RE: [asterisk-users] Security
I make an namp search and the result was its was talking to irc server ,is it normal [EMAIL PROTECTED] ~]# nmap 65.119.192.235 Starting nmap 3.70 ( http://www.insecure.org/nmap/ ) at 2006-08-17 02:34 E Interesting ports on ftp2.c3icare.com (65.119.192.235): (The 1650 ports scanned but not shown below are in state: closed) PORT STATE SERVICE 21/tcp open ftp 135/tcp filtered msrpc 139/tcp filtered netbios-ssn 445/tcp filtered microsoft-ds 990/tcp open ftps 1000/tcp open cadlock 1025/tcp open NFS-or-IIS 3389/tcp open ms-term-serv 4899/tcp open radmin 1/tcp open snet-sensor-mgmt Nmap run completed -- 1 IP address (1 host up) scanned in 352.570 seconds [EMAIL PROTECTED] ~]# nmap 83.133.127.60 Starting nmap 3.70 ( http://www.insecure.org/nmap/ ) at 2006-08-17 02:42 E Interesting ports on webdiscounter.ch (83.133.127.60): (The 1651 ports scanned but not shown below are in state: closed) PORT STATE SERVICE 22/tcp open ssh 135/tcp filtered msrpc 139/tcp filtered netbios-ssn 445/tcp filtered microsoft-ds 1422/tcp open autodesk-lm /tcp open irc-serv 6667/tcp open irc 6668/tcp open irc 7000/tcp open afs3-fileserver Nmap run completed -- 1 IP address (1 host up) scanned in 496.802 seconds From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Khaled Chehab Sent: Thursday, August 17, 2006 9:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Subject: [asterisk-users] Security Dear I am using [EMAIL PROTECTED] V 2.6.I used to sniff the server for 24 hours ,the result was that my server is talking to another servers through port 80 and 22 and 1000,… tcp Ips are 4.254.167.67 65.119.192.235 83.133.127.60 Is there any backdoor or a Trojan ,and how can I fix or debug it to be able to stop it, Remark: My hard drive and my network is completely cleaned from viruses,I used to burn a new image and I get these results . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: what is the real use of AEL?
On Thu, 17 Aug 2006 04:02:15 +0400 , Jean-Michel Hiver wrote:
> On Wed, 2006-08-16 at 18:30 -0700, asterisk-users-
> [EMAIL PROTECTED] wrote:
> > >Steve Murphy joined our development team at the beginning of June. Steve
> > >(murf on IRC/Mantis) had rewritten Asterisk's expression parser and the
> > >AEL language parser as a volunteer community member, along with various
> > >other bug fixes and improvements.
> > >
> > >
> > Which makes me think, what is the real use of AEL. I have taken a look
> > at it, and it makes asterisk's config file almost as unreadable as SER.
> >
> > What exactly does AEL do that a well written AGI / FastAGI app doesn't?
> >
> > I would think (but I'm surely wrong) that it would be better to do work
> > on having well defined APIs that allow us to script Asterisk (such as
> > AGI and the Manager interface) rather than invent Yet Another Pseudo
> > Programming Language - which is going to be an endless task... Don't you
> > think?
> >
> > That being said, just like the rest of the community, I'm very happy
> > with Kevin's exciting announcement!
> >
> > Cheers,
> > Jean-Michel.
In the above, Jean-Michel puts it right on the table: of what
possible use is AEL? Why am I bothering to waste my time with it? It's a
valid question! It deserves some discussion!
(For those of you who don't have any idea about what this discussion is
about, find out all about AEL via the web page,
http://www.voip-info.org/wiki/view/Asterisk+AEL2 )
Many thanks to rushowr for his reply in my defense as pertaining to
speed and readability. Actually, it might be interesting to get a
handle on the real speed of the AEL extension engine vs. the AGI program
at the other end of the pipe. Here is a little AEL script:
context sppedtest {
771 => { // Get the PIPS (priority instructions per second)
// for your current asterisk box.
// perform this on quiescent asterisk box --
// "all your CPU cycles are belong to us"
iterations=10; // Choose iterations wisely--
// around 10 sec. of execution time
// is good, gives good granularity.
Set(time1=${EPOCH});
for(i=1; ${i} < ${iterations}; i = ${i} + 1)
NoOp(Hello);
Set(time2=${EPOCH});
Verbose(The time diff is $[ ${time2} - ${time1} ] seconds);
Verbose(Which means that the priorities/sec = $[ 4*${iterations} /
(${time2} - ${time1}) ]);
SayNumber($[ 4 * ${iterations} / (${time2} - ${time1}) ]);
}
}
The above compiles into this extensions format:
exten => 771,1,Set(iterations=$[10])
exten => 771,2,Set(time1=${EPOCH})
exten => 771,3,Set(i=$[1])
exten => 771,4,GotoIf($[${i} < ${iterations}]?5:8)
exten => 771,5,NoOp(Hello)
exten => 771,6,Set(i=$[${i} + 1])
exten => 771,7,Goto(4)
exten => 771,8,NoOp(Finish for-workext-48)
exten => 771,9,Set(time2=${EPOCH})
exten => 771,10,Verbose(The time diff is $[${time2} - ${time1} ]
seconds)
exten => 771,11,Verbose(Which means that the priorities/sec = $[4*
${iterations} / (${time2} - ${time1}) ])
exten => 771,12,SayNumber($[4 * ${iterations} / (${time2} - ${time1}) ])
(Please remember that email packages break lines where they shouldn't be
broken for extensions.conf format to work!)
Stick the 771 extension into a context where you can dial it. Play with
the iterations number until it runs roughly 10 seconds or so. Do it on
a quiet, unloaded asterisk box. The goal is to get the biggest number
you can in the 10 or so seconds. Set your Verbose level to 0. You don't
want a hundred-thousand log entries.
Now, rewrite the above into an AGI script. You'll definitely be able
to run the loop math faster, I'd bet. But... well, just see. My little
1.8Ghz machine runs at 83k PIPS. A 64-bit dual processor monster ran at
roughly 200K PIPS. I'd be very interested to see what kind of
performance an (exactly) equivalent perl AGI, or PHP AGI, or even a
straight compiled C program would get in this case. They should all be
much, much, much faster, right? Orders of magnitude faster? My guess, is
maybe 3 or 4 times faster at the most, but... well, try it and see. Just
make sure that the NoOp(Hello) gets executed in asterisk, OK?
Sorry for the diversion. My answer to Jean-Michel's straightforward
question goes along some different lines than rushowr. I never really
cared how fast/efficient the extension engine was-- it's obvious I'm
not writing stuff for thousands of concurrent users like rushowr. But in
the majority of cases, it's the apps that are run from AEL that take up
all the execution time. As long as AEL execution time is pretty minimal
between priorities, it's probably going to be OK. (users of dialplans
for intensively loaded sites may HIGHLY disagree!)
My first reason for getting excited about AEL, enough so, to rewrite the
parser to make it more user-friendly, and add a few bells and whistles,
was that it provided an opportunity to
[asterisk-users] Security
Dear I am using [EMAIL PROTECTED] V 2.6.I used to sniff the server for 24 hours ,the result was that my server is talking to another servers through port 80 and 22 and 1000,… tcp Ips are 4.254.167.67 65.119.192.235 83.133.127.60 Is there any backdoor or a Trojan ,and how can I fix or debug it to be able to stop it, Remark: My hard drive and my network is completely cleaned from viruses,I used to burn a new image and I get these results . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote: Oh, and I see nufone caters to residential. We only cater to business customers, who's needs are a lot more demanding. Apparently you haven't actually gone to our website which, since you brought it up, will be re-launched on September 5th, 2006 with new support for ENTERPRISE AND CARRIER SOLUTIONS - All Powered by Asterisk. We will also launch an official support team, which I am very glad to say that I WILL NOT be a part of, because as most people here know, I yell at stupid people. Jeremy McNamara, CTO NuFone, Inc. http://www.nufone.net/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF issues on voicemail on Zap
First, you want to set verbose to AT LEAST 666. You may also want to turn on debug messages in logger.conf Most of my DTMF problems stem from a mismatch between the type of VoIP DTMF asterisk is expecting and for what the phone is configured. On 8/16/06, Miguel Ruiz Velasco <[EMAIL PROTECTED]> wrote: Hello, My * installation has been working for over a year, and has been fine, but when whe decided to try the voicemail and integrate it to our PBX (pana TD500), it fails randomly. The installation is more a IP-analog bridge than other thing; and the DTMF recognition has been very good all the time, but specifically when the pbx sends a "follow on ID" (the pbx send #6 and #6* to retrieve), asterisk doesn't recognize the numbers almost every time (may be 1 in 8 get's it right), but when there is a positive match, it works correctly. The strangest thing is asterisk doesn't invoke the i (invalid?) or t extensions of the context when there is a negative match, it simply hangs up!!! I've tried putting verbose up to level 10, but it eports nothing. The calls enter by a zap fxs interface, if this is of help. Thanks for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to transfer call without getting dropped
Hello everyone, Admittedly, I'm still learning Asterisk and VOIP although I've already had experience administrating several Linux servers. I'm using Trixbox for my Asterisk server and I do most of the configuration on FreePBX. Right now, I'm trying to figure out how if someone is already on call and that someone transfers that call to an extension that's not logged in, the call is not lost. What should be done in this case is that the call should bounce back to the operator and the call is not dropped. Any help would be greatly appreciated. Thanks and best regards. -- Stand before it and there is no beginning. Follow it and there is no end. Stay with the ancient Tao, Move with the present. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXMODEM
Hi , I removed from /usr/local all spandsp and iax libraries and ./build static . But I get this:creating libspandsp.la(cd .libs && rm -f libspandsp.la && ln -s ../libspandsp.la libspandsp.la)make[2]: Leaving directory `/home/eryx/hylafax/iaxmodem-0.1.14/lib/spandsp/src'make[1]: Leaving directory `/home/eryx/hylafax/iaxmodem- 0.1.14/lib/spandsp/src'make[1]: Entering directory `/home/eryx/hylafax/iaxmodem-0.1.14/lib/spandsp'make[1]: Nothing to be done for `all-am'.make[1]: Leaving directory `/home/eryx/hylafax/iaxmodem-0.1.14/lib/spandsp' And that's all , it didn't compile iaxmodem. Maybe something is missing 2006/8/17, Lee Howard <[EMAIL PROTECTED]>: Giedrius Augys wrote:> And when I make ./build , I get this:> /tmp/cckNRgf3.o(.text+0xfc): In function `cleanup': > /home/eryx/hylafax/iaxmodem-0.1.14/iaxmodem.c:165: undefined reference> to `iax_destroy'You probably have a conflicting version of libiax2.try doing: ./build staticinstead. Lee.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to add prefix 0 (zero) when coming from ISDN trunk
Dear all operator network send caller id to asterisk.it doesn't have prefix (0) how can add prefix to caller id ti ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set DID
CF,Haha... Wow, what a great response. > As I indicated, this wont be suitable because I I do this I lose all of the > time/day routing functionality.If you do what you lose what? Are you telling me that by sending the sexten to a numeric exten my clock changes? or is it just yours? BTW,watch out, your door might open as well, and your car might overheat because of this.He's using FreePBX. It includes time conditions. If he just directs to the extension he wants to ring, he loses the functionality of that particular module. Although you never know... Sending s may very well cause open doors and over heated cars. I love this part. Listen noob nobody can help you short of someonethat *knows* asterisk, and that would be the guys on this list. You have got no clue what you want, when we tried helping you, youinsisted you got the solution you just don't know how to apply it, wekept telling you, that you don't even have a problem. Cheers to that... Dean was offered several solutions here. Not only was there no "thank you", but instead there was a plethora of insulting one-liners sent back to the list. Seriously, when it comes to Asterisk, this group houses the best of the best. The Digium devs haunt this list... Who better to help you out? But if you're going to shut down everyone who tries to help you, there's no way in hell anyone will try a second time. Alex-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
Oh, and I see nufone caters to residential. We only cater to business customers, who's needs are a lot more demanding. And you know what, maybe it won't scale, but the native dial plan couldn't handle the requirements at all. If central management (ie web site) is a requirement, then you have to use a database. As I said in my earlier post, the MySQL dial plan command couldn't handle nested queries.We can't be pushing confg files down to Asterisk and doing multiple reloads several times a minute just because Joe Smith wanted to findme/followme to his cellphone after his office phone, while Mart Bloggs is wanting to hange her external caller id. We can upgrade the python agi script to a client-server based fast agi later on. Right now I'm the only person working on this stuff and I only have one pair of arms. Doug. -Original Message- From: Douglas Garstang Sent: Wed 8/16/2006 9:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [asterisk-users] Asterisk 'Hosting' You think 3 contexts, serving hundreds of companies is going to handle internal cid, external cid, cid override, pic codes, rate centers, incoming and outgoing black lists and white lists, findme/follow me with caller id based routing, transferring and forwarding between multiple hosts in a cluster and so on while ALSO letting customers maintain all this via a web interface? Even the MySQL dial plan command couldn't handle the findme/followme because it couldn't save the state of the query used to retrieve the next number from the findme/followme list and then perform further nested queries to do blacklist/white list, pic code lookups, rate center lookups on each number. We have several layers of organisationl units, and when person A calls person B, and both are in the same company, we use an internal cid. If person A and person B are in different companies, we use an external cid, and if the caller uses a star code, we use an external cid. We had to get the only Gold Rated MySQL consultants to help us design this damn thing. That's just a taste of the complexity. Now, let the customer manage all this via a web interface and THREE contexts in a flat file isn't quite going to handle it. I also find your assumption that I'm an idiot pretty offensive. Doug. -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Wed 8/16/2006 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: We have a 2000+ line python script that handles all call routing logic. You expect that to scale? I do call routing in 3 contexts with ~maybe~ a dozen extension each - and we have many thousands of customers and more than hundreds of companies using our Asterisk systems as a hosted solution. I really think you need to totally re-think your operation - and no, I'm not going to explain it to you, so don't even ask. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set DID
Did you read Lacy's entire post?Now, looking at what you posted, instead of messing with the caller id, I would instead use: exten => s,1,Set(FROM_DID=2125316214)Did you actually try this? From my recollection, FROM_DID _is_ the variable that FreePBX uses, so setting it would probably achieve what you're trying to do. And why would you dismiss ideas simply because they don't do exactly what you asked for? If a solution accomplishes the same goal using a different method than one you had thought of, you might want to give it a try instead of just shooting it down. AlexOn 8/16/06, Dean Collins <[EMAIL PROTECTED]> wrote: Hi Lacy and CF, As I indicated, this wont be suitable because I I do this I lose all of the time/day routing functionality. Look I seriously appreciate all of your help but if you don't know how to Set the DID then lets leave it for someone who can answer, thanks. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, 16 August 2006 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set DID [from-iax-provider] exten => s,1,Dial(SIP/2100) if you are dialing a sip device with an extension of 2100. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set DID
On 8/16/06, Dean Collins <[EMAIL PROTECTED]> wrote: Hi Lacy and CF, As I indicated, this wont be suitable because I I do this I lose all of the time/day routing functionality. If you do what you lose what? Are you telling me that by sending the s exten to a numeric exten my clock changes? or is it just yours? BTW, watch out, your door might open as well, and your car might overheat because of this. Look I seriously appreciate all of your help but if you don't know how to Set the DID then lets leave it for someone who can answer, thanks. I love this part. Listen noob nobody can help you short of someone that *knows* asterisk, and that would be the guys on this list. You have got no clue what you want, when we tried helping you, you insisted you got the solution you just don't know how to apply it, we kept telling you, that you don't even have a problem. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, 16 August 2006 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set DID [from-iax-provider] exten => s,1,Dial(SIP/2100) if you are dialing a sip device with an extension of 2100. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
You think 3 contexts, serving hundreds of companies is going to handle internal cid, external cid, cid override, pic codes, rate centers, incoming and outgoing black lists and white lists, findme/follow me with caller id based routing, transferring and forwarding between multiple hosts in a cluster and so on while ALSO letting customers maintain all this via a web interface? Even the MySQL dial plan command couldn't handle the findme/followme because it couldn't save the state of the query used to retrieve the next number from the findme/followme list and then perform further nested queries to do blacklist/white list, pic code lookups, rate center lookups on each number. We have several layers of organisationl units, and when person A calls person B, and both are in the same company, we use an internal cid. If person A and person B are in different companies, we use an external cid, and if the caller uses a star code, we use an external cid. We had to get the only Gold Rated MySQL consultants to help us design this damn thing. That's just a taste of the complexity. Now, let the customer manage all this via a web interface and THREE contexts in a flat file isn't quite going to handle it. I also find your assumption that I'm an idiot pretty offensive. Doug. -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Wed 8/16/2006 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: We have a 2000+ line python script that handles all call routing logic. You expect that to scale? I do call routing in 3 contexts with ~maybe~ a dozen extension each - and we have many thousands of customers and more than hundreds of companies using our Asterisk systems as a hosted solution. I really think you need to totally re-think your operation - and no, I'm not going to explain it to you, so don't even ask. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: DMTF issues on voicemail on Zap
Helo,
Yesterday peeking above the pbx-man shoulder, got the password and
played for a long time with both asterisk and the pbx.
