Displaying 20 results from an estimated 4000 matches similar to: "Packet2packet bridging while in sip.conf canreinvite=no"
2009 Apr 13
10
Asterisk-beginner : cannot make phonecalls using Asterisk
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.
Thought it would go easier...
I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
These are my settings :
sip.conf :
[root at asterisk asterisk]# cat
2009 Apr 17
15
Here is Step by Step Example of Asterisk PBX System Install and configuration
Our small company is replacing Cisco CallManager with Asterisk (because we are tired of sending them money) and I am documenting the process as I go on my blog. I am trying to make the notes as easy as possible in hopes that I can ease someone else's pain. Here is the link:
http://qvlweb.blogspot.com/2009/03/asterisk-pbx-system-install-01-what-i.html
Please feel free to comment on the
2009 Apr 26
1
file.c:655 ast_openstream_full: File /tmp/winkel-gesloten.alaw does not exist in any format
part of extensions.conf:
exten => 11,1,Answer()
exten => 11,n,NoOp(CallerID : ${CALLERID(all)})
exten => 11,n,Playback(/tmp/welkom-tcs.alaw)
exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)
; wordt doorgerouteerd naar context open, maar indien gesloten :
exten => 11,n,NoOp(Oproep tijdens winkel gesloten)
exten => 11,n,Playback(/tmp/winkel-gesloten.alaw)
exten =>
2009 May 19
5
OT: SIP hardphone with multi-color BLF
Hi,
Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors
(or more) ?
This could be very useful to support extended presence, for instance.
Regards
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2007 Mar 08
1
Packet2Packet Bridging Questions
I'm just starting to upgrade some boxes from 1.2.x to 1.4.1 as well
as trying to get some of the RTP traffic offloaded from the network.
I think I'm misunderstanding what the console messages mean when it
says "Packet2Packet Bridding SIP/blah to SIP/blah". I though that
meant that it had successfully (re)INVITED and the media was no
longer going through my Asterisk
2008 Jan 13
2
Packet2Packet bridging occurring when not wanted
Hi,
I have Asterisk set up on Fedora with a single SIP trunk, with a few
handsets configured. The Asterisk box has both public and private
addressing, so "canreinvite=no" is set on both the SIP trunk and handset
configurations so I can get around the nasty NAT issues.
One odd behaviour I am seeing is certain destinations are resulting in
different SIP codes being sent back to Asterisk,
2009 Jun 26
0
Problem loss 2 seconds audio when Packet2Packet bridging
I'm sorry, i send mail in asterisk-bug, but asterisk-users is better for
my problem
Hello,
During a call with canreinvite = no, at the beginning of the call I lose
2 seconds of audio.
is obvious when I call autoattendant.
schema:
SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk 1.4.24.1)
--> Operator SIP
capture of voip1:
- Executing [0825387205 at
2014 Jun 30
0
Fwd: Regarding packet2packet bridging
Dear concern,
I want to configure packet2packet bridging in asterisk.
How could I do this any of the tutorial or instructions will help ?
I found the setting the canreinvite=yes will do the stuff but it is not
working
I am using asterisk 12.3 version
I am very new to asterisk please help me in doing the same.
Thanks in advance.
--
Regards
Sameer Rathod
8109413462
--
Regards
Sameer
2014 Jul 02
1
packet2packet bridging
Hi,
I am new to asterisk I want to configure my asterisk server such that it
only establishes the call
rest the audio must bypass the server and transmitted directly to the peer
In my config file I did changes which are below
canreinvite=yes
nat=force_rtp
dirtectmedia=yes
directsetup=yes
I am using asterisk version 12.3
--
Regards
Sameer Rathod
8109413462
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2010 May 28
0
Dead air between answer and packet2packet bridge (Bug 12708?)
Hi everybody
Hope I picked the right mailing list. If not, please tell me.
We've got a problem with call forwardings. It's exactly the same problem
as described in bug 12708, which is resolved by now.
Situation: Caller -> asterisk -> call forward to mobile (packet2packet
bridge)
Quote from original bug reporter:
'One issue that we have noticed repeatedly is that there is a
2007 Nov 13
0
Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11
Hi,
I'm using a GXP2000 (that's sharing the same GXP2020 firmware file) with the latest 1.1.5.10 beta release. It's working since a week and seems working very well. Before I was using the 1.1.5.3 and I had no problem. 1.1.4.xx versions, instead, are not performing like that one (audio, deadlocks and other minor issues).
You can find a lot of info and old firmware versions at this
2006 Jun 07
1
Music On Hold not working with new 1.2.7.1 install
I have followed the instructions provided at:
http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conf
including installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now.
However, no sound is heard and I get this message from the CLI when accessing MOH:
-- Started music on hold, class 'default', on channel
2004 Nov 25
1
No Music: Queue Hold and MusicOnHold
Hello,
We are working on a new Asterisk installation and have run into some problems related to playing MusicOnHold for a caller when they have been placed on hold by an agent, that took the call from a queue.
A. When pressing the HOLD button on SNOM 190 and Grandstream BudgeTone SIP phones, MusicOnHold works fine when making inbound or outbound direct calls by extension. Music starts to play
2007 Nov 12
1
Grandstream GXP2020 + Asterisk 1.4.11
Hi,
I`m using several GXP2020 phones with newest Firmware 1.1.4.18.
Asterisk Version: 1.4.11.
It happens several times that users complain that the caller cannot hear the
transmitted voice from the phones....
Also now it happens quite often that callers on hold beeing dropped.
Environment: ISDN with chan_misdn and SIP internal calls. No NAT no DNS name
(only IPS configured).
I configured
2007 Nov 13
0
Fwd: Re: Grandstream GXP2020 + Asterisk 1.4.11
Thx John !!
Hmm I found now on voip-info.org a lot of Beta releases which should fix my
problems... Kind of strange whats going on with Grandstream devices and their
firmware ... If you install the latest "official" release you can expect a
few troubles with Asterisk 1.4.11 (one way audio --> randomly, dropped
calls). So you have to install the BETAS whether you want or not...
2007 Sep 25
4
Grandstream GXP2020 / 2000
Hi,
Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a
business graded installation (with really traffic on .... not 3 calls a
day ;-) )
Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall)
Thanks!
Kind Regards,
Erik
2003 Nov 07
8
Putting call on hold
Is there a way to put a call on hold and play music on hold with out
using the park app?
Thanks,
-gcc
2007 Nov 08
2
asterisk and installing chan_h323.so rpm
Hello,
When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors.
Failed dependencies:
libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386
libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386
But i found the same files in
/usr/lib/libh323_linux_x86_r.so.1
/usr/lib/libpt_linux_x86_r.so.1
What to do for asterisk to detect the same
2003 May 13
5
Music on hold, Call Parking, etc
Ok, this falls under the newbie category:
Has anybody created any documentation on using musiconhold or call parking?
I have found sample config files for musiconhold, but I'm not sure how they
work.
[musiconhold.conf]
[classes]
loud=>mp3:/var/lib/asteriks/mohmp3
How do I then set up this feature in extensions.conf?
I can't seem to find an example of what I'm looking for (or I
2017 Apr 06
2
feature codes
hi.
i am running asterisk 11 and i am stuck with the feature codes. how do i
setup them.
Now the system has.
PBX*CLI> features show
Builtin Feature Default Current
--------------- ------- -------
Pickup *8 *8
Blind Transfer # #
Attended Transfer
One Touch Monitor
Disconnect Call * *
Park Call
One Touch MixMonitor
Dynamic Feature Default Current
--------------- ------- -------
(none)
Feature