Re: [asterisk-users] Asterisk not transcoding between installed codecs

[Jaswinder Singh]

iax.conf doesn't take canreinvite=no . It's only for sip . For iax2 its
transfer=yes/no (1.4 ) or notransfer=yes (1.2) , there's one more parameter
in 1.4 with which u can transfer only audio stream . Check voip-info wiki
for all options .

On Thu, Mar 13, 2008 at 7:48 AM, Gonzalo Servat <[EMAIL PROTECTED]> wrote:

> On Wed, Mar 12, 2008 at 12:58 PM, Brent Davidson <
> [EMAIL PROTECTED]> wrote:
>
> >  Do you have canreinvite=no in the sip client configuration?  If not
> > then the two sip phones are probably issuing a reinvite command and taking
> > asterisk out of the call path.  If that happens and the phones can't reach
> > consensus on a codec then you run into audio problems.  If you're not a
> > provider and just using asterisk as a PBX then it's probably better to set
> > the phones up with a matching codec set and allow them to establish a direct
> > connection between each other to keep load off the Asterisk server.
> > Otherwise set canreinvite=no and Asterisk should transcode correctly.
> >
>
> Brent,
>
> Thank you vry much for replying. I thought the message went unseen but
> found your reply when I went to look at the thread :)
>
> You're absolutely right. Looks like the SIP client was messing up (or
> something) when different codecs were used. I tried canreinvite=no and it
> worked perfectly, but as you said, it's best to bypass Asterisk when talking
> between clients on the same network. I tried a different IAX client and it
> had no problems using different codecs (with canreinvite=yes) so all is
> good.
>
> Thanks again!
> Gonzalo
>
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[asterisk-users] Asterisk re-invites and billing

[Jaswinder Singh]

I am using asterisk 1.4.18 (server A ) and have it store records in
mysql database . One of my client uses predictive dialer ( asterisk
1.2.26 based and server B ) which makes many calls  . B registers with
A over sip and there is no nat involved  If i re-invite rtp from
server B  to my carrier ( server A in between )  I saw many calls
having duration of 0,1 or 2 seconds on server A's cdr but surprisingly
all these calls were marked at 15 minutes usage on my provider's
records . My sip route provider himself is re-inviting traffic ahead
to their media gateways . I have gone through asterisk sip.conf and i
don't see any setting limiting anything to around 15 minutes , default
rtp timeout settings are around 60 seconds in asterisk . My provider
says that they don't have any 15 minute limit on their end . The
records on server B also suggests that calls are indeed very small  1
to- 3 seconds . Server A and B both have static ip's and there is no
bandwith problem on server A . If i disable re invites on server A
then this problem isn't present . Did anybody else have this kind of
problem ? Any suggestions

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Re: [asterisk-users] G723 on asterisk 1.4.1

[Jaswinder Singh]

That's strange , i am able to see the *url*  in Martin's reply .

On Sun, Mar 23, 2008 at 6:14 PM, Eric Wieling <[EMAIL PROTECTED]> wrote:
> The only messages I have EVER seen Digium remove from the mailing list
>  archives are discussions about this unlicensed codec.
>
>
>  Martin wrote:
>  > Download an appropriate binary from
>  > [url removed]
>
>
> > and just drop into /usr/lib/asterisk/modules/
>  > add allow=g723 to your sip.conf as necessary and restart asterisk...
>  > Im only not sure how legal is this, you will probably need to obtain
>  > licenses for all concurent channels...
>  > Martin
>  >
>  > - Original Message -
>  > From: "wassim darwish" <[EMAIL PROTECTED]>
>  > To: 
>  > Sent: 22. brezna 2008 15:21
>  > Subject: [asterisk-users] G723 on asterisk 1.4.1
>  >
>  >
>  >> Hi:
>  >> How to install and set up my asterisk server with G723 codec to send and
>  >> receive calls using it.
>  >>
>  >> Thanks in advance;
>  >> Wassim
>  >
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>  >
>  >
>
>  --
>  Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
>  T-1, PRI, Frame Relay, Linux, and network design.  Based near
>  Birmingham, AL.  Now accepting clients worldwide.
>
>
>
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Re: [asterisk-users] g729 encoder/decoder

[Jaswinder Singh]

When g729 phone calls another g729 phone and you are not recording
calls or doing meetme with them  then license is not required ... g729
phone calling g711 will require a license to transcode the g729 side (
no license for g711 side of call ) . In short anytime u need to
convert g729 into some other codec ( transcoding ) you need 1 license
.

On Wed, Apr 2, 2008 at 1:59 AM, Peder @ NetworkOblivion
<[EMAIL PROTECTED]> wrote:
> How does the g729 encoder/decoder count in regards to the total number
>  of licenses and how does it count an encoder/decoder?  I looked on the
>  wiki and don't really see anything that explains it.  In other words,
>  how do the calls below count (assume no reinvite)?
>
>  g729 phone calls into voicemail
>
>  g729 phone calls g711 phone
>
>  g729 phone calls other g729 phone
>
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Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4

[Jaswinder Singh]

asterisk -rx "module load codec_g729.so" or "module load codec_g729" and
shhow translation recalc

On Nov 27, 2007 2:36 AM, Fernando Berretta <[EMAIL PROTECTED]>
wrote:

>  Dear Mindaugas,
>
> Thanks for your promt response
>
> I've already tried this but.. it's not working,, what file do you think I
> should use ? any other idea ?
>
> Best Regards,
> Fernando
>
> Mindaugas Kezys wrote:
>
>  Rename to 
> codec_g729.so
>
> Copy to /usr/lib/asterisk/modules
>
> chmod 777 codec_g729.so
>
>
>
> restart Asterisk
>
> show translations
>
>
>
> Mindaugas Kezys
>
> http://www.kolmisoft.com
>
> Advanced Billing for Asterisk PBX
>
>
>
> *From:* [EMAIL PROTECTED] [
> mailto:[EMAIL PROTECTED]<[EMAIL PROTECTED]>]
> *On Behalf Of *Fernando Berretta
> *Sent:* Monday, November 26, 2007 6:01 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core
> processor 4000 + CENTOS 5 + Asterisk 1.4
>
>
>
> Dear Mindaugas,
>
> I've already download the folowing files for testing
>
> codec_g729-ast14-gcc4-glibc-athlon-sse.so
> codec_g729-ast14-gcc4-glibc-core2.so
> codec_g729-ast14-icc-glibc-x86_64-core2.so
>
> But... no one of them seems to be working
>
>
>
>
>
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Re: [asterisk-users] g729 codec in Athlon 64 x2 Dual core processor 4000 + CENTOS 5 + Asterisk 1.4

[Jaswinder Singh]

Well one of them should work fine ;) . I was not sure if it required .so
extension ( i guess it doesnt ) anyway hitting tab can autocomplete or
atleast give hints . Looks like he is loading wrong module bcoz asterisk
autoloads this on restart  if placed in proper directory with proper
permissions :) .

On Nov 27, 2007 3:33 AM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

> On Tue, Nov 27, 2007 at 03:00:19AM +0530, Jaswinder Singh wrote:
> > asterisk -rx "module load codec_g729.so" or "module load codec_g729" and
> > shhow translation recalc
>
> You ca't really fix a typo without intrducing a new one, eh?
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] SIP port 5060 closed - how do I open it?

[Jaswinder Singh]

Its pretty clear from netstat that asterisk is listening on udp 5060 . It
might be firewall configuration in server thats blocking it . Also you might
have scanned for TCP port 5060 from outside and hence u find it closed ?

On Nov 28, 2007 5:57 AM, Nick Brown <[EMAIL PROTECTED]> wrote:

> Zaheer,
>
> On 28/11/07 9:28 AM, "Zaheer K. Master" <[EMAIL PROTECTED]> wrote:
>
> > Yes I have a sip.conf, contents as follows:
>
> From the CLI can you confirm SIP is running by pasting the results of
> 'module show like sip'
>
> Cheers
> Nick.
>
>
>
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Re: [asterisk-users] SIP call interrupted after 64 seconds

[Jaswinder Singh]

Can you post the part of your dialplan which causes this behaviour ?

On Dec 17, 2007 11:19 PM, Roger Schreiter <[EMAIL PROTECTED]> wrote:
> Hi,
>
> some months ago, I had the problem with an asterisk-1.4.x-
> Version, that some calls (but not all) were interrupted
> 64 seconds after connect (a call limit of 86400 seconds
> was installed using the S()-parameter).
>
> It was just a test machine, and later, I switched to callweaver,
> and the problem had gone. Thus, I never investigated this problem.
>
> Now, I upgraded a machine for production use to asterisk-1.4.8,
> and do encounter the same problem.
>
> I have other asterisk machines running, using the same
> dialplan, without this problem.
>
> Did anyone else observe this strange behaviour of calls ending
> after 64 secondes of uptime?
>
> My os is Suse-Linux 10.2.
>
>
> Thanks for any hints!
> Roger.
>
>
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Re: [asterisk-users] call-limit in database

[Jaswinder Singh]

call-limit is to set number of alternate calls . and L is to limit
duration of each call .

On Dec 22, 2007 2:54 PM, Pezhman Lali <[EMAIL PROTECTED]> wrote:
> Dear
> I am using this function with "L"
> for example in the dbase.
> app=Dial
> appdata="SIP/[EMAIL PROTECTED]|60|L(10)"
> it means dial 1 thru 1.1.1.1, with
> limitation=10 mili-second, and time out=60 sec
>
> best
> Mani
>
> --- Bhrugu Mehta <[EMAIL PROTECTED]> wrote:
>
> > hi, all
> > proble:
> > I have add CALL-LIMIT field in my sip table in
> > mysql.
> > but when i call using sip same error occurred when
> > use simple sip.conf file.
> >
> > is this possible to add CALL-LIMIT field in sip
> > realtime table in mysql.
> > if yes than how
> >
> > Bhrugu Mehta
> >
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Re: [asterisk-users] Your "favorite" Asterisk application.

[Jaswinder Singh]

I like the Echo  application in asterisk ;) . Weird :P

On Jan 24, 2008 7:07 PM, Mark Johnson <[EMAIL PROTECTED]> wrote:

> Ken D'Ambrosio wrote:
> > Hi, all.  I've done some Asterisk recelling, but recently got roped into
> a
> > Sr. SysAdmin position.  Our PBX is c. 1823, and -- well, as pretty much
> > all circuit-based systems do, it sucks.  It sucks to administer, moves
> > suck... you know the drill.  So, I'd love change to an Asterisk system.
> > My boss, who loves to spend money for no particular reason, wants to go
> > proprietary, though.  So I'm going to have to try to sell him.  I
> figured
> > one place to start would be some of the really cool applications that
> > Asterisk has that -- generally, at least -- don't require licensing.
>  Some
> > of my favorites are follow-me, meetme, voicemail-to-e-mail and
> > fax-to-e-mail.  What are some of your favorite features/applications, be
> > ith native or third-party?
> >
> > Thanks,
> >
> > -Ken
>
> We moved from a Cisco Call Manager about 2.5 years ago to Asterisk.  One
> of the hurdles I had was that the Call Manager had a receptionist panel
> so they could see who was on the phone, transfer calls, etc...
>
> I set up a demo of of the Flash Operator Panel and it alleviated that
> sticking point.  It's a little slower than an executable would be, but
> it's web based and flash so it's runs on just about every browser and OS.
>
> You can even do some slick things like pop up windows in the browser to
> provide information about who is calling.  Works good for a CMS system
> where a customer service rep can automatically be shown information
> about the customer who is on the line.
>
> http://www.asternic.org/
>
> --
> Mark Johnson
> http://www.astroshapes.com/information-technology/blog/
>
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Re: [asterisk-users] Realtime problem host='dynamic' in 1.2.26.1

[Jaswinder Singh]

Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve
SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host = 'dynamic'

Quite obvious .. doest sippeers have that row ?

On Jan 24, 2008 6:04 PM, Torbjörn Abrahamsson <
[EMAIL PROTECTED]> wrote:

> Developers and maintainers, any information?
>
> // T
>
> Torbjörn Abrahamsson wrote:
> > Hello!
> >
> > We are using the 1.2 branch, and upgraded to 1.2.26.1. We ran into some
> > problems when using realtime for peers. We connect the PBX to a sip peer
> > at an ITSP, and when we try to dial the peer we get:
> >
> > Jan 23 09:02:07 VERBOSE[2236] logger.c: -- Executing
> > Dial("SIP/dev02-08c36f28", "SIP/[EMAIL PROTECTED]||W") in new stack
> > Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime:
> > Everything is fine.
> > Jan 23 09:02:07 DEBUG[2236] res_config_mysql.c: MySQL RealTime: Retrieve
> > SQL: SELECT * FROM sippeers WHERE name = '989800-out' AND host =
> 'dynamic'
> > Jan 23 09:02:07 WARNING[2236] chan_sip.c: No such host: 989800-out
> > Jan 23 09:02:07 NOTICE[2236] app_dial.c: Unable to create channel of
> > type 'SIP' (cause 3 - No route to destination)
> > Jan 23 09:02:07 VERBOSE[2236] logger.c:   == Everyone is busy/congested
> > at this time (1:0/0/1)
> > Jan 23 09:02:07 DEBUG[2236] app_dial.c: Exiting with
> DIALSTATUS=CHANUNAVAIL.
> >
> > I looked in the archives and found this thread:
> >
> >
> http://lists.digium.com/pipermail/asterisk-users/2007-December/202616.html
> >
> > Here the same problem is discussed for the 1.4 branch, and the result is
> > that the problem should be fixed. But this is still a problem in 1.2branch.
> >
> > Will this be corrected in a new release, or is this not considered a
> > security fix and hence ignored? Actually isn't this a fix for a security
> > fix...
> >
> > BR,
> > Torbjörn Abrahamsson
> >
> >
> >
> >
> >
> >
> >
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Re: [asterisk-users] Disable IAX2 call path optimization

[Jaswinder Singh]

You are usinfg sip or iax ? Its possible to prevent in both cases for sip
under peer definition you can put canreinvite=no and in iax2 you can put
transfer=no for asterisk 1.4 or notranser=yes in asterisk 1.2 .. Search for
this on voip-info.org wiki for more info .

