overlapdial=yes in zapata.conf/chan_dahdi.conf
google it out
Martin
On Fri, Oct 30, 2009 at 6:54 AM, Vieri wrote:
> Hi,
>
> I have a PRI euroisdn link between an Alcatel PBX and Asterisk.
>
> I'm having some trouble with overlap dialing.
>
> Suppose I dial '87
no, I meant this
s,1,Set(H=us)
s,n,Dial(,,g)
s,n,Set(H=them)
h,1,Noop(${H} hanged up)
That might or may not work ... since I didn't actually check it
Martin
On Mon, Oct 26, 2009 at 9:05 AM, Danny Nicholas wrote:
> So this *should* work??
> [outgoing]
> - exten => s,1,Dial(D
--> virtual dahdi
channel1 -- virtual loopback --> virtual dahdi channel2 -- (incoming
call) --> Asterisk -- Dial --> SIP destination or whatever
that way EC can work both ways or you can turn it on one way only ...
via some dialplan application
Martin
You can call application Progress() from within dialplan and it will
cause the Asterisk to send a SIP reply 183
on the call that came to Asterisk.
Martin
On Fri, Oct 23, 2009 at 6:36 AM, Marc Leurent wrote:
> Hello everybody,
> I have 2 users connected on the same Asterisk server th
) is used
also there is a channel variable HANGUPCAUSE and you can check what it
does on the next step
with Dial(,,g) and on h,1 ... since I don't know :)
Martin
On Thu, Oct 22, 2009 at 12:12 PM, B.Masoud @ SH wrote:
> When Asterisk establish a call through an outbound trunk, Is there a
Fax receive not successful - result (13)
Unexpected message received.
For detailed logs please take a look of http://www.pastebin.ca/1634790
Cordialmente,
Martin Cabrera
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asteris
Ring is the state when the device sent 100 Trying after INVITE
When it actually sends 180 Ringing or gets the progress or so message
from another channel
(when used with Dial) then the status changes to Ringing
Martin
On Tue, Oct 20, 2009 at 9:06 PM, Guillaume Yziquel
wrote:
> Hello.
>
or ...
you can also try to use the stun server ... asterisk has it built in
...never used it but saw it's there
Martin
On Tue, Oct 20, 2009 at 1:32 PM, Ott Rose wrote:
> Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at
> your install and they said we ar
;s no error reporting as far as I know
Martin
On Tue, Oct 20, 2009 at 5:26 PM, Eric Chamberlain wrote:
> Hello,
>
> I'd like to implement some public sip uri's that poeple can call into
> and get an echo test. Is there a way to force a codec so that users
> can te
exten => _X.,n,System(sox arg1 ... argN)
Martin
On Sat, Oct 10, 2009 at 5:25 PM, Bart Fisher wrote:
> I'm trying create a feature that allows a callers to add more speech to his
> recording. I think this can be done inside a dialplan, but I can't find an
> exa
is?
>
If you want to send me your patch direct I will make it available through my
website http://www.mycrofters.com and we could also use the forums there to
continue the discussion.
Martin
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iver device that could be used for
this purpose ...
Martin
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asterisk-users mailing list
To UNS
goes from speaker
to microphone of the handset ... that should be cancelled by the sip
phone/device... or someone out there will
hear echo
Martin
On Thu, Oct 1, 2009 at 7:57 PM, John A. Sullivan III
wrote:
> I'm quite new to all this but I was under the impression that most
> electrica
Are you in US ?
do you have the proper keywords in zapata.conf/chan_dahdi.conf like
callprogress=yes etc ?
Martin
On Thu, Oct 1, 2009 at 7:01 PM, Nitesh Divecha wrote:
> Danny,
>
> Thanks for your reply...
>
> Yes these are POTS line and I am not calling myself... Any othe
if a user calling you hears echo of himself then it's the fault of
your sip device/sip phone.
The manufacturer must be using a cheap or an open source echo canceller ...
try getting a different sip device made by some 'normal' company like
polycom or linksys/cisco
Martin
On Thu,
Maybe the GSM carrier is disconnecting you ???
