PSTN <--> Cell Phone
>
> sipphone was able to setup a connection to Cell Phone. When sipphone hangs
> up, Cell Phone also hangs up. However, when Cell Phone hangs up, sipphone
> was not able to hang up.
>
> Could it be that Asterisk was not able to recognise the hangu
y IAXy should be able to talk directly to Teliax.. in
> fact, I just confirmed that. So what's up?
If you grab an iax2 debug on your Asterisk box when it tries to do the native
transfer I can better answer this.
Joshua Colp
Digium
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connection directly, it will make sure that it will work. The
IAXy talks to Teliax, and Teliax talks to the IAXy. What happened is that they
weren't able to talk directly to eachother so they kept the connection through
your Asterisk machine.
Joshua Colp
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bc.so. But it wouldnot help. Please suggest how do I go further with this?
>
> Thankyou all in advance.
Upgrade to the latest version of make and try again.
Joshua Colp
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asteri
is received and authenticated as the
user it is sent into the PBX with that accountcode and the CDR record will
reflect it.
> Doug.
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To
:
>
If you do an Answer and then a Wait(2) before going to VoiceMailMain in your
dialplan does this solve the issue? It might just allow time for everything to
settle but I can't say I've ever heard of someone getting audio like you're
describing.
Joshua Colp
Digium
_
see when memory leaks were fixed. What exactly
does your server do by the way?
> When i reboot the server, the asterisk day per day increase the use of the
> RAM memory.
>
> Any help is appreciated.
>
> Chris.
>
Joshua Colp
Digium
6-08-02 17:07:24 VERBOSE[7027] logger.c: -- Hungup 'Zap/1-1'
>
> I truncated to phones to ring to 3 lines but in reality there are 42
> lines that are supposed to ring at once when *7 is pressed.
>
When you call each
s they used, not everything.
The source for their own SIP client is not there as far as I know.
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ou using your own?
> --
> Carlos Chavez Prats
> Director de Tecnología
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Tel: +52-55-91169161 Ext 2001
>
Joshua Colp
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>
You can use the SIPAddHeader application:
SIPAddHeader(Header: Content)
Adds a header to a SIP call placed with DIAL.
Remember to user the X-header if you are adding non-standard SIP
headers, like "X-Asterisk-Accountcode:". Use this with care.
Adding the wrong headers may jeopardize
nd under what circumstances it is allowed. By
encoding it into the dialed number only the users you control will be able to
do it and the protocol doesn't need to be altered.
Joshua Colp
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lows way down as if
> almost no data is being transmitted.
>
> How do I send a sip debug?
Actually since this happens randomly I doubt that will help. Is there any other
traffic on the network too? Never know... or a faulty switch? Grasping at
random things but nothing really comes to
urity risk
as well. If you really need it why don't you encode it into the dialed number
or something?
> :(
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t really does drop out,
ie: stream actually stops). I know there's some Windows software out there
capable of this as I picked a copy up while at Spring VON but you might need to
look around. OH - can you also send a sip debug with the reinvites? I'm j
I would think that user B would be billed on the originating system, not the
system the call ended up at.
> Doug.
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r mistakes. There
are just some things it wasn't designed to do 'nor does it claim to do. It
wasn't made with the capability to transport accountcode or other arbitrary
Asterisk specific information. Could it be added though? sure.
> Doug.
Joshua Colp
Digium
egy (0s holdtime),
> W:0, C:0, A:0, SL:0.0% within 0s
> No Members
> No Callers
>
> Here's a pastebin to my queues.conf and my extensions.conf sections:
> http://www.pastecode.com/2334
>
> Thanks in advance!
>
> Keith
You are removing a queue member that
- Original Message -
From: Joshua Colp
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion [mailto:[EMAIL PROTECTED]
Sent: Mon, 31 Jul 2006
10:25:38 -0300
Subject: Re: [asterisk-users] music ring (CRBT)
Thanks to BJ Weschke, it's called early media.
progress or something along those
lines. It allows you to send audio to the caller before actually answering the
channel. I know that some providers do support this, but yours may not. I would
first run a test using the Playback application with the noanswer option and
user with extension 191?
A phone is trying to subscribe to get the status of extension 191 (whether it
is in use, etc). In order to make the ERROR go away and to make the
subscription work you need to add this to your internal context:
exten => 191,hint,SIP/191
(If I underst
; I am using [EMAIL PROTECTED] 2.6
>
> Regards
>
It would go into the dialplan in extensions.conf - I don't know exactly how
yours is setup though (especially with [EMAIL PROTECTED] involved) so I can't
tell you where to put it exactly.