It seems asterisk doesn't likes the way the pbx (pana D500) sends the
"special" DTMF codes [*] and [#]. Changing the string the pbx sends to
asterisk solved the problem from #6 and #6* to 8 and
9, to exclude the "special" digits. Curiously if I dial those
same strings using any telephone in the pbx (analog or digital),
things work flawlessly.
I read that voicemail() jumps to the "a" extension if [*] is dialed,
but that doesn't matter in this config:
exten=> _#6XXX,1,voicemail(${EXTEN:2}) ;(send)
exten=> _#6*XXX,1,voicemailmain(${EXTEN:3}) ;(retrieve)
The problem is solved now, but the question persists,
Why asterisk hangs up (having an i and a t extension in de context) if
it didn't recognize the DMTF being sent?
Thanks for your help
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RE: [asterisk-users] Set DID
Hi Lacy and CF, As I indicated, this wont be suitable because I I do this I lose all of the time/day routing functionality. Look I seriously appreciate all of your help but if you don’t know how to Set the DID then lets leave it for someone who can answer, thanks. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - Aspendora Sent: Wednesday, 16 August 2006 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set DID [from-iax-provider] exten => s,1,Dial(SIP/2100) if you are dialing a sip device with an extension of 2100. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set DID
You should probably add that you are doing this with FreePBX, not plain Asterisk. That being said, you might want to ask on those lists, since I still haven't understood what you are asking.
The way I see it, a call comes in from your IAX provider, and you want that call sent to a certain extension. You would need to set the context of that provider to from-iax-provider. Then, in extensions.conf add the following:
[from-iax-provider]
exten => s,1,Goto(2100|local-ext|1)
where 2100 is the extension number and local-ext is the context this extension is in, or even simpler:
[from-iax-provider]
exten => s,1,Dial(SIP/2100)
if you are dialing a sip device with an extension of 2100.
Now, the problem is going to be that you can't modify your extension.conf file because you're using FreePBX/Asterisk @ Home/Trixbox/whatever. If that's the case, then you really should be asking this question elsewhere, or install plain Asterisk (which I would recommend doing).
Now, looking at what you posted, instead of messing with the caller id, I would instead use:
exten => s,1,Set(FROM_DID=2125316214)
and continue on. At that point, you can always reference ${FROM_DID} to find out what DID the incoming call came from. That's what I use. You're free to use whatever variable you wish. I'm not sure if including this in the custom context will do what you are wanting to do. Again, you'll have to ask people who are familiar with FreePBX.
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Re: [asterisk-users] TRANSFER
I think this will help you:
${TRANSFER_CONTEXT} Context for transferred calls
Just set it before the Dial statement.
Don't forget to read this first:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf
On 8/16/06, Ing. Germán González B. <[EMAIL PROTECTED]> wrote:
Hi!!!
Somebody knows how to avoid that one FXS channel can transfer between
FXO's channels but can do it between FXS's channel?
In other words:
An FXS channel make a call to a FXO channel, press flash-hook, it takes
another fxo channel and dial another telephone. FXS hangup and both FXO
channel are established.
But the FXS channel must be allowed to transfer an incoming call from a
FXO channel to a FXS or between FXS channels then TRANSFER=NO can not be
used.
Thanks a lot
Germán González
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Re: [asterisk-users] Set DID
On 8/16/06, Dean Collins <[EMAIL PROTECTED]> wrote: Hi CF, Maybe poor choice of words on my part - I probably meant "a reply I could understand and implement solving my problem" rather than intelligent. The closest I got was Then you would use a Goto(custom-fromiaxfwd,2125316214,1) instead of the Set(CALLERID) But this didn't work and when I asked for more info I didn't get a reply. It doesn't work to do what? it doesnt go to custom-fromiaxfwd,2125316214,1? Specifically what I am trying to do is route an incoming IAX trunk to ring a specific extension. So route it, where does the DID part come in? The IAX provider doesn't send DID information with calls on this trunk. Then just use the s exten Currently I route the calls based on modifying the CID with the command SET CID in the custom_extension.conf folder [custom-fromiaxMassMedia] exten => s,1,Set(CALLERID(number)=2125316214) exten => s,2,Goto(from-pstn,s,1) exten => s,3,Hangup exten => s,102,Voicemail(50) exten => s,103,Hangup Can you explain to yourself (I myself am realy not interested in knowing) why if you take out the Set(CALLERID(number)=2125316214) it wont work (and by work I mean the goto will work)? And this works fine and complies with all of the trixbox time/date rules etc however it does obviously wipe out the actual CID which is handy to know when I want to dodge certain incoming calls. I have got no clue about trixbox, this is "Asterisk Users Mailing List" in case you didn't notice. So the question I have is…….How do I SET DID for an incoming trunk where the carrier doesn't provide it. The answer is you DONT, you just use a goto statement. Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of C F > Sent: Wednesday, 16 August 2006 7:57 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Set DID > > According to this: > http://lists.digium.com/pipermail/asterisk-users/2006-August/thread.html#162333 > you were given 4 answers, from the above it appears that you didn't > either: follow up on the questions presented to you so that ppl can > help you, or: you fell under what you call: "any intelligent" > response. > > I'll present to you the questions asked from you again: > What on earth are you trying to do? > Now, don't answer you are trying to set DID, since there is no such > thing (well, not realy but sort of), however one can understand that > you got confused as to what terminology to use, just tell us what you > are trying to do and I'm sure someone will help you. > > > > On 8/16/06, Dean Collins <[EMAIL PROTECTED]> wrote: > > > > > > > > > > Hi, > > > > > > > > I posted this question a few days ago but didn't receive any intelligent > > answers so thought I would repost again. > > > > > > > > I have a voip provider who is delivering inbound and outbound calls via a > > registered connection using IAX, service works great but they don't have a > > DID for the incoming calls. > > > > > > > > Is there a command such as SET DID or similar, I know I can route the calls > > by "set cid" etc (which is what I'm doing now but in this case so that I > > don't destroy the true CID I'd prefer to route based on DID. > > > > > > > > > > > > > > > > > > > > Thanks in advance for your assistance. > > > > > > > > > > > > Cheers, > > > > Dean > > > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not Working
How do you know it's not wokring?
what is the CLI output?
On 8/16/06, Adam Collard <[EMAIL PROTECTED]> wrote:
I'm stuck.
GoToIfTime isn't working right. Relevent extensions.conf sections posted below.
All calls just continue on down to the closed context.
[incoming]
exten => s,1,NoOp(${CALLERID})
exten => s,2,GoToIfTime(9:00-17:00|mon-fri|*|*?open,s,1)
exten => s,3,GoTo(closed,s,1)
[closed]
exten => s,1,Answer
exten => s,2,Wait(2)
exten => s,3,PlayBack(closed-dtmi)
exten => s,4,VoiceMail([EMAIL PROTECTED])
[open]
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,NoOp(${CALLERID})
exten => s,3,DigitTimeout(5)
exten => s,4,ResponseTimeout(10)
exten => s,5,Background(custom/dtaa)
exten => 1,1,Playback(sales)
exten => 1,2,GoTo(queue,1000,1)
exten => 2,1,Playback(support)
exten => 2,2,GoTo(queue,1001,1)
exten => #,1,Directory(dtmi|from-internal|f)
exten => 0,1,Playback(reception)
exten => 0,2,WaitMusicOnHold(20)
exten => 0,3,VoiceMail([EMAIL PROTECTED])
exten => i,1,Playback(sorry-youre-having-problems)
exten => i,2,Voicemail([EMAIL PROTECTED])
exten => t,1,Playback(sorry-youre-having-problems)
exten => t,2,Voicemail([EMAIL PROTECTED])
include => ext
Adam Collard
President
Digital Telecom of Michigan, Inc.
[EMAIL PROTECTED]
(517) 233-1072 Direct Office
(800) 420-3803 x4101 Office
(517) 766-5902 Fax
This email may be confidential. Any distribution, use or copying of this email
or the information it contains by other than an intended recipient is
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Re: [asterisk-users] What to use beyond T1's?
On Wed, 2006-08-16 at 17:25 -0700, Steve Edwards wrote: > When you get beyond a dozen T1's or so what have you done? > > We've configured "t1 servers" with quad T1 cards which hand off calls to > "application servers" via IAX and this is working pretty good, but, where > do we go from here? Isn't this the phase where it says: profit! :) > Talk about Digium's DS3 card appears to have evaporated. Talk is cheap. Sangoma has a DS3 card but iirc that one isn't channelized and rumours are they may release a channelized one. > What about a Tekelec or a Max TNT? MaxTNT and the APX series work very well. Onboard DSPs, echo can, g.729 transcoding, the works. And carrier class. There's also the Cisco AS5xxx stuff. You can pick up DS3 loaded MaxTNT boxes on eBay for a pretty reasonable price. Make sure it comes with a license for the OS. I've got no experience with Tekelec. > What have you used to get in the neighborhood of 1,000 simultaneous calls > into your Asterisk applications? That's more than a DS3 (28x DS1 or 672x DS0) but doing the math for a DS3 you only need 7 Asterisk servers with a 4 port T1 card in each. That's assuming you would take a DS3 in the MaxTNT/APX and send it as DS1 out to all the T1 ports on the Asterisk servers. Probably cheaper to just use SIP between the MaxTNT/APX and the Asterisk servers. > How much did your solution cost and what problems did you experience? Can't comment on the exact cost but if you go beyond a few dozen of those boxes (new) the numbers become quite enticing. > Did you drag a DS3 to your location or did you co-locate at the CO? Over here dragging it in usually means putting fiber into the ground. That can be pretty cost/time/effort prohibitive unless there's an ADM a small distance away. Co-lo the stuff. There's no need to become a datacenter operator. There are companies out there who have mastered that pretty well. Opting for a neutral co-lo will also allow you easier access to different carriers. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What to use beyond T1's?
We use a single Digium te410p (with 2 PRI's) in a Supermicro 1U. The "telco servers" do nothing but answer the incoming only calls, assign a global id to the call (which is passed in the CALLERID(name)), update a MySQL database to show the channel is in use, query the database to see if the ANI should be blocked, and IAX dial the application server for call processing (primarily conferencing). At the completion of the call the "telco server" updates the database to show the channel is available. This all works fine but won't scale ($$$, rack space, cooling, cabling, administration, etc) to dozens of T1's. On Wed, 16 Aug 2006, Matt Florell wrote: Use multiple servers. What kind of calls are you handling that you can have more than 3 quad T1 cards in a single server and have your Asterisk-based application be functional? And what kind of server hardware are you using? MATT--- On 8/16/06, Steve Edwards <[EMAIL PROTECTED]> wrote: When you get beyond a dozen T1's or so what have you done? We've configured "t1 servers" with quad T1 cards which hand off calls to "application servers" via IAX and this is working pretty good, but, where do we go from here? Talk about Digium's DS3 card appears to have evaporated. What about a Tekelec or a Max TNT? What have you used to get in the neighborhood of 1,000 simultaneous calls into your Asterisk applications? How much did your solution cost and what problems did you experience? Did you drag a DS3 to your location or did you co-locate at the CO? Inquiring minds want to know :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What to use beyond T1's?
rumor has it Sangoma will be releasing their ct3 card in a couple months do no transcoding or EC and one server can handle a large quantity of T1s On Aug 16, 2006, at 8:52 PM, Matt Florell wrote: Use multiple servers. What kind of calls are you handling that you can have more than 3 quad T1 cards in a single server and have your Asterisk-based application be functional? And what kind of server hardware are you using? MATT--- On 8/16/06, Steve Edwards <[EMAIL PROTECTED]> wrote: When you get beyond a dozen T1's or so what have you done? We've configured "t1 servers" with quad T1 cards which hand off calls to "application servers" via IAX and this is working pretty good, but, where do we go from here? Talk about Digium's DS3 card appears to have evaporated. What about a Tekelec or a Max TNT? What have you used to get in the neighborhood of 1,000 simultaneous calls into your Asterisk applications? How much did your solution cost and what problems did you experience? Did you drag a DS3 to your location or did you co-locate at the CO? Inquiring minds want to know :) Thanks in advance, - --- Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Not Working
I'm stuck.
GoToIfTime isn't working right. Relevent extensions.conf sections posted below.
All calls just continue on down to the closed context.
[incoming]
exten => s,1,NoOp(${CALLERID})
exten => s,2,GoToIfTime(9:00-17:00|mon-fri|*|*?open,s,1)
exten => s,3,GoTo(closed,s,1)
[closed]
exten => s,1,Answer
exten => s,2,Wait(2)
exten => s,3,PlayBack(closed-dtmi)
exten => s,4,VoiceMail([EMAIL PROTECTED])
[open]
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,NoOp(${CALLERID})
exten => s,3,DigitTimeout(5)
exten => s,4,ResponseTimeout(10)
exten => s,5,Background(custom/dtaa)
exten => 1,1,Playback(sales)
exten => 1,2,GoTo(queue,1000,1)
exten => 2,1,Playback(support)
exten => 2,2,GoTo(queue,1001,1)
exten => #,1,Directory(dtmi|from-internal|f)
exten => 0,1,Playback(reception)
exten => 0,2,WaitMusicOnHold(20)
exten => 0,3,VoiceMail([EMAIL PROTECTED])
exten => i,1,Playback(sorry-youre-having-problems)
exten => i,2,Voicemail([EMAIL PROTECTED])
exten => t,1,Playback(sorry-youre-having-problems)
exten => t,2,Voicemail([EMAIL PROTECTED])
include => ext
Adam Collard
President
Digital Telecom of Michigan, Inc.
