[asterisk-users] How to write custom functions in AEL2 ,
Hi,
I'm using asterisk 1.6.1 and AEL2.
I'm trying to find the best way to write my own custom functions ?
At the moment, I'm using this pattern (extensions.ael) :
context foo {
123 = {
myfunc(123456);
NoOp(${GOSUB_RETVAL});
};
macro myfunc (arg) {
Return (${arg});
}
1. First, I keep getting warnings like
Warning: file /etc/asterisk/extensions.ael, line 446-446: application call
to Return affects flow of control, and needs to be re-written using AEL if,
while, goto, etc. keywords instead!
and I would like to get rid of them.
2. Secondly, I would like not to use GOSUB_RETVAL and call a custom
function just like I'm calling other functions with statements like :
123 = {
NoOp(TOLOWER(fOo BaR));
NoOp(myfunc(123456));
};
What would you advise me to do ?
Regards
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Re: [asterisk-users] How to write custom functions in AEL2 ,
2009/5/11 Olivier oza-4...@myamail.com
2. Secondly, I would like not to use GOSUB_RETVAL and call a custom
function just like I'm calling other functions with statements like :
123 = {
NoOp(TOLOWER(fOo BaR));
Here, I meant
NoOp(read this ${TOLOWER(fOo BaR)});
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[asterisk-users] Support of /* */ comments in ael.vim
Hello, It seems /* */ comments are not supported in ael.vim (which brings AEL syntax-highlighting to vim). Is it hard to add this feature and have uploaded in vim extensions downloading site ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building a System.
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John F. Ervin wrote: | So, people have recommended building a system from scratch, start with a | CentOS base and installing asterisk and all of the other utilities. | I've only used Trixbox for my business system. I'm wondering what | surprises I'd run into. Right now, I know I'd need the OS, Asterisk, | something like FreePBX, I have a x100p card so I'd need Zaptel, does | that come with asterisk? Fax support, seems to work with Trixbox, but | I've heard that it needs to be loaded. Voicemail etc.? I mean, I don't | know exactly what you'd need because almost everything I need comes with | the Trixbox build. | | Are there (??) instructions for people who are experienced at the | Trixbox level but wish to move on? Here are some nice videos; http://www.asterikast.com/episodes.php - -- Powered by Gentoo GNU/LINUX http://www.linuxcrazy.com pgp.mit.edu -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkoH4usACgkQcZ+z4vAcSsyqwwCeOq3ZSJrGcgyQSSRc44Et1To3 Zq0An3JEy9oIqM8LsBPv1Pyrrf80PJys =Yx5x -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building a System.
On 11/05/09 04:21, John F. Ervin wrote: snip / Are there (??) instructions for people who are experienced at the Trixbox level but wish to move on? Sure, the TFOT book is a great start. If you want to use Ubuntu or Debian rather than Centos then Asterisk is in the Debian and Ubuntu Server Repositories. # sudo apt-get install asterisk # sudo m-a -f get zaptel-source # sudo ECHO_CAN_NAME=OSLEC m-a -t a-i zaptel (The last command might not be needed any more as I believe OSLEC is now the default EC) Should do most of it. I written a few blog articles on Asterisk (building from scratch and installing on Ubuntu etc. Including Zaptel and OSLEC for the x100p) Here's one for getting it running on Ubuntu server: http://www.theopensourcerer.com/2009/02/12/asterisk-zaptel-oslec-and-ubuntu-server/ And here's all of them: http://www.theopensourcerer.com/tag/asterisk/ There are, of course, many more guides and advice out there too. Google is your friend as it the extremely useful http://www.voip-info.org/ wiki.. HTH Alan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Support of /* */ comments in ael.vim
Olivier schrieb: It seems /* */ comments are not supported in ael.vim (which brings AEL syntax-highlighting to vim). Are C-style comments supported in AEL? I don't think so. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Support of /* */ comments in ael.vim
On Mon, May 11, 2009 at 1:55 PM, Philipp Kempgen philipp.kemp...@amooma.de wrote: Olivier schrieb: It seems /* */ comments are not supported in ael.vim (which brings AEL syntax-highlighting to vim). Are C-style comments supported in AEL? I don't think so. They are. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Support of /* */ comments in ael.vim
2009/5/11 Philipp Kempgen philipp.kemp...@amooma.de Olivier schrieb: It seems /* */ comments are not supported in ael.vim (which brings AEL syntax-highlighting to vim). Are C-style comments supported in AEL? I don't think so. This page says it does http://www.voip-info.org/wiki/view/Asterisk+AEL2 In my trials, it seems to work. Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom-330 not displaying line buddy label?
Hello, I have so far Polycom 501 and 403 which displays the label of a key and the name of a buddy near its key. Today I received new 330's and they do not display the name, only 1 2. Besides that they work correctly. Anyone has an idea, or is it a known feature? Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write custom functions in AEL2 ,
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, May 11, 2009 3:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to write custom functions in AEL2
,
Hi,
I'm using asterisk 1.6.1 and AEL2.
I'm trying to find the best way to write my own custom functions
?
At the moment, I'm using this pattern (extensions.ael) :
context foo {
123 = {
myfunc(123456);
NoOp(${GOSUB_RETVAL});
};
macro myfunc (arg) {
Return (${arg});
}
1. First, I keep getting warnings like
Warning: file /etc/asterisk/extensions.ael, line 446-446:
application call to Return affects flow of control, and needs to be
re-written using AEL if, while, goto, etc. keywords instead!
and I would like to get rid of them.
Unfortunately, AEL does not support using return with a value at the
moment. There is a patch on Reviewboard that does this, as well as
*simple* direct assignment from an AEL macro return:
http://reviewboard.digium.com/r/114/
2. Secondly, I would like not to use GOSUB_RETVAL and call a
custom function just like I'm calling other functions with statements
like :
123 = {
NoOp(TOLOWER(fOo BaR));
NoOp(myfunc(123456));
};
What would you advise me to do ?