On Jan 25, 2008 7:03 PM, <[EMAIL PROTECTED]> wrote:

>  I have a call coming in from Asterisk-A going to Asterisk-B where it's
> determined that the called party is in fact yet another number in Asterisk-A
> so a new call is created from B to A and the two calls bridged (by Asterisk)
> at Asterisk-B.
>
>
>
> Originating Caller ==> Asterisk-A  ==> Asterisk-B ==> Asterisk-A
>
>
>
> Now, what happens is that in my case both A and B are on the same network
> and therefore Asterisk-A apparently optimizes the round-trip to Asterisk-B
> out and the original caller talks directly to the extension hosted in
> Asterisk-A without the call path going the round-trip to Asterisk-B.
>
>
>
> Is it possible to prevent this optimization from happening? Any way to
> control if it happens at all, or can it be selected on per-call basis
> somehow?
>
>
>
> Can I find anywhere more details of call path optimization and it's
> configuration, use, functionality and behaviour?
>
>
>
> tnx,
>
> Baldvin
>
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Re: [asterisk-users] Meetme voice quality problems

[Jaswinder Singh]

Ubuntu has a real time kernel in repository "apt-get install linux-rt" . So
you dont need to recompile . I think debian should also have one in
repository , or u can manually compile a real time enabled kernel . Here's
what is shows with real time patched  kernel .

 dmesg|grep ztdummy
[   53.293071] ztdummy: Trying to load High Resolution Timer
[   53.293076] ztdummy: Initialized High Resolution Timer
[   53.293078] ztdummy: Starting High Resolution Timer
[   53.293080] ztdummy: High Resolution Timer started, good to go

 zttest
Opened pseudo zap interface, measuring accuracy...
100.00% 99.987793% 99.792480% 99.780273% 99.987793% 99.975586%
99.987793%
99.987793% 100.00% 100.00% 100.00% 99.987793% 99.987793%
99.987793% 100.00%
100.00% 99.987793% 100.00%
--- Results after 18 passes ---
Best: 100.00 -- Worst: 99.780273 -- Average: 99.969482



On Feb 2, 2008 10:27 PM, Administrator TOOTAI <[EMAIL PROTECTED]> wrote:

> Matthew J. Roth a écrit :
> > Administrator TOOTAI wrote:
> >
> >> This is not true if you're using B410P cards. We always face timing
> >> problem as we can't -Asterisk stability issues- add X100P or TDM400P
> >> with those cards
> >>
> > Daniel,
> >
> > I thought that using an empty TDM400P as a timing source may no longer
> > be the best solution due to the emergence of new stable timing sources
> > (such as HPET), but this is an interesting issue.  Are you stating that
> > you can't put an X100P or a TDM400P with no lines attached alongside a
> > B410P because it impacts the stability of Asterisk?
> Yes
> >  Do you have any
> > idea why?
> No
> >  Can't the B410P be used as a timing source?
> No
> >  What have you
> > done to provide stable timing?
> >
> ztdummy, not always stable :-(
> > I know that's a lot of questions, but I'm genuinely curious.
> ;-)
> >   It seems
> > very strange that a TDM400P in timingonly mode and no lines attached
> > would have any impact on Asterisk's stability.
> >
> I have to add that this is mainly true with 2 B410P in the server or
> with B410P in amd64 machines. Seems also that Debian Etch with a 2.6.18
> kernel is not the best :-(
> --
> Daniel
>
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Re: [asterisk-users] Enterprise or Fedora?

[Jaswinder Singh]

I prefer CentOS barebone install and yumming the way up for dependencies but
manually compile asterisk/zaptel . Ubuntu servers are pretty good too since
its repositories are quite bigger compared to CentOS .

On Feb 2, 2008 11:45 PM, shadowym <[EMAIL PROTECTED]> wrote:

> Actively maintained or actively being broken and fixed with constant
> updates?  Not something suitable for Production IMHO.  Makes more sense
> for
> development and experimentation IMHO.
>
>
> -Original Message-
> From: Benny Amorsen [mailto:[EMAIL PROTECTED]
> Sent: Friday, February 01, 2008 1:54 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Enterprise or Fedora?
>
> "shadowym" <[EMAIL PROTECTED]> writes:
>
> > I cannot think of a single reason to use Fedora for a production
> anything
> > when there are alternatives like CentOS.  Fedora is bleeding edge stuff
> and
> > constantly changing.
>
> The advantage of Fedora is that it is very actively maintained -- and
> asterisk is only a yum install asterisk away!
>
>
> /Benny
>
>
>
>
>
>
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Re: [asterisk-users] Call recording filename

[Jaswinder Singh]

I have figured out a way to include dialed number in recorded
voicefile in freepbx . You have to edit
/var/lib/asterisk/agi-bin/recordingcheck
add this lines after $agi=new AGI()

$temp= $agi->get_variable(DIAL_NUMBER);
$agi->verbose("Number to be dialled is -{$temp["data"]}");

After this you can use variable {$temp["data"]} in outfile names ( set
few line below in same file ) . This is only required for freepbx .

On 30/11/06, Vicky <[EMAIL PROTECTED]> wrote:

No response at all :( . I did a temporary solution . I made cdr mysql to
store unique id into database from this wiki . So i now atleast have
uniquefield common in callfilename and sql  records to tally .

Storing the Unique ID
Q: It would appear that the "uniqueid" field is not being populated in the
MySQL CDR DB. Is this an obsolete field or is a bug?

A: You need to define MYSQL_LOGUNIQUEID at compile time for it to use that
field.

You have two options in /usr/src/asterisk-addons:
1. Add CFLAGS+=-DMYSQL_LOGUNIQUEID to the Makefile.
2. Add a #define MYSQL_LOGUNIQUEID to the top of cdr_addon_mysql.c.

Finally perform the usual make clean, make, make install. Be sure to check
the Makefile for the presence of this flag after having done a CVS update!
You will most probably also want to index the uniqueid field in your cdr
table to improve performance.



On 30/11/06, Nick Hoffman <[EMAIL PROTECTED]> wrote:
> On Wed November 29 2006 05:17, Vicky <[EMAIL PROTECTED]> wrote:
> > I am using asterisk along with freepbx . When recording is enabled for a
> > extension the call record file made in /var/spool/asterisk/monitor
> > contains information like OUT(extension
> > number)-(timestamp)-(uniqueid).wav . This can be a big
mess if there are
> > more than 1000-2000 files in that folder and very hard to locate a call
> > recording based on call time and extension number who dialled. I need to
> > put something like outgoing number dialled within call file name instead
> > of uniqueid .. After watching in console i  opened up
> > /var/lib/asterisk/agi-bin/recordingcheck and saw that
it is setting
> > callfilename variable with extension number,time,unique id , etc. so i
> > edited and instead of $uniqueid i put $DIALEDPEERNUMBER ( saw in
> >
http://www.voip-info.org/wiki/index.php?page=Asterisk+variables
) but
> > its just not giving dialed number and hence callfilename  doesnt contain
> > outgoing number . Any suggestions how can i get outgoing call number in
> > recording file ?
>
>
> Hi Vicky. Did you receive any responses to your email? I'd be interested
in
> anything people suggested.
>
> Cheers,
> -- Nick
> E: [EMAIL PROTECTED]
> P: +61 7 5591 3588
> F: +61 7 5591 6588
>
> If you receive this email by mistake, please notify us and do not make any
> use of the email.  We do not waive any privilege, confidentiality or
> copyright associated with it.
>



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Re: [asterisk-users] auto/forced call

[Jaswinder Singh]

No python code needed . Check .call files at
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

On 23/05/07, Brad Sumrall <[EMAIL PROTECTED]> wrote:

Can anyone guide me to a "how to" on automating a call?

I know a little piece of code (normally python) has to be place some where
and then a file has to be mv into the spooler.

Where do I get the run down?
I have a button on another application that sends an email and I want it to
also send a text message through asterisk!

Brad


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Re: [asterisk-users] Meet me

[Jaswinder Singh]

change conf => 222
to conf => 222
( remove | )

I had same problem as freepbx always put | removing it fixed the problem
On 29/05/07, Khaled Chehab <[EMAIL PROTECTED]> wrote:





I am using asterisk 1.4.4 now and facing a problem with meetme,the code  I
was using with asterisk 1.2 is not functioning with 1.4 ,my code is

conf => 222| at meetme.conf

at meet_me_additional



like this

exten => 21,1,MeetMe(21,dq)

exten => 21,2,Playback(beep)



or this

exten => 222,1,GotoIfTime(*|mon-sun|08-08|may-may?223,1)

exten => 222,n,Playback(vm-goodbye)

exten => 222,n,Hangup

exten =>
STARTMEETME,1,MeetMe(${MEETME_ROOMNUM},${MEETME_OPTS},${PIN})

exten => STARTMEETME,n,Hangup

exten => h,1,Hangup

 exten => 223,1,Set(MEETME_ROOMNUM=222)

 exten => 223,n,GotoIf($[${DIALSTATUS} = ANSWER]?READPIN)

 exten => 223,n,Answer

exten => 223,n,Wait(1)

 exten => 223,n(READPIN),Read(PIN,enter-conf-pin-number,,)

 exten => 223,n,GotoIf($[foo${PIN} = foo]?USER)

 exten => 223,n,GotoIf($[${PIN} = ]?ADMIN)

 exten => 223,n,Playback(conf-invalidpin)

 exten => 223,n,Goto(READPIN)

 exten => 223,n(ADMIN),Set(MEETME_OPTS=aAwciMs)

 exten => 223,n,Goto(STARTMEETME,1)

 exten => 223,n(USER),Set(MEETME_OPTS=ciMs)

 exten => 223,n,Goto(STARTMEETME,1)





please guide me



 
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Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *

[Jaswinder Singh]

I think its a fair decision . 1.2 is very stable and they are not
closing it all together , security issues will still be fixed . They
need to concentrate more on 1.4 to make it bugfree .

On 29/05/07, Stephen Bosch <[EMAIL PROTECTED]> wrote:

John covici wrote:
> I have an install using Rhino cards -- I sure hope they get their act
> together by then.

They have no choice now, do they?

Nothing focuses the attention like a deadline.

-Stephen-
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Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *

[Jaswinder Singh]

Well i guess you just need a good look on logs for why and when you
are getting core dumps . We are having few servers running .1.2.18 and
it has turned out to be most stable  in whole 1.2 branch ( had some
issues with 1.2.13 and 14 ) .


Except that for some users 1.2.18 is NOT stable.  I've had to roll back
to 1.2.15 on my production servers in order to prevent core dumps at
least once per day.  No, I am not willing to turn my production servers
into testing servers to solve this.  Doing so would make me a "former
consultant" for these customers.
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Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *

[Jaswinder Singh]

What you say might be true for small business or home  pbx systems .
But if you have a production server handling sip/iax trunks  over
internet then you need to upgrade to avoid  security related bugs and
exploits that are released .



You seem to miss the idea here.  You work with a version that supports
your "feature needs" and find the sub-version that provides the most
stability for your deployments.  Lets face it these boxes should go in
and run for weeks, months or even years without much intervention
(assuming the mission of the box does not change).  I'm running a
1.2.7.something (i think) that has been running almost nonstop since
installing.  Very reliable and stable for my needs.  Compared to a
Merlin or Nortel or any other system out that I feel I have a much
better product.

Could I benefit from a newer sub-version? Maybe.
Will I upgrade the box in it current roll?  No.

Unless the application I use the box for has a major change (or the
hardware dies) I'll just let it keep on running as it is.

For my future deploys I am working closer with 1.4.  The reason is
clear.  1.4 is the future of asterisk.  When 1.6 or 2.0 comes out I'll
investigate into migrating in that direction at that time because that
will become the future of asterisk.


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Re: [asterisk-users] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFOR THE ASTERISK COMMUNITY - PLEASEREAD NOW *

[Jaswinder Singh]

Well if you are out of luck with asterisk .. How about its fork
callweaver ? I am highly awaiting its stable release to see if it
holds upto what its wiki says .

On 30/05/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

On Tue, May 29, 2007 at 01:06:54PM -0500, Eric ManxPower Wieling wrote:
> Michael Collins wrote:
> >>I think its a fair decision . 1.2 is very stable and they are not
> >>closing it all together , security issues will still be fixed . They
> >>need to concentrate more on 1.4 to make it bugfree .
> >
> >Fair indeed.  I would guess that a completely stable 1.2 w/ security
> >maintenance is acceptable to the majority of users.  Those folks still
> >using 1.0.x certainly aren't clamoring for new features!  The great many
>
> Except that for some users 1.2.18 is NOT stable.  I've had to roll back
> to 1.2.15 on my production servers in order to prevent core dumps at
> least once per day.  No, I am not willing to turn my production servers
> into testing servers to solve this.  Doing so would make me a "former
> consultant" for these customers.

So basically what you're saying is that some efforts should be
concentrated on 1.2 as well.

So let's start with your specific problems. Are there open bugs for them?

--
   Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] channel_find_locked: Avoided deadlock

[Jaswinder Singh]

Is it over iax and there are lot of outgoing channels  ? If yes then
you are not the only person having this ..

On 30/05/07, ram <[EMAIL PROTECTED]> wrote:

Hi

i have 20 people calling agents calling

when ever they calling i get this below error

May 30 00:46:57 WARNING[2840]: channel.c:785 channel_find_locked: Avoided
deadlock for '0x8b2f50', 10 retries!

and the voice go choppy, and voice breakages

iam using Latest SVN, any suggestion to come over this problem

ram

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Re: [asterisk-users] multiple host= in sip.conf

[Jaswinder Singh]

I dont think asterisk supports this . You can have host=dynamic and he
can send calls from different servers . Problem will arise when you
need to call him ( if registrations are enabled then latest
registration will be getting call from you or you can directly send
calls to his ip . )

On 30/05/07, Yusuf <[EMAIL PROTECTED]> wrote:

Hi,

I am running Asterisk 1.4.4, and needed to setup sip accounts for someone to 
call my
server and place calls.  However, he has multiple IP's that he comes from, and 
since I
authenticate him of his IP,  I did this, and it works.

[vz1]
context=outbound
type=friend
host=x.x.x.x
disallow=all
allow=alaw
canreinvite=no

[vz2]
context=outbound
type=friend
host=y.y.y.y
disallow=all
allow=alaw
canreinvite=no

[vz3]
context=outbound
type=friend
host=.z.z.z.z
disallow=all
allow=alaw
canreinvite=no


However, is there anyway I can have just one account for him, with mult host= 
statements,
so I can authenticate him based on his IP in just one place?

--

thanks,
Yusuf
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Re: [asterisk-users] Still cannot make a single call from asterisk via softphone to pstn!!!!!!!

[Jaswinder Singh]

Can you post some output from asterisk cli output while you make call ?

On 30/05/07, BSumrall <[EMAIL PROTECTED]> wrote:





after 18 hours, over 200 pages of reading, a complete reinstall of asterisk
I am down to this.

 extensions.conf

 [globals]
 CONSOLE=Console/dsp
 IAXINFO=guest
 TRUNK=Zap/g2
 TRUNKMSD=1

 [default]
 exten => 8005181896,1,Dial,(IAX2/UXMC)
 exten => s,1,Answer()

 (I tried)
 exten => _1XX,1,DIAL,(IAX2/teliax/${EXTEN},30,tr)
 (as well)

 iax.conf

 [general]
 port=4569
 bandwidth=low
 disallow=lpc10
 jitterbuffer=no
 forcejitterbuffer=no
 tos=lowdelay
 autokill=yes

 register => :[EMAIL PROTECTED]

 [teliax]
 context=default
 type=friend
 host=voip-co3.teliax.com
 auth=md5
 user=
 secret=x
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm

 sip.conf

 [UXMC]
 user=xxx
 context=internal
 type=friend
 qualify=yes
 nat=no
 secret=
 canreinvite=no
 host=dynamic
 nat=no

 If I put back previous config, I can call into the 1800 number and here
that silly chick heckle me from my server!
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Re: [asterisk-users] False ring problem

[Jaswinder Singh]

Do you have 'r' in ur dial command ? 'r' makes asterisk produce ring .

On 30/05/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote:

Hi all,
when a user dials any number, asterisk automatically generates ringing which
caller can hear, and after 2 - 3 rings asterisk detects that the called user
is busy, then caller hears busy tone. for example user hears---
tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing
at the start so that user hears only beep beep beep if the called user is
busy. I have used the R and r options in Dial application but they dont
work.

--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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Re: [asterisk-users] Net2Phone Multiple SIP Trunk Not Working

[Jaswinder Singh]

You just have a 1 call limit on your account on net2phone side .
Making 10 trunk wont let you make 10 account its restriction on your
account not ip . Just change your provider .

On 01/06/07, Salah Eddine ELMRABET <[EMAIL PROTECTED]> wrote:


Hi,

Any help regarding Net2Phone poblem?

BR


On 6/1/07, Andrew Furey <[EMAIL PROTECTED]> wrote:
> On 01/06/07, Salah Eddine ELMRABET <[EMAIL PROTECTED]> wrote:
> > I'm sorry that's because I didn't get a visibility of ny post, I though
that
> > was a network problem (as I cannot see my post on the mailing list)
>
> You never do with mailing lists on Gmail, I presume it hides it based
> on the message ID (since you already have a copy).
>
> Andrew
>
> --
> Linux supports the notion of a command line or a shell for the same
> reason that only children read books with only pictures in them.
> Language, be it English or something else, is the only tool flexible
> enough to accomplish a sufficiently broad range of tasks.
>  -- Bill Garrett
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Re: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD TOO!

