Hi all, and thanks for taking the time to read this.
I am trying to configure Asterisk 10.6 as a T38 Fax gateway. I am
receiving calls through the PSTN and want to send them to my VoIP
carriers as T38. This is my dialplan:
[fax]
exten = _X.,1,Set(FAXOPT(t38gateway)=yes,20)
exten =
I forgot to ask:
Do I have to load res_fax or app_fax to use the T38 gateway capability?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
Hi all,
I am trying to control the whole call using a FastAGI script. To that
effect I launch a FastAGI script (written with asterisk-java).
Basically, I want to DIAL from within the FastAGI script. When the call
ends I want to control the hangup (if executed at the remote end), and
depending on
Hi Tarek,
Yes, after running some more detailed packet captures, it seems that
the SDP sent has the sendonly media attribute. I do not know if it is
the Sonus switch, but the problem is identical to yours.
Unfortunately setting canreinvite=yes for that peer does not solve the
problem. I am
this is related to your carrier's SIP messages as they are sending a
sendonly attribute instead of sendrecv (taking a wild guess here) your
asterisk will act as if the call was placed on hold thus the MOH butts in.
an sip debug log for a similar call will be more helpful?
Thanks for the
Hi all and thanks for reading.
I am having a very strange issue. When dialing out with a certain
carrier, asterisk 1.6.20 will play music on hold instead of a ring
tone, although this behaviour is NOT what I want.
Example dialplan execution:
-- Executing [0021266xxx@test:13]
I am currently suffering various SIP attacks. I am using the following
extension to record the caller's IP address:
exten = h,n,set(CDR(srcip)=${CHANNEL(recvip)})
However, in recent attacks, this IP address is not correct, and I
believe that they are spoofing it. I am using asterisk 1.6.2.15.
Hi all and thanks for reading.
I am experiencing a frustrating issue with asterisk where on some
calls the volume suddenly drops to inaudible o completely fades away
for a time. If you hold on long enough (20 to 30 seconds) the sound
will come back.
My asterisk server is on a public IP, and
I just upgraded my asterisk box from 1.4 + Zaptel to 1.6 + DAHDI and
services I was using perfectly before are suddenly broken.
I have a DISA access configured, and my companies employees use if to
dial into the companies extension from their cell phones.
For example they would dial
On Sat, Apr 24, 2010 at 7:01 AM, David White david.wh...@watchguard.com wrote:
call-id doesn't match?
SIP/2.0 200 OK
...
Call-ID: 2117388659-506...@82.158.83.xxx
...
ACK sip:6615xx...@130.117.xxx.xxx SIP/2.0
...
Call-ID: 2117388659-506...@192.168.1.100
...
I'm not sure, but I think
Hi all.
I am having lots of trouble with random calls dropping after 20
seconds, and I finally managed to capture a full sip trace. I'll paste
it in full below, but I'll give a summary first. It seems that
Asterisk is not recognizing the ACK messages that it receives from the
Grandstream ATA.
Hi all.
I am having lots of trouble with random calls dropping after 20
seconds, and I finally managed to capture a full sip trace. I'll paste
it in full below, but I'll give a summary first. It seems that
Asterisk is not recognizing the ACK messages that it receives from the
Grandstream ATA.
Thanks Tilghman, this immediatley solved the problem.
Perhaps a mention in cdr_adaptive_odbc.conf that the res_odbc.so
module must also be loaded will help newbies like me ;)
--
_
-- Bandwidth and Colocation Provided by
Hi all,
I am having a curious problem. I use two cdr backends, csv and MySQL.
