Sorry to bump such an old post. Which hub is that?
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Hello Everyone,
Today we observed asterisk sending two invites for the initial call before
the call was established (ie, not re-invites). There were no changes made
to the configuration for a very long time, and was kind of confused when
seeing this action. Can someone please suggest where to
Thanks Scott, Restarted all the machines since there uptime was 8 years :).
Everything works ok now.
Kind Regards,
Nick.
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Long story... Would be nice if we can remove this
on BYEs
X-Asterisk-HangupCause: Normal Clearing.
X-Asterisk-HangupCauseCode: 16.
Kind Regards,
Nick.
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Yeah I can do that Anything in sip.conf that we can set?
N.
On Wed, Jul 23, 2014 at 4:39 PM, David Lam software...@gmail.com wrote:
This is defined in chan_sip.c. Simply edit the source file and recompile.
On Wed, Jul 23, 2014 at 1:39 PM, Nick Cameo sym...@gmail.com wrote:
Long story
From sip.conf.sample in 11.10.0
;use_q850_reason = no ; Default no
; Set to yes add Reason header and use Reason header
if it is available.
Using 1.8.7. Shiza
Thanks as always guys.
N.
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Hello Everyone,
We are seeing many instances where we receive 200OK from the
vendors however, asterisk still keeps ringing. Is there anyway to
stop this from happening? I remember reading something about
early media however this seems to be a case of late media?
Kind Regards,
Nick from Toronto.
Did some more testing, what we found was some calls work perfectly to some
phone numbers (ie, two way audio).
For other phone numbers in the UK we are getting 200 OK however:
1) We hear ringing in the handset
2) Call connected but not audio.
This problem is reproducible every time.
Our asterisk
Hello Everyone,
Thank you all for your response. The people I am doing it for run a
non-profit charity,
and are legally able to reach out to their customers. I will wire it
up to the DNC
however, for starters, I would like to get asterisk to:
i) Iterate through a list of numbers
ii) Play a
That's about as simple as it gets.
A call file that goes to the dialplan.
A dialplan that consists of Read (which would play the message) followed a
GotoIf into a mailbox (either voicemail or Dial() to an external number).
One hint for doing unattended dialing like this, make sure you're
Hello Everyone,
We are looking for a simple open source auto dialer with polling
capabilities. What we would like is a program that we can upload
leads to, and have asterisk:
i) Dial numbers
ii) Play pre-recorded
iii) If user presses one, forward the call to an agent
There are so many solutions
On Mon, Apr 21, 2014 at 2:01 PM, James Sharp ja...@fivecats.org wrote:
On 4/21/2014 1:47 PM, Mitul Limbani wrote:
Use vicidial for achieving the same.
Or call files (or AMI originate), a short bit of dialplan logic, and maybe
a call to Queue().
This is a nice and easy solution however,
I'll be attempting this tomorrow for a friend as a favour. Will post the
end result for others
in the future.
Nick from Toronto.
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Is Asterisk not able to gather IP and port info on INVITES in a REGISTER free,
host=dynamic setup? As you all know REGISTERS are resourceful and the
phone can be anywhere..
Kind Regards,
Nick.
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Hello Eric,
On 2/18/14, Eric Wieling ewiel...@nyigc.com wrote:
No. Asterisk will accept calls from unregistered devices, but you have
to enable guests I sip.conf and hope your dialplan is secure. No sane
person does this.
Thank you for your response. Our security layer is abstracted out
I just want to clarify. We are not creating peers automatically. And
we allowguest=no. We do have peer entries in sip_buddies db as you
would expect. As mentioned, we just don't allow phones to REGISTER
every 3600 (for example). Once a valid peer/phone tries to place a
call, we would like asterisk
Hello Ishfaq,
I just tried it and it did create a P-Asserted header however it
contains the extension
of the asterisk peer not what was passed by our switch. From the
previous example:
P-Asserted-Identity: 222 sip:222@192.168.2.10 (which is asterisk
peer extension and not)
P-Asserted-Identity:
Hey Guys, I really appreciate this and I apologize for asking however,
we do not have any way to test in advance outside of our live
environment. Can someone kindly provide a working extension rule that
will retain the following P-Asserted info that is existent from the
inbound-leg to the
Shiza Sounds about right but is it true? Anyone else?
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Hello Markus,
Thank you so much for your response. Our switch is already generating
the needed P-Asserted header:
P-Asserted-Identity: John Doe
sip:14167493...@toronto.location.com; user=phone; nat=yes.
I really did not want to have to rebuild it using `SIPAddHeader`
however, if I have no
Hello Everyone.
Our environment is a register free setup, and our phones are set as
host=dynamic.
The problem we are experiencing is for inbound calls:
Name/username HostDyn Forcerport ACL Port
Status Realtime
222/222 (Unspecified)D N A 0
Hello Everyone,
Our switch is sending P-Asserted info to asterisk however the information
is getting removed when asterisk forks the call. Is it possible to have asterisk
retain the P-Asserted on the leg. This is quite important for valid
functionality of our
network.