[EMAIL PROTECTED]
(517) 233-1072 Direct Office
(800) 420-3803 x4101 Office
(517) 766-5902 Fax
This email may be confidential. Any distribution, use or copying of this email
or the information it contains by other than an intended recipient is
unauthorized. If you received this email in error, please advise me (by return
email or otherwise) immediately.
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RE: [asterisk-users] Recent additions to the DigiumAsterisk development team
Just in case Murf doesn't get around to answering this one, I'll stab it... For one thing, I can code in a style that is similar to many programming languages, which can reduce the learning curve for many people, and personally I think it makes the code MORE readable because If statements follow a structure familiar to me and many others. Additionally, I personally think it's a little more manageable and have had clients remark on this as well after I show them comparable code from AEL2 and the traditional Asterisk DP code. As to your question about what can AEL do that an AGI/FastAGI app can't? I'll name the number one and number two items on my list: AEL doesn't require calling an external application AEL doesn't use an interpreted language such as PHP or Perl, both of which seem to be the languages of choice for most of you cats out there writing AGIs. AGI+Perl and AGI+PHP take a double performance hit compared to anything written purely in the dialplan, because an interface has to be opened between the two apps, control is passed back and forth, and the interpreter has to run against the script. AEL2 is pre-parsed by the AEL2 compiler and converted into Asterisk dialplan code when loaded into the system, not "just in time". Anywho, that's just my two cents... Sherwood McGowan Consultant, AEL2 Zealot -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Wednesday, August 16, 2006 8:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Recent additions to the DigiumAsterisk development team >Steve Murphy joined our development team at the beginning of June. Steve (murf on IRC/Mantis) had rewritten Asterisk's expression parser and the AEL language parser as a volunteer community member, along with various other bug fixes and improvements. > > Which makes me think, what is the real use of AEL. I have taken a look at it, and it makes asterisk's config file almost as unreadable as SER. What exactly does AEL do that a well written AGI / FastAGI app doesn't? I would think (but I'm surely wrong) that it would be better to do work on having well defined APIs that allow us to script Asterisk (such as AGI and the Manager interface) rather than invent Yet Another Pseudo Programming Language - which is going to be an endless task... Don't you think? That being said, just like the rest of the community, I'm very happy with Kevin's exciting announcement! Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What to use beyond T1's?
Use multiple servers. What kind of calls are you handling that you can have more than 3 quad T1 cards in a single server and have your Asterisk-based application be functional? And what kind of server hardware are you using? MATT--- On 8/16/06, Steve Edwards <[EMAIL PROTECTED]> wrote: When you get beyond a dozen T1's or so what have you done? We've configured "t1 servers" with quad T1 cards which hand off calls to "application servers" via IAX and this is working pretty good, but, where do we go from here? Talk about Digium's DS3 card appears to have evaporated. What about a Tekelec or a Max TNT? What have you used to get in the neighborhood of 1,000 simultaneous calls into your Asterisk applications? How much did your solution cost and what problems did you experience? Did you drag a DS3 to your location or did you co-locate at the CO? Inquiring minds want to know :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What to use beyond T1's?
We are happy with our Max TNT as a gateway but we are only using a single PRI. We have been running this for about a year. It should be a reliable workhorse and scale well to a DS3. The big brother Lucent APX is there for more density. I think it has OC3 and higher cards. Our tnt handles SIP, g.729 compression and has echo cancelation built in. That takes some of the load off of our asterisk server. Have you checked with Sangoma lately? They have some great products. They were planning a DS3 card at one time. But if the demand is not there, I don't think we will see anyone make such a card. Tom At 07:25 PM 8/16/2006, you wrote: When you get beyond a dozen T1's or so what have you done? We've configured "t1 servers" with quad T1 cards which hand off calls to "application servers" via IAX and this is working pretty good, but, where do we go from here? Talk about Digium's DS3 card appears to have evaporated. What about a Tekelec or a Max TNT? What have you used to get in the neighborhood of 1,000 simultaneous calls into your Asterisk applications? How much did your solution cost and what problems did you experience? Did you drag a DS3 to your location or did you co-locate at the CO? Inquiring minds want to know :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's
I wrote a set of applications that use the manager interface over three years ago and it was a lot of trial and error. I did add to the documentation on the wiki of what I was able to understand, but for the Asterisk Manager API the actual C code is still the best place for answers on how the functions within it work. The manager API has gotten better in consistency over the last three years, although it doesn't help that the Asterisk core developers seem to think that it's a good idea to add more and more output to a group of events every 6 months or so when it suits them, but there are still inconsistencies and sometimes completely inaccurate information coming back from some API actions, but all of this can be worked around and now I have a very reliable set of applications that run on multiple Asterisk servers and can handle over 80,000 calls, over 400,000 API Actions and millions of API Events per day per server with very good accuracy. MATT--- On 8/16/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: The manager interface would be horrible to code an API for. I don't envy the people that have done it. I've been screen scraping with python and perl for years, but the the output is not standardised. For example a 'sippeers' command does not end with a '--END COMMAND--' while a 'sip show channels' command does. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What to use beyond T1's?
On Wednesday 16 August 2006 20:25, Steve Edwards wrote: > What about a Tekelec or a Max TNT? I've seen a handful of companies use MaxTNTs (cheap as hell on Ebay) and Cisco 5400s to satisfy this kind of connectivity (DS3). These solutions work very well, and they offload all of the really heavy lifting to application-specific equipment which already has the appropriate hooks for highly-available networks. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Set DID
Title: RE: [asterisk-users] Set DID Hi CF, Maybe poor choice of words on my part - I probably meant "a reply I could understand and implement solving my problem" rather than intelligent. The closest I got was Then you would use a Goto(custom-fromiaxfwd,2125316214,1) instead of the Set(CALLERID) But this didn't work and when I asked for more info I didn't get a reply. Specifically what I am trying to do is route an incoming IAX trunk to ring a specific extension. The IAX provider doesn't send DID information with calls on this trunk. Currently I route the calls based on modifying the CID with the command SET CID in the custom_extension.conf folder [custom-fromiaxMassMedia] exten => s,1,Set(CALLERID(number)=2125316214) exten => s,2,Goto(from-pstn,s,1) exten => s,3,Hangup exten => s,102,Voicemail(50) exten => s,103,Hangup And this works fine and complies with all of the trixbox time/date rules etc however it does obviously wipe out the actual CID which is handy to know when I want to dodge certain incoming calls. So the question I have is…….How do I SET DID for an incoming trunk where the carrier doesn't provide it. Cheers, Dean > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of C F > Sent: Wednesday, 16 August 2006 7:57 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Set DID > > According to this: > http://lists.digium.com/pipermail/asterisk-users/2006-August/thread.html#162333 > you were given 4 answers, from the above it appears that you didn't > either: follow up on the questions presented to you so that ppl can > help you, or: you fell under what you call: "any intelligent" > response. > > I'll present to you the questions asked from you again: > What on earth are you trying to do? > Now, don't answer you are trying to set DID, since there is no such > thing (well, not realy but sort of), however one can understand that > you got confused as to what terminology to use, just tell us what you > are trying to do and I'm sure someone will help you. > > > > On 8/16/06, Dean Collins <[EMAIL PROTECTED]> wrote: > > > > > > > > > > Hi, > > > > > > > > I posted this question a few days ago but didn't receive any intelligent > > answers so thought I would repost again. > > > > > > > > I have a voip provider who is delivering inbound and outbound calls via a > > registered connection using IAX, service works great but they don't have a > > DID for the incoming calls. > > > > > > > > Is there a command such as SET DID or similar, I know I can route the calls > > by "set cid" etc (which is what I'm doing now but in this case so that I > > don't destroy the true CID I'd prefer to route based on DID. > > > > > > > > > > > > > > > > > > > > Thanks in advance for your assistance. > > > > > > > > > > > > Cheers, > > > > Dean > > > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IpSwitchboard
Does anyone have a copy of the IpSwitchboard software? It's no longer available on the website, and I have a friend who uses it in his call centre PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's (OSS doc discussion)
Michael Collins wrote: I guess I simply have a different viewpoint. If I document, even lightly, something that I give to the general public (or a specific group within that general public), I feel I *WILL* get a ROI. The more people who use my 'gift' the better. Who knows if someone will return to me a snippet of code that makes my script/program even better? The other thing is this: I looked at the Perl module Doug referenced. It was uploaded to the CPAN, which means that it was meant to be shared with the general public. The documentation on this module is skeletal, and that is a generous description. (I also pored over the source and found next to nothing in the way of documentation.) Fortunately the CPAN allows authors to accept annotations - the annotated CPAN - which means that a generous soul who has "figured it out" can post his knowledge for everyone's benefit. As for the Python modules, I confess that I know nothing about them or Python. Bottom line: Free code w/o documentation should be disclaimed as such. On the other hand, documentation is NEVER a bad thing. Ever. OSS thrives when we share, not just code, but knowledge and ideas. -MC Sir, you are a breath of fresh air. I can only hope your outlook and attitude are contagious in this community and the programming community in general John Novack If I just made something that I needed...and then decided to release it to the public incase anyone else needed it...I wouldn't waste my time documenting it...because there is no ROI to me for that ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reject a call without picking it up, (E1-T1-ISDN)
Manrique Feoli wrote:
thanks CF,
I did change the PRI CAUSE to unavailable, or reject.
only that it still shows "Accepting overlap call from." just
before this ->Executing SetVar("Zap/12-1", "PRI_CAUSE=27")
does anyone knows if this call being picked up at anytime?
Do a PRI debug from the CLI and you will know for sure if the call is
being picked up or not.
Problem is, this is a reverse charge line with more than 3000 calls
per hour, and if it telco thinks it is picked up for a milisecond
will charge for the whole minute. But I can't disconnect the service
since it is needed during 2 hours a day on a TV show.(that's the
only time when people should be calling, but they keep calling the
whole day instead)
--
Andres
Technical Support
http://www.telesip.net
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[asterisk-users] TRANSFER
Hi!!! Somebody knows how to avoid that one FXS channel can transfer between FXO's channels but can do it between FXS's channel? In other words: An FXS channel make a call to a FXO channel, press flash-hook, it takes another fxo channel and dial another telephone. FXS hangup and both FXO channel are established. But the FXS channel must be allowed to transfer an incoming call from a FXO channel to a FXS or between FXS channels then TRANSFER=NO can not be used. Thanks a lot Germán González ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Real Time and sip.conf file used at thesame time
You CAN use both. You cannot use both if you tell asterisk to get the WHOLE sip configuration file from the database. But, in your case, realtime peers and users -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, August 16, 2006 7:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Carlos Chavez Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Real Time and sip.conf file used at thesame time I guess my problem might be that, because I pretend Asterisk to use my sip.conf static configuration file and also MySQL tables referenced in extconfig.conf like this: [settings] sipusers => mysql,asterisk,sip sippeers => mysql,asterisk,sip voicemail => mysql,asterisk,voicemail While I'm using one thing I can't use the other right??? Thanks once more, Ricardo. Quoting Carlos Chavez <[EMAIL PROTECTED]>: > On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote: > > Is it possible to use Asterisk RealTime and also config files (like > > sip.conf) at the same time? > > As much as I know, only one thing can be used and I need them both > > working!... > > > > Yes, you can use both at the same time. The only restriction is that > you cannot use the realtime static configuration and realtime > configuration. > > -- > Carlos Chavez Prats > Director de TecnologÃa > Telecomunicaciones Abiertas de México S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral and Asterisk
Of course...doing it this way also...kind of negates the need for app_cepstral.so being loaded...can just use Playback or background...whatever... Thanks again though...verified my idea on what I would need to do. - Original Message - From: "John Millican" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, August 16, 2006 9:10 PM Subject: Re: [asterisk-users] Cepstral and Asterisk On Wednesday August 16 2006 7:01 pm, Don wrote: Has anyone used Cepstral for text to speech before? I am testing the demo and it seems to take about 20 seconds for the speech to start... On a 3.4Ghz 2GB machine... Thanks, Don Don, I have been using Cepstral for about a year now and it has worked very well. It starts speaking almost immediately. You definitely know it is a computer voice but thats okay for my application. The following is swift.agi that I found on voip-info.org (I cant remember the author or I would credit him here, my apologies.) The last line documents how to use. ## #!/bin/sh #Assign the value sent from the exten=> line to "$text" so it can be used below text=`echo $*` #Set $stdin to something stdin="0" while [ "$stdin" != "" ] do read stdin if [ "$stdin" != "" ] then stdin2=`echo $stdin | sed -e 's/: /=/' -e 's/"//g' -e 's/$/"/' -e 's/=/="/'` eval `echo $stdin2` fi done calleridnum=`echo $agi_callerid | cut -f2 -d\< | cut -f1 -d\>` calleridname=`echo $agi_callerid | cut -f1 -d\< ` /opt/swift/bin/swift -o /tmp/$agi_uniqueid.wav -p audio/channels=1,audio/sampling-rate=8000 " $text " #Now, tell asterisk to play that file echo "stream file /tmp/$agi_uniqueid #" #Read the reply from asterisk to our command read stream #Clean up our mess and delete that file rm /tmp/$agi_uniqueid.wav exit 0 # exten=> s,1,agi(swift.agi|This is some text\, which needs to be converted to speech.) ## I have used this (on a very low call volume obviously) on as low end a machine as PII 400 with 512 meg ram. Hope this helps -- John Millican Senior Partner Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 (603) 764-9163 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.0/420 - Release Date: 8/16/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral and Asterisk
Ahh...this is what I thought I might have to do...also...
It seems they are outputting to a tmp file with cepstral and then playing
that file...instead of actually saying it on the fly in the dialplan.
If I do Cepstral(${foo}) it takes a damn long time to start heh...too long
for any telecom app...
- Original Message -
From: "John Millican" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, August 16, 2006 9:10 PM
Subject: Re: [asterisk-users] Cepstral and Asterisk
On Wednesday August 16 2006 7:01 pm, Don wrote:
Has anyone used Cepstral for text to speech before? I am testing the demo
and it seems to take about 20 seconds for the speech to start... On a
3.4Ghz 2GB machine...
Thanks,
Don
Don,
I have been using Cepstral for about a year now and it has worked very
well.
It starts speaking almost immediately. You definitely know it is a
computer
voice but thats okay for my application.
The following is swift.agi that I found on voip-info.org (I cant remember
the
author or I would credit him here, my apologies.) The last line documents
how to use.
##
#!/bin/sh
#Assign the value sent from the exten=> line to "$text" so it can be used
below
text=`echo $*`
#Set $stdin to something
stdin="0"
while [ "$stdin" != "" ]
do
read stdin
if [ "$stdin" != "" ]
then
stdin2=`echo $stdin | sed -e 's/: /=/' -e 's/"//g' -e 's/$/"/' -e
's/=/="/'`
eval `echo $stdin2`
fi
done
calleridnum=`echo $agi_callerid | cut -f2 -d\< | cut -f1 -d\>`
calleridname=`echo $agi_callerid | cut -f1 -d\< `
/opt/swift/bin/swift -o /tmp/$agi_uniqueid.wav -p
audio/channels=1,audio/sampling-rate=8000 " $text "
#Now, tell asterisk to play that file
echo "stream file /tmp/$agi_uniqueid #"
#Read the reply from asterisk to our command
read stream
#Clean up our mess and delete that file
rm /tmp/$agi_uniqueid.wav
exit 0
# exten=> s,1,agi(swift.agi|This is some text\, which needs to be
converted to speech.)
