Displaying 20 results from an estimated 300 matches similar to: "Problem loss 2 seconds audio when Packet2Packet bridging"
2003 Nov 13
2
IAX trunk monitoring
I have an issue where * tries to route a call over IAX to another server
even if the server is down. I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output. If someone could
tell me what I have configured incorrectly, I would appreciate it.
Thanks,
Stephen
-----------iax.conf on voip2----------
[voip1]
type=friend
username=voip1
host=x.x.x.x (ip
2004 Jul 15
1
zapras - and kernel ??
Hi,
I'm trying to get zapras do work, I had downloaded the pppd-source and the 2
patches.
I succefull compiled and install the patched version of pppd, but got this
error in message-log
Jul 15 11:43:32 voip1 pppd[9296]: In file /etc/ppp/filters: unrecognized
option 'active-filter'
Jul 15 11:43:57 voip1 pppd[9299]: Plugin zaptel.so loaded.
Jul 15 11:43:57 voip1 pppd[9299]: Zaptel
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,
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
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
-------------- next part
2006 Dec 13
0
Help with voicemail
I'm looking to use * for a HQ/branch office topology with fairly few calls
over the WAN. The questions I have all pertain to the following
architectural pic: http://www.45891.com/misc/arch.jpg
I'm looking at a distributed architecture so users are somewhat functional
when the link to HQ is down, with a centralized voicemail server to allow
for transfer of voicemail messages from user to
2006 Oct 27
1
Iax bug ?
Hello,
I'm french, so excuse my poor English.
I'm face to a terrible thing, with has stole a lot of my time.
On the .184 machine, I've the following iax.conf :
[general]
rtcachefriends=yes
bandwidth=high
tos=reliability
jitterbuffer=no
autokill=yes
#include "iax.voip1.conf"
#include "iax.renoir.conf"
The iax.voip1.conf file contains :
[VOIP1]
type=friend
2009 Apr 27
1
Packet2packet bridging while in sip.conf canreinvite=no
I have put canreinvite=no for all my internal SIP-clients in sip.conf
because I want Asterisk to be in the middle of the RTP-stream so he can
provide MusiconHold and so...
Now, what the Asterisk CLI tells me when I make a call from my one
internal SIP-phone to another internal SIP-phone is :
Verbosity is at least 25
== Spawn extension (intern, 51, 1) exited non-zero on
2008 Apr 02
1
show uptime and last reload
Hi,
I just upgraded from 1.2 to 1.4.
In 1.2, when I did a "show uptime" I used to see a
second line telling me the time since the last reload.
Has this been removed in 1.4?
The following is the output of my two test boxes:
Connected to Asterisk 1.4.18.1 currently running on
voip2 (pid = 10605)
Verbosity is at least 3
voip2*CLI> show uptime
System uptime: 15 hours, 55 seconds
2005 Jun 28
0
Asterisk dies with Meetme
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi List
I'm trying to create a conference room using H323 channels.
If i start asterisk normally (service asterisk restart) and connect to
cli using -vvvvvvvr options, when a user enters the Conference,
asterisk says "You are the only ..." and then dies, withou any error
message, nothing at all.
But, if i start asterisk with cli
2014 Feb 16
0
SIP TLS question for asterisk 11
Hi All,
I'm on a middle of an asterisk installation/configuration for my company
and I'm testing the TLS configuration.
For this reason, I used the ast_tls_cert script to build the ssl
certificates for my server.
On sip.conf file:
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
and on
2004 Apr 26
0
Help with connecting 2 servers via iax
I have successfully configured two servers and I am now trying to connect
via iax. When I attempt to call from one ext, 2006(server viop1) to
extension 3006 (server voip2) I receive a timeout or "call failed 403
forbidden.
The information I am receiving from the console is below.
Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type
registered for 'IAX'
2007 Feb 12
4
Zaptel install...
I am having trouble getting Asterisk to compile the zaptel stuff.
Here are the specifics:
Linux Kernel 2.5.9-42.0.8.EL
Asterisk 1.4.0
I compiled libpri, zaptel, asterisk and asterisk-addons (in that
order). This is a fresh install of CentOS. Following the CentOS
install, I did "yum -y update" until there were no updates left.
Here is my src directory:
drwxr-xr-x 24 root root
2011 Nov 17
0
2 same sip extension number on 2 asterisk - call not passing on certain condition
Hi list,
something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both
having an extension [115], one as type peer (caller side 1.4) and one as
friend (callee side 1.8). Phones from both location connect to Asterisk
from LAN. Router are Linux boxes.
Connection between the 2 sites is done like this:
On the callee side
[115] ;callee
type=friend
host=dynamic
secret=otherSecret
2005 Oct 05
0
Unwieldy outbound macro
I have the following pair of macros defined to handle outbound calls from *.
Rather than specifying full dialstrings in the main body of extensions.conf,
outbound dial commands are made using a macro call as follows:
Macro
(outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate
way4)
The final gateway defined is nearly always a fallback to PSTN if none of the
IAX or SIP
2016 Jan 26
2
Samba Hylafax PAM
O, try the following.
Test this first.
ldd /usr/sbin/hfaxd
if you getting libpam.so.. something, then hylafax is compiled with pam support.
Next,
apt-get install libpam-ldap ( just to be sure, i do believe you have installed it already )
create the file :
/etc/pam.d/hylafax
Add :
auth required pam_ldap.so
account required pam_ldap.so
2004 May 20
4
Mystery SIP channels
Has anyone seen this before? This channel is consistently present on
both of my asterisk servers. Sometimes they disappear for a few seconds
and then come back. It always has the same Call ID.
voip1*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter
Format
192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms
UNKN
2005 Jan 04
2
Asterisk stops - why ?
Hi,
Sometimes my asterisk server stops. (after a day or two)
Last output from CLI is:
--------------------------------
-- Registered SIP '000b82017eb7' at 213.237.12.125 port 11620 expires 120
-- Channel 0/26, span 1 got hangup
-- Hungup 'Zap/26-1'
voip1*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk cleanly ending (0).