Joshua Colp
Digium
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> Trikon electronics Pvt. Ltd.
>
> All science is either physics or stamp collecting.
> -- Ernest Rutherford
Joshua Colp
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asterisk-users mai
ot help?
> I also think, but cannot be sure, that this system worked with older
> versions of Asterisk in the 1.2.3 to 1.2.5 era.
>
> Has anyone else seen this?
>
> --
> Alistair Cunningham,
> Integrics Ltd,
> +44 20 799 39 799
> http://integrics.com/
Joshua Colp
Di
-invites in this case?
What do you mean by not used? Even if going through SER it should still be used.
> --
> Alistair Cunningham,
> Integrics Ltd,
> +44 20 799 39 799
> http://integrics.com/
Joshua Colp
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plan. :)
>
> Thx for help.
> Kind regards
> Szolke
>
A full sip debug of the dialog would be very helpful so we would know who is at
fault and what's going on.
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, what other recording formats gives me the benifit of small
> size and good sound quality of mp3?
What most people do for this is they execute a shell script afterwards to
convert it into the MP3 format. As for another format that will work for you I
can't think of any.
> Thank you ve
SIP sessions for each subscriber to one
> or two?
>
The original issue was having multiple SIP devices register to the same
account. Can't exactly limit that since only 1 can be registered at any given
time and the newest registration overrules any previous.
Josh
ccess by IP in my system. As you said,
> there is no way to prevent it, right?
> I am using ARA in the system. Can I detect it if prevention is not
> possible?
>
When it comes down to it - unless you want to go to great extremes, it is not
possible. Sor
- Original Message -
From: Steve Underwood
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion [mailto:[EMAIL PROTECTED]
Sent: Sat, 29 Jul 2006
00:47:44 -0300
Subject: Re: [asterisk-users] CSTA support for asterisk
> Joshua Colp wrote:
>
does it... interesting. You may be able to hack chan_sip up a bit and add
that header in.
Joshua Colp
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7;ve found so far has been with sipsak,
> which wasn't really working out too well for me so I was hoping I could do
> this from inside asterisk rather than execl()ing programs from whithin
> asterisk.
>
What does the packet look like that
really
> appreciate it. I am new to asterisk and i am finding myself pretty lost..
>
>
Joshua Colp
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sted this yet - so please do reply if it works as I have said.
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listening to it just in the dialplan and
not using an AGI at all for streaming it back? We need to eliminate some
variables here and narrow down where the issue might be.
> -- G.
Joshua Colp
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e fact that nobody seems to know what it is - I'd say no. Can you shed
any light on what it is?
> sanchal
>
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it
immediately considers it answered (provided you are using a zaptel analog card).
>
> David Morrow
> Technical Systems Lead
> Autodata Solutions Company
> [EMAIL PROTECTED]
> http://www.autodatasolutions.com
>
> Tel: (519) 9
s one so that
a trunk could support, for example, 2 outbound calls at a time. You would just
see if the group_count is equal to 2 and if so jump to busy, otherwise avail.
> Thanks in advance,
> Aaron
Joshua Colp
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>
> the firmware version for the PAP2T is 3.1.9(LSc)
>
> I am using a dialplan coming from another customer with a similar
> setup, but with PAP2-NA, where it's working fine.
>
> What can I do to fix this.
>
> Regards,
> Olivier
>
Joshua Col
ere's an example for storing it there:
exten => s,1,Set(CDR(userfield)=${SIPCHANINFO(recvip)})
>
> Regards
>
Joshua Colp
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ck how TrixBox has that setup by
default (anyone know?).
> Any configuration is needed to be done in trixbox?
> Thanks
> Victor
>
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To
ing ?
If you do a sip debug you should see two INVITEs to each side after the call is
established with the IP address of the GXP2000 in the SDP. You can also run rtp
debug to see if the RTP audio stream is running through Asterisk.
> I'm trying it with two grandstream gxp2000.
>
>
ock extra registrations is by limiting
the account to a specific IP range for registrations but then if you tried to
register elsewhere with a legitimate attempt, it would be blocked too.
> Thanks.
Joshua Colp
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ing an active call to another extension/context. If this isn't
what you meant, then please do respond with a better explanation.
> Didn't find nothing on voip-info.org
>
> Thanks for your answers
>
> KAI
Joshua Colp
Digium
_
(519) 451-6615
>
> < Lead, follow or get out of the way! >
>
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t; context=c-DID
> dtmfmode=auto
> host=xxx.xxx.xxx.xxx
> insecure=very
> sendrpid=yes
> type=friend
> echo=no
>
> Any suggestions ?