Just a wild guess. They sometimes do that if they have to free
the channel ... for a better paying customer :)
Martin
On Thu, Oct 1, 2009 at 6:09 AM, robert boardman
wrote:
> Hi All
>
> I having an intermittent problem with the above mobil
anyone can just grab the PEF framer datasheet and tweak the driver though...
last I checked there's a whole section devoted to high impedance in
the datasheet
Martin
On Thu, Oct 1, 2009 at 9:56 AM, Kevin P. Fleming wrote:
> Moises Silva wrote:
>
>> May be Martin can help with t
the PBX and do the routing...
Martin
On Wed, Sep 30, 2009 at 7:47 PM, Moises Silva wrote:
>>
>> Is your code vendor locked to Sangoma ???
>>
>
> Hello Martin, not at all. The code is intended to be part of chan_dahdi
> Asterisk channel driver and as such any card
set to high impedance since all
the framers support it...
However I'm pretty much sure it's not part of the drivers as of now.
I had to enable the high impedance mode in the tormenta driver for
myself for tests...
Is your code vendor locked to Sangoma ???
Martin
On Wed, Sep 30, 2009
"pri intense debug span Enables REALLY INTENSE PRI debugging"
add span keyword
or use a tabulator that will do that for you
Martin
On Wed, Sep 30, 2009 at 10:08 AM, Jerry Geis wrote:
> Running asterisk 1.4.26.2
>
> help pri
> pri debug span Enables PRI debug
do you have that user 1006 defined by IP ?
does it have mailbox= also defined ?
my wild guess is that there's unchecked voicemail and asterisk tries
to initialize sending NOTIFY MWI messages
you can't remove these messages they remove themselves after some timeout
Martin
On Sun, Se
u don't change the ${uniquefile} for the second System/Originate
try to add a string to the ${uniquefile} ...
eg
${uniquefile}0
Martin
On Sat, Sep 26, 2009 at 8:05 PM, sean darcy wrote:
> In my quest to actually send a fax, I'm now stuck trying to send the
> confirm.
>
>
rather you could
disallow=alaw
disallow=ulaw
and set dmtfmode=inband
since only g711 codec is clear enough to detect dtmf reliably
Martin
On Fri, Sep 25, 2009 at 10:30 AM, Giedrius Augys wrote:
> Hello,
>
>
> I have one problem and I need to disable dtmf (disable rfc2833, info
find the code in dahdi and put printk so you can see in dmesg or
/var/log/messages
if that gets ever detected
also you may try hanguponpolarityswitch=yes in chan_dahdi.conf
Martin
On Fri, Sep 25, 2009 at 10:19 AM, Stephen Brown Jr
wrote:
> Ok so this is officially driving me crazy. I have
haven't heard of Digium miniPCI transcoding card ... but who knows
maybe they're working on it ...
Martin
On Thu, Sep 24, 2009 at 3:42 AM, Steve Davies wrote:
> Hi,
>
> Given that the Digium transcoding card has no external connections
> (AFAIK), it strikes me that it woul
if you're trying to send the same fax to both parties, then do
exten => s,1,System()
exten => s,2,Sendfax()
step1 will spool the call to dial a number and send a fax
step2 will transmit the fax to the incoming call
Martin
On Wed, Sep 23, 2009 at 7:45 PM, sean darcy wrote:
>
just forget about the dial(a,G()) approach ... you already posted that
it doesn't work ...
either call sendfax on the 1st step
to send fax to the channel that called in to asterisk or
use that call to trigger sending a fax with originate/system
Martin
On Wed, Sep 23, 2009 at 7:45 PM, sean
more lines for originate app
Martin
On Wed, Sep 23, 2009 at 11:00 AM, Jared Smith wrote:
> On Wed, 2009-09-23 at 10:17 -0500, Martin wrote:
>> BTW there should be an Originate app executable from dialplan ...
