Matt, thank you very much for your help!
I'm just going to comment on the 'directmedia'/'canreinvite' points here.
1) There is no 'reinvite' setting in chan_sip. If you patched
Asterisk, than your mileage may vary.
I didn't patch. Just using vanilla Asterisk from your website ...
2)
Just a thought regarding testing.
Create a suitable TIFF file with more than 30 seconds worth of data and
send it from Asterisk using SendFAX() to convince yourself whether
Asterisk will work with your ITSP, you may still need to enable session
timers.
Have you considered setting up an
Hi All
I am trying to get voicemail switched over to ARA on version 13 and notice
that the password is not stored in the db when it is changed.
A little hair pulling and playing around and I think the problem is in the
function ast_update2_realtime in main/config.c.
Issued source is ==
int
Como integrar asterisk con Ldap.??
Saludos
--
Dario Javier Estupiñan Vallejo
darioestupi...@soygenial.co
Investigación y Desarrollo - Neiva
Corporación Politécnica Nacional
.
--
Hello all
Over the last couple of months weve been experiencing a strange problem,
which Ive been unable to solve.
We have an Asterisk 1.4.19 thats been running happily for the last several
years. All calls go through an AGI server from dialplan.
On average we have appr. 3000
On 16-12-14 14:00, Dario Estupinan wrote:
Como integrar asterisk con Ldap.??
https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver
Best,
Patrick
ps this mailing list uses the English language
--
_
-- Bandwidth and
Hi,
On Mon, Dec 15, 2014 at 9:03 AM, Recursive li...@binarus.de wrote:
I would be grateful if you could refer to my message from some minutes ago. I
have provided all the details there.
According to the detailed trace asterisk is indeed retransmitting SIP
OK messages:
snip
Session
Thanks George.
I will give it a try.
Have a great day!
Dan
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 11:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
=as18f69e58.
Call-ID: 452af6610540b7cf0d4c49f372d46779@192.168.10.75.
CSeq: 2 REFER.
Server: IPTAM PBX (Version 20141216/6814).
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Contact: sip:180@192.168.10.75:25060.
Content-Length: 0
I corrected my local_net setting (based on advice from network admin).
I have tried several different values for the from_user and still have the same
problem.
Asterisk receives the OK from Vitelity.
Asterisk sends the ACK (without a Contact header).
Vitelity doesn’t seem to process it, so they
On Tue, Dec 16, 2014 at 9:00 AM, Dan Cropp d...@amtelco.com wrote:
I corrected my local_net setting (based on advice from network admin).
I have tried several different values for the from_user and still have the
same problem.
Asterisk receives the OK from Vitelity.
Asterisk sends the
Dan Cropp wrote:
I corrected my local_net setting (based on advice from network admin).
I have tried several different values for the from_user and still have
the same problem.
Asterisk receives the OK from Vitelity.
Asterisk sends the ACK (without a Contact header).
A Contact header is not
Hello Binni,
It is hard to say anything without more information.
You need to understand what happens in those dropped calls.
Logs would help. Traces might help also. Try mirror the traffic to another
server and capture it using tcpdump, or even run tcpdump on the server
itself.
On Tue, Dec
Thank you George and Joshua.
This can cause major problems. I've rarely (if ever) come across an ALG
(that's what that is) that didn't break something.
I am contacting the network admin and seeing if he can modify the firewall.
I'm a lifelong programmer. Code and programming, I understand.
On Tue, Dec 16, 2014 at 6:25 AM, Paddy Grice pa...@wizaner.com wrote:
Hi All
I am trying to get voicemail switched over to ARA on version 13 and notice
that the password is not stored in the db when it is changed.
A little hair pulling and playing around and I think the problem is in the
On 15 Dec 2014 22:49, Jonathan White uvac...@googlemail.com wrote:
Does anyone know if it is possible to disable asterisknow-version from
writing over my issues file?
Alternatively is it really required to have it as a dependency in asterisk
13? surly everyone has upgraded from the old
I am running a heartbeat... Asterisk 11.15.0 - same behaviour is noticed on
1.4.43 also
I issue a call through the API that does the below. just UserEvent and
Hangup
-- Executing [s@heartbeat:1] UserEvent(Local/s@heartbeat-000f;2,
HeartBeat, Noop) in new stack
-- Executing
Here's an update...
My network admin would not turn off the ALG because it would cause several
other problems to other phone systems we have.
He looked at the sip trace. What he found is the PJSIP trace showed a
different IP address than the older chan_sip so he had me change the aor
contact
On Tue, Dec 16, 2014 at 11:45 AM, Dan Cropp d...@amtelco.com wrote:
Here's an update...
My network admin would not turn off the ALG because it would cause several
other problems to other phone systems we have.
He looked at the sip trace. What he found is the PJSIP trace showed a
different
Tonight several community services will have intermittent availability
due to maintenance. This maintenance will begin at approximately 10:00
PM CST[1] and should last no longer than two
hours, ending around 12:00 AM CST.
The affected services are:
* issues.asterisk.org
* crowd.asterisk.org
*
On Tue, Dec 16, 2014 at 1:04 PM, Jerry Geis ge...@pagestation.com wrote:
I am running a heartbeat... Asterisk 11.15.0 - same behaviour is noticed
on 1.4.43 also
I issue a call through the API that does the below. just UserEvent and
Hangup
-- Executing [s@heartbeat:1]
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