[Jaswinder Singh]

Well freepbx runs fine with asterisk 1.4.4 ( atleast for me ) i think
some changes was introduced in 1.4 ( 1.4.4 ?)  for some backward
compatibility...  like show channels  now work in 1.4.4 instead of
core show channels but it gives a notice that 'show channels' is
deprecated bla bla .Freepbx works completely fine with asterisk 1.4
for me .


On 31/05/07, shadowym <[EMAIL PROTECTED]> wrote:

If anything this should "motivate" the FreePBX developers a bit more.

-Original Message-
From: Jared Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 30, 2007 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] *End Of Life ASTERISK 1.2.X THEN AMP IS DEAD
TOO!

On 5/30/07, BSumrall <[EMAIL PROTECTED]> wrote:
> AMP does not support 1.4 and will not until AMP 2.3 is released!

I'm sorry to hear you think our decision (I say "our", as I was at the
Asterisk Developers' Conference where the decision was made) will kill the
AMP project.  Personally, I don't think the situation is as dire as you say.
I'm quite sure the AMP developers will step up to the plate and support
Asterisk 1.4 in due time.  When that will be I can't say, as I'm not active
in the AMP community. I can't image it would take that long to move over to
Asterisk 1.4, as the dialplan changes aren't *that* extensive between 1.2
and 1.4. (Obviously any code that ties into the internal C APIs of Asterisk
will take longer to port.)

> Bet you guys didn't think about that one!

Actually, we did.  As a matter of fact, I was *very* vocal at the conference
in stating that we needed to give users, integrators, and projects like AMP
a substantial warning before putting Asterisk 1.2 in security maintenance
mode, as they need time to react.

At the same time, I don't think anyone should expect the Asterisk developers
to base all their decisions completely on the timetables of outside projects
(like AMP).  There is a plethora of projects and programs out there that tie
into Asterisk, and if we as developers waited for every single one to move
over to Asterisk 1.4, we'd never accomplish anything.  There's simply a
finite set of resources (developers and bug marshalls in this case), and a
decision had to be made on how best to use those resources.  Personally, I
think it would be great if there were more communication between the outside
projects and the Asterisk developers, so that there isn't so much animosity
when decisions like this are made.

In short, the decision is probably going to cause some short-term discomfort
for some people, but I truly believe it's a good decision for the long-term
health and sanity of the Asterisk developers and Asterisk community in
general.  No, we're not trying to kill off AMP or any other outside project
-- we're trying to make Asterisk (and by extension, anything that uses or
adds on to Asterisk) as great as possible.

-Jared


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Re: [asterisk-users] yum om centos

[Jaswinder Singh]

independently install each rpm via rpm command :-/

On 04/06/07, Khaled Chehab <[EMAIL PROTECTED]> wrote:





I have 2 servers, one connected to internet and the other is on a private
lan have no access to internet.

On the first server I update the kernel by yum update

And installed asterisk prerequisite module

yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf \ libtool
make automake automake14 automake15 automake16 automake17 \ bison byacc flex
libtermcap libtermcap-devel newt newt-devel \ ncurses ncurses-devel
openssl-devel zlib zlib-devel krb5-devel





I zipped /var/cache/yum from the first server and extract it on the second
server at the same directory.



On the second server I tried to update using

yum update

 but the yum update failed.





How can I do that with out connecting the second server to internet .







Khaled

Regards





 
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Re: [asterisk-users] any codec passthru mode

[Jaswinder Singh]

Asterisk by default uses the codec preferred by other device/client  .
Asterisk 1.2 ( dunno abt 1.4 specifically)  is not intelligent enough
to check if it can avoid transcoding by forcing same codec on other
side of conversation . If both sides prefer g729 then asterisk does
not do transcoding but if one side prefer gsm and other prefers g729
and the gsm side can also support g729 still asterisk will transcode .
Someone posted a patch to this in mantis bug tracking system at digium
for 1.2 .. google for it and maybe you can find  .

On 31/05/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote:

Does anybody has any documentation on codec negotiation within asterisk?

Well im using free g729 codec for testing purposes. i mentioned g729 just as
an example. whatever codec is mentioned in user perefernce, asterisk uses
ulaw to throw out the call.


On 5/30/07, Marco Mouta <[EMAIL PROTECTED]> wrote:
> so you r sure you have g729 licences installed and ur * is transcoding
your RTP streaming?
>
> Test the work flow with disallow=all and allow=g729, can be my mistake but
I remember to read somewhere on the net any issue about codec negotiating
precedence when you use allow=all.
>
> good luck
>
>
>
> On 5/30/07, Rizwan Hisham < [EMAIL PROTECTED]> wrote:
> >
> > Hi all,
> > My configuration is:
> > USER (connects to)> ASTERISK---(connects to)--->CARRIER-OUT
> >
> > i want the user preffered codec to pass thru asterisk to carrier-out.
what i mean is:
> > USER (user uses g729)> ASTERISK---(asterisk should use g729 for
dialing out)--->CARRIER-OUT
> >
> > instead, this is what happens
> > USER (user uses g729)> ASTERISK---(asterisk uses
g711u)--->CARRIER-OUT
> >
> > How can i force asterisk to use user preffered codec for dialing out so
that my asterisk machine saves time by no conversion
> > USER PREFERENCE IS
> > disallow=all
> > allow=g729
> >
> > CARRIER PREFERENCE IS
> > allow=all
> >
> > Anybody who can help?
> >
> > --
> > Rizwan Hisham
> > Software Engineer
> > AXVOICE Inc.
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> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
>
>
> --
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> This e-mail message is intended only for individual(s) to whom it is
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>



--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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Re: [asterisk-users] ringback detection

[Jaswinder Singh]

It just might be that your carrier is not sending ring . You can use
'r' in asterisk dial command in extensions.conf to generate ring from
asterisk .

On 31/05/07, dima <[EMAIL PROTECTED]> wrote:

Hello, everyone.
Could anyone explain me how does ringback detection works in asterisk.
Sometimes, when making a call, my asterisk box doesn't detect a ringback
and I just hear silence until the other party picks up the phone. I've
checked the SIP messages and they are ok (I'm getting 183 "session in
progress"), so I guess I should be debugging the RTP packets. From then
on I'm stuck. Does anyone know what type of packets I should be looking
for?

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Re: [asterisk-users] g729

[Jaswinder Singh]

Are you trying to record the conversation as well ?

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:





I installed a hardware g729 codec card in my asterisk, and I'm getting the
following error when calling from a g729 sip extension to a SIP trunk also
set to g729.  The call goes through just fine, but these error messages keep
flying by until I disconnect the call.



Any ideas?



ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies

Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
Translation to slin failed, dropping frame for spies
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Re: [asterisk-users] g729

[Jaswinder Singh]

I think asterisk first converts audio stream to slin for recording to
a wav file . Since you are using hardware g729 transcoder i think this
is what is causing the problem . Is the calla actually being recorded
?  I suggest that you use monitor application since it can directly
record g729 audio stream and run some cron script with sox mixing the
IN and OUT files in 1 file .

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:

Yes

This is my extensions.conf entry.

exten => _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
exten =>
_1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
RID}-${EXTEN}-${TIMESTAMP}-OUT)
exten =>
_1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
OUT)
exten => _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
exten => _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
exten => _1NXXNXX,6,Set(CALLERID(number)=14073844200)
exten => _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
exten => _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 4:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729

Are you trying to record the conversation as well ?

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
>
>
>
>
> I installed a hardware g729 codec card in my asterisk, and I'm getting the
> following error when calling from a g729 sip extension to a SIP trunk also
> set to g729.  The call goes through just fine, but these error messages
keep
> flying by until I disconnect the call.
>
>
>
> Any ideas?
>
>
>
> ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
> failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
>
> Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> Translation to slin failed, dropping frame for spies
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>
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>
>
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Re: [asterisk-users] Voip-info.org

[Jaswinder Singh]

Yep its down for me tooo .

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:





Is anyone else having trouble going into voip-info.org today?
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Re: [asterisk-users] g729

[Jaswinder Singh]

Just read somewhere that you can use extension as g729 even in
mixmonitor so it will record g729 stream and later you can convert it
to mp3 or wav using sox . If this fails then try monitor application .


On 06/06/07, Jaswinder Singh <[EMAIL PROTECTED]> wrote:

I think asterisk first converts audio stream to slin for recording to
a wav file . Since you are using hardware g729 transcoder i think this
is what is causing the problem . Is the calla actually being recorded
?  I suggest that you use monitor application since it can directly
record g729 audio stream and run some cron script with sox mixing the
IN and OUT files in 1 file .

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
> Yes
>
> This is my extensions.conf entry.
>
> exten => _1NXNXXX,1,Set(DYNAMIC_FEATURES=automon)
> exten =>
> _1NXXNXX,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/CONTINEX-${CALLE
> RID}-${EXTEN}-${TIMESTAMP}-OUT)
> exten =>
> _1NXXNXX,3,Set(TOUCH_MONITOR=CONTINEX-${CALLERID}-${EXTEN}-${TIMESTAMP}-
> OUT)
> exten => _1NXXNXX,4,Set(CDR(accountCode)=${CALLERID}-${TIMESTAMP})
> exten => _1NXXNXX,5,Set(CDR(UserField)=${MONITOR_FILENAME})
> exten => _1NXXNXX,6,Set(CALLERID(number)=14073844200)
> exten => _1NXXNXX,7,MixMonitor(${CALLFILENAME}.wav49)
> exten => _1NXXNXX,8,Dial(SIP/[EMAIL PROTECTED],,wW)
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
> Singh
> Sent: Wednesday, June 06, 2007 4:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] g729
>
> Are you trying to record the conversation as well ?
>
> On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> >
> > I installed a hardware g729 codec card in my asterisk, and I'm getting the
> > following error when calling from a g729 sip extension to a SIP trunk also
> > set to g729.  The call goes through just fine, but these error messages
> keep
> > flying by until I disconnect the call.
> >
> >
> >
> > Any ideas?
> >
> >
> >
> > ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin
> > failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11864]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11851]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> >
> > Jun  5 18:24:01 ERROR[11871]: channel.c:1316 queue_frame_to_spies:
> > Translation to slin failed, dropping frame for spies
> > ___
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> >
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> >
> >
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Re: [asterisk-users] any codec passthru mode

[Jaswinder Singh]

Yes it might be dumb but since asterisk is a pbx and not a sip proxy
it has to perform many other functions as well .  But i do think that
asterisk should act little smart in this case


SIP wrote:
> That just seems really, REALLY dumb for a program of this magnitude.
>
> I know this has been patched here and there by one person or another,
> but does anyone know if any of these patches to make CODEC negotiation
> actually, you know, negotiate a CODEC will ever make it into the core
> src?
>
>
> Jaswinder Singh wrote:
>> Asterisk by default uses the codec preferred by other device/client  .
>> Asterisk 1.2 ( dunno abt 1.4 specifically)  is not intelligent enough
>> to check if it can avoid transcoding by forcing same codec on other
>> side of conversation . If both sides prefer g729 then asterisk does
>> not do transcoding but if one side prefer gsm and other prefers g729
>> and the gsm side can also support g729 still asterisk will transcode .
>> Someone posted a patch to this in mantis bug tracking system at digium
>> for 1.2 .. google for it and maybe you can find  .
>>
>> On 31/05/07, Rizwan Hisham <[EMAIL PROTECTED]> wrote:
>>> Does anybody has any documentation on codec negotiation within
>>> asterisk?
>>>
>>> Well im using free g729 codec for testing purposes. i mentioned g729
>>> just as
>>> an example. whatever codec is mentioned in user perefernce, asterisk
>>> uses
>>> ulaw to throw out the call.
>>>
>>>
>>> On 5/30/07, Marco Mouta <[EMAIL PROTECTED]> wrote:
>>> > so you r sure you have g729 licences installed and ur * is
>>> transcoding
>>> your RTP streaming?
>>> >
>>> > Test the work flow with disallow=all and allow=g729, can be my
>>> mistake but
>>> I remember to read somewhere on the net any issue about codec
>>> negotiating
>>> precedence when you use allow=all.
>>> >
>>> > good luck
>>> >
>>> >
>>> >
>>> > On 5/30/07, Rizwan Hisham < [EMAIL PROTECTED]> wrote:
>>> > >
>>> > > Hi all,
>>> > > My configuration is:
>>> > > USER (connects to)> ASTERISK---(connects to)--->CARRIER-OUT
>>> > >
>>> > > i want the user preffered codec to pass thru asterisk to
>>> carrier-out.
>>> what i mean is:
>>> > > USER (user uses g729)> ASTERISK---(asterisk should use
>>> g729 for
>>> dialing out)--->CARRIER-OUT
>>> > >
>>> > > instead, this is what happens
>>> > > USER (user uses g729)> ASTERISK---(asterisk uses
>>> g711u)--->CARRIER-OUT
>>> > >
>>> > > How can i force asterisk to use user preffered codec for dialing
>>> out so
>>> that my asterisk machine saves time by no conversion
>>> > > USER PREFERENCE IS
>>> > > disallow=all
>>> > > allow=g729
>>> > >
>>> > > CARRIER PREFERENCE IS
>>> > > allow=all
>>> > >
>>> > > Anybody who can help?
>>> > >
>>> > > --
>>> > > Rizwan Hisham
>>> > > Software Engineer
>>> > > AXVOICE 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
>>> > >
>>> > >
>>> >
>>> >
>>> >
>>> > --
>>> > Esta mensagem (incluindo quaisquer anexos) pode conter informação
>>> confidencial para uso exclusivo do destinatário. Se não for o
>>> destinatário
>>> pretendido, não deverá usar, distribuir ou copiar este e-mail. Se
>>> recebeu
>>> esta mensagem por engano, por favor informe o emissor e elimine-a
>>> imediatamente. Obrigado.
>>> >
>>> > This e-mail message is intended only for individual(s) to whom it is
>>> addressed and may contain information that is privileged, confidential,
>>> proprietary, or otherwise exempt from disclosure under applicable
>>> law. If
>>> you believe you have received this message in error, please advise the
>>> sender by return e-mail and delete it fro

Re: [asterisk-users] TCP<->UDP SIP proxy?

[Jaswinder Singh]

I think there is a patch for sip over tcp in asterisk but not sure if
its stable or not

try this http://bugs.digium.com/view.php?id=4903

You can also install openser as sip proxy . it supports sip over tcp .

On Wed,  6 Jun 2007 19:14 +0300, Yehavi Bourvine +972-8-9489444
<[EMAIL PROTECTED]> wrote:

Hello,

   One of our faculties have Microsoft's LCS and would like to connect it to
our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS
talks SIP over TCP with TLS. Anyone can recommend a gateway between these two
protocols?

  Thanks! __Yehavi:
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Re: [asterisk-users] Voip-info.org

[Jaswinder Singh]

Its up and working now .

On 06/06/07, Compnet Bobby <[EMAIL PROTECTED]> wrote:

Same in southern cali!




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder
Singh
Sent: Wednesday, June 06, 2007 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voip-info.org

Yep its down for me tooo .