That requires rather a lot more work than the above patch, but if you
use the direct assignment at least you needn't worry about GOSUB_RETVAL.
Regards,
- Brad
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[asterisk-users] No CDR generated for calls to queues with no agents
Hi, I am using Asterisk 1.6.0.9. I have calls coming from another asterisk server via IAX and lands in a queue. I have noticed that if there are no agents logged in the queue no CDR is generated. If there is one agent logged in then the phone rings and a CDR is generated even if the call was pickedup or not. Looking at the bug db http://bugs.digium.com/view.php?id=13691 is similar to the problem I am facing. Btw queue_log is showing all entries as expected. My extensions.conf is as follows: [general] static=yes writeprotect=no [globals] [inbound-calls] exten = queue, 1, Queue(genenq) exten = queue, n, Hangup [sip] #include dialplan/1xxx.conf cdr.conf is [general] enable=yes unanswered = yes batch=no safeshutdown=yes [csv] usegmtime=no loguniqueid=yes loguserfield=yes [custom] loguniqueid=yes loguserfield=yes is this a bug or am I doing some thing stupid ? raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF received twice
Hi all,
I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from
my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so
I can use the GW. For this I use:
exten = s,1,NoOp(One of our workers (${CALLERID(number)}) is calling
office) ;callerID is the one of the calling mobile phone
exten = s,n,Background(silence/1)
; Nokia E65 send digits in DTMF mode, no need to take care about input
corrections
;
exten = s,n(enterDigits),Read(myExten,pls-entr-num-uwish2-call,0,,,3)
exten = s,n,GotoIf($[${myExten}=]?enterDigits)
[...]
Problem is that received DTMF digits in ${myExten} are received twice eg
for 1234 ${myExten} has 11223344. I correct the extension by dialplan
but I think it's not really a solution.
In sip.conf, the dtmfmode is set to auto. If I set it to rfc2833, the
same behaviour.
Can somebody confirm this before I open a bug, thanks.
Regards
--
Daniel
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Re: [asterisk-users] Building a System.
This is the most useful script anyone has published on this list for a long time. Thanks David, * stars on this.. I can finally have our clients move from trixbox to an asterisk vanilla system in no time now. Ps.. here are a few suggestions.. àMove ntpdate ntp.bri.connect.com.au out of the shell and into a var of aster-vars àMaybe print out locations of interesting files after install àBring iptables back up.. with sip rules included But its a 5/5 for me, will try it in virtualization also. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaverstyn, David C Sent: May-11-09 12:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Building a System. Hi John, Im not sure if this will help you or not but I created a script that will install Asterisk with all the required components for DAHDI, Faxing, fax to email, LDAPget, CDR, FOP etc. It can even include text to speech applications. I created it because I wanted to install Asterisk multiple times and as quickly as possible. It does the exact same steps as one would do when doing an install manually. I created the bash script aster-install http://www.klaverstyn.com.au/wiki/index.php?title=Aster-install . http://www.klaverstyn.com.au/wiki/index.php?title=Aster-install Regards David. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John F. Ervin Sent: Monday, 11 May 2009 1:22 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Building a System. So, people have recommended building a system from scratch, start with a CentOS base and installing asterisk and all of the other utilities. I've only used Trixbox for my business system. I'm wondering what surprises I'd run into. Right now, I know I'd need the OS, Asterisk, something like FreePBX, I have a x100p card so I'd need Zaptel, does that come with asterisk? Fax support, seems to work with Trixbox, but I've heard that it needs to be loaded. Voicemail etc.? I mean, I don't know exactly what you'd need because almost everything I need comes with the Trixbox build. Are there (??) instructions for people who are experienced at the Trixbox level but wish to move on? -- John F. Ervin Central Florida TeleSource, LLC. 4270 Aloma Ave #124-69C Winter Park, FL 32792 (W) 407-679-6238 (F) 866-566-1282 (F) 321-445-0781 jer...@jervin.com http://jervin.com/cft ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write custom functions in AEL2 ,
2009/5/11 Watkins, Bradley bradley.watk...@compuware.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Monday, May 11, 2009 3:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to write custom functions in AEL2
,
Hi,
I'm using asterisk 1.6.1 and AEL2.
I'm trying to find the best way to write my own custom functions
?
At the moment, I'm using this pattern (extensions.ael) :
context foo {
123 = {
myfunc(123456);
NoOp(${GOSUB_RETVAL});
};
macro myfunc (arg) {
Return (${arg});
}
1. First, I keep getting warnings like
Warning: file /etc/asterisk/extensions.ael, line 446-446:
application call to Return affects flow of control, and needs to be
re-written using AEL if, while, goto, etc. keywords instead!
and I would like to get rid of them.
Unfortunately, AEL does not support using return with a value at the
moment. There is a patch on Reviewboard that does this, as well as
*simple* direct assignment from an AEL macro return:
http://reviewboard.digium.com/r/114/
Fine : that's exactly what I was looking for !
Unfortunately, at the moment, this feature is still worked on.
http://reviewboard.digium.com/r/114/
2. Secondly, I would like not to use GOSUB_RETVAL and call a
custom function just like I'm calling other functions with statements
like :
123 = {
NoOp(TOLOWER(fOo BaR));
NoOp(myfunc(123456));
};
What would you advise me to do ?
That requires rather a lot more work than the above patch, but if you
use the direct assignment at least you needn't worry about GOSUB_RETVAL.
