One thing i have noticed is that your profile-id don't match and therefore you
would get no video. Asterisk is not a problem
On Jan 9, 2012, at 1:20 AM, Durgesh Mishra wrote:
Hi,
I am facing an issue while testing the video mail service of Asterisk. I have
two different setup on one setup
, uri and authorization user name. if response is the same
between the two, I am lost.
On Nov 24, 2011, at 2:11 PM, Jonas Kellens wrote:
On 11/22/2011 06:13 PM, Alex Vishnev wrote:
it is strange that Aastra acks 401, sends another invite but does not
increase CSeq. Is that the same behavior
do you see the register messages? if your device is not registered, INVITE
would be challenged. You should check to see if register messages are being
properly acknowledge with 200OK.
On Nov 22, 2011, at 10:29 AM, Jonas Kellens wrote:
On 11/22/2011 04:25 PM, Bruce Ferrell wrote:
Jonas,
Your registration should have also have the flow
PEER ASTERISK
REGISTER---
--401
REGISTER(nonce) -
200OK
Then the server controls the life of the registration and 200 Expires Header
gives you this timeout. If the invite is sent
I doubt it. Unknown headers should be ignored by UAS. is it possible to post
the trace?
On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote:
On 11/22/2011 05:31 PM, Alex Vishnev wrote:
Your registration should have also have the flow
PEER ASTERISK
REGISTER
it is strange that Aastra acks 401, sends another invite but does not increase
CSeq. Is that the same behavior with others?
On Nov 22, 2011, at 11:51 AM, Jonas Kellens wrote:
On 11/22/2011 05:42 PM, Alex Vishnev wrote:
I doubt it. Unknown headers should be ignored by UAS. is it possible
the best way to handle large sip client base is using provisioning interface.
Even though you can create configuration files and server them with
asterisk+extensions, you need to consider security aspects of this approach as
well. Using tftp or simple protocols to server config files works on
no, you need a tag i.e From: sip:p...@xx.xx.xx.xx;tag=xxx, where xx
is a unique identifier
see the definition of SIP Dialog
Dialog: A dialog is a peer-to-peer SIP relationship between two
UAs that persists for some time. A dialog is established by
SIP messages, such
this could be an unsupported codec. Do you know if Office365 supports PCMU? I
would also try to get rid of 101 (rfc2833) and see if that makes a difference
On Aug 15, 2011, at 8:40 PM, o o wrote:
Trying to make this work, and Office 365 support is useless, giving me the
following response
for configuring CCM 8.0 with Office 365, it states that they
support 711ulaw . I also tried setting dtmfmode=auto/rfc2833/info/inband with
no luck.
Trying to get someone with a brain at MS to work with me on this.
From: Alex Vishnev alex9...@gmail.com
To: o o bj_5...@yahoo.com; Asterisk Users
you can monitor queue_log file for ADDMEMBER or REMOVEMEMBER events. from that
point on, you can store them or take any other action.
the other way is to use AMI an monitor for Agent login/logoff events
On Aug 12, 2011, at 7:06 AM, Michael wrote:
Hello,
Is there a way to either store
wrote:
Am 18.07.11 16:15, schrieb Alex Vishnev:
I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an
attended transfer. The transfer is going to an outbound number (normally AA
on another IP PBX). the audio on the first transfer is fine. But if the user
requests
I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an
attended transfer. The transfer is going to an outbound number (normally AA on
another IP PBX). the audio on the first transfer is fine. But if the user
requests a transfer from AA to another department, I loose audio
Does anyone know why i would get this RINGNOANSWER events in queue_log when
clearly the agent is busy and call-waiting is disabled.
1309550595|1309550570.399965|2253|Local/05@from-internal/n|CONNECT|2|1309550593.399966|0
Hello
I have a small call center with about 7 queues. all agents are dynamic and they
login to each queue via a dialplan. When you perform queue show you will see
that all agents are able to service all queues. All queues have the same
weight/priority. While monitoring a system I can see that
Does anyone know why i would get this RINGNOANSWER events in queue_log when
clearly the agent is busy and call-waiting is disabled.