On 06/06/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
>
>
>
>
> Is anyone else having trouble going into voip-info.org today?
> ___
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Re: [asterisk-users] Needed changes in Asterisk to change the SIP port to 5062

[Jaswinder Singh]

In sip.conf it should be bindport=5062

On 06/06/07, Crazy Boy <[EMAIL PROTECTED]> wrote:

Hi Friends,
I want to use 5062 port for SIP protocol. I made the below modifications in
my server to use 5062 port.
Polycom phone: port=5062
 Trunk settings: port=5062
 sip.conf: bindaddr=5062
 Extension configuration details: 5062
Our VoIP provider told me that they are allowing the SIP traffic through
5060 to 5064. I observed on my server console that my server is registered
with our VoIP provider with 5062 port. But, I am unable to make outgoing
calls.
Do I need to modify any other settings in Asterisk?
Look forward to your response. Thank you.
Regards,
 Chandra.

 
Need a vacation? Get great deals to amazing places on Yahoo! Travel.


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Re: [asterisk-users] IAX trunk with dynamic IPs

[Jaswinder Singh]

In your no-ip client set it to update ip every 2 minutes or so . and
/etc/asterisk/dnsmgr.conf  set refresh interval as 30-40 by defualt
its 300 ( 5 minutes)

On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote:

Hi Matt,

Every time I do that, IAX stop sending the POKE messages (necessary for
trunk management).
Do you know what could be happening?

Thanks.
Ronaldo.

Matt wrote:
>
> *set "enable=yes" in the "[general]" section of
> /etc/asterisk/dnsmgr.conf*
>
>
> 
>
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Re: [asterisk-users] IAX trunk with dynamic IPs

[Jaswinder Singh]

Hello
You should use qualify=310 ( any value in millisec ) .. qualify=yes
is not proper .

I am not sure about how asterisk's dnsmgr manages dns refreshing but
maybe someone else can answer that question .

On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote:

Hi Jaswinder,

That is what I did. The thing now is, when I set "enable=yes" in
/etc/asterisk/dnsmgr, Asterisk stops sending IAX POKE messages to the
remote peer (to keep the trunk up).
I've searched on the Internet but I couldn't find any documentation
about how "DNS update manager" works for Asterisk. Do you have any?

Ronaldo.

Jaswinder Singh wrote:
> In your no-ip client set it to update ip every 2 minutes or so . and
> /etc/asterisk/dnsmgr.conf  set refresh interval as 30-40 by defualt
> its 300 ( 5 minutes)
>
> On 10/06/07, Ronaldo Z. Afonso <[EMAIL PROTECTED]> wrote:
>> Hi Matt,
>>
>> Every time I do that, IAX stop sending the POKE messages (necessary for
>> trunk management).
>> Do you know what could be happening?
>>
>> Thanks.
>> Ronaldo.
>>
>> Matt wrote:
>> >
>> > *set "enable=yes" in the "[general]" section of
>> > /etc/asterisk/dnsmgr.conf*
>> >
>> >
>> >
>> 
>> >
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Re: [asterisk-users] No audio after Dial with G option

[Jaswinder Singh]

Remove Answer() and try .

On 12/06/07, Rosalinda Trevino Cadena <[EMAIL PROTECTED]> wrote:


 I'm using the Dial application in the extensions file with the G option
to execute an AGI script after the Dial (I need to track the call status) as
follows:

exten => _X.,3, Dial({DIAL_STRING},,G(_X.^4))
exten => _X.,4, Answer()
exten => _X.,5,AGI,agiScript.php

The problem is that testing between two internal phones (with two ATA) I
loose the audio when I include the G option in the Dial application, while
the audio is restored if I remove the G option, but that way I can't execute
the AGI script wile the call is up.


Any ideas on how to solve the problem or on what the cause might be?

Thanks,
Rosalinda

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Re: [asterisk-users] Qualify renders all SIP peers unreachable

[Jaswinder Singh]

What does "sip show peers" output ? Also set a timeout in millisec like
qualify=200 instead of qualify=yes

On 14/06/07, randulo <[EMAIL PROTECTED]> wrote:


I totally puzzled by this situation. I have asterisk 1.4.4 behind NAT.
All SIP peers are working properly to place or receive calls.
Any SIP peer or friend whether NATted or not will become UNREACHABLE
if qualify=yes.

I have identical peers on the other asterisk 1.2.16 production server.
In fact, two of the phones (linksys 941 and Polycom ip500) are using
one line for each asterisk. The 1.2 one works normally, the 1.4 does
not.

The sip confgs from "sip show settings" are identical on the two servers.

The sip.conf peer entries were moved over exactly.

Ports 5060 to 5065 are forwarded to the asterisk server.

Looking at sip debug, I notice a few differences:

REGISTER from phone:

"Authorization: Digest username="Poly", realm="asterisk",..."

does not show on the 1.4 server.

Trying (sent by *):

"Supported: replaces"

The Via lines are the same (internal ip addresses) on both servers,
but there is a "Sending to 192.168..." on the 1.2 message where there
is none on the 1.4.

What is "supported: replaces" ?

What config setting generates the "Authorization: Digest..." message ?
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Re: [asterisk-users] SIP Peering--call terminated prematurely

[Jaswinder Singh]

Please do not post same thing again and again . It wont help you get better
replies , Post you asterisk cli output while call is in progress and when it
disconnects prematurely .

On 18/06/07, Don Kelly <[EMAIL PROTECTED]> wrote:


I am attempting to establish SIP peering between Asterisk and an AltiGen
soft PBX. This is my first experience with SIP peering.

I can successfully make both inbound and outbound calls to/from a
softphone
on the AltiGen system (network access is provided by a PRI on the Asterisk
system), but they are disconnected unexpectedly.

The attachment is a redirect of the Asterisk CLI during a call that is
disconnected prematurely.

Here's what's in SIP.conf:

[altigen]
type=friend
username=altigen
secret=coolbeans
host=dynamic
deny=0.0.0.0/0.0.0.0
permit=10.0.2.150/255.255.255.255
qualify=yes
disallow=all
allow=ulaw
context=altigen-inbound
dtmfmode=rfc2833

The machines are a couple feet apart on a LAN through a 100MB switch.

I'd appreciate any help.

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax


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Re: [asterisk-users] Binding to multiple addresses

[Jaswinder Singh]

You can use bindaddr=0.0.0.0  to bind to all interfaces in sip.conf and
iax.conf .

On 23/06/07, Jordan Novak <[EMAIL PROTECTED]> wrote:


 I have a simliar problem as the port binding question.
I have a four port parelell processing NIC, I would like to team them
together. Can I do this in asterisk if they are not actually "teamed" in
hardware. I would be binding to several addresses simultaniously.

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Re: [asterisk-users] Use of ChanSpy

[Jaswinder Singh]

exten =>*76,1,Answer
exten => *76,2,Chanspy(|qb) ; q for quiet and b for only bridged calls
exten => *76,3,Hangup

Now you can spy on any call ,. All you need to do is press * again and again
to change calls . Like if 3 calls are going  then you can switch between
calls by pressing *  and # increases or decreases volume
This will spy on only sip calls:
.exten => *76,2,Chanspy(SIP/|qb)

for iax2:
.exten => *76,2,Chanspy(IAX2/|qb)

for a certain extension ( eg: sip extension 4455)
.exten => *76,2,Chanspy(SIP/4455|qb)

Hope this helps

On 24/06/07, Oscar Carriles <[EMAIL PROTECTED]> wrote:


 Maybe this helps



; spy on agent

exten => *7792,1,Playback(agent-newlocation)

exten => *7792,2,Read(EXT)

exten => *7792,3,Chanspy(Agent/${EXT}|q)

exten => *7792,4,Hangup



 ; spy on sip

exten => *7797,1,Playback(agent-newlocation)

exten => *7797,2,Read(EXT)

exten => *7797,3,Chanspy(SIP/${EXT}|q)

exten => *7797,4,Hangup



; spy on everybody

exten => _**779.,1,Chanspy(${EXTEN:5}|q)

exten => _**779.,2,Hangup



-Mensaje original-
*De:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *En nombre de *Carlos Garcia
Mujica
*Enviado el:* Jueves, 21 de Junio de 2007 04:17 p.m.
*Para:* asterisk-users@lists.digium.com
*Asunto:* [asterisk-users] Use of ChanSpy



How can I use the Asterisk command ChanSpy If I need to spy on a call?

I already added the function to the extensions.conf, and I get the beeps,
but then what do I do??? I don't understand the use of this function.


Best Regards

No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.472 / Virus Database: 269.9.6/865 - Release Date: 24/06/2007
08:33 a.m.

No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.472 / Virus Database: 269.9.6/865 - Release Date: 24/06/2007
08:33 a.m.

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Re: [asterisk-users] CDR Records "s" as dst

[Jaswinder Singh]

This is due to changes in cdr in asterisk 1.4.5 so in all outgoing
dial from macro it shows 's' in cdr . Could this be a bug ? I doubt if
it was intended to be that way .

On 25/06/07, Troy - Purple Oranges <[EMAIL PROTECTED]> wrote:
> I am using VoiceOne http://voiceone.it/ as my management interface.
>
> I am not 100% sure when it started, but my CDR is now full of "s" as
> the DST instead of the actual dialed number.
>
> As I understand it - it is because it is being recorded in the CDR
> while in a macro (as below).
>
> Is there any work around so that I can record the actual dialed number?
>
> [macro-dialout]
> exten = s,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1})
> exten = s,n,NoOp(CID_NAME  : ${CID_NAME})
> exten = s,n,NoOp(CID_NUMBER: ${CID_NUMBER})
> exten = s,n,NoOp(CID_CLIR  : ${CID_CLIR})
> exten = s,n,NoOp(TRUNK : ${TRUNK})
> exten = s,n,Set(CALLERID(name)=${CID_NAME})
> exten = s,n,Set(CALLERID(num)=${CID_NUMBER})
> exten = 
> s,n,Set(PRESENTATION=${IF($["${CID_CLIR}"="1"]?prohib_not_screened:allowed_not_screened)})
> exten = s,n,SetCallerPres(${PRESENTATION})
> exten = s,n,GotoIf(${ISNULL(${TRUNK})}?s-CONGESTION,1)
> exten = s,n,Dial(${TRUNK}/${ARG1}${TRUNKOPTIONS}||gTW)   ;Ring the interface
> exten = s,n,NoOp(DIALSTATUS = ${DIALSTATUS})
> exten = s,n,Goto(s-${DIALSTATUS},1)  ;Jump based on status
> (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
> exten = s-BUSY,1,Playtones(busy)
> exten = s-CONGESTION,1,Playtones(congestion)
> exten = _s-.,1,Goto(s-CONGESTION,1)  ;Treat anything else as no answer
>
> --
> Regards,
> Troy Kelly
> Director
> Purple Oranges Pty Ltd
> http://purpleoranges.com/
> --
> Brisbane (07) 3018 2840
> Fax (07)  3105 5987
> 
> Disclaimer - This email and any files transmitted with it are
> confidential and contain privileged or copyright information. You must
> not present this message to another party without gaining permission
> from the sender. If you are not the intended recipient you must not
> copy, distribute or use this email or the information contained in it
> for any purpose other than to notify us.
>
> Any views expressed in this message are those of the individual
> sender, except where the sender specifically states them to be the
> views of Purple Oranges Pty Ltd.
>
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Re: [asterisk-users] Asterisk + hinting presence + macro

[Jaswinder Singh]

It was due to changes in cdr in asterisk 1.4.5 previous version does not do
it .there is a fix on bugs.digium.com or you can wait till next release or
use asterisk 1.4.4

On 28/06/07, Rob Schall <[EMAIL PROTECTED]> wrote:


I currently have about 50 polycom 501 phones on my asterisk setup. The
dialplan is set to work with mysql (realtime), and all of the extensions
for the phones route through the same macro (stdexten). This all works
fine until I tried to set up notify status.

On voip-info, they say do something like...

,hint,SIP/
,1,Dial(SIP/)
blah blah blah

This functionality works fine. But what if you have a macro
s,hint,SIP/${ARG1}
s,1,Dial(SIP/${ARG1}

this adds a "s" hint which obviously doesn't work, instead of a hint for
 as it should.

Any ideas?

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Re: [asterisk-users] Asterisk + hinting presence + macro

[Jaswinder Singh]

Sorry i didnt read your mail properly . I thought your problem is with
cdr's. Here's link to cdr problem  :)

http://lists.digium.com/pipermail/asterisk-dev/2007-June/028085.html

see the next message for patch .

On 29/06/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:


Rob Schall wrote:
> Eric "ManxPower" Wieling wrote:
>> Rob Schall wrote:
>>
>>> I currently have about 50 polycom 501 phones on my asterisk setup.
>>> The dialplan is set to work with mysql (realtime), and all of the
>>> extensions for the phones route through the same macro (stdexten).
>>> This all works fine until I tried to set up notify status.
>>>
>>> On voip-info, they say do something like...
>>>
>>> ,hint,SIP/
>>> ,1,Dial(SIP/)
>>> blah blah blah
>>>
>>> This functionality works fine. But what if you have a macro
>>> s,hint,SIP/${ARG1}
>>> s,1,Dial(SIP/${ARG1}
>>>
>>> this adds a "s" hint which obviously doesn't work, instead of a hint
>>> for  as it should.
>>>
>>
>> Yes.  Put in the correct hint.  There is no reason that
>> ",hint,SIP/" would not work in a macro.
>>
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>>
> So, if I understand you correctly, my macro would look something vaguely
> like...
>
> [macro-stdexten]
> ${ARG1},hint,SIP/${ARG1}
> s,1,Dial(${ARG1})?
>
> This will work? My understand was that by going into a macro, you were
> going to be using the "s" extension. I'm not sure how that hint would
> get called if its not inside the s extension.

I have no idea, but as I understand it, Hints are separate from
extensions.

I guess you could do something like:

[macro-stdexten]
exten => s,1,Goto(${MACRO_EXTEN},1)

exten => _,hint,SIP/${ARG1}
exten => _,1,Dial(${ARG1})

I do this sort of thing in many of my macros that Dial somewhere.  I
seem to remember something about hints not working for pattern matching.
or working weirdly.

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Re: [asterisk-users] v1.4.x ready yet?

[Jaswinder Singh]

I jumped into asterisk 1.4 and its pretty stable .. i never got a core dump
but it did  halt while reloading a few times . I am back on asterisk 1.2 now
but i think asterisk 1.4 is stable .

On 29/06/07, Bruce Reeves <[EMAIL PROTECTED]> wrote:


While I have not jumped all my systems to 1.4, there were some that I have
moved to 1.4 and I have found it to be as stable as 1.2 was on those
machines.One of the systems is a 10 user office with Sangoma cards and
another is a 70+ user pure voip system. In both cases I have warning
messages about my dialplan usage of realtime and the fact that it will be
depreciated in the next release, but everything works as it should and the
upgrades.txt guided me through the changes to my dialplan. Hope that
helps.

On 6/29/07, shadowym <[EMAIL PROTECTED] > wrote:
>
>
>
> Hi All,
>
> Eagerly waiting for v1.4.x to mature a bit before getting serious about
> it.
> Is it ready for production yet?  If that's too general, where is it in
> terms
> of stability compared to where 1.2.x is now.  Anyone running it
> successfully
> in production environment and if so what sort of config do you have?
>
>
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--
Bruce Reeves
Nortex Networks
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Re: [asterisk-users] Question about dnsmgr

[Jaswinder Singh]

Are you sure calls were dropped with change in IP ?? I think it should let
current calls run and use new IP for new connections . However if
destination serv drops calls then it's a different story .

On 03/07/07, Henry J. Cobb <[EMAIL PROTECTED]> wrote:


Asterisk 1.4.5 full log:
[Jul  2 09:31:16] VERBOSE[2682] logger.c:   == Refreshing DNS lookups.
[Jul  2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net'
changed from 64.2.142.17 to 64.2.142.29
[Jul  2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed
Jitterbuf max 600 timeslots

And the calls are dropped.