Regards,
- Brad
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[asterisk-users] Asterisk 1.6.2.0-beta2 Now Available
The Asterisk Development Team is pleased to announce the second beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta2 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release merges in changes to the device state code which caused a performance regression in Asterisk 1.6.1 and 1.6.2. The result of this device state code review is that performance has been positively affected while maintaining the new distributed device state functionality. Additional information about these changes can be found on reviewboard at http://reviewboard.digium.com/r/205/. In addition, this release also resolves several community reported issues. For a full list of changes in this beta, please see the ChangeLog: http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta2/ChangeLog You can get more information about the new features and various changes in Asterisk 1.6.2.0 at: http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta2/CHANGES And if you're upgrading from previous versions of Asterisk see this file: http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta2/UPGRADE.txt Issues discovered in testing of this beta can be reported at http://bugs.digium.com. Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building a System.
On Mon, May 11, 2009 at 02:16:29PM +1000, Klaverstyn, David C wrote: Hi John, I'm not sure if this will help you or not but I created a script that will install Asterisk with all the required components for DAHDI, Faxing, fax to email, LDAPget, CDR, FOP etc. It can even include text to speech applications. I created it because I wanted to install Asterisk multiple times and as quickly as possible. It does the exact same steps as one would do when doing an install manually. I created the bash script aster-install http://www.klaverstyn.com.au/wiki/index.php?title=Aster-install . http://www.klaverstyn.com.au/wiki/index.php?title=Aster-install A scripted installation is indeed doable (I did it myself: http://updates.xorcom.com/astribank/bristuff/1.4) - I had enough iterations to improve it and enough pressure to make it usable. But what happens after you install it? How do you follow up with newer versions? E.g. many people still have an ancient version of mpg123 installed. With many known security holes. This is because at the time it was the black magic working and recommended version for getting music-on-hold to work. mpg123 has moved on. But it is still installed on many systems. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF received twice
out there is a file to change the dtmf duration
where are you? or from where is your cellphone?
from other phones like lkand lines it works well?
David
2009/5/11 Administrator TOOTAI ad...@tootai.net
Hi all,
I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from
my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so
I can use the GW. For this I use:
exten = s,1,NoOp(One of our workers (${CALLERID(number)}) is calling
office) ;callerID is the one of the calling mobile phone
exten = s,n,Background(silence/1)
; Nokia E65 send digits in DTMF mode, no need to take care about input
corrections
;
exten = s,n(enterDigits),Read(myExten,pls-entr-num-uwish2-call,0,,,3)
exten = s,n,GotoIf($[${myExten}=]?enterDigits)
[...]
Problem is that received DTMF digits in ${myExten} are received twice eg
for 1234 ${myExten} has 11223344. I correct the extension by dialplan
but I think it's not really a solution.
In sip.conf, the dtmfmode is set to auto. If I set it to rfc2833, the
same behaviour.
Can somebody confirm this before I open a bug, thanks.
Regards
--
Daniel
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--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
()_()signature to help him gain world domination.
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Re: [asterisk-users] VoIP over satellite internet
Message: 10 Date: Fri, 8 May 2009 20:30:11 -0700 From: Eric Fort eric.f...@gmail.com Subject: [asterisk-users] VoIP over satellite internet To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 2ad2af430905082030w389822aduc877f8b0a1afe...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 Could those on the list who have used or tried to use VoIP over a satellite internet connection comment on how well it works or if it even works at all in a reliable way. What is the effect of latency on the VoIP path and how much is generally tolerable? routing via satellite adds about a quarter second of latency to the path. Is that too much? It is possible-- barely-- but you have to be able to put up with two to six second lags between replies and lots of stepping on each other in conversations. The feasibility will also depend on the traffic shaping/filtering of the provider and whether they black hole VoIP ports/packets. There will be a lot of delay and echo which can be compounded by an imbalance in upstream and downstream bandwidth. If you're using dialup for upstream the bandwidth _will_ be an issue. If you're looking for point-to-point communication a client-to-client push-to-talk solution like Speak Freely [1] might be a better choice. You may also want to consider-- if you're trying to use Asterisk-- a narrowband codec such as Speex. [2] I used Speak Freely over 28.8 dialup links to have conversations between Florida and Ontario almost fifteen years ago. It's more like a two-way radio than a telephone but it works very well and is win/lin cross-platform. [1] http://speak-freely.sourceforge.net/ [2] http://speex.org/ Thanks, Josh Fuller josh.ful...@telus.com The views expressed in this e-mail are mine alone and do not necessarily reflect the views of my employer. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write custom functions in AEL2 ,
On Mon, May 11, 2009 at 1:30 AM, Olivier oza-4...@myamail.com wrote:
Hi,
I'm using asterisk 1.6.1 and AEL2.
I'm trying to find the best way to write my own custom functions ?
At the moment, I'm using this pattern (extensions.ael) :
context foo {
123 = {
myfunc(123456);
NoOp(${GOSUB_RETVAL});
};
macro myfunc (arg) {
Return (${arg});
}
1. First, I keep getting warnings like
Warning: file /etc/asterisk/extensions.ael, line 446-446: application call
to Return affects flow of control, and needs to be re-written using AEL if,
while, goto, etc. keywords instead!
and I would like to get rid of them.
This is easily done. Return() is calling the Return application; 'return',
however, is the keyword the AEL uses. Note the lack of a capital R at the
beginning of
the word return. AEL is case sensitive and Return is not equal to
return.
Also note that, as a previous reply mentions, that return takes no args,
that there is a patch available to upgrade to do that.
You don't need the patch to do what the patch does, tho. But, not having
refreshed my memory on the particulars, I will say
no more!
2. Secondly, I would like not to use GOSUB_RETVAL and call a custom
function just like I'm calling other functions with statements like :
123 = {
NoOp(TOLOWER(fOo BaR));
NoOp(myfunc(123456));
};
Again, check your version of Asterisk against whether AEL uses Gosub() to
implement macros.
The AEL2 wiki page on voip-info.org (
http://voip-info.org/wiki/view/Asterisk+AEL2 ) can also
be quite helpful at times!