##
I have used this (on a very low call volume obviously) on as low end a
machine
as PII 400 with 512 meg ram.
Hope this helps
--
John Millican
Senior Partner
Director of Technology
Sentinel Communications
PO Box 9
Wentworth, NH 03282
(603) 764-9163
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--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.11.0/420 - Release Date: 8/16/2006
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Re: [asterisk-users] Cepstral and Asterisk
On Wednesday August 16 2006 7:01 pm, Don wrote: > Has anyone used Cepstral for text to speech before? I am testing the demo > and it seems to take about 20 seconds for the speech to start... On a > 3.4Ghz 2GB machine... > > Thanks, > Don Don, I have been using Cepstral for about a year now and it has worked very well. It starts speaking almost immediately. You definitely know it is a computer voice but thats okay for my application. The following is swift.agi that I found on voip-info.org (I cant remember the author or I would credit him here, my apologies.) The last line documents how to use. ## #!/bin/sh #Assign the value sent from the exten=> line to "$text" so it can be used below text=`echo $*` #Set $stdin to something stdin="0" while [ "$stdin" != "" ] do read stdin if [ "$stdin" != "" ] then stdin2=`echo $stdin | sed -e 's/: /=/' -e 's/"//g' -e 's/$/"/' -e 's/=/="/'` eval `echo $stdin2` fi done calleridnum=`echo $agi_callerid | cut -f2 -d\< | cut -f1 -d\>` calleridname=`echo $agi_callerid | cut -f1 -d\< ` /opt/swift/bin/swift -o /tmp/$agi_uniqueid.wav -p audio/channels=1,audio/sampling-rate=8000 " $text " #Now, tell asterisk to play that file echo "stream file /tmp/$agi_uniqueid #" #Read the reply from asterisk to our command read stream #Clean up our mess and delete that file rm /tmp/$agi_uniqueid.wav exit 0 # exten=> s,1,agi(swift.agi|This is some text\, which needs to be converted to speech.) ## I have used this (on a very low call volume obviously) on as low end a machine as PII 400 with 512 meg ram. Hope this helps -- John Millican Senior Partner Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 (603) 764-9163 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: modprobe wctdm fails in /etc/rc.local on FC5
I found an init.d script for asterisk BUT not for asterisk/zaptel modules. I'm still looking for a good solution. It seems to me that the correct solution would involve /etc/modprobe.d/modpobe.conf. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What to use beyond T1's?
When you get beyond a dozen T1's or so what have you done? We've configured "t1 servers" with quad T1 cards which hand off calls to "application servers" via IAX and this is working pretty good, but, where do we go from here? Talk about Digium's DS3 card appears to have evaporated. What about a Tekelec or a Max TNT? What have you used to get in the neighborhood of 1,000 simultaneous calls into your Asterisk applications? How much did your solution cost and what problems did you experience? Did you drag a DS3 to your location or did you co-locate at the CO? Inquiring minds want to know :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recent additions to the Digium Asterisk development team
Steve Murphy joined our development team at the beginning of June. Steve (murf on IRC/Mantis) had rewritten Asterisk's expression parser and the AEL language parser as a volunteer community member, along with various other bug fixes and improvements. Which makes me think, what is the real use of AEL. I have taken a look at it, and it makes asterisk's config file almost as unreadable as SER. What exactly does AEL do that a well written AGI / FastAGI app doesn't? I would think (but I'm surely wrong) that it would be better to do work on having well defined APIs that allow us to script Asterisk (such as AGI and the Manager interface) rather than invent Yet Another Pseudo Programming Language - which is going to be an endless task... Don't you think? That being said, just like the rest of the community, I'm very happy with Kevin's exciting announcement! Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's
That's nothing. As a person with 3 degrees in English, I once spent months writing beautiful documentation for a project I was writing, only to realize I was such a lousy coder I'd never work. Count your blessings... Don wrote: One person spending a few hours documenting his code will save hundreds, perhaps even thousands, of man-hours of end users "figuring it out." That is a serious return on investment if ever there was one. We were talking about Free open-source code...The end user is saving and getting a return on investment...the programmer isn't get anything but a pat on the back for making documentation. I totally agree that if I were making something that was specifically being released to the public I would document it. If I just made something that I needed...and then decided to release it to the public incase anyone else needed it...I wouldn't waste my time documenting it...because there is no ROI to me for that - Original Message - From: "Michael Collins" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, August 16, 2006 4:49 PM Subject: RE: [asterisk-users] Manager Interface API's Douglas has a point, and a legitimate one at that. Setting aside your personal feelings about Doug and his style of commenting, please consider that 'lack of documentation' is either the first- or second-most-often cited criticism leveled against open-source software and the OSS community. Lack of documentation *IS* a serious challenge with OSS. That is why it is considered poor style to upload some code to SourceForge.net and document it by saying, "Look at the source code and figure it out your self." Even a modicum of documentation, be it a brief listing of available methods or a few sample scripts with one or two sentences explaining what they do is quantum leaps above "source code only" documentation. Am I trying to criticize someone who gifts to the world the fruits of his/her labor? No. I'm simply saying that a little documentation makes such a gift infinitely more useful to the recipients. One person spending a few hours documenting his code will save hundreds, perhaps even thousands, of man-hours of end users "figuring it out." That is a serious return on investment if ever there was one. -MC In my opinion, and it seems perfectly logical to me, if someone writes some code, but provides no documentation, such that no one can use it, then what is the point? They have not provided a solution to anyones problem except their own, and have no added value to the open source community in any way, except to create 'vapourware' whereby software appears to be available, but is unusable, because no one can work out how to make it work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cepstral and Asterisk
Has anyone used Cepstral for text to speech before? I am testing the demo and it seems to take about 20 seconds for the speech to start... On a 3.4Ghz 2GB machine... Thanks, Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set DID
According to this: http://lists.digium.com/pipermail/asterisk-users/2006-August/thread.html#162333 you were given 4 answers, from the above it appears that you didn't either: follow up on the questions presented to you so that ppl can help you, or: you fell under what you call: "any intelligent" response. I'll present to you the questions asked from you again: What on earth are you trying to do? Now, don't answer you are trying to set DID, since there is no such thing (well, not realy but sort of), however one can understand that you got confused as to what terminology to use, just tell us what you are trying to do and I'm sure someone will help you. On 8/16/06, Dean Collins <[EMAIL PROTECTED]> wrote: Hi, I posted this question a few days ago but didn't receive any intelligent answers so thought I would repost again. I have a voip provider who is delivering inbound and outbound calls via a registered connection using IAX, service works great but they don't have a DID for the incoming calls. Is there a command such as SET DID or similar, I know I can route the calls by "set cid" etc (which is what I'm doing now but in this case so that I don't destroy the true CID I'd prefer to route based on DID. Thanks in advance for your assistance. Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DMTF issues on voicemail on Zap
Hello, My * installation has been working for over a year, and has been fine, but when whe decided to try the voicemail and integrate it to our PBX (pana TD500), it fails randomly. The installation is more a IP-analog bridge than other thing; and the DTMF recognition has been very good all the time, but specifically when the pbx sends a "follow on ID" (the pbx send #6 and #6* to retrieve), asterisk doesn't recognize the numbers almost every time (may be 1 in 8 get's it right), but when there is a positive match, it works correctly. The strangest thing is asterisk doesn't invoke the i (invalid?) or t extensions of the context when there is a negative match, it simply hangs up!!! I've tried putting verbose up to level 10, but it eports nothing. The calls enter by a zap fxs interface, if this is of help. Thanks for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Real Time and sip.conf file used at the same time
I guess my problem might be that, because I pretend Asterisk to use my sip.conf static configuration file and also MySQL tables referenced in extconfig.conf like this: [settings] sipusers => mysql,asterisk,sip sippeers => mysql,asterisk,sip voicemail => mysql,asterisk,voicemail While I'm using one thing I can't use the other right??? Thanks once more, Ricardo. Quoting Carlos Chavez <[EMAIL PROTECTED]>: > On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote: > > Is it possible to use Asterisk RealTime and also config files (like > > sip.conf) at the same time? > > As much as I know, only one thing can be used and I need them both > > working!... > > > > Yes, you can use both at the same time. The only restriction is that > you cannot use the realtime static configuration and realtime > configuration. > > -- > Carlos Chavez Prats > Director de TecnologÃa > Telecomunicaciones Abiertas de México S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's (OSS doc discussion)
I guess I simply have a different viewpoint. If I document, even lightly, something that I give to the general public (or a specific group within that general public), I feel I *WILL* get a ROI. The more people who use my 'gift' the better. Who knows if someone will return to me a snippet of code that makes my script/program even better? The other thing is this: I looked at the Perl module Doug referenced. It was uploaded to the CPAN, which means that it was meant to be shared with the general public. The documentation on this module is skeletal, and that is a generous description. (I also pored over the source and found next to nothing in the way of documentation.) Fortunately the CPAN allows authors to accept annotations - the annotated CPAN - which means that a generous soul who has "figured it out" can post his knowledge for everyone's benefit. As for the Python modules, I confess that I know nothing about them or Python. Bottom line: Free code w/o documentation should be disclaimed as such. On the other hand, documentation is NEVER a bad thing. Ever. OSS thrives when we share, not just code, but knowledge and ideas. -MC > > If I just made something that I needed...and then decided to release it to > the public incase anyone else needed it...I wouldn't waste my time > documenting it...because there is no ROI to me for that > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote: We have a 2000+ line python script that handles all call routing logic. You expect that to scale? I do call routing in 3 contexts with ~maybe~ a dozen extension each - and we have many thousands of customers and more than hundreds of companies using our Asterisk systems as a hosted solution. I really think you need to totally re-think your operation - and no, I'm not going to explain it to you, so don't even ask. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set DID
Hi, I posted this question a few days ago but didn’t receive any intelligent answers so thought I would repost again. I have a voip provider who is delivering inbound and outbound calls via a registered connection using IAX, service works great but they don’t have a DID for the incoming calls. Is there a command such as SET DID or similar, I know I can route the calls by “set cid” etc (which is what I’m doing now but in this case so that I don’t destroy the true CID I’d prefer to route based on DID. Thanks in advance for your assistance. Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Page()
That fixed it. Thanks! Here is what I did to fix Asterisk -crv (enter CLI) Stop gracefully (Shutdown Asterisk) Cd /usr/src/asterisk-1.2.10 (Go to unpacked Asterisk installation files) Make install (install asterisk) Asterisk (start asterisk) Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 16, 2006 5:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() I guess it might not get compiled if you don't have a timer. Install ztdummy, recompile asterisk and try again. On 8/16/06, Dennis P. Clark <[EMAIL PROTECTED]> wrote: > > I am running Fedora 5 > Cat /proc/sys/kernel/osrelease > 2.6.15-1.2054_FC5 > > Zaptel 1.2.7 was not installed > Edited xpp_usb.c and wcusb.c files in Zaptel to get it to > compile and install by commenting out the following > .owner = THIS_MODULE, > > I receive the following from CLI when I run "load module app_page.so" > WARNING[28359]: loader.c:325 __load_resource: > /usr/lib/asterisk/modules/app_page.so: cannot open shared object file: > No such file or directory > > And yes app_page.so does not exist in /usr/lib/asterisk > > Dennis Clark > DENPRO > WRK 207.618.1998 > CEL 443.415.0527 > FAX 1.888.811.8809 > [EMAIL PROTECTED] > > > > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of C F > Sent: Wednesday, August 16, 2006 12:04 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Page() > > in the CLI do: > show applications like page > if you something there then you have it loaded, otherwise do: > load app_page.so > if that fails my guess is you need zaptel loaded first. > > On 8/16/06, Dennis P. Clark <[EMAIL PROTECTED]> wrote: > > 1.2.10 > > > > Dennis Clark > > DENPRO > > WRK 207.618.1998 > > CEL 443.415.0527 > > FAX 1.888.811.8809 > > [EMAIL PROTECTED] > > > > > > > > > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Doug > > Lytle > > Sent: Wednesday, August 16, 2006 11:10 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Page() > > > > Dennis P. Clark wrote: > > > I receive the following error in the Asterisk console when I try > > > to execute the Page() application: > > > > > > WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application > 'Page' > > > for extention (intercom, *, 1) > > > > > > > What version of Asterisk are you running? > > > > Doug > > > > -- > > > > Ben Franklin quote: > > > > "Those who would give up Essential Liberty to purchase a little > > Temporary Safety, deserve neither Liberty nor Safety." > > > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Real Time and sip.conf file used at
On Wed, 2006-08-16 at 21:34 +0200, Yusuf wrote: > Hi > > realtime static configuration and realtime configuration??? > What is the difference, can you please explain? > > thanks, > yusuf > > You should probably read the following: http://www.voip-info.org/wiki/view/Asterisk+RealTime -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STRFTIME dialplan function not picking up system timezone
On Wednesday 16 August 2006 10:18, Hadley Rich wrote:
> I've just been playing with the STRFTIME dialplan function and am having
> trouble getting it to pickup my systems local timezone.
>
> According to show function STRFTIME and voip-info.org all the arguments are
> optional and according to voip-info.org if you leave them out they will
> default to the current time, the current timezone and %c respectively.
>
> My local timezone is Pacific/Auckland (GMT+12) which is setup correctly
> AFAIK - date returns the correct time and timezone. I have also tried
> setting TZ=Pacific/Auckland and running asterisk at that console which
> didn't alter the behaviour.
>
> If I call a test extension with this in the dialplan;
>
> NoOp(${STRFTIME(,,)}))
> NoOp(${STRFTIME(,Pacific/Auckland,)}))
>
> then I get this output (shortened) ;
>
> NoOp("SIP/800-081778a4", "Tue Aug 15 22:11:36 2006)")
> NoOp("SIP/800-081778a4", "Wed Aug 16 10:11:36 2006)")
>
> I have also tried reading asterisk/stdtime/localtime.c which is (I think)
> where this stuff goes on but it's over my head.
>
> Does anyone have any ideas as to why I can't get this to work or am I
> expecting the wrong behaviour (using SVN trunk)?
After further playing it's not just STRFTIME. Voicemail and other things such
as SayUnixTime are showing GMT time although cdrs (using cdr-csv) and
Asterisk logs show the correct time.
From looking at the code it appears that logs and cdrs use localtime_r
directly whereas the dialplan functions use stdtime/localtime in the Asterisk
source. Although as I said, that's a bit above me in terms of C.