1309550595|1309550570.399965|2253|Local/05@from-internal/n|CONNECT|2|1309550593.399966|0
I have a Bridged call with 2 parties. I want to redirect one party to a
conference room and the other party to an outside number. I tried doing that
with a dialplan. I used ChannelRedirect in the dialplan and redirected the
first channel to the conference room. however, the second channel
Hello,
I am implementing a small ACD system on Asterisk 1.6.2.17.2 I need help with
ChannelRedirect. I have a caller connected to an agent. The agent may request
additional help by consulting another department. I can't use manual process
with blind or directed transfer as the agent have many
Doug,
If you stop complaining and listen to what people are saying, you would be
able to accomplish your goals. Some of your points have merit, but you are
asking for help in all the wrong ways. Please remember, this list is for
users, not developers. The user community is quite extensive in
Adam,
I personally think that replacing hard-wired network and going with Sats is
a mistake. Judging from pure round-trip delay you measured the packet round
trip seems sufficient to have a good conversation, but pinging is not enough
to trouble shoot the network problems. You will need to do a
I briefly looked thru the code and I don't believe there is a way to
separate the context or really make them independent. I know exactly what
you want to accomplish. I think it could be done with a little trick. For
example, every customer on hosted pbx would be given some kind of unique
Actually that is not true. You can have a short time where audio path is
open prior to answering of the call. This depends on the provider, switch
and software. I think the largest window I have seen is 90 seconds.
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
-Commercial Discussion
Subject: RE: [Asterisk-Users] custom ring tone
Audio both ways? Sure would beat the collect call game :P
On Thu, 2005-09-22 at 21:15 -0400, Alex Vishnev wrote:
Actually that is not true. You can have a short time where audio path is
open prior to answering of the call
sipsak (www.sipsak.org. ) is an excellent tool for
this.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan
Sent: Friday, August 26, 2005
10:48 AM
To: 'Asterisk Developers Mailing
List'
Cc: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Hello,
I am using asterisk CVS-head from 6/28. I am also using chan_oh323 that
comes with asterisk. I tried to place a call from h323 device into asterisk.
in extensions.conf, I routed the call to my sip phone. The sip phone was
already registered with asterisk. all the signaling looks ok to me.
Charlie,
I am interested. Can you contact me off-list with details.
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos A
Maimone @ GAUSS
Sent: Thursday, June 23, 2005 3:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
You may also want to check the following link
http://www.voip-info.org/wiki-MSN%20PHP. This is work in progress, but I
think it may help you. it is based on IM messaging protocol to/from MSN
Messenger. I don't believe there is a redirect to hard phones, but I think
that could be part of command
Mike,
The cable needs to be a cross-over cable when connecting directly between 2
T1s, bypassing PSTN. One side of isdn has to be configured as TE and the
other as NT. Only 4 wires are needed (not full 8 wires) to build a T1
cross-over. If you are connecting the systems thru pstn, you need
Pizco,
SER is definitely better suited to deal with NAT issues then ASTERISK is. I
suggest looking at SER and NAT helpers like media proxy application (part of
SER). I also recommend looking at NAT devices at SER wiki page to make sure
that your router/nat device is compatible. In general, this
I think the word crap is a pretty strong word and is not fare to the
authors. Everyone have their own requirements of how billing should or
should not work. Everyone is exposed to a different way a pre-paid calling
card platform should behave. I have been in pre-paid environment for almost
15
Eivind
Most obvious solution is snmp. Using snmp you can collect statistics and
provision your remote systems. However, SNMP is an enabler and not the full
solution. You still need to write SMUX agents and develop application MIBS
that allow you to get/store application specific data. To my
Tomas,
Yes, BT100 is a little picky on the use of Stun Servers. For example, it
will not work at all with Vovida Stun server. Also, Stun negotiation takes
some time. So if you rebooted the phone, I would suggest waiting 15-30
seconds until phone syncs up with Stun server and requests binding. You
source Stun servers are there? And if there are none, what
commercial one have you found to work well with BT100?
Thanks again,
Tomas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Sent: Saturday, April 16, 2005 12:00 PM
To: 'Asterisk Users
First of all I hope you realize you can't have the same context activated at
the same time for the same host as * does not support this. So I am just
thinking the configuration below are just examples of what you tried. I
strongly suggest using dtmfmode=rfc2833 and dtmfmode=info instead of inband.