I fixed this by turning off enable in dnsmgr.conf

My question is:

Do you attempt to move existing IAX connections when you see a DNS change
or do you leave the existing connections the fnord alone on their
current IP addresses and simply use the DNS change for new
connections?

-HJC


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Re: [asterisk-users] Google acquires Grand Central

[Jaswinder Singh]

Think about voicesense which will sense what you are talking and pop in a
*relevant* voice ad  to spice up conversation :P  .

On 03/07/07, Dean Collins <[EMAIL PROTECTED]> wrote:


 Ooops did Google just become a carrier :)
http://googleblog.blogspot.com/2007/07/all-aboard.html

I hear stocks crumbling worldwide as I type.


Cheers,
Dean



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Re: [asterisk-users] Upgrade Asterisk

[Jaswinder Singh]

Yes that is write order . libpri then zaptel then asterisk . Remember that
zaptel compilation is not required if you are using asterisk for  voip only
environment .But it's always good to install it before asterisk if you want
to use conferencing abilities of asterisk .

Regards,
Jaswinder Singh

On 05/07/07, Vidura Senadeera <[EMAIL PROTECTED]> wrote:



Hi,

Try first installing latest release of libpri, then zaptel

Try install asterisk after then. ope you will be able to compile it
without any probs.

--
Thanks & Regards,
Vidura Senadeera,
Network Engineer,
Debug Solutions
Sri Lanka.
Tel - +94114520036
Mobile - +9466596
Web - www.debug.lk

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Re: [asterisk-users] Asterisk console filtering and logging

[Jaswinder Singh]

This feature would be really great but i dont think asterisk supports it .
It either shows dialplan execution of all extensions when verbosity is
increased or of none when set to 0 . You can set verbose 0 and sip debug a
single peer but you cant enable dialplan execution viewing for single
extension/peer ( please correct me if i am wrong ).

Regards,
Jaswinder Singh.

On 05/07/07, Eugene Prokopiev <[EMAIL PROTECTED]> wrote:


Hi,

Is it possible to filter messages on asterisk console, which was started
with -, to see messages only for one extensions? By default there
are all messages for any extensions displayed so dialplan debuging is
very difficult.

Is it possible to log such console messages:

...
 -- Executing Set("SIP/10.0.0.1-0061f5d0", "CDR(userfield)=2422718")
 -- Executing Dial("SIP/10.0.0.1-0061f5d0", "SIP/708,25,tT")
...

to file. I can't find any suitable option in logger.conf

--
Thanks,
Eugene Prokopiev

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Re: [asterisk-users] Upgrade Asterisk

[Jaswinder Singh]

Yes just download new version of asterisk,zaptel,libpri  . "make install"
for all 3 ( first libpri , then zaptel, then asterisk ) . It is recommended
to stop asterisk b4r doing "make install" of new version . Do not do "make
samples" or it will overwrite you config's . After installing newer zaptel
do " rmmod ztdummy zaptel zttranscode" then modprobe 3 of them ( or a
restart of server will do ) . Now just start asterisk again and it will read
all the prior  config's you made as they are in /etc/asterisk . It's that
easy :) .

or just do "make install" for all 3 packages and restart server once ( it
will load new kernel modules after restart automatically and you dont need
to do that rmmod and modprobe stuff ) .

On 04/07/07, Christian Victor <[EMAIL PROTECTED]> wrote:


Hi!

Just ashort question - obviously I am too stupid too find the answer on
the net. :-)

I want to upgrade a running Asterisk 1.4.2 to 1.4.6 - what will I have
to do? Just install it over the existing version? Do I need to backup
the configuration? Will I need to reconfigure the source or will the new
version "import" my old settings? Will I need to update Zaptel and
Libpri too?

Argh - I installed like 50 asterisk systems but this one is the first
production machine with issues so heavy that I have to upgrade it.

Please point me to a update/upgrade howto etc. if available on the net.

Thanks a ton
Christian

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Re: [asterisk-users] Simple CDRs w/Asterisk/OpenSER.

[Jaswinder Singh]

Asterisk is poor with codec negotiation . It does not check if it can avoid
transcoding  by forcing codec available to both sides .. instead it will
read it's config file and will select first allowed codec that  is also
available on other device on each leg of call and happily transcode between
them .There was a patch on digium submitted by someone for asterisk 1.2.12
or so but it isnt updated from long time .  I am sure guys at digium are
aware about it and working on it . It's not  a bug  since asterisk is not a
sip proxy and tries to keep media path through it to offer its pbx features
but it would be a great feature nonetheless if implemented .

On 05/07/07, Alex Balashov <[EMAIL PROTECTED]> wrote:



Suggestions on how to use Asterisk to collect CDRs from a OpenSER-based
proxy / call routing setup?  I need to get simple CDRs;  not for detailed
settlement/rating, but just for reconciliation with an ultimate TDM
carrier just to make sure we only get billed for what we're actually
using.

I'd use the often-heralded approach of dumping a call from OpenSER into
Asterisk and having it bounce right back out toward the proxy by way of
REINVITEs.  I don't want the media running through Asterisk or Asterisk
being a limiting factor in that regard.

The problem is I don't have native G.729 support - we have no need for
it because neither the customer's network elements nor ours lack an
implementation of their own they can negotiate on just fine.  But
unfortunately Asterisk insists on natively homogenising the SDP from
both sides even if it subsequently removes itself from the media path!

So, I end up with situations where on the one side, I get, say:

Customer MGW --> OpenSER --> Asterisk - sends call as G.729.

Asterisk --> OpenSER --> Our MGW - our MGW prefers G.711a.

Now, if customer MGW <-> Our MGW were talking directly, as they do
when the deal is brokered through the OpenSER proxy, they would simply
negotiate upon what they agree.  But for some reason with Asterisk
this does not seem to be working as advertised;  we get lots of failed
calls if we pass them through Asterisk because one leg is one codec
and the other is another.  I am not sure how it arrives at that
conclusion despite the overlap of shared codecs (G.729 on both sides,
I would expect it to pass thru licence-free), and to be honest, I
don't particularly care if it's a bug or a feature, I just need it
not to introduce codec issues if I use it as a billing target.

Any help or insight would be greatly appreciated.

Thanks,

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] Which features are lost when canreinvite is turned on ?

[Jaswinder Singh]

If you manage to get everything working with canreinvite=yes ( i suppose u
figure out nat issues ) then you cant play music on hold , can't record
calls , and can't do most of pbx stuff asterisk is capable of .. but dont
worry asterisk doesnt disable all this features if canreinvite=on .. like if
you have call recording enabled in configuration and also have
canreinvite=yes then asterisk wont send reinvite's and media stream will
pass thorugh asterisk  . For  most of pbx  canreinvite should be kept off
unless  you have latency issues , or you are just connecting 2 pbx systems
and doing something like billing in between and not touching media stream .

On 09/07/07, Olivier <[EMAIL PROTECTED]> wrote:


You mean I'm heading to NAT issues ?
And what about Record-Route options ? Will it really help to be notified
of call endings ?


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Re: [asterisk-users] Best and easiest soft phone for my Dad..

[Jaswinder Singh]

Get portsip ( www.portsip.com  ) its realtively easy to configure ( just
push in user/password and server name at startup ) .. there might be NAT
issue so make sure you have nat=yes in ur asterisk's sip.conf for the peer
definition . If it still doesnt work then you need to find a iax phone like
zoiper ( http://www.zoiper.com/  previously idefisk ).

On 21/07/07, WipeOut <[EMAIL PROTECTED]> wrote:


Hi,

Here is the situation.. My Dad is working on contract in overseas.. He
has internet access in his hotel.. He wants to be able to talk to my Mum
but the calls are expensive..

I have an asterisk box setup for my business and it has a public IP
etc.. My Mum has access to a working phone extension on this box..

I got my Dad to install X-Lite but for some reason it won't register and
trying to talk him through working out whats wrong is proving to be
difficult.. Also I haven't used a softphone in years.. It could be the
NAT in the hotel, it could be a firewall or any number of things that
can cause these issues.. It could even be X-Lite or something running on
his PC..

So I am looking for a softphone thats really simple to setup and as
foolproof as possible..

If SIP is likely to be problematic to setup then I have no problem
getting him to use IAX but will need suggestions of which IAX softphone
to use and also how to configure it in the iax.conf (haven't done this
before)..

Any suggestions welcome and appreciated..

Thanks..

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Re: [asterisk-users] asterisk is not sip proxy

[Jaswinder Singh]

Asterisk is not a sip proxy but it *can* partly act as a sip proxy if
reinvites are enabled ( canreinvite=yes in sip.conf ) only then asterisk
connects 2 end points directly and does signalling between them .
Asterisk is a PBX now suppose u need to record all calls ..do conferencing
stuff  then rtp stream need to pass from asterisk (  openser cant do this
bcoz it just connects 2 endpoints  and only does signalling ) .. If you do
canreinvite=yes in sip.conf for both peers then asterisk does only
signalling ( also dial command should not have transfer parameters tT .. ) .
If both peers are behind NAT then asterisk reinvites may not work properly .

On 23/07/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:


Dear Edgar;

I am little bit confused, do u mean that asterisk does
not work in that way:

RTP (media) to be from the sournce to the destination
directly while signaling to be via asterisk?

So, what he parameter canreinvite is doing?

Regards,

ITS
Ip Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460




Hello Asteriskers,

I'm confused about why Asterisk is not a SIP proxy and
why exactly
this can affect the performance of a large Asterisk
system.

I know that Asterisk acts as a useragent endpoint, but
my doubt is why
exactly Asterisk could overload the call flow if the
RTP voice stream
goes from the caller to the called party.

Does someone know how many calls or pencentaje that
could handle a SER
or OpenSER in comparison with Asterisk?








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Re: [asterisk-users] SIP IP Trunk, between Asterisk and Softswitch

[Jaswinder Singh]

In your case it will send calls without registering to softswitch . Btw what
does your softswitch expects from asterisk ? like is it configured to
authenticate by username alone , user/pass or ip address ?? People here  can
help you better if you post that info .


On 24/07/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:


Dear List;

I am trying to create a link between Asterisk and My
softswitch, the link to be SIP Trunk.

I did the below configuration and I do not know if any
one can help me and advise me to have better
configuration to be sure that link is fine. But I do
not know how to determine the SIP username to be sent
for my softswitch as sometimes the softswitch need to
check it.

Also, does asterisk request to register on the
softswitch or it can send directly without
registeration? (Note: the trunk is SIP).

Please check the below configuration and advise me if
it is correct:

[aloonet]
type=peer
qualify=yes
host=193.111.196.240 ; IP Address of the softswitch
canreinvite=yes
context=outbound
disallow=all
allow=g723
nat=no

Is it OK? Will it register on my softswitch or will
send call directly without registeration on it?

Regards
Bilal





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Re: [asterisk-users] What is the best softphone work with Asterisk

[Jaswinder Singh]

Idefisk/zoiper softphone is for IAX2 and it works fine almost everytime .
However there is  more variety in sip softphones . I think zoiper is much
better than other iax2 softphones .

On 25/07/07, bilal ghayyad <[EMAIL PROTECTED]> wrote:


Hi All;

Thanks for all replies :) -

But that means, softphone in Asterisk is not that
good, I see all complains. Any advise?

Please Mr. "Time Bandit": What do u mean by "my IAX2"?
Is it your code or what?

Also Mr. "Rayan": I am noticing that you are advising
for SIP, what about IAX? Nothing suitable? If this is
the case, then where is the main advantage of IAX
protocol as specifically the IAX softphone does not
work fine?

Any help?
Regards
Bilal


I've had decent luck with PhonerLite, connecting via
SIP.  The
interface
is not the best, but I've been able to connect
reliably and make calls.

-Ryan

bilal ghayyad wrote:
> Hi List;
>
> I need to configure a softphone to be client and use
> it with Asterisk, which is the recommended one? Is
it
> iax2?
>
> Regards
> Bilal



  

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Re: [asterisk-users] What is the best softphone work with Asterisk

[Jaswinder Singh]

Idefisk is now renamed to zoiper . http://www.zoiper.com/ :)

On 26/07/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:


Am Mittwoch, den 25.07.2007, 12:13 -0700 schrieb bilal ghayyad:
> Hi BaharatSamaria;
>
> Thanks for your kindly email.
>
> Are (Xlite and phoner) IAX or SIP? From where I can
> download them (Xlite and phoner)?

I googled for "xlite". One of the first matches was a wiki page on
voip-info.org, which in turn linked me to the X-Lite manufacturer's
homepage. 
CounterPath's X-Lite 3.0 is the market's leading free SIP based
softphone available for download.
.

The first link in the google search list for "phoner" immediately led me
to the phoner homepage, 
- VoIP support for SIP connections
Phoner is freeware, so this program can be used and distributed without
any restrictions. Distribution has to be free of charge.


I think you will have no trouble to find the URIs yourself, probably
within about 30 seconds. In doubt you might consult
http://www.googleguide.com/ to learn about google.

Anselm


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Re: [asterisk-users] Lines Not being Hung UP Major

[Jaswinder Singh]

Btw are the phones behind NAT ?? Also you can try some softphone and make
sure that this problem is caused by snom phones or some other factors ..

On 25/07/07, Michael J. Liberatore <[EMAIL PROTECTED]>
wrote:


I thought it was the fios service but now I realize it's the snom 360!
It doesn't hang up random outgoing calls.  It seems like it only happens
on outbound calls from phones that have been updated to 6.5.12 or
6.5.10.  It didn't happen before, but I don't remember what version
firmware it was before, maybe 6.2.3 or so.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron Arts
Sent: Monday, July 16, 2007 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Lines Not being Hung UP Major

Do your SNOM phones sometimes use answer-after:0, and do they have
keyboard LEDs subscribed to their own extensions?
Do those people hangup calls by puttig down the handset instead of
pressing the X key?

We are seeing hanging channels in this particular case.

Ron


Michael J. Liberatore wrote:
> Hi all, i am having a major asterisk problem.  I think it started
> around
> 1.2.19 but could also be 1.2.18 zaptel or 6.5.10 snom 360.  basically
> we start getting busy signals, all our 4 line hunt group is busy, i
> then check the channels and there are open calls that were hung up
long ago.
> i thought it was a zap problem but then i saw the same problem with
> iax2 calls.  its becoming a huge issue because if i dont reboot
> asterisk several times a day, all our lines get filled up with dead
> calls.  I am now running 1.2.21.1 asterisk with the same problem.
Please help.
>
> Mike
>
>
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Re: [asterisk-users] Locking a device to a codec

[Jaswinder Singh]

in ur sip.conf under the device definition you can set it

for example device name is asterisk is pap2

[pap2]
username=pap2
secret=blabla
type=friend
disallow=all
allow=g729

Then asterisk will only use g729 for incoming as well as outgoing calls on
this device .