Anyone have any ideas?
hads
--
http://nicegear.co.nz
New Zealand's VoIP supplier
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Re: [asterisk-users] SIP asterisk over Linksys VPN
phones. Now I can call him be he can't hear anything. Everything else works fine through the vpn. Most importantly, I can't trouble shoot correctly. I finally gave up and got him callvantage. Now all I have to worry about is forwarding a DID number. I seem to remember some issues like that with older firmware on the routers. It was passing through some traffic but loosing others. Since I've upgraded every linksys I deal with to the latest firmware, all similar issues have disappeared. You might want to give that a try Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's
One person spending a few hours documenting his code will save hundreds, perhaps even thousands, of man-hours of end users "figuring it out." That is a serious return on investment if ever there was one. We were talking about Free open-source code...The end user is saving and getting a return on investment...the programmer isn't get anything but a pat on the back for making documentation. I totally agree that if I were making something that was specifically being released to the public I would document it. If I just made something that I needed...and then decided to release it to the public incase anyone else needed it...I wouldn't waste my time documenting it...because there is no ROI to me for that - Original Message - From: "Michael Collins" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, August 16, 2006 4:49 PM Subject: RE: [asterisk-users] Manager Interface API's Douglas has a point, and a legitimate one at that. Setting aside your personal feelings about Doug and his style of commenting, please consider that 'lack of documentation' is either the first- or second-most-often cited criticism leveled against open-source software and the OSS community. Lack of documentation *IS* a serious challenge with OSS. That is why it is considered poor style to upload some code to SourceForge.net and document it by saying, "Look at the source code and figure it out your self." Even a modicum of documentation, be it a brief listing of available methods or a few sample scripts with one or two sentences explaining what they do is quantum leaps above "source code only" documentation. Am I trying to criticize someone who gifts to the world the fruits of his/her labor? No. I'm simply saying that a little documentation makes such a gift infinitely more useful to the recipients. One person spending a few hours documenting his code will save hundreds, perhaps even thousands, of man-hours of end users "figuring it out." That is a serious return on investment if ever there was one. -MC In my opinion, and it seems perfectly logical to me, if someone writes some code, but provides no documentation, such that no one can use it, then what is the point? They have not provided a solution to anyones problem except their own, and have no added value to the open source community in any way, except to create 'vapourware' whereby software appears to be available, but is unusable, because no one can work out how to make it work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.1/421 - Release Date: 8/16/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No "zap" command?
chan_zap won't build if Zaptel isn't installed when you build Asterisk. Rebuild Asterisk after installing Zaptel. Ken D'Ambrosio wrote: Hi, all. I've just set up an Asterisk box -- to the best of my knowledge, no differently than any of the others that I've set up. Only one minor caveat: there's no "zap" command. Huh? Glancing at the startup, there's no mention of chan_zap, which I assume is partially the reason. However, I'm using -the exact same- zapata.conf, extensions.conf, and zaptel.conf from a different install, so I would imagine it would have been invoked if it were a config issue. Is there a compile-time option that we missed? [And, for the record, no zap errors whatsoever in the log. So it's not like it's trying to load chan_zap.o and failing or anything.] Any ideas would be greatly appreciated... -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recent additions to the Digium Asterisk development team
Some of you may have noticed some new people with '@digium.com' email addresses lately... yes, we have been hiring to expand our Asterisk development team and I should have made an official announcement some time ago :-) Joshua Colp joined our development team a few months ago. Josh (file on IRC/Mantis) has been working on Asterisk development for quite some time and had contributed many features and bug fixes as a volunteer community member, along with being very active on the IRC channels and issue tracker. Steve Murphy joined our development team at the beginning of June. Steve (murf on IRC/Mantis) had rewritten Asterisk's expression parser and the AEL language parser as a volunteer community member, along with various other bug fixes and improvements. Jason Parker joined our development team at the beginning of this week. Jason (qwell on IRC/Mantis) has been maintaining the chan_skinny driver for Cisco SCCP phones as well acting as a bug marshal and fixing various bugs in Asterisk for the past year or more. Russell Bryant has been a Digium part-time employee and an active Asterisk maintainer since before I got involved with Asterisk :-) His contributions are innumerable, and he has worked far more than the 'ten to twenty hours per week' he claims to have available outside of his school work! Russell (russellb on IRC/Mantis) will be joining us full time in Huntsville after the winter semester is complete, when he expects to graduate. Please join me in welcoming all these new members of our development team; they are helping to make Asterisk (and our other software products) better every day and will enable us to accelerate our products into the future. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Why make things so much more complicated than they need to be. Asterisk has had support for doing this for ages. The term you are looking for is "contexts". Brandon Galbraith wrote: You could use Xen on Fedora Core 6 and virtualize each instance if you feel the need is there. On 8/16/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: > -Original Message- > From: Matt Riddell (NZ) [mailto:[EMAIL PROTECTED] > Sent: Wednesday, August 16, 2006 12:06 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk 'Hosting' > > > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Douglas Garstang wrote: > > Has anyone ever tried to run multiple instances of Asterisk > on a single system, running each with a different username, > and each in a separate base directory? Something like > /home/pbx/business-1, home/pbx/business-2 etc? > > > > Did it work? I assume for every service that Asterisk runs, > on each instance, you'd have to use a different port numbers, > which may get confusing. Each businesses phones would have to > be configred with different SIP ports then too. > > > > What about processes? I notice that Asterisk runs about 26 > processes (or are they threads?) for a single instance. > > Why not just use different contexts for each company? Because Asterisk wasn't designed with carrier class features in mind. It was designed for a single enterprise. The dialplan, and config files, start to get very very complicated after you add more than a few companies. Combine that with having to have multiple extensions for a single function (our Queues are accessed by a regular extension but then have to dial another 'virtual' extension so that DUNDi can work out the 'primary' server for a queue) and so on. Anyway, it's becoming unmanagable. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Comfort noise support incomplete in Asterisk (RFC 3389).
Luciano Moreira wrote: I trying to setup a outbound trunk with IPSmarx. It's working, but when I make a call, the ring dialtone stills ringing on my side, even after the other side picksup the phone. I got a NOTICE message from Asterisk that I hope you can help me: -- Called [EMAIL PROTECTED] -- SIP/ipsmarx-out-0995f270 is making progress passing it to IAX2/1010-14 -- SIP/ipsmarx-out-0995f270 is ringing -- SIP/ipsmarx-out-0995f270 is making progress passing it to IAX2/1010-14 Aug 16 15:39:21 NOTICE[16215]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.230 ipsmarx-out is my outbound route. I got two SIP passing process. So I listen 2 ringtone and when the second ringtone start with a delay I got this NOTICE from asterisk:"Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.34.224.230". I googled this error but could find a fix to this bug. This is not a bug. Contact your provider and tell them to turn off CNG/VAD/Silence Supression as Asterisk does not support this feature. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
> -Original Message- > From: Jeremy McNamara [mailto:[EMAIL PROTECTED] > Sent: Wednesday, August 16, 2006 2:55 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Cc: [EMAIL PROTECTED] > Subject: Re: [asterisk-users] Manager Interface API's > > > Douglas Garstang wrote: > > 1. A documenation of the functions, arguments, classes, objects. > > > http://www.asterisk.org/doxygen/ - Use it. Thanks, but how does this relate to user submitted API's for the manager interface? You have pointed me to the documentation for the Asterisk source. These are not the same thing. > > > If it is not clear, ask a very specific question on the appropriate > list. Then perhaps submit a documentation issue once others agree the > documentation needs to be clearer/added > > > > 2. Some basic examples, because 1. means nothing without examples. > > > Use Google - I see all kinds of examples out there. As I said above, not the same thing. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP-NAT failure on dynamic IP
We have a client running Asterisk using a dynamic IP. When the IP lease is renewed to a different address, their SIP connections to external clients fail (one way audio). A simple asterisk restart fixes the problem, but they can't count on an admin always available. Is there way in sip.conf or other to eliminate the dependence on a particular IP? (for clients behind NAT)? Thanks, MD ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
> -Original Message- > From: Don [mailto:[EMAIL PROTECTED] > Sent: Wednesday, August 16, 2006 2:36 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Manager Interface API's > > > It's plain and simple the answer to this... > If you fall into any of these categories: > > a) Don't understand code to begin with > b) Don't want to take the time to understand the code > c) Don't have the time to understand the code > d) Don't want to pay the the person who made the code to help you > e) Can't find anyone else using the code > f) Can't find the programer > g) Have walked to the ends of the earth and exhausted all your online > resources for finding more info out about the code and still no luck. > > You basically need to go with some commercial company and pay > them. Go to > the .biz list > > I for one have used numerous open source free code...and > sometimes it was a > struggle...but most of the time you can find the programmer's > email in the > sourcecode...a lot of them are willing to give you some help/info. > > I haven't seen you mention any particular code by name you are having > trouble with either. > > Maybe if you did...someone here has already figured it out > and would be > willing to help. I did mention the code. I mentioned the two python API's, and the perl API. http://sourceforge.net/projects/pyst http://py-asterisk.berlios.de/py-asterisk.php http://search.cpan.org/~xantus/POE-Component-Client-Asterisk-Manager/Manager.pm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
The manager interface would be horrible to code an API for. I don't envy the people that have done it. I've been screen scraping with python and perl for years, but the the output is not standardised. For example a 'sippeers' command does not end with a '--END COMMAND--' while a 'sip show channels' command does. > -Original Message- > From: Don Fanning [mailto:[EMAIL PROTECTED] > Sent: Wednesday, August 16, 2006 1:30 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Manager Interface API's > > > The manager interface isn't some mystical beast that can't be > overcome. > Try the wiki if you're lost. Really people scripting isn't that hard. > If you don't like the way people do code, there's nothing stopping you > from writing something new (except for lack of skill but that's why > people do it for a living). > > You buy them books, send them to school and all they do is eat the > pages. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Douglas > Garstang > Sent: Wednesday, August 16, 2006 12:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion; > [EMAIL PROTECTED] > Subject: RE: [asterisk-users] Manager Interface API's > > Actually, because there's no documentation, I don't have > anything that I > can use. > > > -Original Message- > > From: Dovid Bender [mailto:[EMAIL PROTECTED] > > Sent: Tuesday, August 15, 2006 12:54 PM > > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - > > Non-Commercial Discussion > > Subject: Re: [asterisk-users] Manager Interface API's > > > > > > Some of them write it for them selves and out of the goodness > > of thier heart > > will put out there for free. They dont need doc's since they > > wrote it them > > selves. Be happy that you got it for free. Do you want > people to stop > > releasing code because others complain ? > > - Original Message - > > From: "John Novack" <[EMAIL PROTECTED]> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > Sent: Tuesday, August 15, 2006 12:39 PM > > Subject: Re: [asterisk-users] Manager Interface API's > > > > > > > I, for one, didn't take his comment as anything other than > > constructive > > > Lack of documentation is an issue, open source or not. > > > It is an unfortunate situation that many very smart coders > > understand what > > > they have created, but are unwilling or unable to supply enough > > > information for many others to make effective use of > their creation > > > How many have struggled through the years with uncommented > > or poorly > > > commented code when the original creator is off to > greener pastures? > > > > > > JMO > > > > > > John Novack > > > > > > > > > Moises Silva wrote: > > >> Douglas. Please take this as a constructive comment. I > > have followed > > >> your questions in asterisk-dev and users lists, and you > > always seem to > > >> make non constructive comments about the people giving > > code/work for > > >> Free. And you focus in the negative part, never giving > > importance to > > >> the positive things about it. > > >> > > >> If you dont like something, then change it yourself, they are not > > >> providing a payed service. The source is available AS-IS > > if you want > > >> it, and if you like it, take it; If you dont, just ignore > > it, try to > > >> not make peyorative comments. > > >> > > >> Regards > > >> > > >> On 8/15/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: > > >>> > > >>> > > >>> Well, I don't know about you, but if I have to read the > > source code to > > >>> work > > >>> out how it works, I'm going to go and look at someone > > elses, that may > > >>> have > > >>> some BASIC documentation and examples. > > >>> > > >>> -Original Message- > > >>> From: Don [mailto:[EMAIL PROTECTED] > > >>> Sent: Tuesday, August 15, 2006 9:09 AM > > >>> To: Asterisk Users Mailing List - Non-Commercial Discussion > > >>> Subject: Re: [asterisk-users] Manager Interface API's > > >>> > > >>> > > >>> > > >>> Probably cause it is someone like most of us sitting at > home doing > > >>> it...releasing it for free...so why would we write pages of > > >>> documentation > > >>> for it? > > >>> If it's open source and it's free...Then offer them some > > money to make > > >>> documentation for it hehe... > > >>> > > >>> > > >>> - Original Message - > > >>> From: Douglas Garstang > > >>> To: Asterisk Users Mailing List - Non-Commercial Discussion > > >>> Sent: Tuesday, August 15, 2006 11:05 AM > > >>> Subject: [asterisk-users] Manager Interface API's > > >>> > > >>> > > >>> Can anyone recommend the best Manager Interface API, > > putting language > > >>> preferences aside? > > >>> > > >>> The python and perl ones have bupkiss documentation. I > > can't understand > > >>> why > > >>> anyone would even write an api and make it publically > > available without > > >>> documenting it.
[asterisk-users] No "zap" command?