>> But since there's none you can do
>
> There is an Originat
re's none you can do
exten => _X.,n,System(echo -e "Channel: SIP/num...@gateway\\ncontext:
send\\nExtension: s\\nPriority: 1\\n" >
/var/spool/asterisk/outgoing/call-${UNIQUEID})
and at send,s,1 call sendfax
Martin
On Wed, Sep 23, 2009 at 1:44 AM, sean darcy wrote:
> Martin w
) too much load on the small CPU
loose a few frames or deliver late and your voice TDMoE won't work right
I just speculate here
Martin
On Wed, Sep 23, 2009 at 7:56 AM, Jeff LaCoursiere wrote:
>
> On Wed, 23 Sep 2009, Tzafrir Cohen wrote:
>
>> On Tue, Sep 22, 2009 at 07:43:51PM
is used. You cannot use
any additional
action post answer options in conjunction with this option.
your priority+1 is Hangup ...
is that it ?
Martin
On Tue, Sep 22, 2009 at 7:32 PM, sean darcy wrote:
> Using Digium fax I've tried a simple dialplan:
>
> '
I do not know if fonebridge would work here since it sends/receives
the ~2 Mbps (for each circuit/port)
of data over ethernet ... constantly. That could choke the USB ...
Martin
On Tue, Sep 22, 2009 at 5:54 PM, wrote:
> Yes. Using Ubuntu, Asterisk with Dahdi. USB to Ethernet HUB and a Redf
also sprach Torintino T [2009.09.19.1356 +0200]:
> Try to put qualify=yes.
I had qualify=2000, but even with the default, the problem prevails.
Thanks for taking the time to reply,
--
martin | http://madduck.net/ | http://two.sentenc.es/
"den stil verbessern, das heißt den
tried this before, but no change to the behaviour. :(
Thank you for taking the time to reply.
--
martin | http://madduck.net/ | http://two.sentenc.es/
"give a man a fish, and you'll feed him for a day. teach a man to
fish, and he'll buy a funny hat. talk to a hungry man abou
The question is why is there a monthly fee ? Is this transcription
server done automatically
or using the amazon turks ? If it was software then they could afford
selling it for a license fee ...
there's always upgrades / maintenance they can charge...
Martin
On Sat, Sep 12, 2009 at 11:
Contact: ;expires=3600
Date: Sat, 12 Sep 2009 07:56:31 GMT
Content-Length: 0
<--- SIP read from UDP:86.197.113.72:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
77.109.139.86:5060;branch=z9hG4bK63402845;rport=5060;received=77.109.139.86
To: ;tag=8497k6qgg9hc6ve50s89
From: "asterisk" ;tag=as1
yes, could
you please post your experiences? Were there any issues you may have
encountered that I should be looking out for?
Best regards,
Martin W. Capdevielle
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fine with previous versions of asterisk.
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office: 888-865-0065 x202
Mobile: 585-705-1400
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PCI Express x1 card will work and will fit in the x8 slot
PCI-X slots are usually 3.3V
Martin
On Mon, Sep 7, 2009 at 10:35 AM, mancyb...@gmail.com wrote:
> On Mon, 7 Sep 2009 08:48:25 -0500
> "Juan Cardoza" wrote:
>
>> Hello
>>
>> What is your Asterisk pr
that's probably for ADSI phones ... chan_local confuses the VoiceMailMain app
and you hear it ... Why do you need to call it via chan_local ? Can't
you do Macro or just
call VoiceMailMain directly ?
Martin
On Fri, Sep 4, 2009 at 3:28 AM, Santiago
Gimeno wrote:
> Hello,
>
>
I put in "mwisendtype=nofsk"
in chan_dahdi.conf anyway, and all features like faxdetect and
transfer are turned off.
Has anyone else experienced this issue and fixed it?
Thanks.
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Roches
F response as my modem does not
seem to know the network names. I also needed to increase the storage size for
this field to prevent data corruption.
The problem is I can't dial out or accept incoming calls.
-- Executing [3...@home:2] Dial("SIP/martin-007ab0c8",
"sebi/h
so the solution
really has to work.
Does anyone else on the list have a PRI to VoIP failover setup that's
worked for them in a high volume environment?
Thanks!