I think there are a couple of things you can do:
1. Switch the provider to get a stable internet connection ;-)
2. convert your lookups to IP addresses instead of domains. However, if you
clients register with address like [EMAIL PROTECTED], then dns will be used to
resolve blah.com and then
Try setting externip=(asterisk public ip address)
Hth
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo
Arratia
Sent: Friday, April 15, 2005 12:56 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users]
-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk behind NAT
I have... Externip=x.x.x.xand nothing... Does not seem to help in
anything. Still my provider sees the private IP.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Vishnev
Can you capture Ethernet traffic with ethereal or similar tools and show
what is happening?
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
http://www.0xdecafbad.com
Sent: Thursday, April 14, 2005 1:56 AM
To: asterisk-users@lists.digium.com
I have seen the same problem as well. If don't think this is a problem with
BT100. I think the problem is with public STUN server. I think sometimes,
the server is too overloaded and can't provide the translation. That is when
you are getting the problem with your clients behind NAT. the only
The demo does not seem to be working, I am doing something wrong. It is
constantly complaints that file placed in 'File' field is not found. Please
let me know how to resolve this.
Thanks
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of lenz
Sent:
Magnus,
As far as I remember, Festival is only Text-to-speech, not voice
recognition. In order to do what you want you need a voice recognition
application. Also, compression gives voice recognition quite a challenge, as
the speech samples arriving at the voip voice recognition engine is not the
If you setup host=dynamic in sip.conf, then the registration does not depend
on ip address. It depends on sip user name of sip URI. You need to provision
sip user name inside each phone. Please bare in mind that it is different
then sip authentication name.
Hope this helps
Alex
-Original
Scott,
First, you need to get the most recent os
for the pix, otherwise you will have a lot of problems with udp packets and
translations (including bad checksum on your udp packets). I am running both
pix515 and pix501 without a problem with sip and h323. you dont need to
open any
: Re: [Asterisk-Users] Registration to multiple GKs
Is it possible to run Asterisk with another GKs using Neighbor mode?
If it is possible, we can run asterisk with several gnugks.
On Apr 2, 2005 10:41 PM, Alex Vishnev [EMAIL PROTECTED] wrote:
I don't think you can. The rules of h323 is so
Cenk,
Are you sure that remote will handle H245
tunneling? If the remote does not know how to do that, you will get transport
failure. I would suggest doing FastStart instead and
see if you are getting the same results. Of course, you can verify that the
remote can handle faststart as
I don't think you can. The rules of h323 is so that you can register with a
single gk at a time.
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of VoIP Newbie
Sent: Saturday, April 02, 2005 6:37 AM
To: Asterisk-Users@lists.digium.com
Subject:
I have seen that before when you mismatch the type of pri flavor. For
example, if you carrier gives you 4ess and you put 5ess in your config.
There are subtle differences in packets. I would check the configuration on
your carrier side and * side.
-Original Message-
From: [EMAIL
Clay,
It looks like you have the order of the codecs in [general] section as g729, then ulaw. Try reversing them and see if it helps. You may also
view the order in the friend section as well. If that works, you may have to
setup 2 peers in sip.conf. one
for faxing with ulaw, and one
You dont have any codecs configured in your oh323 conf. also FastStart with H245 tunneling should be enabled to get the
best call-setup out of h323.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cenk Yabas
Sent: Thursday, March 31, 2005 7:18
AM
To:
Checkout http://www.voip-wiki.org as it relates to asterisk. There are a
number of useful guides on how to setup and run asterisk. Btw, all the
config files should be located in /etc/asterisk. RH9 should be fine to run
asterisk.
Alex
-Original Message-
From: [EMAIL PROTECTED]
Simon,
I am not sure if I understand you question
properly. However, you can configure password for each user (peer or friend) in
corresponding channel configuration file (i.e. sip.conf)
HTH
Alex
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon
Stephen,
You should be able to setup what you want. For example, asterisk sip peer
will register with your provider. The IP/analog phones will attempt outbound
calls which will be sent to this provider. What you need to determine is how
your provider bills for the calls. If they bill flat, then
All,
I need some help figuring out the best way to write applications for
asterisk. I am trying to implement something similar to astcc pre-paid
application where the application will need to play voice prompts, collect
tones and perform queries over TCP sockets. It will also need to redirect
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