What would you advise me to do ?
Regards
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--
Steve Murphy
ParseTree Corp
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Re: [asterisk-users] VoIP over satellite internet
snip ...routing via satellite adds about a quarter second of latency to the path. Is that too much? /snip Eric, I believe that you are mistaken. Routing via satellite adds about a quarter second of latency PER TRIP from earth to orbit. This is simply due to the distance a satellite is from the ground and the speed of light (interference not withstanding). Traceroutes and pings to satellite providers can be misleading because they cache some content on the birds in order to decrease latency. As I recall they even intercept some pings to accomplish the same. A *real* round trip for a VOIP call and/or non-interfered TCP connection would look like this: 1. Your device up to the bird (~250ms) 2. The bird back to the ground (~250ms) 3. The ground station out to the internet (~Nms) 4. The internet back to the ground station (~Nms) 5. The ground station back to the bird (~250ms) 6. The bird back to your device (~250ms) As you can see, even the one way udp stream will take approximately 500ms beyond any latency introduced by things such as your wireless network and the internet. VOIP over satellite, as Josh indicated, will be painful. You'll be talking all over one another due to the delay assuming that the stream can even be sustained with that much latency. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over satellite internet
David Gibbons wrote: snip ...routing via satellite adds about a quarter second of latency to the path. Is that too much? /snip Eric, I believe that you are mistaken. Routing via satellite adds about a quarter second of latency PER TRIP from earth to orbit. This is simply due to the distance a satellite is from the ground and the speed of light (interference not withstanding). Traceroutes and pings to satellite providers can be misleading because they cache some content on the birds in order to decrease latency. As I recall they even intercept some pings to accomplish the same. A *real* round trip for a VOIP call and/or non-interfered TCP connection would look like this: 1. Your device up to the bird (~250ms) 2. The bird back to the ground (~250ms) 3. The ground station out to the internet (~Nms) 4. The internet back to the ground station (~Nms) 5. The ground station back to the bird (~250ms) 6. The bird back to your device (~250ms) As you can see, even the one way udp stream will take approximately 500ms beyond any latency introduced by things such as your wireless network and the internet. VOIP over satellite, as Josh indicated, will be painful. You'll be talking all over one another due to the delay assuming that the stream can even be sustained with that much latency. -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Of course, that's assuming your satellite is in geosynchronous orbit. If its in LEO, then its much better. Singer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over satellite internet
On Fri, May 08, 2009 at 11:56:42PM -0500, Frank Bulk wrote: If people don't mind taking turns talking, it will work. It's just going to be like talking on a CB. Reminds me of talking to my grandparents in the Europe as a child in the early 80's. Just recall that VoIP can generally live with high latency (in the worst case the parties take turns). OTOH, jitter (the variance of the latency) can be a real problem. If packets are guaranteed to be delieved with a delay of 1000ms +- 2ms, you'd probably get a decent quality (assuming you got rid of echo). If packets get delivered at 500ms +-250ms, the endpoints will have a much harder time producing a good call from that. Dropped packets are also a major pain, naturally. IIRC ilbc is relatively resiliant to dropped packets, as in it each packet's decoding is independent of other packets (that might get dropped). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk w/ Nokia e Series Handsets
Anyone using Nokia E Series handsets with Asterisk? I'm trying to deploy some e71's and am having an issue. I can get a single handset working, but when I try to create a SIP profile on the second phone, it won't allow me to save the profile, saying that devices in the same realm must have identical username and password. Anyone have a workaround for this to add a second Nokia phone under the Asterisk realm with a different userid and PW? Thanks Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com mailto:br...@voipsupply.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers mailto:bsay...@voipsupply.com , CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over satellite internet
snipOf course, that's assuming your satellite is in geosynchronous orbit. If It's in LEO.../snip Singer, You are of course correct, low earth orbit will have lower latency. I was assuming that this user would be using a stationary link on the ground, not a portable sat phone or an aimable dish to make these calls. That may be an incorrect assumption. Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk w/ Nokia e Series Handsets
Cory Andrews a écrit : Anyone using Nokia E Series handsets with Asterisk? I'm trying to deploy some e71's and am having an issue. I can get a single handset working, but when I try to create a SIP profile on the second phone, it won't allow me to save the profile, saying that devices in the same realm must have identical username and password. Anyone have a workaround for this to add a second Nokia phone under the Asterisk realm with a different userid and PW? Few of our customers are running severals E70/E65/E66 connected to the same Asterisk, no problem. If you have some troubles on creating a setup with an E serie, switching the phone off/on is generally helpful. Also, be sure you have the latest firmware for your phone. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF received twice
David fire a écrit :
out there is a file to change the dtmf duration
where are you?
France
[...]
from other phones like lkand lines it works well?
No, the same. The called number is a number received by a trunk SIP, the
GW is also setted as dtmfmode=auto. Calling from mobile phone or
landline to other services using DTMF -like banks- is OK.
I make further tests and so that setting dtmfmode=info for this GW make
DTMF working correctly! Is this the normal behaviour?
Our dialplan works great for others GW's, if this is normal we have to
adapt it in case of dtmfmode=info. From where can we get the dtmf type?
For me it looks like a bug.
Thanks for your help.
2009/5/11 Administrator TOOTAI ad...@tootai.net
Hi all,
I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from
my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so
I can use the GW. For this I use:
exten = s,1,NoOp(One of our workers (${CALLERID(number)}) is calling
office) ;callerID is the one of the calling mobile phone
exten = s,n,Background(silence/1)
; Nokia E65 send digits in DTMF mode, no need to take care about input
corrections
;
exten = s,n(enterDigits),Read(myExten,pls-entr-num-uwish2-call,0,,,3)
exten = s,n,GotoIf($[${myExten}=]?enterDigits)
[...]