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office: 888-865-0065 x202
Am Donnerstag, den 09.07.2009, 11:26 +0200 schrieb ESGLinux:
> Hi all,
>
>
> I´m a beginner with asterisk and I want to know if with asterisk I can
> send sms to a mobile, I´m on Spain, and I don´t know this can be a
> problem (with the operators...)
Hi,
the SMS code in Asterisk is - afaik - o
loose a few 20 ms frames and you'll be fine. But the digital
data has to be so it would
be treated as control frames so to speak.
Martin
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To U
I'm sure he meant UDP not RTP.
In order to guarantee the delivery you can simply do what IAX already
does ... ACK the
frames. This is what TCP does and ISDN PRI protocol layer 2 on the T1/E1.
But why does he want to do it ? Share secret / illegal files LOL ?
Martin
On Fri, Jun 26, 2009 at
; Attended transfer
;parkcall => #7; Park call (one step parking)
--
martin | http://madduck.net/ | http://two.sentenc.es/
it may look like i'm just sitting here doing nothing.
but i'm really actively waiting
for all my problems to go away.
spamtraps: madduck.bo...@m
et e.g. parkedcallrecording=caller, doesn't that mean
that the caller will achieve recording rights by parking and
unparking, even if s/he originally didn't have recording abilities?
--
martin | http://madduck.net/ | http://two.sentenc.es/
"i must get out of these wet clothes a
ing callback feature seems broken in 1.6.
In the end, it seems that when I dial 701 to pick up the call, the
dial flags of the original channel aren't restored. I don't know how
to verify or further debug this though.
Cheers,
--
martin | http://madduck.net/ | http://two.sentenc.es/
the version of Asterisk that you're working with.
Sorry. This is with the (experimental) Debian packages from Xorcom,
version 1:1.6.1.0~dfsg-1.7248
> I can make the patch available on request.
Yes, please. It's good to know that this is a known bug.
--
martin | http://madduck.ne
tures happen,
neither during a normal call, nor during a conference.
I've tried with multiple phones.
What could be the problem?
--
martin | http://madduck.net/ | http://two.sentenc.es/
"and if the cloud bursts, thunder in your ear
you shout and no one seems to hear
and if the band
the problem?
--
martin | http://madduck.net/ | http://two.sentenc.es/
"man sagt nicht 'nichts!', man sagt dafür 'jenseits' oder 'gott'."
- friedrich nietzsche
spamtraps: madduck.bo...@madduck.net
digita
prevent that
and just let it play?
--
martin | http://madduck.net/ | http://two.sentenc.es/
http://lavender.cime.net/~ricky/badgers.txt
spamtraps: madduck.bo...@madduck.net
digital_signature_gpg.asc
Description: Digital signature (see http://martin-krafft.net/gpg
so who's writing the channel driver for it ?
Martin
On Thu, Jun 4, 2009 at 2:26 PM, John Todd wrote:
>
> Michael Graves bounced this to me this morning - it looks interesting
> as a possible device for which an Asterisk channel driver could be
> written:
>
> http:
cifying the exact numberes
Martin
On Wed, May 27, 2009 at 11:19 AM, Stefan-Michael Guenther
wrote:
> Hi,
>
> I have set "context=default" both in /etc/asterisk/dahdi-channels.conf
> and /etc/asterisk/chan_dahdi.conf, and created the necessary context
> with extens for b
;react" to all numbers that come on that circuit and do
Echo app on incoming calls
Martin
On Tue, May 26, 2009 at 1:30 PM, Stefan-Michael Guenther
wrote:
> Hi,
>
> these are my first steps with DAHDI and I finally managed to get
> asterisk to load chan_dahdi (after I found out, that I
Yes, you can share it as long as you designate say the first 2 ports
to the first asterisk instance and the other 2 to another one.
Martin
On Sun, May 24, 2009 at 2:01 PM, Julian Lyndon-Smith wrote:
> Can I run two instances of asterisk sharing a single te412p ?
>
> I want to be abl
I think you should request to get it fixed via free digium tech support
Martin
On Fri, May 22, 2009 at 12:51 PM, Russell Brown wrote:
>
> I've having problems with a BT 2 span ISDN-30/Digium TE205P asterisk
> setup with outgoing calls not completing and requiring an Asterisk reset
e time now (6+ years) and this
sounds like a pretty basic problem
that could cause a lot of failed calls with some SIP MTAs.
