didn't the ISA bus go away with PC99 specs? It is getting extremely rare to find ISA motherboards now days. Don't waste your time on it. If you consider your time with more than minimum wage then the time spent making it work will be more than buying a digium card. -- 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] 403 FORBIDDEN Help!
is the sip extension on the vocal sip server also 1234? if not, that could be why it's not working... when you're dialing sip, you have to use the format: exten = LOCEXT,1,Dial(SIP/[EMAIL PROTECTED]:port) so it would be something like exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]:5060) where REMEXT is the sip extension you're trying to dial. pardon me with the context stuff, i just woke up recently, and didn't think to ask if the remote extension was the same. - Original Message - From: Bartosz Jozwiak To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 9:50 AM Subject: Re: [Asterisk-Users] 403 FORBIDDEN Help! Asterix PBX is loggin to Vocal and the extension number is also loggin on the same vocal server. I cannot make it work :( - Original Message - From: Josh Roberson To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 11:43 AM Subject: Re: [Asterisk-Users] 403 FORBIDDEN Help! I'm new too, but alot of my 403 forbidden messages when adding extensions were due to context rules.. make sure that the client dialing the extension is included in the same context your extension is in. just my thoughts on it, as it resolved a lot of 403 errors for me. - Original Message - From: Bartosz Jozwiak To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 9:31 AM Subject: [Asterisk-Users] 403 FORBIDDEN Help! Hello, I have a question. I set up an extension to 1234 exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]:5060) And when I dial that extension I got in SIP message "403 FORBIDDEN" Can somebody tell me why I cannot call that extension? When I am not using Asterisk I can call that extension without any problems. My SIP proxy is VOCAL. I am new here so I do not know a lot yet. Thank you in advance. Bartosz Jozwiak