Hi, all. I've just set up an Asterisk box -- to the best of my knowledge, no differently than any of the others that I've set up. Only one minor caveat: there's no "zap" command. Huh? Glancing at the startup, there's no mention of chan_zap, which I assume is partially the reason. However, I'm using -the exact same- zapata.conf, extensions.conf, and zaptel.conf from a different install, so I would imagine it would have been invoked if it were a config issue. Is there a compile-time option that we missed? [And, for the record, no zap errors whatsoever in the log. So it's not like it's trying to load chan_zap.o and failing or anything.] Any ideas would be greatly appreciated... Thanks, -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Page()
I guess it might not get compiled if you don't have a timer. Install ztdummy, recompile asterisk and try again. On 8/16/06, Dennis P. Clark <[EMAIL PROTECTED]> wrote: I am running Fedora 5 Cat /proc/sys/kernel/osrelease 2.6.15-1.2054_FC5 Zaptel 1.2.7 was not installed Edited xpp_usb.c and wcusb.c files in Zaptel to get it to compile and install by commenting out the following .owner = THIS_MODULE, I receive the following from CLI when I run "load module app_page.so" WARNING[28359]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_page.so: cannot open shared object file: No such file or directory And yes app_page.so does not exist in /usr/lib/asterisk Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 16, 2006 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() in the CLI do: show applications like page if you something there then you have it loaded, otherwise do: load app_page.so if that fails my guess is you need zaptel loaded first. On 8/16/06, Dennis P. Clark <[EMAIL PROTECTED]> wrote: > 1.2.10 > > Dennis Clark > DENPRO > WRK 207.618.1998 > CEL 443.415.0527 > FAX 1.888.811.8809 > [EMAIL PROTECTED] > > > > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Doug > Lytle > Sent: Wednesday, August 16, 2006 11:10 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Page() > > Dennis P. Clark wrote: > > I receive the following error in the Asterisk console when I try to > > execute the Page() application: > > > > WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' > > for extention (intercom, *, 1) > > > > What version of Asterisk are you running? > > Doug > > -- > > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little > Temporary Safety, deserve neither Liberty nor Safety." > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's
- Douglas Garstang <[EMAIL PROTECTED]> wrote: > Green pastures for sure. I think people develop the code, thinking > they will write docs later on. By the time they get close to releasing > their code, they've lost interest, or the priority of this project has > decreased. It's human nature. The open source community then ends up > with software thats unusable. > > Is it so ludicrous that if you develop an API that you document it? > We're not talking about developing a fahrenheight-celcius converter in > basic here. We're talking about an pplication rogramming > interface! It's a programming interface. It's not the same as some > GUI where you can get an idea of how it works by using it. If an API > doesn't have any docs, it's completely useless. I don't think anybody completely disagrees with you. Documentation is important, and people like Jared Smith and Leif Madsen are doing a great job with doing documentation after the fact. But, just to play devils advocate (no, not the pinball game), you point out that "API" stands for "Application Programming Interface". If you are able to write a program which uses the API, you should also be able to read the comments in the code and other code that uses the API, to fairly easily figure out how it works. :) If the documentation was so horrible that nobody could figure it out, there would be no programs that use it - and we both know that to be false. > > Doug. -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP asterisk over Linksys VPN
Dan, what Linksys device was it? On 8/16/06, Dan Casey <[EMAIL PROTECTED]> wrote: I tried to do sip over vpn with with a linksys router handle just one phone. When I tried it, it worked fine. Once i shipped it out we had all types of problems. at first it was fine, then 1 out of 5 calls would sound like cell phones. Now I can call him be he can't hear anything. Everything else works fine through the vpn. Most importantly, I can't trouble shoot correctly. I finally gave up and got him callvantage. Now all I have to worry about is forwarding a DID number. Raymond McKay wrote: > I usually run the RV series of router for this. Much better > thoughourput on the VPN. Remember these low end devices can usually > only handle about 1Mbps - 3Mbps of encryption max depending on the > unit. Other than that, I have had up to 8 behind a VPN such as this. > I do generally recommend though that a small appliance style asterisk > box sit on any side of a remote connection with a 1 port FXO card > installed for timing, emergency 911 capability, and trucking and > jitterbuffer support over IAX2. This, IMHO, tends to provide for > better reliability. I generally recommend some kind of HDD less > Compact Flash based system. Less mechanicals to break and you can pick > one up generally for $600-$800 with the digium card depending on speed > and number of phones to support. > > Regards, > > Raymond McKay > President > RAYNET Technologies LLC > http://www.raynettech.com > (860) 693-2226 x 31 > Toll Free (877) 693-2226 > - Original Message - From: "C F" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, August 15, 2006 1:23 PM > Subject: [asterisk-users] SIP asterisk over Linksys VPN > > >> Has anybody tried using a VPN and around 10 phones behind the tunnel >> to connect to an asterisk server using Linksys VPN routers? >> Like this one: >> http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1115416832495&pagename=Linksys%2FCommon%2FVisitorWrapper >> >> ___ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speed dials on Polycom IP601?
Empty line keys will be filled with speed dial entries in the phone's directory - when creating a directory entry, set the speed dial value to 1 for the first, 2 for the next.. etc. CP On 16-Aug-06, at 11:23 AM, Warren ((mailing lists)) wrote: I just got my first IP601 and put together my first * system (yay!) I have the first 2 buttons set up to be for the extension for the phone. I was wondering how I could make the remaining 4 into speed dials? IE: label button 3 "Sales mgr" and have it dial extension 246. TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXMODEM
Giedrius Augys wrote: And when I make ./build , I get this: /tmp/cckNRgf3.o(.text+0xfc): In function `cleanup': /home/eryx/hylafax/iaxmodem-0.1.14/iaxmodem.c:165: undefined reference to `iax_destroy' You probably have a conflicting version of libiax2. try doing: ./build static instead. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
Doug, I'm sure that you are not the only one who considers an API w/o docs to be of limited or no value. I just doubt that many people have use for a management API because they don't use the Asterisk manager interface very much. On a side note: some folks who have limited programming abilities but who otherwise want to contribute to the community have been know to "figure it out" and then document it. If there's a great API w/o any docs, someone might volunteer to do the dirty work. If the management API in question seems useful then I would ask: is anyone interested in creating some documentation? It could be submitted to the module's author for review, which might stimulate him to get involved in the documentation process. Just a thought... -MC > > I must be about the only person who thinks that an API without > documentation is completely useless. As I said in another post, this isn't > some GUI where you can poke around with it and work out how it works by > experimentation. Documentation of the API is essential. > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's
Douglas Garstang wrote: 1. A documenation of the functions, arguments, classes, objects. http://www.asterisk.org/doxygen/ - Use it. If it is not clear, ask a very specific question on the appropriate list. Then perhaps submit a documentation issue once others agree the documentation needs to be clearer/added 2. Some basic examples, because 1. means nothing without examples. Use Google - I see all kinds of examples out there. Jeremy McNamara P.S. Where in the GPL does it say that documentation is required? If you are not prepared to be neck deep into the code, then open-source is not for you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
Douglas has a point, and a legitimate one at that. Setting aside your personal feelings about Doug and his style of commenting, please consider that 'lack of documentation' is either the first- or second-most-often cited criticism leveled against open-source software and the OSS community. Lack of documentation *IS* a serious challenge with OSS. That is why it is considered poor style to upload some code to SourceForge.net and document it by saying, "Look at the source code and figure it out your self." Even a modicum of documentation, be it a brief listing of available methods or a few sample scripts with one or two sentences explaining what they do is quantum leaps above "source code only" documentation. Am I trying to criticize someone who gifts to the world the fruits of his/her labor? No. I'm simply saying that a little documentation makes such a gift infinitely more useful to the recipients. One person spending a few hours documenting his code will save hundreds, perhaps even thousands, of man-hours of end users "figuring it out." That is a serious return on investment if ever there was one. -MC > > In my opinion, and it seems perfectly logical to me, if someone writes > some code, but provides no documentation, such that no one can use it, > then what is the point? They have not provided a solution to anyones > problem except their own, and have no added value to the open source > community in any way, except to create 'vapourware' whereby software > appears to be available, but is unusable, because no one can work out how > to make it work. > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
> -Original Message- > From: Matt Riddell (NZ) [mailto:[EMAIL PROTECTED] > Sent: Wednesday, August 16, 2006 1:35 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk 'Hosting' > > > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Douglas Garstang wrote: > > > It's obvious that Asterisk was designed more for the > enterprise (ie a single company), rather than for the carrier > (ie multiple companies). It's a bit hard to explain here, but > even with more than a few companies, the config files and > dial plan start to become horribly complex. > > > > Our first customer has 15 contexts (right now) in > extensions.conf (we've broken each company into a separate > files included from extensions.conf and sip.conf for some > manageability). At several hundred companies, that's several > thousand contexts. We have three Asterisk boxes, and can add > more, but the config is (almost) idential between them for > redundancy, and this means that each Asterisk box has to have > a dialplan configured for all companies. > > And so you're thinking it would be better to run several hundred > Asterisk instances?! > > Good luck. > > I think your project would work a lot better if you worked like this: > > 1) Get requirements > 2) Map features and limitations of products > 3) Write PseudoCode > 4) Work out ways to load test your ideas > 5) Write real code > 6) Load test again with real code > > Hint: Layer your system so that each component is not doing too much > > Hint #2: Read: http://www.astricon.net/files/David_Zimmer_EUR06.pdf Interesting, but there's no meat. Do you know if they've actually implemented this yet, or are still designing it? This is essentially the same thing as we are doing (except I'm expected to do it all myself), except we aren't using realtime because it's not documented, and the asterisk dialplan language isn't advanced enough to deliver all the feature sets we need. We have a 2000+ line python script that handles all call routing logic. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's
In my opinion, and it seems perfectly logical to me, if someone writes some code, but provides no documentation, such that no one can use it, >> then what is the point? Because they probably wrote it for something they needed and released it to the general public...to those that want to take the time to figure out how such and such handled the problem they had. If people wanted to try and understand it fine...if they didn't fine...is how they looked at it...I mean if I write something specifically for me and released it cause I thought someone with a high understanding of what they were doing would want to use it...I wouldn't write documentation for it either. If I wrote something specifically to release to the public...I would write documentation for it. - Original Message - From: "Douglas Garstang" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, August 16, 2006 3:34 PM Subject: RE: [asterisk-users] Manager Interface API's -Original Message- From: Moises Silva [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 15, 2006 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager Interface API's Douglas. Please take this as a constructive comment. I have followed your questions in asterisk-dev and users lists, and you always seem to make non constructive comments about the people giving code/work for Free. And you focus in the negative part, never giving importance to the positive things about it. In my opinion, and it seems perfectly logical to me, if someone writes some code, but provides no documentation, such that no one can use it, then what is the point? They have not provided a solution to anyones problem except their own, and have no added value to the open source community in any way, except to create 'vapourware' whereby software appears to be available, but is unusable, because no one can work out how to make it work. If you dont like something, then change it yourself, they are not providing a payed service. The source is available AS-IS if you want it, and if you like it, take it; If you dont, just ignore it, try to not make peyorative comments. I'll refer to my opinion above. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.1/421 - Release Date: 8/16/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-942 TFTP Provisioning
I'm trying to provision some spa-942 phones via TFTP. The phones get their address from a dhcp server which sends it option 66 (address of the tftp server). After spending some time with the phones and even breaking down to sniff traffic from the phones I see that they are not requesting their config from tftp. I can kind of "fake" the phones into grabbing their configs by doing something like: A little off-topic but how do you create the provisioning file for Linksys/Sipura devices? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Interface API's
It's plain and simple the answer to this... If you fall into any of these categories: a) Don't understand code to begin with b) Don't want to take the time to understand the code c) Don't have the time to understand the code d) Don't want to pay the the person who made the code to help you e) Can't find anyone else using the code f) Can't find the programer g) Have walked to the ends of the earth and exhausted all your online resources for finding more info out about the code and still no luck. You basically need to go with some commercial company and pay them. Go to the .biz list I for one have used numerous open source free code...and sometimes it was a struggle...but most of the time you can find the programmer's email in the sourcecode...a lot of them are willing to give you some help/info. I haven't seen you mention any particular code by name you are having trouble with either. Maybe if you did...someone here has already figured it out and would be willing to help. - Original Message - From: "Douglas Garstang" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" ; <[EMAIL PROTECTED]> Sent: Wednesday, August 16, 2006 3:28 PM Subject: RE: [asterisk-users] Manager Interface API's Actually, because there's no documentation, I don't have anything that I can use. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 15, 2006 12:54 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Manager Interface API's Some of them write it for them selves and out of the goodness of thier heart will put out there for free. They dont need doc's since they wrote it them selves. Be happy that you got it for free. Do you want people to stop releasing code because others complain ? - Original Message - From: "John Novack" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, August 15, 2006 12:39 PM Subject: Re: [asterisk-users] Manager Interface API's > I, for one, didn't take his comment as anything other than constructive > Lack of documentation is an issue, open source or not. > It is an unfortunate situation that many very smart coders understand what > they have created, but are unwilling or unable to supply enough > information for many others to make effective use of their creation > How many have struggled through the years with uncommented or poorly > commented code when the original creator is off to greener pastures? > > JMO > > John Novack > > > Moises Silva wrote: >> Douglas. Please take this as a constructive comment. I have followed >> your questions in asterisk-dev and users lists, and you always seem to >> make non constructive comments about the people giving code/work for >> Free. And you focus in the negative part, never giving importance to >> the positive things about it. >> >> If you dont like something, then change it yourself, they are not >> providing a payed service. The source is available AS-IS if you want >> it, and if you like it, take it; If you dont, just ignore it, try to >> not make peyorative comments. >> >> Regards >> >> On 8/15/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: >>> >>> >>> Well, I don't know about you, but if I have to read the source code to >>> work >>> out how it works, I'm going to go and look at someone elses, that may >>> have >>> some BASIC documentation and examples. >>> >>> -Original Message- >>> From: Don [mailto:[EMAIL PROTECTED] >>> Sent: Tuesday, August 15, 2006 9:09 AM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: Re: [asterisk-users] Manager Interface API's >>> >>> >>> >>> Probably cause it is someone like most of us sitting at home doing >>> it...releasing it for free...so why would we write pages of >>> documentation >>> for it? >>> If it's open source and it's free...Then offer them some money to make >>> documentation for it hehe... >>> >>> >>> - Original Message - >>> From: Douglas Garstang >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Sent: Tuesday, August 15, 2006 11:05 AM >>> Subject: [asterisk-users] Manager Interface API's >>> >>> >>> Can anyone recommend the best Manager Interface API, putting language >>> preferences aside? >>> >>> The python and perl ones have bupkiss documentation. I can't understand >>> why >>> anyone would even write an api and make it publically available without >>> documenting it. >>> >>> Doug. >>> >>> >>> >>> >>> >>> ___ >>> --Bandwidth and Colocation provided by Easynews.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> >>> >>> >>> No virus found in this incoming message. >>> Checked by AVG Free Edition. >>> Versi
RE: [asterisk-users] Manager Interface API's
The manager interface isn't some mystical beast that can't be overcome. Try the wiki if you're lost. Really people scripting isn't that hard. If you don't like the way people do code, there's nothing stopping you from writing something new (except for lack of skill but that's why people do it for a living). You buy them books, send them to school and all they do is eat the pages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, August 16, 2006 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: RE: [asterisk-users] Manager Interface API's Actually, because there's no documentation, I don't have anything that I can use. > -Original Message- > From: Dovid Bender [mailto:[EMAIL PROTECTED] > Sent: Tuesday, August 15, 2006 12:54 PM > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: Re: [asterisk-users] Manager Interface API's > > > Some of them write it for them selves and out of the goodness > of thier heart > will put out there for free. They dont need doc's since they > wrote it them > selves. Be happy that you got it for free. Do you want people to stop > releasing code because others complain ? > - Original Message - > From: "John Novack" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, August 15, 2006 12:39 PM > Subject: Re: [asterisk-users] Manager Interface API's > > > > I, for one, didn't take his comment as anything other than > constructive > > Lack of documentation is an issue, open source or not. > > It is an unfortunate situation that many very smart coders > understand what > > they have created, but are unwilling or unable to supply enough > > information for many others to make effective use of their creation > > How many have struggled through the years with uncommented > or poorly > > commented code when the original creator is off to greener pastures? > > > > JMO > > > > John Novack > > > > > > Moises Silva wrote: > >> Douglas. Please take this as a constructive comment. I > have followed > >> your questions in asterisk-dev and users lists, and you > always seem to > >> make non constructive comments about the people giving > code/work for > >> Free. And you focus in the negative part, never giving > importance to > >> the positive things about it. > >> > >> If you dont like something, then change it yourself, they are not > >> providing a payed service. The source is available AS-IS > if you want > >> it, and if you like it, take it; If you dont, just ignore > it, try to > >> not make peyorative comments. > >> > >> Regards > >> > >> On 8/15/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: > >>> > >>> > >>> Well, I don't know about you, but if I have to read the > source code to > >>> work > >>> out how it works, I'm going to go and look at someone > elses, that may > >>> have > >>> some BASIC documentation and examples. > >>> > >>> -Original Message- > >>> From: Don [mailto:[EMAIL PROTECTED] > >>> Sent: Tuesday, August 15, 2006 9:09 AM > >>> To: Asterisk Users Mailing List - Non-Commercial Discussion > >>> Subject: Re: [asterisk-users] Manager Interface API's > >>> > >>> > >>> > >>> Probably cause it is someone like most of us sitting at home doing > >>> it...releasing it for free...so why would we write pages of > >>> documentation > >>> for it? > >>> If it's open source and it's free...Then offer them some > money to make > >>> documentation for it hehe... > >>> > >>> > >>> - Original Message - > >>> From: Douglas Garstang > >>> To: Asterisk Users Mailing List - Non-Commercial Discussion > >>> Sent: Tuesday, August 15, 2006 11:05 AM > >>> Subject: [asterisk-users] Manager Interface API's > >>> > >>> > >>> Can anyone recommend the best Manager Interface API, > putting language > >>> preferences aside? > >>> > >>> The python and perl ones have bupkiss documentation. I > can't understand > >>> why > >>> anyone would even write an api and make it publically > available without > >>> documenting it. > >>> > >>> Doug. > >>> > >>> > >>> > >>> > >>> > >>> ___ > >>> --Bandwidth and Colocation provided by Easynews.com -- > >>> > >>> asterisk-users mailing list > >>> To UNSUBSCRIBE or update options visit: > >>> > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >>> > >>> > >>> > >>> > >>> No virus found in this incoming message. > >>> Checked by AVG Free Edition. > >>> Version: 7.1.405 / Virus Database: 268.10.10/419 - Release Date: > >>> 8/15/2006 > >>> > >>> > >>> > >>> > >>> > >>> ___ > >>> --Bandwidth and Colocation provided by Easynews.com -- > >>> > >>> asterisk-users mailing list > >>> To UNSUBSCRIBE or update options visit: > >>> > >>>
[asterisk-users] Strange CLI Output
While the console is in monitor mode (asterisk -r) I see duplicate messages from Asterisk one after the other - But not when I connect via SSH. Example: -- Starting simple switch on Zap/65-1 -- Starting simple switch on Zap/65-1 What could cause this? Thanks Bart ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trixbox Fax to PDF
Well, I have my trixbox receiving faxes and e-mailing them as PDFs. Here's my setup:[General Settings]Extension of fax machine for receiving faxes: systemEmail to have faxes sent to:
[EMAIL PROTECTED]Email to have faxes appear they are sent from: [EMAIL PROTECTED][Miscellaneous Destinations]Description: incoming fax
Dial: {core:simu_fax}(No featurecode set.)[Inbound Routes]DID Number: a unique DIDFax Extension: system (have also tried freePBX default
)Fax Email: [EMAIL PROTECTED]Fax Detection Type: NVFaxPause after answer: 6(Set Destination)Misc Destinations: incoming fax
My
problem is this: when I send a fax to this route, asterisk will e-mail
the PDF to email1 instead of email2. If I only wanted one fax inbox,
this would be fine, but I'm looking to make this Trixbox a fax server
that will be able to handle multiple fax inboxes.Any idea what I've got wrong here?