I would expect this kind of problem from an Asterisk version before 1.0.0
Martin
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urse of action here? While I can happily
> construct dialplans and stuff, this level of ISDN is completely beyond
> my experience.
>
I think you should request to get it fixed via free digium tech
support. It's a libpri/Asterisk problem
Martin
NFO arrives, we no longer have any memory of the
> Cseq
> of the INVITE that the phone sent.
well then Asterisk now behaves as a poor written hand script that
handles SIP calls ...
INFO can arrive at any time when dtmfmode=info
Martin
___
--
for some reason (someone would have to look deeper) your SIP peer
sends ACK to 200 OK and Asterisk doesn't "get it"
so it retransmits 200 OK a couple times and then assumes there's noone there
Martin
On Fri, May 22, 2009 at 12:36 PM, James Lamanna wrote:
> Hi,
> I ha
this command doesn't show the codecs present in the system do you
have g723 compiled too ?
try core show translations or something like that
Martin
On Fri, May 22, 2009 at 2:25 AM, Chris Maciejewski wrote:
> Hi Martin,
>
> Yes, I do have GSM compiled for sure.
>
> $
it should work just fine; do you have the GSM codec compiled/loaded
core show modules like codec_gsm ... ?
OR that particular version has a BUG...
Martin
On Thu, May 21, 2009 at 3:56 AM, Chris Maciejewski wrote:
> Hi,
>
> I am not sure if I am doing something wrong, but I
Y,
Because the scheduler usually uses the dahdi timer to run ... and if
the timer has stopped
then the frames/events will not go out and finally you get the scheduler full
Martin
On Thu, May 21, 2009 at 9:14 AM, Hose wrote:
> What you say...Martin (asteriskl...@callthem.info):
>
>&
check if your dahdi card still takes interrupts at this point
dahdi_test should return some numbers close to 99%
Martin
On Wed, May 20, 2009 at 3:10 PM, Hose wrote:
> What you say...Hose (hose+aster...@bluemaggottowel.com):
>
>> Hi,
>>
>> I'm getting the following e
1) it'll be hard to get 120 g729 calls with software codec unless you
have a super server with alot of logical CPU units ...
in that case it might be cost efficient to buy the transcoding card
2) you have to pay for the g729 codec licenses unless you want to use
it illegally
Martin
On Wed
cript and
> act on it? (what to look for).
>
There is no such option right now since libpri/Asterisk would ignore
the 2nd PROGRESS message.
Of course this can be custom coded especially that it probably is not
the default behavior on all mobile operat
how about you grep all your multiple scripts for something like
killall -9 asterisk
OR
asterisk -rx "stop now"
OR
asterisk -rx "restart now"
OR
something of that kind;
Asterisk by itself will not disconnect ...
Martin
On Tue, May 19, 2009 at 5:26 PM, David @ULC wrote:
>
ok, if 18xx are to go through analog lines and the rest through PRI
then it's simply
exten => _18XXNXX,1,Dial(zap/g1/${EXTEN})
exten => _18XXNXX,n,Hangup
exten => _1[2-79]XXNXX,1,Dial(zap/g0/${EXTEN})
exten => _1[2-79]XXNXX,n,Hangup()
Martin
On Tue, May 19,
Can you clarify ? Do you want the calls first go through analogs and
when they're all in use
then through the PRI ? Is that why you're putting the priority 101 in
the PRI context ?
Martin
On Tue, May 19, 2009 at 10:37 AM, Tim Nelson wrote:
> Greetings!
>
> I'm hoping
BTW Is vicidial related to http://www.contacttel.com/ ?
http://www.contacttel.com/
http://www.vicidial.com/
the same female face is looking from these websites :)
Martin
On Mon, May 18, 2009 at 5:07 PM, Matt Florell wrote:
> OK, enough with the ViciDial bashing.
>
> Have you taken
do what it is supposed to do?
