Displaying 20 results from an estimated 3000 matches similar to: "Sporadic one way audio problem"
2015 Mar 18
2
4 Port PRI
Hi Guys
I have a 4 port PRI card that I need to setup each port in their own
group.
In chan_dahdi.conf I have the following which works for one port
How do I add the rest of the ports in their own groups so that I can have
different signaling on each?
[channels]
language=en
switchtype=euroisdn
pridialplan=unknown
resetinterval=600
echocancel=yes
echotraining=yes
2015 Mar 18
1
4 Port PRI
4 Port PRI sangoma a104
From: jg [mailto:webaccounts173 at jgoettgens.de]
Sent: Wednesday, March 18, 2015 2:09 PM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 4 Port PRI
I have a 4 port PRI card that I need to setup each port in their own group.
In chan_dahdi.conf I have the following which works for one port
How do I add the rest
2009 Dec 30
1
Force Jitter Buffer for SIP to SIP calls
We have a customer on a wireless connection that has very bad jitter. They
can hear people fine, but people have a very hard time hearing them. They
are connected via a SPA-2102.
It is a SIP client going to a SIP trunk.
Something like this in sip.conf [general] would be in effect for all SIP
clients:
jbenable = yes
jbmaxsize = 150
jbresyncthreshold = 1000
jbimpl = fixed
jblog = yes
I only want
2020 Feb 14
1
Predictive call - agent talking to a customer, then suddenly talking to another customer
Hi, do you have NAT between Asterisk and agent phones?
S pozdravem
Tomáš Holý
Hi Tomas
Thanks for replying.
Yes, the phones are in one location in a LAN and are then NATed to enable them to contact the Asterisk which is hosted in the cloud.
A typical sip.conf phone configuration on the remote server for the site is
[general]
session-timers=refuse
disallow=all
allow=g729:20
allow=ulaw
2013 Jun 16
0
define extension to send calls to gatekeeper
hello every one,
i have an asterisk system and want to act as gateway and send calls to
cisco gatekeeper.
this is my h323.conf file:
[general]
port=1720
binaddr=192.168.0.YY
context=from-trunk
faststart=yes
h245tunneling=yes
gatekeeper=192.168.0.XX //cisco address
progress_setup=8
progress_alert=8
dtmfmode=rfc2833
jbenable=yes
jbforce=no
jbmaxsize=200
jbresyncthreshold=1000
jbimpl=fixed
jblog=no
2009 Sep 08
0
Intermittent metallic voice SIP->ISDN ISDN<-SIP
Hi all,
I'm fighting with a really strange problem that is really busting me.
I have an asterisk 1.4.22 ( from a trixbox 2.6.2 ) and mISDN 1.1.7
3 extension on hardphone and 3 extension in softphone ( zoiper )
What happens is that sometimes the people on the other side of communication hear my
voice as metallic and chopped. This happen either on incoming call than on outgoing
call.
If I
2015 Mar 18
0
4 Port PRI
> I have a 4 port PRI card that I need to setup each port in their own group.
>
> In chan_dahdi.conf I have the following which works for one port
>
> How do I add the rest of the ports in their own groups so that I can have different signaling
> on each?
>
> [channels]
>
> language=en
>
> switchtype=euroisdn
>
> pridialplan=unknown
>
>
2014 Mar 24
1
Problem with TLS/SRTP with Asterisk 11.8.1
Hi,
I followed the TLS/SRTP tutorial on the wiki [0] using Asterisk 11.8.1
on CentOS 6.5 x86_64 and CSipSimple on a Nexus with Android 4.4.x local
wifi. The phone seems to register but directly after that things fall
apart (turning SELinux off made no difference):
*CLI> -- Registered SIP 'encrypted' at 10.0.0.137:58079
> Saved useragent
2015 Jan 29
1
JITTERBUFFER function
> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to
call
> > the JITTERBUFFER function?
>
> You only need to use the JITTERBUFFER function.
>
> The jbenable option will enable a jitter buffer on every channel
> created for that peer (or, if global, for every peer in the system).