Problem is that received DTMF digits in ${myExten} are received twice eg
for 1234 ${myExten} has 11223344. I correct the extension by dialplan
but I think it's not really a solution.
In sip.conf, the dtmfmode is set to auto. If I set it to rfc2833, the
same behaviour.
Can somebody confirm this before I open a bug, thanks.
Regards
--
Daniel
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[asterisk-users] Problems with res_odbc
Good morning,
I'm having suddenly cut-offs and I don`t know why. It's been hapenning since
I enabled cdr_odbc/func_odbc in my system.
We use func_odbc to register some queue member's events (login, pause, etc.)
at an external DB ('remoto' connector) and to uptade local tables at a local
DB ('local' connector).
Currently we are usind cdr_odbc to Postgresql and cdr_addon to MySQL (wich
we are planing to eliminate soon)
Mayor issue is that SIP peers dissapear for 1 minute until modules get
reloaded.
Asterisk Version: 1.4.21.1
Local BD: Postgresql 7.4.19
External BD: Postgresql 7.4.19
This is what I see at logs:
*Problem is begginig:*
[May 11 09:02:57] NOTICE[17727] *loader.c: 2 modules will be loaded*.
[May 11 09:02:57] NOTICE[17727] res_odbc.c: Adding ENV var:
INFORMIXSERVER=my_special_database
[May 11 09:02:57] NOTICE[17727] res_odbc.c: Adding ENV var:
INFORMIXDIR=/opt/informix
[May 11 09:02:57] NOTICE[17727] res_odbc.c: Connecting local
[May 11 09:02:57] NOTICE[17727] res_odbc.c: res_odbc: Connected to local [*
conector-loca*l]
[May 11 09:02:57] NOTICE[17727] res_odbc.c: Registered ODBC class 'local'
dsn-[conector-local]
[May 11 09:02:57] NOTICE[17727] res_odbc.c: Connecting remoto
[May 11 09:02:57] NOTICE[17727] res_odbc.c: res_odbc: Connected to remoto [*
connector-remoto*]
[May 11 09:02:57] NOTICE[17727] res_odbc.c: Registered ODBC class 'remoto'
dsn-[connector-remoto]
[May 11 09:02:57] NOTICE[17727] res_odbc.c: res_odbc loaded.
[May 11 09:02:57] NOTICE[17727] config.c: Registered Config Engine odbc
[May 11 09:02:57] NOTICE[17727] cdr.c: CDR simple logging enabled.
*Loader.c again!*May 11 09:02:57] NOTICE[17727*] loader.c: 154 modules will
be loaded.*
*I have my realtime peers with MySQL not Postgresql, but...*
May 11 09:02:57] ERROR[17727] res_config_pgsql.c: Postgresql RealTime:
Failed to connect database server asterisk on . Check debug for more info.
[May 11 09:02:57] NOTICE[17727] config.c: Registered Config Engine pgsql
[May 11 09:02:58] NOTICE[17727] config.c: Registered Config Engine mysql
[May 11 09:02:58] NOTICE[17727] app_queue.c: Queue members successfully
reloaded from database.
[May 11 09:02:58] NOTICE[17727] chan_ooh323.c:
-
--- *** IMPORTANT NOTE ***
---
--- This module is currently unsupported. Use it at your own risk.
---
-
[May 11 09:02:58] NOTICE[17727] chan_ooh323.c: Unable to load config
ooh323.conf, OOH323 disabled
[May 11 09:02:58] NOTICE[17727] pbx_ael.c: Starting AEL load process.
[May 11 09:02:58] NOTICE[17727] pbx_ael.c: AEL load process: calculated
config file name '/etc/asterisk/extensions.ael'.
[May 11 09:02:58] NOTICE[17727] pbx_ael.c: File /etc/asterisk/extensions.ael
not found; AEL declining load
[May 11 09:02:58] ERROR[17813] chan_zap.c: !! Got S-frame while link down
*Realtime peers are reacheble again, why they got unreachable?*
[May 11 09:02:59] NOTICE[17835] chan_sip.c: Peer '870' is now Reachable.
(27ms / 2000ms)
[May 11 09:03:00] NOTICE[17835] chan_sip.c: Peer '901' is now Reachable.
(26ms / 2000ms)
[May 11 09:03:00] NOTICE[17835] chan_sip.c: Peer '982' is now Reachable.
(62ms / 2000ms)
[May 11 09:03:07] NOTICE[17835] chan_sip.c: Peer '937' is now Reachable.
(62ms / 2000ms)
[May 11 09:03:11] NOTICE[17835] chan_sip.c: Peer '855' is now Reachable.
(62ms / 2000ms)
[May 11 09:03:13] NOTICE[17835] chan_sip.c: Peer '914' is now Reachable.
(61ms / 2000ms)
[May 11 09:03:13] NOTICE[17835] chan_sip.c: Peer '804' is now Reachable.
(62ms / 2000ms)
I'll be very gratefull by your suggestions.
Elder Arohuanca Lagos
Phone: +51 1 994149553
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Re: [asterisk-users] Asterisk w/ Nokia e Series Handsets
Hi Cory, I believe you meant that you can't add a second SIP profile on the same phone, right? There seems to be a bug with the latest Nokia E-series that has this problem. It complains even if the userid and password are identical. I worked around this by just changing the realm name. Nokia and Asterisk doesn't seem to mind this. As long as your Public User Name uses the correct domain/realm name, it seems to work. In my case, I was trying to add more than one SIP profiles for the same user account, but with different access point. In your case, I think you are trying to add another SIP profile with a different user account to the same phone? If that is the case, I haven't tried that before. Regards, Steve Cory Andrews wrote: Anyone using Nokia “E” Series handsets with Asterisk? I’m trying to deploy some e71’s and am having an issue. I can get a single handset working, but when I try to create a SIP profile on the second phone, it won’t allow me to save the profile, saying that devices in the same “realm” must have identical username and password. Anyone have a workaround for this to add a second Nokia phone under the Asterisk “realm” with a different userid and PW? Thanks *Cory J. Andrews* Director New Market Initiatives *Sayers Media Group* *VoIP Supply, LLC* 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE _candr...@sayersmedia.com mailto:br...@voipsupply.com_ _ _ _ _ Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers mailto:bsay...@voipsupply.com, CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PauseMonitor() Hanging Up Call
Hi All,
I'm at the end of my tether here and would really appreciate some help.