One final note, 'echo' is not a valid option.
> Thanks
>
>
>
Joshua Colp
Digium
__
dial.so is not loaded, so the Dial dialplan application does not exist. You
can load it from the CLI by doing load app_dial.so or explicitly putting it in
your /etc/asterisk/modules.conf to be loaded when Asterisk starts.
>
> Dave
>
Joshua Colp
Digium
_X.,1,Dial(SIP/[EMAIL PROTECTED],30,trg)
exten => _X.,2,Hangup
Notice the . after the X? It means match any extension starting with 0 through
9, of any length.
> If anyone would be so kind as to shed some insight into the matter it'd
>
n't find any documentation about
> how to configure it though.
I don't believe there's anything configurable but if you open app_voicemail.c
there's two declarations, VOICEMAIL_DIR_MODE and VOICEMAIL_FILE_MODE which set
the p
ure dtmfmode in sip.conf for
each entry is set to the one you want.
> Thanks.
>
> Jason.
>
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link, I'm afraid there could
> be a bandwidth problem.
chan_sip requests the count fairly frequently, dunno how much traffic it would
actually generate though.
> JY
Joshua Colp
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s the same as your s extension to see if this is it. Or even:
exten => _X.,1,Noop(Hey they called ${EXTEN})
exten => _X.,n,Hangup
>
> --
> Thx
> MAG
>
Joshua Colp
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ute an outside application that exists right now
and using your own method to communicate back to turn on MWI (maybe generating
a SIP NOTIFY to poke the phone with?).
3. Share the voicemail directory over something like NFS.
> Thanks
> JY
Joshua Colp
Digium
__
em with
> SIP?
>
We need more information in order to give you an answer (if there is one). Do
you mean that when you are using SIP to your provider it sometimes fails? As
well, console output would be nice so we could see what your Asterisk is doing.
> Looking forward to your re
now?) have
done a good job on their SIP stack and appear to have done a good job on their
T.38 implementation too. If you do end up giving them a try, definitely report
back so others will have some feedback.
Have a great day!
Joshua Colp
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ones solved
it and narrowed down the problem ;)
> Thanks for any help you can give me
You're welcome and hopefully some others can give some insight and maybe
information on their own deployments similar to what you wish to do.
> Carlos Bernat
>
>
Joshua Colp
Digiu
like a regular call, how can an AGI script tell that
> it's a transferred call?
>
> Doug.
I know of no way for it to.
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;
> Is this documented somewhere?
No, the new INVITE does not have that info... I even just tested it from my
Polycom IP600, it was a regular normal INVITE. As for documented about the call
flow... probably somewhere on the internet, it
ars out of both calls.
With a blind transfer the phone can simply say hey channel... this is your new
extension and context.
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gt;
> How can we do this?
>
> Doug.
>
What type of transfer? blind or attended?
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ht
e able to place calls.
That should be fine.
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different. In the future you can use sip show peer to see what is
happening.
>
> Doug.
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>>> http:
Stephen Bosch wrote:
So -- to clarify that -- it's technically possible to have a single DID
that allows multiple calls to be set up. The DID is just the line
identifier, but we could have say three simultaneous calls, as long as
the provider allows it -- correct?
You got it.
--
Joshua
s may limit how many you
can actually have up simultaneously but that's a feature of their system.
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I have one word for your response: NICE.
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Alberto Sagredo wrote:
Umm. Maybe i have left some asterisk 1.2.9.1 modules.. and it has not
been replaced.
By i made a make install after i compiled it, so it would be replaced?.
I will check it.
Thanks
Joshua Colp escribió:
Alberto Sagredo wrote:
I have the same problem on some modules
/usr/lib/asterisk/modules? Sounds like you have some
leftover modules as that was converted to a dialplan function and
app_math taken away - in trunk.
--
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Joshua Colp wrote:
Colin Anderson wrote:
What I'd love to see is a reasonably grunty DSP available on the cards
that is _user programmable_. There's some stuff a host processor isn't
particularly good at (at least at present... most CPUs have an inbuilt
FPU, but when do we get
ll could not be completed as dialed. Please check
the number and try your call again later. This is a recording. 2CY.
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I'd just like to note that AEL2 was brought over into Asterisk trunk
(what will become 1.4) and the old AEL removed. That's where most
development is taking place on AEL2, and why you don't see patches on
the bug tracker.
--
Joshua Colp
Software Developer
Digium
odifications to
bring the module up to code with the loader. You can't just build and load
any old module, it HAS to be updated. Trunk itself is not broken, all
modules in the official source tree have been updated to the new loader
standards. It do
m.com/mailman/listinfo/asterisk-users
That's because the extension dialed was 1. Using the Pickup application
you can't do a Pickup on the device called (ie: SIP/10 or SIP/11) but
the extension, which is 1.