> Depending on the version of Asterisk, it will also place the
2007 Feb 07
0
Connection problem w/ Attended Transfer
Hi all,
I'm new posting here, though not to perusing. I'm having an issue
with attended transfer and was wondering if anyone had heard of the
problem/had any suggestions... Apologies in advance if this post is
excessively newb-oid.
- An incoming call C is passed to A, a POTS telephone connected via a
Handytone 286 ATA.
- A presses atxfer key, then dials B, a Win XP laptop running
2008 Nov 11
0
help with call with no sound via PSTN
Hello guys, I am having some problems with calls comming from the PSTN
lines, when somebody calls people can't hear me, but I can hear them, every
day I have to do a /etc/init.d/asterisk stop && /etc/init.d/dahdi restart to
have calls with sound again, wich cli dubug commands can I use to see what
is going on, here I have my chan_dahdi.conf and sip.conf, I am using 1.6
Thanks a lot!
2008 Apr 30
0
Jitter buffer not used in SIP -> chan_local -> ZAP path even with /nj for local channels
Hi,
Asterisk 1.4
Working (jitter buffers created as expected):
ZAP -> SIP
SIP -> ZAP
Not working (no jitter buffers created):
SIP -> chan_local (with /nj) -> ZAP
SIP -> chan_local (with /j) -> ZAP
SIP -> chan_local (with no flags) -> ZAP
I have this in zapata.conf:
jbenable=yes
jbforce=no
jbimpl=fixed
jbmaxsize=300
Is there something I haven't tried that will make
2010 Nov 03
1
inbound call issue...
Can anyone tell me why my inbound calls keep getting rejected with 401?
Here's the debug information:
<--- SIP read from UDP:147.135.32.221:5060 --->
INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0
Call-ID: 31007e-31 at 147.135.32.221
CSeq: 1 INVITE
From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc
To: "Gregory Malsack"<sip:s at
2014 May 27
0
dahdi-dahdi native bridging and audio level
Hello!
I use asterisk with TE420 as PRI switch for two channels :
;panasonic uplink
group=3
context=panasuplink
; relaxdtmf=yes
; immediate=yes
rxgain=0.0
txgain=0.0
mohsuggest=default
jbenable = no
; jbenable = yes
; jbmaxsize = 200
; display_send=name_initial
display_send=name
2008 Feb 08
1
(no subject)
Hi,
I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also.
But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized
2011 Sep 14
1
Sip re-register / delay problem.
Hello,
For the moment I have the following settings in my sip.conf. I want to
optimize them to archive the following things:
- for the moment all my users will re-register too often. I want that only
lagged users to re-register quickly.
- check from time to time all users but no too often to see if is logged and
can be called.
Overall i want only lagged users to reregister and users with good
2007 Dec 27
1
SIP Channel jitter buffer issue
Hi,
I have a SIP client which is registered to asterisk. Asterisk is
registered to a SIP trunk and also handles the media. Now since my client
has some issues in its RTP Tx, which seems to have some amount of jitter
(mean jitter as per ethereal trace is about 17ms, max jitter is 20 ms and
max delta is 85 ms), to over come that I have enabled jitter buffer in the
SIP channel by setting sip.conf
2014 Jul 02
1
Webrtc Not acceptable here
Hi,
I am getting
*Can't provide secure audio requested in SDP offer*
with sipml5 client hosted on my local system
[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=sameer ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF
2015 May 11
0
"Retransmission Timeout" results in dropped calls after 32 seconds
----- Original Message -----
> From: "Joshua Colp" <jcolp at digium.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
> Sent: Monday, May 11, 2015 1:24:53 PM
> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds
>
> > Could this
2013 Jun 02
1
Asterisk T.38 Pass-Through doesn't work
What I have is:
* Asterisk 1.8.10.1~dfsg-1ubuntu1,
* SPA112 ATA with analog fax in 1-st FXS port connected,
* SIP trunk with provider supporting T.38.
My network looks like this:
* spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in
neighbouring LANs,
* Asterisk connects to the provider (80.75.130.136) via router
(82.200.7.184). Router has full DNAT to Asterisk server.
What happens?