___
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Re: [asterisk-users] Digium TDM400P Vs Sangoma A200
Rich Adamson wrote: Correction... it only plugs into "one" pci slot, but anything beyond four ports covers up additional pci slots (even though it doesn't plug into the pci connector). So, a 24 port a200 card would essentially render "all" pci slots unusable due to size. Depends on where the MB lives in the case, to be sure. Some ( rack mount, to be sure ) cases have many more openings for cards than modern MB's And growing much beyond 8 ports would seem to dictate another solution, perhaps a T1? Not having a need I haven't looked closely at the economics, but a Sangoma T1 card and a used Adtran would be in the neighborhood. Regardless the A200 works with a much larger selection of MB's and Sangoma doesn't tell you to "try another motherboard"! JN The tdm2400 card consumes a single pci slot, but its length is so long that on many systems one would need to use a tin's snip to cut the fixed disk drive bays out of the way to make it fit. John Novack wrote: The A200 is a far better card. More forgiving of Motherboards, MUCH more expandable, slightly lower cost. Only real drawback is modules are in pairs, so if you want 4 FXO, you need to buy 4. It expands to 24 ports using one PCI slot Also, if you ever are rich, hardware echo cancel is an option John Novack Jonathan Borden wrote: I was wondering which of these cards would be better for a 1-2 line SOHO. I would like room to grow as well as I am concerned with voice quality and life expectancy of the product. Any input into which one and why would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Force immediate re-registration on sip reload
Is there any way to force Asterisk to re-register after a sip reload is issued? At the moment, after a sip reload is issued, sip show registry reports all sip UA entries as "Unregistered". How can I get Asterisk to immediately send out a registration request to the proxy? Similarly all SIP peers lose their registration status with Asterisk. So when the device is used to make a call immediately after the SIP reload the call is not processed by Asterisk. It takes about 2 minutes 20 seconds before Asterisk starts processing SIP register requests from UAs and before it sends out the registration request to the proxy. How can I reduce this time? Any suggestions greatfully received. Regards Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP asterisk over Linksys VPN
Thanks for your answer, thruoughput was the only thing I was realy worried about. After I posted the question I decided that this one might be best mainly because of the QoS support it has: http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1150490914408&pagename=Linksys%2FCommon%2FVisitorWrapper as well as the price being quite reasonable. Have you tried that one before? Thank you On 8/16/06, Raymond McKay <[EMAIL PROTECTED]> wrote: I usually run the RV series of router for this. Much better thoughourput on the VPN. Remember these low end devices can usually only handle about 1Mbps - 3Mbps of encryption max depending on the unit. Other than that, I have had up to 8 behind a VPN such as this. I do generally recommend though that a small appliance style asterisk box sit on any side of a remote connection with a 1 port FXO card installed for timing, emergency 911 capability, and trucking and jitterbuffer support over IAX2. This, IMHO, tends to provide for better reliability. I generally recommend some kind of HDD less Compact Flash based system. Less mechanicals to break and you can pick one up generally for $600-$800 with the digium card depending on speed and number of phones to support. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 - Original Message - From: "C F" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, August 15, 2006 1:23 PM Subject: [asterisk-users] SIP asterisk over Linksys VPN > Has anybody tried using a VPN and around 10 phones behind the tunnel > to connect to an asterisk server using Linksys VPN routers? > Like this one: > http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1115416832495&pagename=Linksys%2FCommon%2FVisitorWrapper > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAXMODEM
Hi,I want to build fax server with asterisk uder CentOS 4.3. The problem is , that I can't to compile iaxmodem-0.1.14.I'm making these steps:To install libiax2: (if you're not going to 'build static')# cd lib/libiax2 # ./configure# make# make installTo install spandsp: (if you're not going to 'build static')# cd lib/spandsp# ./configure# make# make installAnd when I make ./build , I get this: /tmp/cckNRgf3.o(.text+0xfc): In function `cleanup':/home/eryx/hylafax/iaxmodem-0.1.14/iaxmodem.c:165: undefined reference to `iax_destroy'/tmp/cckNRgf3.o(.text+0x10d):/home/eryx/hylafax/iaxmodem-0.1.14/iaxmodem.c:166: undefined reference to `iax_destroy' /tmp/cckNRgf3.o(.text+0xe78): In function `t31_call_control_handler':/home/eryx/hylafax/iaxmodem-0.1.14/iaxmodem.c:436: undefined reference to `iax_destroy'/tmp/cckNRgf3.o(.text+0x1806): In function `iaxmodem': /home/eryx/hylafax/iaxmodem-0.1.14/iaxmodem.c:678: undefined reference to `iax_disable_jitterbuffer'/tmp/cckNRgf3.o(.text+0x180b):/home/eryx/hylafax/iaxmodem-0.1.14/iaxmodem.c:683: undefined reference to `iax_disable_debug' /tmp/cckNRgf3.o(.text+0x2224):/home/eryx/hylafax/iaxmodem-0.1.14/iaxmodem.c:899: undefined reference to `iax_destroy'/tmp/cckNRgf3.o(.text+0x2558):/home/eryx/hylafax/iaxmodem-0.1.14/iaxmodem.c:966: undefined reference to `iax_congestion' collect2: ld returned 1 exit statusMaybe somebody can help me to solve this problem?Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TDM400P Vs Sangoma A200
Correction... it only plugs into "one" pci slot, but anything beyond four ports covers up additional pci slots (even though it doesn't plug into the pci connector). So, a 24 port a200 card would essentially render "all" pci slots unusable due to size. The tdm2400 card consumes a single pci slot, but its length is so long that on many systems one would need to use a tin's snip to cut the fixed disk drive bays out of the way to make it fit. John Novack wrote: The A200 is a far better card. More forgiving of Motherboards, MUCH more expandable, slightly lower cost. Only real drawback is modules are in pairs, so if you want 4 FXO, you need to buy 4. It expands to 24 ports using one PCI slot Also, if you ever are rich, hardware echo cancel is an option John Novack Jonathan Borden wrote: I was wondering which of these cards would be better for a 1-2 line SOHO. I would like room to grow as well as I am concerned with voice quality and life expectancy of the product. Any input into which one and why would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Page()
I am running Fedora 5 Cat /proc/sys/kernel/osrelease 2.6.15-1.2054_FC5 Zaptel 1.2.7 was not installed Edited xpp_usb.c and wcusb.c files in Zaptel to get it to compile and install by commenting out the following .owner = THIS_MODULE, I receive the following from CLI when I run "load module app_page.so" WARNING[28359]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_page.so: cannot open shared object file: No such file or directory And yes app_page.so does not exist in /usr/lib/asterisk Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 16, 2006 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() in the CLI do: show applications like page if you something there then you have it loaded, otherwise do: load app_page.so if that fails my guess is you need zaptel loaded first. On 8/16/06, Dennis P. Clark <[EMAIL PROTECTED]> wrote: > 1.2.10 > > Dennis Clark > DENPRO > WRK 207.618.1998 > CEL 443.415.0527 > FAX 1.888.811.8809 > [EMAIL PROTECTED] > > > > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Doug > Lytle > Sent: Wednesday, August 16, 2006 11:10 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Page() > > Dennis P. Clark wrote: > > I receive the following error in the Asterisk console when I try to > > execute the Page() application: > > > > WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' > > for extention (intercom, *, 1) > > > > What version of Asterisk are you running? > > Doug > > -- > > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little > Temporary Safety, deserve neither Liberty nor Safety." > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
> -Original Message- > From: Brian Capouch [mailto:[EMAIL PROTECTED] > Sent: Tuesday, August 15, 2006 11:15 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Manager Interface API's > > > Douglas Garstang wrote: > > Can anyone recommend the best Manager Interface API, > putting language > > preferences aside? > > > > The python and perl ones have bupkiss documentation. I > can't understand > > why anyone would even write an api and make it publically available > > without documenting it. > > > > Have you taken your "be nice on the lists" pill today? > > The most likely explanation is that people have written these > interfaces > primarily for their own use, and when they decided to share > with others, > only had/made time to minimally document them. > > Do you understand that? You've got me doubting you can't understand > such things, so I wonder why you *say* you don't understand. > > Unless you enjoy being a troll. I must be about the only person who thinks that an API without documentation is completely useless. As I said in another post, this isn't some GUI where you can poke around with it and work out how it works by experimentation. Documentation of the API is essential. Why did they have time to write it, but suddenly run out of time when it came time to document? It's human nature to start something and lose interest. Given that documentation always happens at the end, it's what usually suffers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
> -Original Message- > From: Stefan Reuter [mailto:[EMAIL PROTECTED] > Sent: Tuesday, August 15, 2006 11:10 AM > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: Re: [asterisk-users] Manager Interface API's > > > John Novack wrote: > > I, for one, didn't take his comment as anything other than > constructive > > Lack of documentation is an issue, open source or not. > > To make this thread even more constructive: > What kind of documentation do you expect from a Manager API package? 1. A documenation of the functions, arguments, classes, objects. 2. Some basic examples, because 1. means nothing without examples. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Real Time and sip.conf file used at
> On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote: >> Is it possible to use Asterisk RealTime and also config files (like >> sip.conf) at the same time? >> As much as I know, only one thing can be used and I need them both >> working!... >> > > Yes, you can use both at the same time. The only restriction is that > you cannot use the realtime static configuration and realtime > configuration. > > -- > Hi realtime static configuration and realtime configuration??? What is the difference, can you please explain? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
> -Original Message- > From: John Novack [mailto:[EMAIL PROTECTED] > Sent: Tuesday, August 15, 2006 10:40 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Manager Interface API's > > > I, for one, didn't take his comment as anything other than > constructive > Lack of documentation is an issue, open source or not. > It is an unfortunate situation that many very smart coders understand > what they have created, but are unwilling or unable to supply enough > information for many others to make effective use of their creation > How many have struggled through the years with uncommented or poorly > commented code when the original creator is off to greener pastures? Green pastures for sure. I think people develop the code, thinking they will write docs later on. By the time they get close to releasing their code, they've lost interest, or the priority of this project has decreased. It's human nature. The open source community then ends up with software thats unusable. Is it so ludicrous that if you develop an API that you document it? We're not talking about developing a fahrenheight-celcius converter in basic here. We're talking about an pplication rogramming interface! It's a programming interface. It's not the same as some GUI where you can get an idea of how it works by using it. If an API doesn't have any docs, it's completely useless. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
Douglas Garstang wrote: Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. Ok - And the problem is? Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: > It's obvious that Asterisk was designed more for the enterprise (ie a single > company), rather than for the carrier (ie multiple companies). It's a bit > hard to explain here, but even with more than a few companies, the config > files and dial plan start to become horribly complex. > > Our first customer has 15 contexts (right now) in extensions.conf (we've > broken each company into a separate files included from extensions.conf and > sip.conf for some manageability). At several hundred companies, that's > several thousand contexts. We have three Asterisk boxes, and can add more, > but the config is (almost) idential between them for redundancy, and this > means that each Asterisk box has to have a dialplan configured for all > companies. And so you're thinking it would be better to run several hundred Asterisk instances?! Good luck. I think your project would work a lot better if you worked like this: 1) Get requirements 2) Map features and limitations of products 3) Write PseudoCode 4) Work out ways to load test your ideas 5) Write real code 6) Load test again with real code Hint: Layer your system so that each component is not doing too much Hint #2: Read: http://www.astricon.net/files/David_Zimmer_EUR06.pdf - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE43N2S6d5vy0jeVcRAj0OAJ4vgp3aMeBiEsVsU+zqhyouu8CPlgCffPAv 0SccdLfefS8GUtkxZpIMpU4= =ciMO -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
Use a virtual private asterisk system. You'll be happier if you did. http://www.telephreak.org/papers/vpa/ > > Has anyone ever tried to run multiple instances of Asterisk > on a single system, running each with a different username, > and each in a separate base directory? Something like > /home/pbx/business-1, home/pbx/business-2 etc? > > > > Did it work? I assume for every service that Asterisk runs, > on each instance, you'd have to use a different port numbers, > which may get confusing. Each businesses phones would have to > be configred with different SIP ports then too. > > > > What about processes? I notice that Asterisk runs about 26 > processes (or are they threads?) for a single instance. > > It's obvious that Asterisk was designed more for the enterprise (ie a single company), rather than for the carrier (ie multiple companies). It's a bit hard to explain here, but even with more than a few companies, the config files and dial plan start to become horribly complex. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager Interface API's
> -Original Message- > From: Moises Silva [mailto:[EMAIL PROTECTED] > Sent: Tuesday, August 15, 2006 10:20 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Manager Interface API's > > > Douglas. Please take this as a constructive comment. I have followed > your questions in asterisk-dev and users lists, and you always seem to > make non constructive comments about the people giving code/work for > Free. And you focus in the negative part, never giving importance to > the positive things about it. In my opinion, and it seems perfectly logical to me, if someone writes some code, but provides no documentation, such that no one can use it, then what is the point? They have not provided a solution to anyones problem except their own, and have no added value to the open source community in any way, except to create 'vapourware' whereby software appears to be available, but is unusable, because no one can work out how to make it work. > > If you dont like something, then change it yourself, they are not > providing a payed service. The source is available AS-IS if you want > it, and if you like it, take it; If you dont, just ignore it, try to > not make peyorative comments. I'll refer to my opinion above. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] calling in-out
Hello people, I am having some issues with my new SIP provider.