These are my settings:
Call Detail Record (CDR) settings
--
Logging:Enabled
Mode: Batch
Log unanswered calls: Yes
* Batch Mode
Vieri
Check out the early dial feature in the Grandstream products (if you
enabled it)
and play with the pedantic option.
thanks, already made sure I use pedantic=no and earlydial is off in my GW
Peder
Like the poster below said, do a sip debug on a call and see which end sends
the bye
Hi all,
I am using cdr_adaptive_odbc and it works fine. I am trying to save
the q931 hangupcause to a cdr record. My diaplan looks like this.
exten = _X.,1,Dial(${EXTEN})
exten = h,1,Set(CDR(q931)=${HANGUPCAUSE})
exten = h,2,Verbose(${HANGUPCAUSE})
However, as I can see by the verbose
However, as I can see by the verbose command, ${HANGUPCAUSE} is always
0. I thought it was a channel variable that contained the hangupcause?
Just an update, if the call is established, then there is a
hangupcause received.
The above problem only happens if the caller hangs up before pickup.
Doug, thanks for the help, already looked it up, but it does not seem
to be a NAT issue (which is what most posters suggest when googling)
Danny, those are billsec durations, the call has been established and
media is being passed for 20 seconds.
Thanks again!
Alex
--
Hi all,
I am having trouble getting cdr_adaptive_odbc to work.
I have correctly configured the odbc drivers and dsn (I have tested
this by connecting directly using isql). I have also configured
/etc/asterisk/cdr_adaptive_odbc.conf like so:
[test-asterisk]
connection=test-asterisk-odbc
Hi all,
This issue is giving me a lot of grief with my customers. I have 5
asterisk servers running in production, each one with almost 70
simultaneous calls at peak hour. Most of my customers complain that
their calls drop after 20 seconds or so.
After running through my cdr's, I see that the
Hi all,
I am worried because on my production asterisk servers, I am receiving
these errors every 2-3 minutes. my log files are full of them:
WARNING[xxx] app_dial.c: Unable to forward voice or dtmf
and also, less frequent:
WARNING[xxx] app_dial.c: Unable to write frame
How can I find out
Hi all,
I've been trying to add a custom mysql field to my CDR's, but I must
be doing something wrong.
I am using asterisk 1.4 and asterisk 1.6, in extensions.conf I add:
exten = h,1,Set(CDR(q931)=${HANGUPCAUSE})
This extension is executed, I can see it in the asterisk console.
I have added a
I'm trying to get my asterisk server to reinvite. I have two asterisk
servers with public IP's. My users (behind NAT) register on one server
(I'll call it server 1), and for some calls they are transfered over
to the other server (server 2), because that server has the E1's.
I want server 1 to be
Thank you Steve, that's a good idea.
If I use a global variable like
-- IF GLB 2 GLB = 0
dial(iax2/isp${GLB}/${EXTEN})
-- GLB = GLB +1
I believe this could cause a race condition if two calls are sent to
the carrier at the same time?
--
Hello everybody.
I have a provider that has 3 asterisk boxes which I must balance my
calls against. At the moment, I route different destinations to
different boxes but this causes lots of problems.
Without resorting to OpenSER or other proxies (as my provider also
uses IAX), is there a way I
Wow, can't believe I missed that.
Thanks so much!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Much to my surprise I tried to debug an AGI script today with agi
debug on the Asterisk CLI and it did not work. Plus, I could find no
reference on lie of it being removed.
Is there another name for that command? I scanned the CLI help but
found nothing similar. Both my 1.6 boxes do not have the
Hello everybody,
When I execute the sip show peers command in the asterisk console I
always get the following notice, repeated 15 times after the sip show
peers output.
[Dec 21 03:38:31] NOTICE[12693]: utils.c:1074 ast_wait_for_output:
Timed out trying to write
This happens on a freshly
*Darrick Hartman:*
NO! If you're using a specific 'branch' of asterisk, the latest release
in that branch is the recommended version. There are almost certainly
bugs/issues with earlier versions. 1.6.1.9 is the recommended version
of Asterisk 1.6.1.x.
*Danny Nicholas:*
RC's are
Hello all,
I have a pretty much standard installation of an Asterisk 1.6.1.6 with no
PRI cards of any type (full VoIP).
Occasionally (it happens every 2 weeks or so), it just stops running. I was
using safe_asterisk but it seems that safe_asterisk did not restart it. I do
have the core dump file
30 matches
Mail list logo