I'm trying to implement DTMF based pause/resume of call recording. I'm
using Asterisk 1.4.22.1.
Here's the scenario:
The caller (SIP or ISDN, doesn't matter) dials into the asterisk which
executes the following code:
exten = _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb)
exten = _X.,n,Set(__DYNAMIC_FEATURES=in-pauseMonitor#in-resumeMonitor)
exten = _X.,n,Dial(SIP/myphone,300,tTo)
My [applicationmap] in features.conf is setup as follows:
in-pauseMonitor = *7,self/callee,Macro,pause-record
in-resumeMonitor = *9,self/callee,Macro,resume-record
I also have the following contexts setup in extensions.conf:
[macro-pause-record]
exten = s,1,Playback(sounds/recPaused)
exten = s,n,PauseMonitor()
exten = s,n,MacroExit
[macro-resume-record]
exten = s,1,Playback(sounds/recResumed)
exten = s,n,UnPauseMonitor()
exten = s,n,MacroExit
Now, if I setup the call and hit *7 on the callee phone, the call is hungup
every time! No error message, just simply hangs up, as follows:
Executing [...@macro-pause-record:2]
PauseMonitor(SIP/myphone-09d26e60, ) in new stack
== Spawn extension (macro-pause-record, s, 2) exited non-zero on
'SIP/myphone-09d26e60' in macro 'pause-record'
== Auto fallthrough, channel 'SIP/xlite-09d18fc0' status is 'ANSWER'
If I change the [applicationmap] entries in features.conf to allow
pause/resume from the caller phone, e.g.:
in-pauseMonitor = *7,self/caller,Macro,pause-record
in-resumeMonitor = *9,self/caller,Macro,resume-record
Then it works like a charm!
Seems there's an issue with pause/resume from callee side.
Can anyone shed any light on what I'm doing wrong here please?
Regards,
Jon Morgan.
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Re: [asterisk-users] PauseMonitor() Hanging Up Call
Jon Morgan wrote:
Hi All,
I’m at the end of my tether here and would really appreciate some help.
I’m trying to implement DTMF based pause/resume of call recording. I’m
using Asterisk 1.4.22.1.
Here’s the scenario:
The caller (SIP or ISDN, doesn’t matter) dials into the asterisk which
executes the following code:
exten = _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb)
exten = _X.,n,Set(__DYNAMIC_FEATURES=in-pauseMonitor#in-resumeMonitor)
exten = _X.,n,Dial(SIP/myphone,300,tTo)
My [applicationmap] in features.conf is setup as follows:
in-pauseMonitor = *7,self/callee,Macro,pause-record
in-resumeMonitor = *9,self/callee,Macro,resume-record
I also have the following contexts setup in extensions.conf:
[macro-pause-record]
exten = s,1,Playback(sounds/recPaused)
exten = s,n,PauseMonitor()
exten = s,n,MacroExit
[macro-resume-record]
exten = s,1,Playback(sounds/recResumed)
exten = s,n,UnPauseMonitor()
exten = s,n,MacroExit
Now, if I setup the call and hit *7 on the callee phone, the call is
hungup every time! No error message, just simply hangs up, as follows:
Executing [...@macro-pause-record:2]
PauseMonitor(SIP/myphone-09d26e60, ) in new stack
== Spawn extension (macro-pause-record, s, 2) exited non-zero on
'SIP/myphone-09d26e60' in macro 'pause-record'
== Auto fallthrough, channel 'SIP/xlite-09d18fc0' status is 'ANSWER'
If I change the [applicationmap] entries in features.conf to allow
pause/resume from the caller phone, e.g.:
in-pauseMonitor = *7,self/*caller*,Macro,pause-record
in-resumeMonitor = *9,self/*caller*,Macro,resume-record
Then it works like a charm!
Seems there’s an issue with pause/resume from callee side.
Can anyone shed any light on what I’m doing wrong here please?
Regards,
Jon Morgan.
The problem is that the callee's channel does not have a monitor on it, just
the
caller's channel. The PauseMonitor application has the unfortunate effect that
if the channel on which it is called has no monitor attached, then the
application returns as if an error occurred and the dialplan stops. I
unfortunately do not see a direct way to tell from the dialplan if a channel
has
a monitor attached (there is a MONITORED channel variable, but it will be true
for both channels of a monitored call).
I can think of ways to work around the problem of the call being hung up, but
the problem is that even with the workarounds in place, calling PauseMonitor on
the callee's channel will not result in the monitor actually becoming paused
since, once again, there is no monitor attached to the callee's channel.
So for the purposes of your setup, the only way you're going to be able to get
what you want working, short of actually changing the source code, is to only
allow the caller to be able to pause the monitor.