The sip provider gives me only an IP address to configure my sip account,
since they do allow by IP address and not by username password.
This all configuration appears to work well, since I can originate a call
and it will ring the destination, and I can originate a call from PSTN and *
will "see" it. But none of both call difections will be stabilished.
If I originate a call from * to a PSTN number, with a sip debug I get:
Destroying call '[EMAIL PROTECTED]'
pbx*CLI>
<-- SIP read from 200.123.190.50:5060:
SIP/2.0 500 Server Internal Error
To: ;tag=3364745030-621025
From: "CrossFone" ;tag=as4d4398b9
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Contact: sip:[EMAIL PROTECTED]:5060
Via: SIP/2.0/UDP 200.59.45.210:5060;branch=z9hG4bK6c94441d;rport
Content-Length: 0
--- (8 headers 0 lines)---
-- Got SIP response 500 "Server Internal Error" back from 200.123.190.50
Transmitting (no NAT) to 200.123.190.50:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 200.59.45.210:5060;branch=z9hG4bK6c94441d;rport
From: "CrossFone" ;tag=as4d4398b9
To: ;tag=3364745030-621025
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/CrossFone-087b3e40 is circuit-busy
Destroying call '[EMAIL PROTECTED]'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Goto("SIP/1501-087acbf8", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing NoOp("SIP/1501-087acbf8", "Dial failed due to CONGESTION")
in new stack
-- Executing Macro("SIP/1501-087acbf8", "outisbusy|") in new stack
-- Executing Playback("SIP/1501-087acbf8", "all-circuits-busy-now") in
new stack
If I make a call to my SIP number, it will ring till I pickup the phone,
when I pickup the phone, I get:
<-- SIP read from 201.216.206.221:62477:
--- (0 headers 1 lines)---
pbx*CLI>
<-- SIP read from 200.123.190.50:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Max-Forwards: 70
Session-Expires: 3600;Refresher=uac
Supported: timer
To: 1159174200
From: ;tag=3364745421-27664
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
Via: SIP/2.0/UDP 200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe
Contact: sip:[EMAIL PROTECTED]:5060
Content-Type: application/sdp
Content-Length: 170
v=0
o=NexTone-MSW 1234 0 IN IP4 200.123.190.53
s=sip call
c=IN IP4 200.123.190.53
t=0 0
m=audio 21660 RTP/AVP 18 4 4 0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
--- (12 headers 8 lines)---
Using INVITE request as basis request -
[EMAIL PROTECTED]
Sending to 200.123.190.50 : 5060 (non-NAT)
Found peer 'CrossFone'
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 4
Found RTP audio format 0
Peer audio RTP is at port 200.123.190.53:21660
Found description format G729
Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x105
(g723|ulaw|g729)/video=0x0 (nothing), combined - 0x105 (g723|ulaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Looking for 1159174200 in from-sip-external (domain 200.59.45.210)
list_route: hop:
Transmitting (no NAT) to 200.123.190.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe;received=200.123
.190.50
From: ;tag=3364745421-27664
To: 1159174200
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Length: 0
---
-- Executing NoOp("SIP/5060-087ace18", "Received incoming SIP connection
from unknown peer to 1159174200") in new stack
-- Executing Set("SIP/5060-087ace18", "DID=1159174200") in new stack
-- Executing Goto("SIP/5060-087ace18", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing GotoIf("SIP/5060-087ace18", "0?from-trunk|1159174200|1") in
new stack
-- Executing Set("SIP/5060-087ace18", "TIMEOUT(absolute)=15") in new
stack
-- Channel will hangup at 2006-08-16 19:27:24 UTC.
-- Executing Answer("SIP/5060-087ace18", "") in new stack
We're at 200.59.45.210 port 19920
Adding codec 0x100 (g729) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 200.123.190.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
200.123.190.50:5060;branch=b86b531fb60caa03195a218a6e8947fe;received=200.123
.190.50
From: ;tag=3364745421-27664
To: 1159174200 ;tag=as7635cbf2
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Type: application/sdp
Content-Length: 231
v=0
o=root 2815 2815 IN IP4 200.59.45.210
s=session
c=IN IP4 200.59.45.210
t=0 0
m=audio 19920 RTP/AVP 18 4 0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
---
-- Executing Wait("SIP/5060-087ace18", "2") in new stack
pbx*CLI>
<-- SIP read
RE: [asterisk-users] Manager Interface API's
Actually, because there's no documentation, I don't have anything that I can use. > -Original Message- > From: Dovid Bender [mailto:[EMAIL PROTECTED] > Sent: Tuesday, August 15, 2006 12:54 PM > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: Re: [asterisk-users] Manager Interface API's > > > Some of them write it for them selves and out of the goodness > of thier heart > will put out there for free. They dont need doc's since they > wrote it them > selves. Be happy that you got it for free. Do you want people to stop > releasing code because others complain ? > - Original Message - > From: "John Novack" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, August 15, 2006 12:39 PM > Subject: Re: [asterisk-users] Manager Interface API's > > > > I, for one, didn't take his comment as anything other than > constructive > > Lack of documentation is an issue, open source or not. > > It is an unfortunate situation that many very smart coders > understand what > > they have created, but are unwilling or unable to supply enough > > information for many others to make effective use of their creation > > How many have struggled through the years with uncommented > or poorly > > commented code when the original creator is off to greener pastures? > > > > JMO > > > > John Novack > > > > > > Moises Silva wrote: > >> Douglas. Please take this as a constructive comment. I > have followed > >> your questions in asterisk-dev and users lists, and you > always seem to > >> make non constructive comments about the people giving > code/work for > >> Free. And you focus in the negative part, never giving > importance to > >> the positive things about it. > >> > >> If you dont like something, then change it yourself, they are not > >> providing a payed service. The source is available AS-IS > if you want > >> it, and if you like it, take it; If you dont, just ignore > it, try to > >> not make peyorative comments. > >> > >> Regards > >> > >> On 8/15/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: > >>> > >>> > >>> Well, I don't know about you, but if I have to read the > source code to > >>> work > >>> out how it works, I'm going to go and look at someone > elses, that may > >>> have > >>> some BASIC documentation and examples. > >>> > >>> -Original Message- > >>> From: Don [mailto:[EMAIL PROTECTED] > >>> Sent: Tuesday, August 15, 2006 9:09 AM > >>> To: Asterisk Users Mailing List - Non-Commercial Discussion > >>> Subject: Re: [asterisk-users] Manager Interface API's > >>> > >>> > >>> > >>> Probably cause it is someone like most of us sitting at home doing > >>> it...releasing it for free...so why would we write pages of > >>> documentation > >>> for it? > >>> If it's open source and it's free...Then offer them some > money to make > >>> documentation for it hehe... > >>> > >>> > >>> - Original Message - > >>> From: Douglas Garstang > >>> To: Asterisk Users Mailing List - Non-Commercial Discussion > >>> Sent: Tuesday, August 15, 2006 11:05 AM > >>> Subject: [asterisk-users] Manager Interface API's > >>> > >>> > >>> Can anyone recommend the best Manager Interface API, > putting language > >>> preferences aside? > >>> > >>> The python and perl ones have bupkiss documentation. I > can't understand > >>> why > >>> anyone would even write an api and make it publically > available without > >>> documenting it. > >>> > >>> Doug. > >>> > >>> > >>> > >>> > >>> > >>> ___ > >>> --Bandwidth and Colocation provided by Easynews.com -- > >>> > >>> asterisk-users mailing list > >>> To UNSUBSCRIBE or update options visit: > >>> > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >>> > >>> > >>> > >>> > >>> No virus found in this incoming message. > >>> Checked by AVG Free Edition. > >>> Version: 7.1.405 / Virus Database: 268.10.10/419 - Release Date: > >>> 8/15/2006 > >>> > >>> > >>> > >>> > >>> > >>> ___ > >>> --Bandwidth and Colocation provided by Easynews.com -- > >>> > >>> asterisk-users mailing list > >>> To UNSUBSCRIBE or update options visit: > >>> > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >>> > >>> > >> > >> > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocati
RE: [asterisk-users] Asterisk 'Hosting'
Brandon, Thanks. We're a litle past that stage of complexity. I'm just throwing the question out there because it's becoming obvious that trying to provision hundreds of customers on a cluster of Asterisk systems is going to be very hard to manage. -Original Message-From: Brandon Galbraith [mailto:[EMAIL PROTECTED]Sent: Wednesday, August 16, 2006 12:53 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk 'Hosting'Doug,I'd suggest using contexts, but then having two servers for redundancy also. That way, if one asterisk box goes down, you don't have 50-100 clients completely down.-brandon On 8/16/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. -Original Message-From: David Freeman [mailto:[EMAIL PROTECTED]]Sent: Wednesday, August 16, 2006 11:36 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk 'Hosting'You might be able to use virtual NICs to eliminate the problem with "non-standard" ports for a company's SIP phones. Or real NICs using a couple of multi-homed cards.I haven't tried it, though. On 8/16/06, Douglas Garstang <[EMAIL PROTECTED]> wrote: Has anyone ever tried to run multiple instances of Asterisk on a single system, running each with a different username, and each in a separate base directory? Something like /home/pbx/business-1, home/pbx/business-2 etc?Did it work? I assume for every service that Asterisk runs, on each instance, you'd have to use a different port numbers, which may get confusing. Each businesses phones would have to be configred with different SIP ports then too. What about processes? I notice that Asterisk runs about 26 processes (or are they threads?) for a single instance. Doug. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED] AIM: brandong00Voice: 630.400.6992"A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost" ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP asterisk over Linksys VPN
I tried to do sip over vpn with with a linksys router handle just one phone. When I tried it, it worked fine. Once i shipped it out we had all types of problems. at first it was fine, then 1 out of 5 calls would sound like cell phones. Now I can call him be he can't hear anything. Everything else works fine through the vpn. Most importantly, I can't trouble shoot correctly. I finally gave up and got him callvantage. Now all I have to worry about is forwarding a DID number. Raymond McKay wrote: > I usually run the RV series of router for this. Much better > thoughourput on the VPN. Remember these low end devices can usually > only handle about 1Mbps - 3Mbps of encryption max depending on the > unit. Other than that, I have had up to 8 behind a VPN such as this. > I do generally recommend though that a small appliance style asterisk > box sit on any side of a remote connection with a 1 port FXO card > installed for timing, emergency 911 capability, and trucking and > jitterbuffer support over IAX2. This, IMHO, tends to provide for > better reliability. I generally recommend some kind of HDD less > Compact Flash based system. Less mechanicals to break and you can pick > one up generally for $600-$800 with the digium card depending on speed > and number of phones to support. > > Regards, > > Raymond McKay > President > RAYNET Technologies LLC > http://www.raynettech.com > (860) 693-2226 x 31 > Toll Free (877) 693-2226 > - Original Message - From: "C F" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, August 15, 2006 1:23 PM > Subject: [asterisk-users] SIP asterisk over Linksys VPN > > >> Has anybody tried using a VPN and around 10 phones behind the tunnel >> to connect to an asterisk server using Linksys VPN routers? >> Like this one: >> http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1115416832495&pagename=Linksys%2FCommon%2FVisitorWrapper >> >> ___ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: > Because Asterisk wasn't designed with carrier class features in mind. It was > designed for a single enterprise. The dialplan, and config files, start to > get very very complicated after you add more than a few companies. Combine > that with having to have multiple extensions for a single function (our > Queues are accessed by a regular extension but then have to dial another > 'virtual' extension so that DUNDi can work out the 'primary' server for a > queue) and so on. Anyway, it's becoming unmanagable. So write better management software, that's what we've and many others have done. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE43DHS6d5vy0jeVcRAnEyAJ9yNLv+vDF2esy4S6Hik8C46POiDQCeKd5X 6BND4aXxRw5nxifVC1oQM6U= =pfPk -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
> and each in a separate base directory? Something like > /home/pbx/business-1, home/pbx/business-2 etc? > > Use VPSs, like www.openvz.org Pablo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
You could use Xen on Fedora Core 6 and virtualize each instance if you feel the need is there.On 8/16/06, Douglas Garstang < [EMAIL PROTECTED]> wrote:> -Original Message-> From: Matt Riddell (NZ) [mailto: [EMAIL PROTECTED]]> Sent: Wednesday, August 16, 2006 12:06 PM> To: Asterisk Users Mailing List - Non-Commercial Discussion> Subject: Re: [asterisk-users] Asterisk 'Hosting' >>> -BEGIN PGP SIGNED MESSAGE-> Hash: SHA1>> Douglas Garstang wrote:> > Has anyone ever tried to run multiple instances of Asterisk> on a single system, running each with a different username, > and each in a separate base directory? Something like> /home/pbx/business-1, home/pbx/business-2 etc?> >> > Did it work? I assume for every service that Asterisk runs,> on each instance, you'd have to use a different port numbers, > which may get confusing. Each businesses phones would have to> be configred with different SIP ports then too.> >> > What about processes? I notice that Asterisk runs about 26> processes (or are they threads?) for a single instance. >> Why not just use different contexts for each company?Because Asterisk wasn't designed with carrier class features in mind. It was designed for a single enterprise. The dialplan, and config files, start to get very very complicated after you add more than a few companies. Combine that with having to have multiple extensions for a single function (our Queues are accessed by a regular extension but then have to dial another 'virtual' extension so that DUNDi can work out the 'primary' server for a queue) and so on. Anyway, it's becoming unmanagable. Doug.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED]AIM: brandong00Voice: 630.400.6992"A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost" ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 'Hosting'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Douglas Garstang wrote: > Well, we're talking about several dozen, maybe 100, companies, per Asterisk > box here. Surely all the more reason to do it with contexts than instances. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFE422LS6d5vy0jeVcRAhQQAJ9XLDlNHe2Xv7oBA568nvaPbnKI1wCeM+t4 5geXNT+XaPj1gSxdSROQKYE= =AZoL -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk load testing
Nitin Gupta wrote: > Hi, > did anyone try do load-testing on asterisk, for sip channel calls? > I want to have a rough estimate about - how many calls, an asterisk server, > running on say dual 240 opteron with 1 GB memory, can handle? > Also how much internet bandwidth does a typical call requires? I heard around > 20Kbps with typical codecs, is that right? > > Thanks in advance, > Nitin > http://www.erlang.com/calculator/lipb/ Go read up on some codecs http://www.vocal.com/data_sheets/codecs_voip.html Too many variables to answer your question. -- =+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743 sil infiltrated . net http://www.infiltrated.net "How a man plays the game shows something of his character - how he loses shows all" - Mr. Luckey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