Tried setting `sendrpid =
We use opensips as a type of firewall as well and don't need to set
qualify=yes.
N
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Hello Laszlo,
That you for your response. Just to confirm callerid=whatever will only
effect the private numbers? And will
not have any effect on FROM headers with valid CIDs, as is intended?
N.
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Hello Everyone,
Calls that are private name private number have the following TO header:
From: Unavailable sip:aster...@server.com;tag=as120a1079.
Don't tell anyone, but we are trying to put on a We're big enough to own
the pricey softswitch look. Even though I would pick a OpenSIPS +
Asterisk
Correction, and by TO, I mean FROM header :)
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Hello Everyone,
Just getting in a new cisco router, and would really like to get it up and
running as soon
as possible. Everything is configured from what we can see. This is a NAT
setup.
After 2 seconds on a successfully established call we reach retrans max,
and asterisk
disconnects the call.
Hello Eric, I knew this problem all so well however, never knew CISCO sip
alg was enabled by
default. The following settings got us up and going shortly after the email:
no ip nat service sip udp port 5060
ip nat inside source static udp 192.168.2.5 5060 interface Dialer0 5060
access-list 130
God Bless and Merry Christmas to All!
Nick.
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Hello Everyone,
I have a strange issue where the caller's phone keeps ringing even after
the 200 OK. I am using the latest version of Asterisk 1.8, and wanted
to know if anyone could give me any pointers before posting the SIP
debug messages.
Kind Regards,
Nick from Toronto.
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Hello Everyone,
I was just wondering if the cli command file convert supports wildcard or
entire directories? I am looking at a very long list right now and anxiously
waiting a response :).
Kind Regards,
Nick from Toronto.
--
DIR=sounds directory/*
cd $DIR
for f in $DIR; do
/usr/local/asterisk/sbin/asterisk -rx file convert $f ${f%.*}.g729
done
Thank you come again.
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Hello Everyone,
I am using the following dialplan to allow users to check their
messages from PSTN world:
; Internal Routing
exten = _1XX,1,Dial(SIP/${EXTEN}, 10)
exten = _1XX,n,Wait(1)
exten = _1XX,n,Answer
exten = _1XX,n,Wait(1)
exten = _1XX,n,Voicemail(${EXTEN},us)
exten = _1XX,n,Hangup
The
Guys,
Sorry, I pasted the wrong dialplan. What we are using is the following:
exten = 6474770990,1,Answer
exten = 6474770990,n,Wait(1)
exten = 6474770990,n,Voicemail(1001,us)
exten = 6474770990,n,VoicemailMain(1001)
exten = 6474770990,n,Hangup
So, we are using VoicemailMain, but wile trying to
Thanks guys, it worked perfectly. No inadvertent message added when
attempting
to check messages from outside:
exten = 1001,1,Answer
exten = 1001,n,Wait(1)
exten = 1001,n,Voicemail(1001,us)
exten = a,1,VoicemailMain(1001)
exten = 1001,n,Hangup
Press * without pound, and voice mail main fires.
Hello Everyone,
I have a new problem where when placing the call, asterisk will
automatically go into music on hold until the call is connected (ie,
no ringing). It was kind of confusing because sometimes `SESSION
PROGRESS` takes longer than others, during this time we are in MOH.
The call does
There is two way audio, it's just during ringing that this happens.
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Hello Jg,
Thank your for your response. No m option on dial. I think it's a RTP
relay issue however, do not know how to diagnose the SDP payload. Any
help would be appreciated.
N.
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Yeah of course. Still digging into it :). Will post the solution if I
find it. a2billing forum takes for ever to answer...
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I have no idea where the `m` is coming from. I even looked into the
A2Billing script. Still digging
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Yes of course, I just did not want to overwhelm you guys with SIP
trace. Before that though, I realized something:
[Sep 10 12:03:30] WARNING[8178]: res_musiconhold.c:802 set_moh_exec:
SetMusicOnHold application is deprecated and will be removed. Use
Set(CHANNEL(musicclass)=...) instead
-- AGI
Hope this helps someone save a day of running around.
So my issue was with a2billing. The warning `No remote address on RTP
instance '0xb6d16a28' so dropping frame`
was not related to the music on hold coming on during ringing.
The Problem:
We have a script that loads rates into
Oh scandalous Instead of playing the MOH, I would like to play the
ringtone that is on the machine. Ummm, where is it? :)
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I ran a test call with trace can be found here:
http://pastebin.com/f8MuxaFV
I also wanted to mention that yes we have * setup with disallow=all
and allow=g729 for testing,
maybe permanently if we can successfully setup G729 pass through. That
being said, the same problem is still there using
Yeah!!! The Dial command setting:
http://forum.asterisk2billing.org/viewtopic.php?f=2t=1704
I know this is not an a2billing mailing list, and I am sorry however,
I do think that the No remote address on RTP instance may have
something to do with it. Maybe the ringtone from downstream is not
You ok sir? Are you going to make it?
N.
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