Mark Michelson
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[asterisk-users] Ready to put the box on the net
I loaded PBX in a flash and I have a simple dialplan setup. I'm guessing this needs to go on the DMZ of my router for anyone to get to it correct? Is there any way to keep it behind the router and map to it or is that more trouble than it is worth?Thanks!Ronny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ready to put the box on the net
On Monday 11 May 2009 19.54.47 k4...@bellsouth.net wrote: I loaded PBX in a flash and I have a simple dialplan setup. I'm guessing this needs to go on the DMZ of my router for anyone to get to it correct? Is there any way to keep it behind the router and map to it or is that more trouble than it is worth? Thanks! Ronny You may experience problems if you enable DMZ to the asterisk server with another computers on your network! I recommend only forwarding udp port 5060 for sip and port 4569 for iax2 and udp port from 11000-12000 for data. Remember you have to set this range (11000-12000) in rtp.conf. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ready to put the box on the net
For my information (and anyone else interested), how much of the information at this link - http://www.voip-info.org/wiki/view/Asterisk+config+rtp.conf is still valid? According to that information, the setup you describe would basically allow for 250 or so concurrent calls. Also, I expect that even though that doc states that the 12000 should actually be 11999, it would go by with a warning if that. So in a really tight environment, you could set this up for as small of a range as 11000-11003? Thanks in Advance. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Puskás Zsolt Sent: Monday, May 11, 2009 1:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ready to put the box on the net On Monday 11 May 2009 19.54.47 k4...@bellsouth.net wrote: I loaded PBX in a flash and I have a simple dialplan setup. I'm guessing this needs to go on the DMZ of my router for anyone to get to it correct? Is there any way to keep it behind the router and map to it or is that more trouble than it is worth? Thanks! Ronny You may experience problems if you enable DMZ to the asterisk server with another computers on your network! I recommend only forwarding udp port 5060 for sip and port 4569 for iax2 and udp port from 11000-12000 for data. Remember you have to set this range (11000-12000) in rtp.conf. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone with a working pfSense firewall configuration?
Other SIP clients behind the firewall (not using STUN, work). We have a SIP client using STUN and ICE behind a pfSense firewall. The firewall is behaving oddly. REGISTER packets work fine. But when the client tries to make a call, the first INVITE packet from the client pass through the firewall and makes it to the Asterisk server. The Asterisk server sends back a 401 client sends ACK, traffic passes fine. When the client then sends the INVITE with the authentication information, the INVITE packet never makes it to the Asterisk server. A packet trace on the WAN interface of the firewall shows the INVITE going out, but the packets never make it to the Asterisk server. Any ideas on how to configure pfSense to work with a SIP client using STUN and ICE, without having to install siproxyd? -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP over satellite internet
On Fri, 2009-05-08 at 20:30 -0700, Eric Fort wrote: Could those on the list who have used or tried to use VoIP over a satellite internet connection comment on how well it works or if it even works at all in a reliable way. What is the effect of latency on the VoIP path and how much is generally tolerable? routing via satellite adds about a quarter second of latency to the path. Is that too much? Yes, done it, several times. Although that was done over a leased satelite link. Any other ip-traffic had a lower priority... Had a number of hardphones (snom's utstar's) at one end, and asterisk via ipsec-tunnels at the other end. In between a C-band satelite and a small tracking antenna on moving vehicles (cars and ships) Latency is noticable, but acceptable. Choise of codecs determine the number of available channel, given an amount of bandwith, obviously ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone with a working pfSense firewall configuration?
- Eric Chamberlain e...@rf.com wrote: Other SIP clients behind the firewall (not using STUN, work). We have a SIP client using STUN and ICE behind a pfSense firewall. The firewall is behaving oddly. REGISTER packets work fine. But when the client tries to make a call, the first INVITE packet from the client pass through the firewall and makes it to the Asterisk server. The Asterisk server sends back a 401 client sends ACK, traffic passes fine. When the client then sends the INVITE with the authentication information, the INVITE packet never makes it to the Asterisk server. A packet trace on the WAN interface of the firewall shows the INVITE going out, but the packets never make it to the Asterisk server. Any ideas on how to configure pfSense to work with a SIP client using STUN and ICE, without having to install siproxyd? -- Eric Chamberlain, Founder RF.com - http://RF.com/ pfSense employs source-port randomization by default. You may want to enable advanced outbound NAT which turns this behavior off. While I'm not sure this is the source of your problems, I've seen it ruin otherwise acceptable SIP situations. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with res_odbc
On Monday 11 May 2009 10:54:48 am Daniel - Asterisk wrote: *Realtime peers are reacheble again, why they got unreachable?* [May 11 09:02:59] NOTICE[17835] chan_sip.c: Peer '870' is now Reachable. (27ms / 2000ms) Unless you saw a message that said the peers became unreachable, then they were never that way. The delay you're seeing is the time between when the reload occurred (which purges SIP peers from memory) and when the SIP clients next contacted Asterisk (which is when Asterisk again loads the hosts into memory). There is no problem that I can see here, at all, unless you have other symptoms that you have not specified. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF received twice
Administrator TOOTAI wrote:
David fire a écrit :
out there is a file to change the dtmf duration
where are you?
France
[...]
from other phones like lkand lines it works well?
No, the same. The called number is a number received by a trunk SIP, the
GW is also setted as dtmfmode=auto. Calling from mobile phone or
landline to other services using DTMF -like banks- is OK.
I make further tests and so that setting dtmfmode=info for this GW make
DTMF working correctly! Is this the normal behaviour?
Our dialplan works great for others GW's, if this is normal we have to
adapt it in case of dtmfmode=info. From where can we get the dtmf type?
For me it looks like a bug.
Thanks for your help.
2009/5/11 Administrator TOOTAI ad...@tootai.net
Hi all,
I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from
my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so
I can use the GW. For this I use:
exten = s,1,NoOp(One of our workers (${CALLERID(number)}) is calling
office) ;callerID is the one of the calling mobile phone
exten = s,n,Background(silence/1)
; Nokia E65 send digits in DTMF mode, no need to take care about input
corrections
;
exten = s,n(enterDigits),Read(myExten,pls-entr-num-uwish2-call,0,,,3)
exten = s,n,GotoIf($[${myExten}=]?enterDigits)
[...]
Problem is that received DTMF digits in ${myExten} are received twice eg
for 1234 ${myExten} has 11223344. I correct the extension by dialplan
but I think it's not really a solution.