On 27/07/07, Matt <[EMAIL PROTECTED]> wrote:
>
> Right.. what I'm asking is:
>
> If I set my PAP2T to use G723 or G729 outgoing calls from that
> device go in that format.
> However, incoming calls to the device from asterisk are running at
> G711u.  The PBX will access any format G711u, G723, G729 or GSM.
> What do I need to do to make asterisk use the same codec back to the
> ATA as it is using to the PBX?
>
> On 7/27/07, dave cantera <[EMAIL PROTECTED]> wrote:
> >
> >  baji, mhoppes,
> >  remember, if you have Only the g729 codec allowed or if this is the
> only
> > allow= entry in the sip.conf file, callers requesting any other codec
> will
> > be rejected
> >  daveC
> >
> >
> >  Baji Panchumarti wrote:
> >  On 7/27/07, Matt <[EMAIL PROTECTED]> wrote:
> >
> >
> >  Can someone comfirm my logic here?
> >
> > If I want a phone to use G729 I can set it to use G729... do I
> > also need to set it in Asterisk? I'm thinking no... as long as
> > asterisk WILL do G729... if that's all the device accepts it should go
> > to that codec, yes?
> >
> >  (based on my understanding, take it for what it is worth)
> >
> >  if allow=all or allow=g729 is in your
> >  asterisk configuration (sip.conf / iax.conf ) then asterisk will
> >  stream packets in g729 (assuming you have any licesnses
> >  needed in place).
> >
> >  -baji.
> >
> > --
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> >
> >
> >
> >
> >  --
> > My wife's sister is in California.
> > I should buy her a Videophone2008!
> >
> > Truly, The Next Best Thing to Being There!
> > --
> >
> > WorldWideVideoPhones.com
> > 856.380.0894
> >
> >
> >
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Re: [asterisk-users] SIP "Max Channels" Setup

[Jaswinder Singh]

http://www.voip-info.org/wiki-Asterisk+config+sip.conf

* 
call-limit
* = number : Number of simultaneous calls through this user/peer

On 27/07/07, Nicholas Blasgen <[EMAIL PROTECTED]> wrote:
>
> I'm running Asterisk without FreePBX or any of the other managers.  I'm
> trying to figure out how to set the maximum number of channels allowed on a
> single line?  I'd just rather not have Asterisk try the line when I know
> I'll recieve a CONGESTION back from the SIP phone provider (ViaTalk in this
> case).  Is there a configuration option I can't find that sets the maximum
> number of connections a SIP channel can handle at a given moment?  I expect
> the line to be something simple, but I can't find it detailed on the Wiki.
>
> --
> /Nick
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Re: [asterisk-users] 1and1 dedicated servers have been down for a few hours .

[Jaswinder Singh]

Is this a web hosting forum or mailing list ?

On 31/07/07, Asterisk guy <[EMAIL PROTECTED]> wrote:
>
> 1and1 dedicated server's service  has  been down for a few hours  , unable
> to reach them by phone or email. do anyone know what is going on there ?
>
> Mario
>
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Re: [asterisk-users] Silly MeetMe() question.

[Jaswinder Singh]

Do you have proper version of zaptel installed corresponding to your
asterisk version ?

On 31/07/07, Knud Müller <[EMAIL PROTECTED]> wrote:
>
> Alex Balashov schrieb:
> > On Mon, 30 Jul 2007, Knud Müller wrote:
> >
> >> what does your modules directory contain? Can you find a file
> >> /usr/lib/asterisk/modules/app_meetme.so after make install?
> >
> >   No.  I know it needs to be compiled, but it is not being compiled no
> > matter what I seem to do in the way of arguments to ./configure,
> > installations of zaptel, etc.
> >
> Better have a loot at the apps directory, there is a Makefile that lists
> all apps to be compiled. app_meetme depends on a flag called
> WITHOUT_ZAPTEL.
> I have not tried to use meetme without zaptel, but its worth a try add
> meetme explicitly.
> > --
> > Alex Balashov
> > Evariste Systems
> > Web: http://www.evaristesys.com/
> > Tel: +1-678-954-0670
> > Direct : +1-678-954-0671
> > 
> >
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>
>
> --
> Knud A. Müller
> Geschäftsführer
> Tel.: 040/398053-11
> Fax: 040/398053-29
> e-Mail: [EMAIL PROTECTED]
>
> portrix.net GmbH
> Stresemannstr. 375
> 22761 Hamburg
> HRB 79850 (Amtsgericht Hamburg)
> Geschäftsführer: Knud Alex Müller, Henning Voss, Niclas Schroeder
>
> http://www.portrix.net
>
>
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Re: [asterisk-users] partial ChanSpy

[Jaswinder Singh]

Chanspy() app allows spying live channel but you will get 2 way voice  in it
. I dont think any other app allows to spy on one side of call .

On 03/08/07, nik600 <[EMAIL PROTECTED]> wrote:
>
> Hi
>
> is it possible to spy (not record, spy) partially on a channel?
>
> for exaple, i'd like to listen only the input or the output voice.
>
> is it possible?
> thanks
>
>
> --
> /*/
> nik600
> https://sourceforge.net/projects/ccmanager
> https://sourceforge.net/projects/nikstresser
>
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Re: [asterisk-users] iax2 registration being rejected

[Jaswinder Singh]

>
>
> Yes, since IAX2 only uses one port, this is correct.  Another thing to
> keep in mind is to set a low qualify value in Asterisk since some
> routers will tear down the connection pretty quickly.  The qualify acts
> as a keep-alive and prevents the router from closing the port and losing
> the map.
>
> Thanks,
> Steve



But if you set timeout lower than actual latency to peer .. it will result
in asterisk not sending any calls to peer at all so keeping it too low will
create  more problem  .. however peer will be able to make outgoing calls .
I think asterisk doesnt rely on qualify= parameter to keep connection open .
Main purpose of qualify option is to make sure peer is not lagged then
specified timeout period else call quality will be pathetic .. qualify=200
seems ok  . Btw i have never seen a device losing registration when qualify
value is set huge ( i keep  qualify = 2000 for a very dirty connection
sometimes :D  so that asterisk will show latency when i do "sip show peers"
and "iax2 show peers" in cli  )


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Re: [asterisk-users] CDR/MySQL basic config

[Jaswinder Singh]

sock=/tmp/mysql.sock

Is this path for socket correct ?
In some distro it is /var/lib/mysql/mysql.sock . Type "locate mysql.sock" in
shell .  Also remove  uncomment port=3306  if using socket to connect .

On 07/08/07, Alessandro Russo <[EMAIL PROTECTED]> wrote:
>
> Hi, try to login as asteriskcdruser to mysql
>
> 
> # mysql -u asteriskcdruser -p
> Enter password: password
> Welcome to the MySQL monitor.  Commands end with ; or \g.
> Your MySQL connection id is 12
> Server version: 5.0.32-Debian_7etch1-log Debian etch distribution
>
> Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
>
> mysql>
>
>
> 
> Can you login with asteriskcdruser?
> If you cannot login there are some problems with privileges or...I don't
> know :(
>
>
> On 8/7/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:
> >
> >  Hi Alessandro,
> >
> >
> >
> > Thanks for that.. I'm pretty sure about the user. I used Webmin to
> > confirm the user configs, but I ran your commands anyway:
> >
> >
> >
> >
> >
> > mysql> use mysql ;
> >
> > Reading table information for completion of table and column names
> >
> > You can turn off this feature to get a quicker startup with -A
> >
> >
> >
> > Database changed
> >
> > mysql> select Host from user where User = 'asteriskcdruser' ;
> >
> > +---+
> >
> > | Host  |
> >
> > +---+
> >
> > | localhost |
> >
> > +---+
> >
> > 1 row in set (0.00 sec)
> >
> >
> >
> > mysql> grant insert on asteriskcdrdb.* to [EMAIL PROTECTED] by 
> > 'asteriskcdruser';
> >
> > Query OK, 0 rows affected (0.00 sec)
> >
> >
> >
> > But I still get the failure:
> >
> >
> >
> > [Aug  7 15:14:10] ERROR[29103]: cdr_addon_mysql.c:436 my_load_module:
> > Failed to connect to mysql database asteriskcdrdb on localhost.
> >
> > cdr_addon_mysql.so => (MySQL CDR Backend)
> >
> > [Aug  7 15:14:10] ERROR[29103]: res_config_mysql.c:627 mysql_reconnect:
> > MySQL RealTime: Failed to connect database server  on  (err 2002). Check
> > debug for more info.
> >
> > [Aug  7 15:14:10] WARNING[29103]: res_config_mysql.c:474 load_module:
> > MySQL RealTime: Couldn't establish connection. Check debug.
> >
> > [Aug  7 15:14:10] NOTICE[29103]: config.c:1171
> > ast_config_engine_register: Registered Config Engine mysql
> >
> > MySQL RealTime driver loaded.
> >
> > res_config_mysql.so => (MySQL RealTime Configuration Driver)
> >
> >
> >
> > This box also das Cacti installed on it, which makes use of the MySql
> > server as well (and all is ok there).
> >
> >
> >
> >
> >
> > Adrian Marsh
> >
> >
> >   --
> >
> > *From:* [EMAIL PROTECTED] [mailto:
> > [EMAIL PROTECTED] *On Behalf Of *Alessandro Russo
> > *Sent:* 07 August 2007 14:13
> > *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> > *Subject:* Re: [asterisk-users] CDR/MySQL basic config
> >
> >
> >
> > Hi,
> > first step is correct
> >
> > Hmm.. This is what I get:
> >
> > [EMAIL PROTECTED] ~]# mysql -u root -p
> > Enter password:
> > Welcome to the MySQL monitor.  Commands end with ; or \g.
> > Your MySQL connection id is 187143 to server version: 4.1.20
> >
> > Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
> >
> >  You make an errore here : mysql> use asteriskcdrdb
> >
> > users' information are stored in mysql db
> >
> > mysql> use mysql;
> > Reading table information for completion of table and column names
> > You can turn off this feature to get a quicker startup with -A
> >
> > Database changed
> > mysql
> >
> > mysql> select Host from user where User = 'asteriskcdruser' ;
> > +---+
> > | Host  |
> > +---+
> > | localhost |
> > +---+
> > 1 row in set (0.00 sec)
> >
> > mysql>
> >
> > Are you sure that user 'asteriskcdruser' has the privileges to insert
> > record in DB "asteriskcdrdb"?
> > If not...allow 'asteriskcdruser' to insert record ^_^
> >
> > mysql> grant insert on asteriskcdrdb.* to [EMAIL PROTECTED] by 
> > 'asteriskcdruser';
> > mysql> exit
> >
> > Reload asterisk and try
> >
> >
> >  On 8/7/07, *Adrian Marsh* < [EMAIL PROTECTED]> wrote:
> >
> >
> > Hmm.. This is what I get:
> >
> > [EMAIL PROTECTED] ~]# mysql -u root -p
> > Enter password:
> > Welcome to the MySQL monitor.  Commands end with ; or \g.
> > Your MySQL connection id is 187143 to server version: 4.1.20
> >
> > Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
> >
> > mysql> use asteriskcdrdb ;
> > Reading table information for completion of table and column names
> > You can turn off this feature to get a quicker startup with -A
> >
> > Database changed
> > mysql> select Host from user where User = 'asteriskcdruser' ;
> > ERROR 1146 (42S02): Table 'asteriskcdrdb.user' doesn't exist
> > mysql>
> >
> >
> > Adrian Marsh
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [m

Re: [asterisk-users] Use of context=... in [default] section of sip.conf

[Jaswinder Singh]

When you make calls then context=xxx of the peer you are using ( your
extension ) will matter , the context=yyy line of your trunk wont matter .
If you dont specify a context= for  a peer then it is considered to be in
[default] context  .

On 07/08/07, Jared Smith <[EMAIL PROTECTED]> wrote:
>
> On Tue, 2007-08-07 at 11:18 -0400, Filipe Brandenburger wrote:
> > If I have [myprovider] section with context=something. When I do an
> > outgoing call by using Dial(SIP/myprovider/464646)", does context=...
> > affect anything? As I understand it, it only affects incoming calls, but
> > I might be wrong.
>
> That's correct.  The context is only there to tell Asterisk where in the
> dialplan to send *incoming* calls.
>
> > Another thing, the setting of context=... on [default] section will
> > affect all [provider] sections without context=..., right? What if I
> > don't specify any context on [default], what would be the default
> > context?
>
> My guess would be the [default] context, but I could be wrong.
>
> > What if there's no context or an invalid context on a section,
> > what would happen to incoming calls that match that section?
>
> The calls would most likely be rejected by Asterisk.
>
>
>
> --
> Jared Smith
> Community Relations Manager
> Digium, Inc.
>
>
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Re: [asterisk-users] asterisk wait for traling digits

[Jaswinder Singh]

This should be configured in phone system instead of asterisk :) .

On 08/08/2007, Michael Rice <[EMAIL PROTECTED]> wrote:
>
> This is part f the phones dial plan. Our aastra phones do the same
> thing. Most phones allow you to configure the dial plan on them.
>
> satish patel wrote:
> > i have only one single 16XX dialplan for reached to avaya system then
> > why i have to wait for more digit
> >
> > satish patel
> >
> > */Don Pobanz <[EMAIL PROTECTED]>/* wrote:
> >
> >  > satish patel said
> >  >
> >  > I have asterisk setup now what happend
> >  > when i dial 4 digit number my asterisk wait for few digit why
> >  > when i press # key it is dialing fast but without # wait for
> >  > few number is there any configuration for dialplan
> >
> > This part of the dial plan looks like it should dial without the
> wait.
> > Could there be another part of your dial plan that starts with '16'?
> If
> > not have you reloaded extenions.conf either by restarting asterisk
> or
> > doing an 'extensions reload'?
> >
> > Don Pobanz
> >
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> >
> >
> > 
> > Pinpoint customers
> > <
> http://portal.mxlogic.com/redir/?atTQSjhOUqenT3qtXTvhvp7ndw0SWt53ySAWRVvfcPeoujvLw1g0tfSdyqKNa_ek2f5J9RHO-r5rablxiIvgF-NIj5j9EVU8AGD1cojjjsqIGIs1Z9RGRqpAUgmy30RGxM7qECsd3rh0V-VK_nLt6WtQXTdTdXivNBgGnrFYq5O5mUm-wafBitegAhASHOVJNdwQsCQknD7TAm1P1JZAS2_id41FrSA_zaxkKTjUQdbFEwSA_zaxkQg6dBcQgeRyq89NQ-k29EwgAhBexKvxYYmfSk3q9J4SDtBxBwQszDC3vZCceKlB
> >who
> > are looking for what you sell.
> >
> >
> > 
> >
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>
> --
> Michael Rice
> Systems Administrator
> Office: 210-366-2500
> Ext.  : 231
> Direct: 210-293-6231
> McClelland and Hine, Inc.
>
>
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Re: [asterisk-users] les.net losing DID's

[Jaswinder Singh]

Please stop advertising your forums/services on every single chance u get on
users list .

On 08/08/07, Al Bochter <[EMAIL PROTECTED]> wrote:
>
>  That is why you need to start posting info about the providers at
>
> http://www.bochterservices.com/phpbb/
>
> so everyone knows
> This is a FREE SERVICE provided by Bochter Services and it is not going
> away any time soon.
> There will be more added by your request
>
> Best regards,
>
> Al Bochter
> http://www.BochterServices.com
>
> ---
> See what we are selling at auction
> http://www.epier.com/auctions.asp?bochterservices
> ---
> Take a look at our online store
> http://www.bochterservices.com/onlinestore/
> ---
> Join our forum. This is where you can talk about VOIP
> You can overview some providers others have used.
> http://bochterservices.com/phpbb/
> ---
>
>
>
> Stephen Bosch wrote:
>
> Mail list wrote:
>
>  Just got mail from them saying my NY DID will be deactivated in few days
> . Funny thing is their site is still showing orderable DID's of  same
> area code . Anybody else got this ?
>
>
> Wow. That is totally unacceptable.
>
> Are they going to give you the option of porting the DID?
>
> -Stephen-
>
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>
>
>
>
> 
> Inbound (clean). Database: 000764-2, 08/08/2007 - 8/8/2007 5:31:56 PM
>
>
>
>
>
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Re: [asterisk-users] pick sip channel whn two party talking

[Jaswinder Singh]

google for ASTERISK CMD CHANSPY and follow voip-info link in search results
.