>
> Is there something better out there that does the same thing and is open
> source?
Good point. What are the other OS alternatives ??? Although it's
already OT but I heard from a few customers there were using Vicidial
and we
sterisk-biz
My bad ... Well his logic was to get the biggest possible exposure
(looking for users ??? :)
Martin
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What kind of business is non-commercial ? Please be so kind to explain
that to me ...
Martin
On Mon, May 18, 2009 at 3:47 PM, Steve Edwards
wrote:
> On Mon, 18 May 2009, ContactTel Business wrote:
>
>> This is a global message to all to announce our xx / xx /
>>
?) - but the fax is received OK. Any
> reason to worry? Anything to do?
your WARNING prints after the DAHDI channel hanged up.
It's possible the receivefax app would want to do a hangup itself instead
of being hanged up. It might be a normal behavior since the app is
disc
Steve,
On Thu, Apr 30, 2009 at 5:05 AM, Steve Howes wrote:
> On 30 Apr 2009, at 04:41, Martin wrote:
>> No more questions. This all can be done in 2-3 hrs [PERIOD].
>
> Then do it.
Then pay me $500
Also I see from your previous posts you like to send your little
useless com
No more questions. This all can be done in 2-3 hrs [PERIOD].
Martin
On Wed, Apr 29, 2009 at 8:02 PM, Steve Edwards
wrote:
> Un-top-posting...
>
>> On Wed, Apr 29, 2009 at 1:32 PM, Steve Edwards
>> wrote:
>
>> On Wed, 29 Apr 2009, Alistair Cunningham wrote:
>
>
You're saying this is worth $5k ? This can be done in 2-3 hrs so are
you really charging
$1666-2500 an hour ?
Martin
On Wed, Apr 29, 2009 at 1:32 PM, Steve Edwards
wrote:
>> If anyone would like to write this, and it gets accepted into the
>> Asterisk subversion repository for
run a "sip debug" and check whether it's asterisk disconnecting the
calls (usually a SIP BYE message)
or whether Asterisk is getting the disconnect from your Cisco GW
Martin
On Wed, Apr 22, 2009 at 10:56 AM, Cristi Iconaru
wrote:
> Hello all,
>
> I have some issues wit
a favor and port his "hack" to
the A 1.4 or 1.6 ?
Martin
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ok,
just in case check if you have /usr/share/zoneinfo/UTC
also if you still have the coredump file ... enter gdb
and do
frame 0
print p
print name
Martin
On Sun, Apr 19, 2009 at 3:31 AM, Justin Piszcz wrote:
>
>
> On Sat, 18 Apr 2009, Martin wrote:
>
>> Hi,
>>
&
x7f7f84076ba0, zone=0x0) at stdtime/localtime.c:1142
ast_localtime is called with zone=NULL
and yet ast_tzset is called with zone = "UTC"
you must have downloaded some version with hardcoded "UTC" timezone ...
or there's a major memory problem ...
Martin
On Sat, Ap
tical to Fax for Asterisk nonwithstanding the
> 1 channel limitation?
it's identical
>
> 3. Can any purchase of Fax for Asterisk count as channel 2+, when used in
> conjunction with Free Fax for Asterisk?
I believe so but I didn't test it
>
> 4. When (If ever) is
I can do it as a paid bounty if there's noone "volunteering".
Would need access to the box with the live circuit including TBCT enabled.
PM me if interested
Martin
On Wed, Apr 15, 2009 at 10:45 AM, Don Kelly wrote:
> Someone referred to a facility message when the TBCT
code
grep PRI_2BCT * -r
channels/chan_dahdi.c:#ifdef PRI_2BCT
channels/chan_dahdi.c:#ifdef PRI_2BCT
it might actually only work in the version of Asterisk it was introduced for ...
Martin
On Wed, Apr 15, 2009 at 8:24 AM, Ron Joffe wrote:
> On Tuesday 14 April 2009 18:41, Jared Smith wrote:
>
issue like NAT or so
Martin
On Wed, Apr 15, 2009 at 4:30 AM, Gerald Harshany wrote:
> Hi
> Last couple of days I received the subject "WARNING" message on a
> home-based asterisk pbx.