In sip.conf, the dtmfmode is set to auto. If I set it to rfc2833, the
same behaviour.
Can somebody confirm this before I open a bug, thanks.
Regards
--
Daniel
I've seen a couple of examples of this on the list where a provider
sends DTMF in multiple formats and Asterisk with dtmfmode=auto picks up
all the digits sent in all formats. Maybe there should be a code change
so that dtmfmode=auto makes asterisk lock on to the mode of the first
digit received for a session and ignores all other formats for that
particular session? Does that make sense to anybody?
___
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Re: [asterisk-users] Anyone with a working pfSense firewall configuration?
On May 11, 2009, at 2:30 PM, Tim Nelson wrote: pfSense employs source-port randomization by default. You may want to enable advanced outbound NAT which turns this behavior off. While I'm not sure this is the source of your problems, I've seen it ruin otherwise acceptable SIP situations. Thanks, I just tried using the static-port option. The source ports aren't randomized any more, but the INVITEs still disappear after the initial 401-INVITE response. -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-user] Which policy for ISDN BRI support in NT/PtMP ?
For those also need NT over PtMP, i started a initial patch for it. Very limited at the moment, only one incoming call to chan_dahdi from one device is possible. But i was pleasantly surprised that NT-ptmp is working anyway Get the patch here: http://bugs.digium.com/view.php?id=15048 Kristijan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Codecs
I got very excited when I read the title of this email - I was hoping someone had learnt to speak g729. Ah well. PaulH Adrian Marsh wrote: Hi, I’m having problems with an asterisk server that’s not offering Codecs for ulaw and alaw as it should. I’ve three servers in total: a1, a2 and “b” A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see A1 is offering codecs: [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 14958 Adding codec 0x2000 (amr) to SDP *Adding codec 0x4 (ulaw) to SDP* *Adding codec 0x8 (alaw) to SDP* Adding non-codec 0x1 (telephone-event) to SDP But when A2 makes the same call to B, it only offers amr: [May 6 16:38:44] WARNING[20408]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 15554 Adding codec 0x2000 (amr) to SDP Adding non-codec 0x1 (telephone-event) to SDP Its not building ulaw or alaw into its list. Server B doesn’t support AMR, so rejects the call. (I’ve no idea about the 0x4000 error – but I see it on both the good and bad servers, so I don’t think its related). The odd thing is that the sip.conf files for A1 and A2 are exactly the same (save IP info). The build of the Asterisk server is from a 1.4.15 private build to add AMR, but, it’s the same source built on both A1 and A2. I’m trying to figure out why A2 isnt offering ulaw and alaw. The codec seems ok, and is listed in the show codecs: 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 8192 (1 13) (0x2000) audio amr (AMR) But I cant see why its not transcoding across to ulaw/alaw. Thanks, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding Codecs
All, I think we've found what was blocking us. It seems that SElinux, for some unknown reason, didn't like the AMR codec, and did something to block it. Set that to passive, and the problem goes away... Would still like to learn more about asterisk codec translation though, if anyone has any pointers. Adrian From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 07 May 2009 09:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Understanding Codecs Hi All, My theory on the codec translation deepens: Doing a core show translation on the A1 server (working) I get: g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 amr g723- ---- -- -- -- -- - gsm- -222 21 3- - 11 2- 45 ulaw- 2-12 21 3- - 11 2- 45 alaw- 21-2 21 3- - 11 2- 45 g726aal2- 222- 21 3- - 11 1- 45 adpcm- 2222 -1 3- - 11 2- 45 slin- 1111 1- 2- - 10 1- 44 lpc10- 2222 21 -- - 11 2- 45 g729- ---- -- -- -- -- - speex- ---- -- -- -- -- - ilbc- 2222 21 3- -- 2- 45 g726- 2221 21 3- - 11 -- 45 g722- ---- -- -- -- -- - amr- 13 13 13 1313 1214- - 22 13- - But on the new server it gives: g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 amr g723- ---- -- -- -- -- - gsm- -222 21 2- - 11 2- - ulaw- 2-12 21 2- - 11 2- - alaw- 21-2 21 2- - 11 2- - g726aal2- 222- 21 2- - 11 1- - adpcm- 2222 -1 2- - 11 2- - slin- 1111 1- 1- - 10 1- - lpc10- 2222 21 -- - 11 2- - g729- ---- -- -- -- -- - speex- ---- -- -- -- -- - ilbc- 2222 21 2- -- 2- - g726- 2221 21 2- - 11 -- - g722- ---- -- -- -- -- - amr- ---- -- -- -- -- - So where are the codec translations set? Thanks Adrian From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 06 May 2009 18:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Understanding Codecs Forgot to add: sip.conf for both A1 and A2 has the following global codec definitions: disallow=all allow=clear allow=amr allow=ulaw allow=alaw The Asterisk build is a private build that adds the clear and AMR codec setups. The two servers are running Fedora, though A1s on 6 and A2s on 10. I cant see why that would make a difference though. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 06 May 2009 17:53 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Understanding Codecs Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and b A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see A1 is offering codecs: [May 6 16:43:19] WARNING[25404]: channel.c:720 ast_best_codec: Don't know any of 0x4000 formats Audio is at IP HIDDEN port 14958 Adding codec 0x2000 (amr) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP But when A2 makes the same call to
[asterisk-users] Asterisk Manager API Action Originate
Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Now in Asterisk 1.4.23 I get some error code 5's but since they're so few I tend not to worry. But what is concerning is the number of Error 0's I get. Error 0 is No Such Extension (ie, Failure I assume) but my Provider's CDR log shows No Answer. (I would show you my CDR but it seems Originate doesn't log in the CDR like every other call for some reason). Any ideas to correct this issue? Or is there a better updated version of that list that would fix my understanding of what the error codes were? Nicholas Blasgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