On 08/08/07, satish patel <[EMAIL PROTECTED]> wrote:
>
> Dear all
>
>   i need this feature in asterisk whn 2 party calling that
> time i pickup call and listen conversation of that party spoofing like is it
> possible in asterisk
>
> Rgds
>
> satish patel
>
> --
> Choose the right car based on your needs. Check out Yahoo! Autos new Car
> Finder 
> tool.
>
>
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Re: [asterisk-users] CDR-CSV Processing

[Jaswinder Singh]

Enable mysql loggin of cdr's by installing asterisk-addons and use
asterisk-stat
http://www.areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54

On 13/08/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> On Mon, Aug 13, 2007 at 08:49:11AM -0500, Jeremy Mann wrote:
> > Does anyone have any tools to process CDR-CSV files into reports?
>
> Throw them into a near-by spreadheet.
>
> --
>Tzafrir Cohen
> icq#16849755jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] CDR billsec greater than duration

[Jaswinder Singh]

I made same thread few months ago and many people said that they dont have
such records in plain asterisk install ( no freepbx ) . I was also using
freepbx when i had  this problem . Heres mine :

mysql> select count(*) from cdr where billsec > duration;
+--+
| count(*) |
+--+
|  124 |
+--+

this is out of 1749216 cdr records .

I am also using freepbx btw . In all such cdr's duration is always 0 but
billsec varies .

On 15/08/07, Edoardo Serra <[EMAIL PROTECTED]> wrote:
>
> Hi all,
> I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1
>
> Doing a select in the CDR table I noticed there are some calls with
> billsec greater than duration, duration is always 0 in those calls.
>
> How can this happens ? Am I missing something ?
>
> Tnx in advance
>
> Regards
>
> Edoardo Serra
> WeBRainstorm S.r.l.
>
>
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Re: [asterisk-users] CDR billsec greater than duration

[Jaswinder Singh]

Yes it maybe a hung channel problem .. but question is no matter how much
billsec ... should duration be more than that ?

On 16/08/07, Anthony Francis <[EMAIL PROTECTED]> wrote:
>
> You are a victim of hung channels, just write a script that corrects this.
>
> Anthony
>
> Mail list wrote:
> > The destination numbers are valid in almost all cases . But i do think
> > it might be when someone is on call and on client side internet
> > connection  goes off .. I am really not sure about this one but i just
> > saw that maximum such records are from one of my customer who has a
> > very bad connection .
> >
> > On 16/08/07, *Edoardo Serra* <[EMAIL PROTECTED]
> > <mailto:[EMAIL PROTECTED]>> wrote:
> >
> > I noticed that fpbx calls ResetCDR on call hangup (don't know why
> this
> > choice)
> >
> > Could it be related to that ??
> >
> > Tnx
> >
> > E.
> >
> > Jaswinder Singh ha scritto:
> > > I made same thread few months ago and many people said that they
> > dont
> > > have such records in plain asterisk install ( no freepbx ) . I
> > was also
> > > using freepbx when i had  this problem . Heres mine :
> > >
> > > mysql> select count(*) from cdr where billsec > duration;
> > > +--+
> > > | count(*) |
> > > +--+
> > > |  124 |
> > > +--+
> > >
> > > this is out of 1749216 cdr records .
> > >
> > > I am also using freepbx btw . In all such cdr's duration is
> > always 0 but
> > > billsec varies .
> > >
> > > On 15/08/07, *Edoardo Serra* <[EMAIL PROTECTED]
> > <mailto:[EMAIL PROTECTED]>
> > > <mailto:[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>> wrote:
> > >
> > > Hi all,
> > > I have a strange situation on a Asterisk 1.2.17 with
> > FreePBX
> > > 2.2.1
> > >
> > > Doing a select in the CDR table I noticed there are some
> > calls with
> > > billsec greater than duration, duration is always 0 in those
> > calls.
> > >
> > > How can this happens ? Am I missing something ?
> > >
> > > Tnx in advance
> > >
> > > Regards
> > >
> > > Edoardo Serra
> > > WeBRainstorm S.r.l.
> > >
> > >
> > > ___
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> > >
> > >
> > >
> > >
> >
> 
> >
> > >
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Re: [asterisk-users] asterisk multiport

[Jaswinder Singh]

What i actually do is make asterisk listen on some other port like 5097 and
redirect port 5060 to it with iptables  like this
/sbin/iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5060 -j DNAT --to
YOURIPHERE:5097

This works very well . If i make asterisk listen on 5060 and redirect say
5097 to 5060 i had lot of problems with firewalled systems ( blocked 5060 by
isp ) . Also on blocked end its recommended to use some softphone like xlite
which  completely allows you to set custom ports on machine itself to
listen, taking 5060 completely out of picture .


On 17/08/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
> Steven wrote:
> > I am curious.
> >
> > Why would one need to do this?
> >
> > If a phone connect to 5060 from another port number, asterisk happily
> works, so why use multiple port on asterisk?
> >
>
> I cannot see the thread history but from the context, I would say
> because many ISPs block 5060, 25, and others.
>
> Thanks,
> Steve Totaro
>
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Re: [asterisk-users] A102d sangoma's card and ztdummy

[Jaswinder Singh]

Sin you have sangoma card , it will act as timer . You need to install
meetme ( app_conference is not very stable last time i read ) .

On 01/09/07, fateme fatah <[EMAIL PROTECTED]> wrote:
>
> Hi:
> I want to have conference call service and I use A102d sangoma's card.Do I
> should install ztdummy or app-conference?
> Best regards.
>
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Re: [asterisk-users] Configure extension by software

[Jaswinder Singh]

You can use asterisk realtime which can read sip config from database (
mysql/pgsql) . Your application can just write info to database and asterisk
will read it and make peers . You can also include a custom config file
within sip.conf and make your application write peer settings to  that file
and reload asterisk by using asterisk management interface .

On 08/09/2007, phananhvu <[EMAIL PROTECTED]> wrote:
>
> Before an IP Phones can be registered to an Asterisk server, the extension
> for it must be configured in Asterisk. Usually, Asterisk adminintor must add
> the extension by hand. Is there any library, API to do this by software???
>
> For example, i want to develope a software that add new extensions to
> Asterisk system, sothat, any IP Phones can use that extensions to establish
> a call.
>
>
> I'm digging on Asterisk-Java but this library seems not support this.
>
> Anybody has dealed with this before ??
>
>
> Phan Anh Vu
> DT12.K49.HUT
> RDLab ( C9.410 ) HUT
>
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Re: [asterisk-users] Difference in show channels

[Jaswinder Singh]

'show channels' shows only running calls  while 'sip show channels' shows
all running sip sessions including phones trying to register .

On 09/09/2007, ram <[EMAIL PROTECTED]> wrote:
>
> Hi all
>
> what is the difference between
>
> show channels
>
> sip show channles
>
> i see the difference in both
>
> show channels show me 30 channels
>
> sip show channels shows me 221 channels
>
> any description on this
>
> ram
>
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Re: [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

[Jaswinder Singh]

I prefer centos , debian/ubuntu are also a good option . It just depends on
which distribution you are comfortable with . We also have asterisk running
very stable on slackware .

On 12/09/2007, Gordon Henderson <[EMAIL PROTECTED]> wrote:
>
> On Wed, 12 Sep 2007, Euler Pereira wrote:
>
> > Hey all!
> >
> >I'm newbie in the Asterisk World but old in other telephony systems
> like
> > Lucent/Avaya, Sopho, Siemens and Linux/Unix system.
> >
> >I'm in doubt, as based system, should I install Fedora, Debian,
> > Slackware, FreeBSD our Sun Solaris? Which is more robust for a small
> > Asterisk system, about 8 extensions, 4 hardphone and 4 softphone?
>
> Which of Fedora, Debian or Slackware do you know best?
>
> I use Debian, but that's because it's the one I know best.
>
> Gordon
>
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Re: [asterisk-users] errors messages in asterisk CLI

[Jaswinder Singh]

Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql:
Unknown connection error:
(2006) MySQL server has gone away

This part is more like  mistake in /etc/asterisk/cdr_mysql.conf . Check it
once and relaod asterisk , then you can type "cdr mysql status" in cli to
check if it connects to mysql properly .




Sep 21 20:31:49 ERROR[3774]: cdr_csv.c:237 csv_log: Unable to re-open master
file
/var/log/asterisk//cdr-csv//Master.csv : Permission denied

Permission error try chowning the directory to user  which asterisk runs on
OR chmod 777 /var/log/asterisk/* -R

Sep 21 20:31:49 ERROR[3774]: cdr_custom.c:127 custom_log: Unable to re-open
master file
/var/log/asterisk/cdr-custom/Master.csv : Permission denied

Same permission error .


On 22/09/2007, Jody Gugelhupf <[EMAIL PROTECTED]> wrote:
>
> hi ppl, i have a problem, i get these messages in the asterisk CLI:
>
> Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql:
> Unknown connection error:
> (2006) MySQL server has gone away
>
> Sep 21 20:31:49 ERROR[3774]: cdr_csv.c:237 csv_log: Unable to re-open
> master file
> /var/log/asterisk//cdr-csv//Master.csv : Permission denied
>
> Sep 21 20:31:49 ERROR[3774]: cdr_custom.c:127 custom_log: Unable to
> re-open master file
> /var/log/asterisk/cdr-custom/Master.csv : Permission denied
>
> how can i fix these errors?
> here some info about my system:
>
> debian etch 4.0
> kernel:
> 2.6.18-4-686
> Asterisk 1.2.13
> VoiceOne version is v. 0.5.0 using plugin subsystem v. 0.4pre3
> mysql Ver 14.12 Distrib 5.0.32, for pc-linux-gnu (i486) using readline 5.2
> PHP 4.4.4-8+etch4 (cli) (built: Jun 30 2007 21:02:54)
> using grandstream(handytone) 486 as sip device, no other devices or PSTN
> connected, only using
> sip/voip providers
> behind router/NAT
>
> thx in advance :)
> jody :)
>
>
>   
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> favourite sites. Download it now at
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Re: [asterisk-users] prepaid application recommendation

[Jaswinder Singh]

A2billing is very versatile and good solution for asterisk prepaid/postpaid
billing .

On 22/09/2007, Apa Minerala <[EMAIL PROTECTED]> wrote:
>
> You should make sure you know how to install it yourself.
>
> And you should also test it very very VERY carefully.
>
> I can't underline very enough.
>
> And if ever you ask for service, get a real company, with a real person
> behind the desk, who is doing only this.
>
> I have had my sad story with the A2Billing people.
>
> Tudor
>
> *Sarfaraz Chougule <[EMAIL PROTECTED]>* wrote:
>
> I would recomend using Areski's billing solution :
> http://www.areski.net/a2billing
>
>
>
> On 9/21/07, Rilawich Ango <[EMAIL PROTECTED]> wrote:
> >
> > Hi all,
> > I am looking for a prepaid application.  I found that there are many
> > applications in the page
> > http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications.
> > Anyone recommendation among them?
> > ango
> >
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>
>
> --
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> **
>Sarfaraz Chougule
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Re: [asterisk-users] SIP and Firewall

[Jaswinder Singh]

Here you go http://www.voip-info.org/wiki/view/Asterisk+firewall+rules  .
You can also set your rtp.conf properly and open very few rtp ports instead
of all 1-2 udp ports .

On 22/09/2007, Guenther Sohler <[EMAIL PROTECTED]> wrote:
>
> Hallo,
>
> I'd like to correctly set up my firewall in my system for udp and asterisk
>
> I have got a server, which has got one static ip adress to the internet.
> Asterisks is running on this server.
> It registers at sipgate.at and mujtelefon.com
> The Server also does nat to the my intranet, where my pc and my hardware
> sip
> phone sits. The Hardware sip phone registers to asterisk on my server
> from its intranet ip adress. Everything works fine.
>
> The question is just: How to code good stateful firewall rules with
> iptables
> and netfilter_sip ?
> What would be apropriate to my system ?
>
> rds
>
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Re: [asterisk-users] Multiple Home system with SIP

[Jaswinder Singh]

since asterisk is only using operating system's routing ability , you can
always set static routes using route command in linux .

On 26/09/2007, Jeremy Mann <[EMAIL PROTECTED]> wrote:
>
> Why did you waste time with this reply?  You do realize some users don't
> have control over their Exchange servers, and asinine footers are placed
> into an email without their intervention or control right?
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] On Behalf Of Benny Amorsen
> Sent: Tuesday, September 25, 2007 1:55 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Multiple Home system with SIP
>
> > "JM" == Jeremy Mann <[EMAIL PROTECTED]> writes:
>
> I would have answered, but I was prohibited from quoting properly:
>
> JM> If you are the intended recipient, further disclosures are
> JM> prohibited without proper authorization.
>
>
> /Benny
>
>
> This e-mail, facsimile, or letter and any files or attachments transmitted
> with it contains information that is confidential and privileged. This
> information is intended only for the use of the individual(s) and
> entity(ies) to whom it is addressed. If you are the intended recipient,
> further disclosures are prohibited without proper authorization. If you are
> not the intended recipient, any disclosure, copying, printing, or use of
> this information is strictly prohibited and possibly a violation of federal
> or state law and regulations. If you have received this information in
> error, please notify Texas Health Management Group immediately at
> 1-817-310-4999. Texas Health Management Group, its subsidiaries, and
> affiliates hereby claim all applicable privileges related to this
> information.
>
> --
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> dangerous content by MailScanner, and is
> believed to be clean.
>
>
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Re: [asterisk-users] echo problems

[Jaswinder Singh]

Also many people using softphone turn's on mic boost in windows xp which
also makes echo if it is set to very loud .

On 30/09/2007, Philipp Kempgen <[EMAIL PROTECTED]> wrote:
>
> http://linux.sgms-centre.com/misc/netiquette.php#threading
> http://linux.sgms-centre.com/misc/netiquette.php#toppost
> SCNR
>
> Regards,
>   Philipp Kempgen
>
> --
> amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
> Let's use IT to solve problems and not to create new ones.
>   Asterisk? -> http://www.das-asterisk-buch.de
>   My pick of the month: rfc 2822 3.6.5
>
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Re: [asterisk-users] Injecting a sound file into a bridged call

[Jaswinder Singh]

See chanspy in asterisk 1.4 , it also has a whisper mode and you can talk to
one party http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy . But i
dont know  how to play  a recorded file in it .

On 08/10/2007, Girts Graudins <[EMAIL PROTECTED]> wrote:
>
>  Hello everyone,
>
>
>
> I'm looking for a way to play a sound file to an already established
> bridged call.  It is meant for one party, but it's ok if both parties would
> hear it.  Ideally, I'd like to be able to trigger this from the Management
> Interface with something like:
>
>
>
> Action: Playback
>
> File: tt-weasels
>
> Channel: Zap/nn
>
>
>
> However, I haven't seen anything like that being available, so I'm looking
> for other suggestions.
>
>
>
> The critical pieces are as follows:
>
> 1)  I need to be able to initiate this as an outside event/command;
> like I said, MI would be ideal;
>
> 2)  I've seen "whisper"-type of functionality associated with meetme
> rooms, but I'd rather not set up a dynamic meetme room for each call I'm
> bridging;
>
> 3)  Obviously there's Playback() and Background() available in the
> dialplan, but I need to be able to trigger the sound at will after the
> call's already been established.
>
>
>
> This sounds like a simple thing to wish for, yet I don't see a ready
> answer.  Any tips would be appreciated.
>
>
>
> TIA,
>
>
>
>
>
> Girts
>
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Re: [asterisk-users] asterisk.conf and it's impact on CLI

[Jaswinder Singh]

astrundir => /var/run
 Change this to astrundir => /var/run/asterisk  on 1.4 server and chmod
/var/run/asterisk to 777 . make sure u create that directory as well .