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[EM
Joshua Colp wrote:
Peter J Dean wrote:
Ummm, not sure if I am missing anything, but I have never experienced
this before, where the asterisk release didn't compile.
Downloaded all the newest releases from Digium, I compile everything
first before installing to minimise downtime of the
tinfo/asterisk-users
Did you do a make install on zaptel and libpri?
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Not in the 1.2 release series, no. It only receives bug fixes and this
would not be a bug fix.
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Software Developer
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P - 256-428-
Matthias Fechner wrote:
Hello Joshua,
Joshua Colp wrote:
[portunity-out]
type=friend
host=iax.iaxport.de
username=XXX
secret=YY
context=incoming-portunity
notransfer=yes
Only if the username is specified as portunity-out when the other side dials
you. Otherwise your Asterisk has no
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Only if the username i
in /var/log/asterisk just
before the machine hung the messages posted avobe(is the first time we see
it).
Please contact Digium support about this issue.
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[EMAIL PROTECTED
19 05:14:36
> EDT 2006 i686 i686 i386 GNU/Linux
>
> I re-downloaded the codec and attempted the i686 and i586 version wiht
> no luck.
> md5sum codec_g729a.so
> 92b64cc5be4a3e622c91357b116d99e3 codec_g729a.so
>
> Thanks -Jason
>
>
>
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fine, multiple threads are started.
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users
Why don't you use something like the chan_local channel driver to send
the call into the dialplan where it will then execute the extension? If
you don't you're going to see what you're getting above. You're looping
an outbound call back inbound to the
r. Thanks!
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It's changed in sip.conf in the [general] section. It's the useragent
setting.
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[EMAIL
0) loaded RTLD_LOCAL
Apr 20 08:27:58 WARNING[13559]: loader.c:744
__load_resource: Key routine returned NULL in module
/usr/lib/asterisk/modules/res_snmp.so
Apr 20 08:27:58 WARNING[13559]: loader.c:753
__load_resource: 5 errors loading module
/
loaded RTLD_LOCAL
Apr 20 08:27:58 WARNING[13559]: loader.c:744
__load_resource: Key routine returned NULL in module
/usr/lib/asterisk/modules/res_snmp.so
Apr 20 08:27:58 WARNING[13559]: loader.c:753
__load_resource: 5 errors loading module
/usr/lib/asterisk/modules/res_snmp.so, aborted
does not support sending messages like this at this moment. Sorry!
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To UNSU
mended to use it
instead of depending on CVS. Information about using SVN to check out
things is available at http://www.asterisk.org/download
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stinfo/asterisk-users
You won't know that the phone is going to forward the call to an
alternate number until you send the call to it. You could theoretically
cache the information... but then it wouldn't be real time up to date
with the phone's settings.
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ptions visit:
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Some people have problems, some people don't. There is no way you can be
prepared for every situation out there. We try our best.
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P - 2
results so others will know!
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phones to register, or calls to be placed. One solution is to run a DNS server on the same machine, and cache results or use IP addresses instead of hostnames.
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solution as there is no billing
solution out there for Asterisk that fits all. Usually you end up making
tweaks here and there even if you do use a prebuilt solution.
--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
___
rate of G726. I also do not know the legal
implications of it. It may be illegal... Who knows, not I.
--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
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UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
You do this all the time on asterisk-biz, that's fine but please do not
do this on asterisk-users. This is not the place for it.
--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878
m is that one
side is using G723.1 and the other is ALAW, and Asterisk can not transcode
between the two.
--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
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cret=212121
trunk=yes
host=dynamic
notransfer=yes
--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
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Asterisk-Users mailing list
To UNSUBSCRIBE or up
m.com/mailman/listinfo/asterisk-users
Check everything you can: username, passwords, etc.
--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
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Asterisk-
ation takes an extension, not a dial string.
Take out the SIP/ and see if it works.
Ala:
exten => _*.,1,Pickup(${EXTEN:1})
--
Joshua Colp
Software Developer
Digium
P - 256-428-6066
C - 506-878-0147
[EMAIL PROTECTED]
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Marco Mouta wrote:
Password and username are ok.
On 4/4/06, Joshua Colp <[EMAIL PROTECTED]> wrote:
Marco Mouta wrote:
Hi all,
I've 2 * tryning to connect each other
Server A is already registred on server B
But server B never registers in server A
I always get this:
Tx-Fram
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