>
> Is someone spoofing a "register" method on port 5060? Or, is this "
Y, it can be that someone wants to register with a sniffed SIP packet.
it's basically the nonce="" value is not the same Asterisk sent for
that REGISTER session
Martin
On Tue, Apr 14, 2009 at 11:10 AM, Danny Nicholas wrote:
> http://lists.digium.com/pipermail/asterisk-user
#x27;m not
sure if it's already there
Martin
On Mon, Apr 13, 2009 at 5:51 PM, Martin wrote:
> Y
>
> On Mon, Apr 13, 2009 at 6:05 AM, Sai P. Varanasi
> wrote:
>> Hi,
>> I have a requirement where an IVR application on asterisk has to play a
>> audio file
Y
On Mon, Apr 13, 2009 at 6:05 AM, Sai P. Varanasi
wrote:
> Hi,
> I have a requirement where an IVR application on asterisk has to play a
> audio file in g729 and when a digit is pressed, the call should switch to
> another codec (say ulaw). So, What can I do in the extensions.conf to
> trigger
1) your asterisk box talks to OpenSIPS
2) in that case OpenSIPS should traverse NAT
3) you should not do nat=yes for that device since Asterisk talks to
OpenSIPS (but then it might not matter)
Either take OpenSIPS out of the way or configure NAT traversal w/media
and it should work
Martin
On
I thought so. Unless someone can write a buffer overrun code to email
them the sip.conf or other config files
then you should be fine if you don't provision unsecured contexts to
dial out to PSTN ...
there was a buffer overrun in chan_sip but it was a couple years ago
Martin
On Tue, Apr 7,
Can you give more information about this vulnerability ?
Martin
On Mon, Apr 6, 2009 at 2:55 PM, Jeremy Mann wrote:
> Just FYI:
>
>
>
> IP address 89.248.168.176 has been trying to use the recently release SIP
> vulnerability in Asterisk to make outbound calls via our box. Th
to remove the already running timer
so the T309 could be scheduled since
anyways all other timers do not matter since without T309 the call is
hanged up anyways.
Martin
On Mon, Apr 6, 2009 at 12:23 PM, Afonso Zimmermann wrote:
> Martin escreveu:
>
> Based on the Asterisk logs
That's because you have to create a user account in sip.conf ... +
Asterisk is sensitive about it.
What should help is if you register the phone with that sip account first.
Martin
On Mon, Apr 6, 2009 at 5:00 AM, Rilawich Ango wrote:
> HI,
>
> Recently, I found that asterisk f
Have you looked at the syntax of register => keyword ?
register => [transport://]user[:secret[:authuse...@host[:port][/extension]
; If no extension is given, the 's' extension is used.
There you have it ... Contact: wrote:
> I have an ITSP we are trying to work with that has an "Unusual" way of
It's SIP in rfc (RFC2833) then SIP INFO and then if you can't do
anything else inband audio (only G711)
Martin
On Mon, Apr 6, 2009 at 2:24 AM, Cesc Santa wrote:
> Hi,
>
> I know it is a bit off-topic, but I'd like to ask the community what is the
> current most supp
ringing to A-leg
has to be disabled.
Martin
On Mon, Apr 6, 2009 at 2:38 AM, Khaled W. Chehab wrote:
> Dear Martin
>
> Can you inform me how to make the patch or from where I can get it otherwise
> if there is an application can generate it?
> Or if its relate to chan_sip.c ,plea
Based on the Asterisk logs you posted the Asterisk doesn't have it
implemented per:
"The implementation of timer T309 in the user side is optional"
Martin
On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann wrote:
> Martin escreveu:
>
> What is the specification for T30
me in iax.conf with no password to access
the unsecured context.
Martin
On Sat, Apr 4, 2009 at 3:42 PM, Todd Reese wrote:
> Hi All,
>
> Coming in to day, the logs on the asterisk server showed several entries
> such as:
>
> [Apr 4 15:25:16] NOTICE[9280]: chan_sip.c:14627 handle
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