On 20/10/2007, Al lists <[EMAIL PROTECTED]> wrote:
>
> this message is basically tells you asterisk is not running.
> can you check and see if asterisk is running and present in memory?
> something like
> ps -ef | grep asterisk
>
>
> On 10/20/07, Dominic Son <[EMAIL PROTECTED]> wrote:
>
> > I was previous using Asterisk 1.2.9.1  and decided to get some real
> > servers outside of my house. It was time for Asterisk 1.4.4.
> > I figured since all the conf files were in /etc/asterisk form the old
> > box, i'd just copy tha directory over to the new server. My SIP DID AGI
> > stuff worked, except running 'asterisk -r' doesn't. It tells me
> >
> > ' Unable to connect to remote asterisk (does
> > /var/run/asterisk/asterisk.ctl exist?)'
> >
> > Basically, the difference between 'asterisk.conf' file is as follows:
> >
> > v 1.2.9 (installed through trixbox)
> > astrundir => /var/run/asterisk
> >
> > v 1.4.4
> > astrundir => /var/run
> >
> > So in my new servers, if i keep it as '/var/run/asterisk, my DID phone
> > will work with stanaphone (in which i'm crapping in my pants if they'll
> > exist cause they never return emails). Though CLI won't work.
> >
> > if i do '/var/run', my DID won't work, but CLI will...
> >
> > I've tried just coping over the extensions_additional.conf and
> > sip_additional.conf files from my old setup to my new one, and that didn't
> > work. Maybe I should just install my previous version. Are there QoS
> > differences though? I'd rather not regress if that were the case.
> >
> >
> > --
> > Anything else, let me know.
> >
> > - Dominic
> >
> >
> > "It is not the force of a stroke that makes fine art"
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Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite

[Jaswinder Singh]

Latest version of X-lite does not have GSM codec . Downgrde your versiona dn
you will get  gsm codec . I read on their forums that next version will
again be including GSM codec .

On 03/11/2007, Julio Tejera <[EMAIL PROTECTED]> wrote:
>
> Latest version of X-Lite does not
> support GSM codecs any more
>
> It could be a good idea that you post
> on the rigth place not here :o)
>
> jat
>
> - Original Message -
> From: "Alejandro Cabrera Obed" <[EMAIL PROTECTED]>
> To: "asterisk Users Mailing List" 
> Sent: Friday, November 02, 2007 2:05 PM
> Subject: Re: [asterisk-users] Off-Topic: add GSM codec to X-Lite
>
>
> > SIP wrote:
> >> Alejandro Cabrera Obed wrote:
> >>> Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip
> server
> >>> connected to Twinkle and X-Lite clients. I have to use the GSM codec
> for
> >>> all of my clients, and it was set up in the sip.conf specifically in
> >>> "allow=gsm" line.
> >>>
> >>> Twinkle has GSM codec built in, but when I open X-Lite audio codecs
> >>> settings I can't see the GSM codec, being that the official web site
> and
> >>> the PDF manual  of X-Lite 3.0 say it has GSM builtin support.
> >>>
> >>> Do you know what's the matter with X-Lite and GSM ??? Can I add it ???
> >>>
> >>> Really thanks
> >>>
> >>> Alejandro
> >>>
> >>>
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> >>>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>
> >> It lists GSM on my audio codec settings. Perhaps there's something
> >> wrong with your install? Try disabling the Zero Touch bandwidth
> >> detection. It has, in the past, interfered with my selection of codecs.
> >>
> >> N.
> > Thanks for your support...I've uninstalled my X-Lite 3.0 softphone and
> > after that I've downloaded the X-Lite 3.0 again from the official web
> > site. But when I go to audio codecs settings, the GSM codec is not
> > present. I disable the "zero touch bandwith detection" and restart the
> > softphone, but the GSM codec is not present at all.
> >
> > Any idea ???
> >
> > Thanks
> >
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Re: [asterisk-users] Asterisk unable to register to tnet.it

[Jaswinder Singh]

Check dns server entries in asterisk box . /etc/resolv.conf . Put
opendns servers ip there just to test . opendns ip's are
208.67.220.220 and 208.67.222.222

On Tue, Jul 15, 2008 at 2:19 PM, map <[EMAIL PROTECTED]> wrote:
> Hi Giorgio,
>
> RE my point 2:
> You should test a sip client, whatever you want, on your linux/asterisk box
> just to double check that this box works fine.
> If you are abel to connect with a sip client from tour asterisk box we will
> be sure that the network configuration is ok.
> You have no natt but maybe your routing table is not correct :-)
>
> Do you already test to just ping to tnet.it port 5060 ?
>
>
> Marino
>
> On Tue, Jul 15, 2008 at 10:27 AM, Giorgio Incantalupo
> <[EMAIL PROTECTED]> wrote:
>>
>> Hi Marino,
>>
>> 1) yes I can connect using the account
>> 2) no, I'm running zoiper on a different machine. I'm using an Asterisk
>> server which is not behind nat as for the machine zoiper is runnin' on.
>> The Asterisk server is directly connected to internet, I wanted to avoid
>> nat problems, that's why.
>> Moreover I tried to create a simpler account on my zoiper using
>> username, password and domain name only and it works even without
>> setting  the sip proxy.
>> I changed the Asterisk server too: now I'm using a test one where I can
>> ping tnet.it from... but nothing changes.
>> I'm using this string:
>> register => 0442410280:provapolika:[EMAIL PROTECTED]/0442410280
>> I changed it in many other forms following the wiki pages but nothing.
>> I see sip packets are sent to tnet.it (I set up sip debug) but I always
>> get this message:
>>
>> Jul 15 10:06:39 NOTICE[3281]: chan_sip.c:5495 sip_reg_timeout:--
>> Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #1)
>>
>> I wonder why I had no problems with the other provider we are using
>> while tnet.it is making me get crazy
>>
>> Thank you.
>>
>> Giorgio
>>
>>
>> map wrote:
>> > Hi Giorgio,
>> >
>> > Just to recap:
>> > 1) you are able to connect to tnet.it  by using the
>> > same account of your asterisk box. There is no issue related to your
>> > account.
>> > 2) Could you please confirm that you are running zoiper from the same
>> > box used by asterisk? If yes we can exclude some generic network issues.
>> >
>> >
>> > From your previous email :
>> > ...
>> > Activating "sip debug" shows the register packets but nothing in return.
>> > ...
>> >
>> > I think that this is a network related issue, but you have to solve it
>> > by using a Asterisk config file.
>> >
>> > Unfortunately I think that the faster way to solve your problem is
>> > trying to understand if sip messages are correctly sent to tnet.
>> > I strongly suggest to use http://www.wireshark.org/ previoulsly named
>> > Ethereal in order to check sip messages.
>> > I have to sniff both asterisk and zoiper sip messages.
>> > I know that this can be tricky but this can help you to understand
>> > what is wrong in sip messages.
>> >
>> > Please let me know if you need more detail.
>> >
>> >
>> > Marino
>> >
>> > On Tue, Jul 15, 2008 at 9:31 AM, Giorgio Incantalupo
>> > <[EMAIL PROTECTED] >
>> > wrote:
>> >
>> > Hi Marino,
>> >
>> > I tried to connect zoiper directly to the provider with the same
>> > account
>> > parameters I'm using with Asterisk. Zoiper connects without
>> > problems. It
>> > is true tnet.it  is not resolvable but I can use
>> > the proxy URL
>> > sip.tnet.it  which seems to work with Zoiper
>> > but not with Asterisk. I'm
>> > trying to understand where is the problem. I thought I had to
>> > specify
>> > the outboundproxy parameter in the general section of sip.conf to
>> > make
>> > Asterisk correctly work but it seems that's not enough.
>> >
>> >
>> > Thank you.
>> >
>> > Giorgio
>> >
>> >
>> > map wrote:
>> > > Hi Giorgio,
>> > >
>> > > From your email seems clear that your Asterisk box can not resolve
>> > > tnet.it   and SIP register
>> > messages are not replied.
>> > > I suggested to check if your Asterisk box is really sending SIP
>> > > messages, you can use a net sniffer.
>> > > Did you alerady used different sip client with the same sip
>> > account of
>> > > your Asterisk box?
>> > >
>> > > Did you use zoiper from the same box?
>> > >
>> > > Marino
>> > >
>> > > p.s.
>> > > Are you Italian?
>> > >
>> > >
>> > > On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo
>> > > <[EMAIL PROTECTED]
>> > 
>> > > > >>
>> > > wrote:
>> > >
>> > > Hi Marino,
>> > > Asterisk gives a timeout on registration and a "no such
>> > host" because
>> > > cannot resolve tnet.it   but
>> > that server address is
>> >  

Re: [asterisk-users] DID number

[Jaswinder Singh]

[442033553]
user=442033553
type=pusers
secret=1234
host=dynamic
context=users
nat=yes


make it context=stations , i am assuming this is how your DID provider
is sending u calls ?

Let us know if your DID provider is just sending calls to your ip
address or you are registering asterisk server with the, . Keep
context=stations in extensions.conf  global section .

On Thu, Sep 4, 2008 at 2:41 AM, Igor Hernandez <[EMAIL PROTECTED]> wrote:
> Hey,
>
> Did you reload asterisk after changing the extensions.conf?
>
> Also, if you try it with "sip set debug" on the console what do you see?
>
>
> michel freiha wrote:
>> Hello Air,
>>
>> I did what you asked for but I got the following error:
>>
>> extensions.conf:
>>
>> [stations]
>> exten => 442033553,1,Answer
>> exten => 442033553,n,Playback(demo-nogo)
>>
>> Error message:
>> [Sep  3 20:43:02] NOTICE[14092]: chan_sip.c:14035 handle_request_invite:
>> Call from '' to extension '442033553' rejected because extension not found.
>> Regards
>> On Wed, Sep 3, 2008 at 11:36 PM, Igor Hernandez <[EMAIL PROTECTED]
>> > wrote:
>>
>> michel freiha wrote:
>> > Hi All,
>> > I bought a DID number from VOxbone...this number could be dialed from
>> > any PSTN line and could be forwarded to any SIP server like asterisk
>> > server...Now I need to forward this number to my asterisk server
>> so when
>> > a customer dial this number from his GSM or Land line PSTN number the
>> > call will be forwarde to my asterisk server and I need to play a wav
>> > file for example..
>> > Can you please give me some tips about how to accomplish this task?
>> >
>> > Regards
>> >
>> >
>> >
>> 
>> >
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>>
>> Hello,
>>
>> I have never used that provider but usually either the provider knows
>> your switch's ip and routes the did traffic to it or you have asterisk
>> register with the provider so that it knows where to route the calls.
>>
>> Once thats done you can do something like
>>
>> exten => XX,1,Answer
>> exten => XX,n,Playback(file)
>>
>> Where the x's are the number that you see coming in from your provider.
>> If you're routed all your dids from what looks like one
>> number(callcentric does this) then you might need to use the sip header
>> to route your did to the particular extension you want. You shouldn't
>> have to bother with this if you only have one did.
>>
>>
>> Regards,
>>
>> --
>> Igor Hernandez
>> Escape Communications
>> http://www.escapetel.com 
>>
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>>
>>
>>
>> 
>>
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>
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Re: [asterisk-users] A request for your input.

[Jaswinder Singh]

ITS Open source related .

On 22/03/07, Bill Hackensack <[EMAIL PROTECTED]> wrote:


On 3/22/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
>
> Hello
>
>
> P.S The program that I am using is open source, of course
> ( www.phpsurveyor.org)!


What part of the survey is running Asterisk?



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Re: [asterisk-users] Outbound SIP call from asterisk extension

[Jaswinder Singh]

[outgoing]
exten => 1234,1,Dial(SIP/[EMAIL PROTECTED])


Whats the dialplan number to ring to userA on server (ONDO) ?
if u know that try
exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]
)

since ur sip.conf has [*192.xxx.xxx.xxx*-out].
i am not sure why you use Sip/test to call to userA . Is that how server
ONDO is configured ?
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Re: [asterisk-users] Failure acknowledgement time

[Jaswinder Singh]

You can use qualify=(time in ms) option in sip.conf but its phone's
configuration that should register to asterisk everytime its reconnected .

On 26/03/07, cimsi <[EMAIL PROTECTED]> wrote:


Hi,
I've noticed that if I disconnect or reconnect a phone from the net,
Asterisk take long time to realize that (even more then 10 minutes). Is
there a way to reduce this time, working on the configuration files?
Thank you.

silvia



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Re: [asterisk-users] error in FreePBX

[Jaswinder Singh]

disable voicemail for  that extension .. apply settings .. re-enable
voicemail .. re-apply settings . this helped me once before.
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Re: [asterisk-users] Announcement: Asterisk Service Provider Edition v1.0 Beta

[Jaswinder Singh]

Wow i need a tftp client to download it now .
Nice April  1 joke  :P .
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Re: [asterisk-users] Make an iso image or a kickstart-Really its too urgent

[Jaswinder Singh]

Why not use a asterisk specific live cd distribution like www.astlinux.org
? It is also installable on usb . You can copy your whole dialplan and
settings ( all files in /etc/asterisk ) on a pendrive .

On 25/04/07, Khaled Chehab <[EMAIL PROTECTED]> wrote:


 Dears  its too urgent

Can anyone guide me ……

I want to put  my asterisk system  on an iso image like trixbox ,or how to
make a.



how can I do that ,I am using centos 4.4 final







Regards






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Re: [asterisk-users] Calls in ulaw, not gsm as desired

[Jaswinder Singh]

Try ilbc if the phone supports (free) or g729  ( better but your asterisk
will need license if you want to transcode calls from g729 to other codecs
or want to record calls ) .  Also check your phones config if its support
multiple codecs . .

On 02/05/07, Rob Schall <[EMAIL PROTECTED]> wrote:


 So I reloaded things and had just gsm set for 2 of my polycom 501 phones.
However, the logs say "No codec found", which leads me to believe that
polycom 501 phones can't use gsm. Does anyone have something like this
working? If not gsm, is there a better option with these phones over a low
bandwidth situation?

Rob

Ed Nuñez wrote:

 Reload will reload your sip.conf file!  As well as iax.conf,
extensions.conf, queues.conf, voicemail.conf, users.conf











*From:* [EMAIL PROTECTED] [
mailto:[EMAIL PROTECTED]<[EMAIL PROTECTED]>]
*On Behalf Of *Rob Schall
*Sent:* Tuesday, May 01, 2007 2:06 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Calls in ulaw, not gsm as desired



I was in the asterisk console and I typed "reload". Is this not enough to
reload the sip.conf file?

Rob

Andreas Sikkema wrote:

However, even once I reloaded the extensions, its still only

using ulaw.





You didn't reload the sip config? Maybe that's your problem?







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[asterisk-users] 0 duration but non-zero billsec in mysql cdr

[Jaswinder Singh]

I was just going through my call records ( stored in mysql database
by cdr_MYSQL module ) and saw a record having duration = 0 and billsec
of more than 50 seconds . I did a query on cdr where duration <
billsec and saw that there were infact some 250 records with duration
less than billsecond ( table had around 4,00,000 records) . Did anyone
came across this ?
I also checked csv files and they had same record with duration 0 and
higher bill seconds .

Happen with both asterisk 1.2.17 as well as 1.2.18
All sip to iax/sip calls  . Destination numbers were valid.
Dialplan maintained with freepbx .
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Re: [asterisk-users] 0 duration but non-zero billsec in mysql cdr

[Jaswinder Singh]

Someone in -biz list pointed out that this could be a freepbx problem
so i think i will go check there .

@ Salvatore Giudice:

how can i intentionally do it ? Damn i need a app that can make sure
customer phone doesnt  hangup for the time i specify .. even if
customer breaks his phone  . lol
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