Hi Carlos
Le 07/08/2020 à 06:33, Carlos Chavez a écrit :
I am having a strange problem with a new provider. We already
have a couple SIP trunks working fine. We are trying a new provider
but we are having one way audio problems with outgoing calls. Incoming
calls do have two way audio,
On Mon, Feb 24, 2020 at 10:59 PM Ira wrote:
> Hello Asterisk,
>
> I've been running a CENTOS 5 box with Asterisk 14 and am trying to
> move to Asterisk 17.2 on a new Fedora Server 31 box. I built Asterisk
> from Source as I've always done and copied all the configuration files
> and other stuff
On Sat, 15 Aug 2015 12:42:38 -0300
Joshua Colp jc...@digium.com wrote:
I am not sure why this hasn't bit anyone else. Perhaps most
Asterisk systems are in one of two classes, connecting to all NAT
phones or connecting to all public phones, and I am in a minority
situation where I am
Not 100% ure, but maybe play with the canreinvite or directmedia settings.
On Wed, Aug 12, 2015 at 3:10 AM, D'Arcy J.M. Cain da...@vex.net wrote:
I have been banging my head against the wall for weeks now on this
one. I have a switch running NetBSD and Asterisk 11.19.0 although I
have had
On Sat, 15 Aug 2015 16:30:39 +0800
Michael Dupree mich...@easybitllc.com wrote:
Not 100% ure, but maybe play with the canreinvite or directmedia
settings.
Yes! That was it. Just for future searches here is what I did. I
added directmedia = no in sip.conf. This fixed the issue.
I believe
On Sat, Aug 15, 2015, at 12:08 PM, D'Arcy J.M. Cain wrote:
On Sat, 15 Aug 2015 16:30:39 +0800
Michael Dupree mich...@easybitllc.com wrote:
Not 100% ure, but maybe play with the canreinvite or directmedia
settings.
Yes! That was it. Just for future searches here is what I did. I
added
Hi D'Arcy
that the server IP for RTP as specified in the initial SIP is correct?
Both the server and client are outside of NAT so I don't know what this
might mean. They both have public IPs.
This was a problem we had when the RTP server negotiated in SIP with our
VOIP ITSP on one side of the
On Thu, 13 Aug 2015 10:41:31 +0200
Stefan Viljoen viljo...@verishare.co.za wrote:
Have you checked your RTP port ranges (I'm sure you have), and also
Yes. The ATA is using a range well within the range open on the server.
that the server IP for RTP as specified in the initial SIP is correct?
On Tue, Aug 11, 2015, at 04:10 PM, D'Arcy J.M. Cain wrote:
I have been banging my head against the wall for weeks now on this
one. I have a switch running NetBSD and Asterisk 11.19.0 although I
have had this problem on older versions as well. I, and my users, can
call out, we can receive
On Friday 21 Nov 2014, Andrew Colin wrote:
Hi All
We have a strange issue with our hosted asterisk server running on Debian
Internal calls btween extensions using g729 give one way audio
As soon as we change the codec to ALAW the issues goes away.
Any ideas how to fix this?
Outbound
You probably do not have enough g729 channels license.
On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Friday 21 Nov 2014, Andrew Colin wrote:
Hi All
We have a strange issue with our hosted asterisk server running on Debian
Internal calls btween extensions
Subject: Re: [asterisk-users] One way audio internal
You probably do not have enough g729 channels license.
On 21-Nov-2014 4:17 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Friday 21 Nov 2014, Andrew Colin wrote:
Hi All
We have a strange issue with our hosted asterisk server running
[mailto:mi...@enterux.in]
*Sent:* Friday, November 21, 2014 12:51 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Cc:* Andrew Colin
*Subject:* Re: [asterisk-users] One way audio internal
You probably do not have enough g729 channels license.
On 21-Nov-2014 4:17 PM, A J
I currently am running on a
Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz
Codec im using is
codec_g729-ast18-icc-glibc-x86_64-core2.so
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
On Friday 21 Nov 2014, Andrew Colin wrote:
I am using the free g729
OK, so there shouldn't be any licencing problems (unless for some reason your
Asterisk is wanting to use the paid-for g.729 aot the Free one. Look at the
CLI output very, very carefully to see if this might be happening).
: Friday, November 21, 2014 1:04 PM
To: Andrew Colin
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] One way audio internal
Then something to do with your codec selection priority.
On 21-Nov-2014 4:26 PM, Andrew Colin and...@convergedgroup.net wrote:
I am
port: 65021
Thanks again for your time!
Kind Regards,
Gary Shergill
- Original Message -
From: Amit Patkar a...@avhan.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, May 21, 2014 4:55:57 PM
Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external
asterisk
Hi Gary
You need to check if ICE / STUN is configured.
How are these extensions configured? If you are in private network, you
might have to disable DirectMedia / reInvite for calls going between 2
asterisk boxes.
I hope this helps to resolve your issue.
*Thanks Regards,*
Amit Patkar
On
:50 AM
Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)
Hi Gary
You need to check if ICE / STUN is configured.
How are these extensions configured? If you are in private network, you
might have to disable DirectMedia / reInvite for calls going between 2
asterisk
Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 21, 2014 3:36:54 PM
Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)
Hi Amit,
ICE/STUN is configured correctly. The extension for the webrtc user is defined
in sip.conf on the asteriskrtc.local server
-users@lists.digium.com
Sent: Wednesday, May 21, 2014 04:41:50 AM
Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)
Hi Gary
You need to check if ICE / STUN is configured.
How are these extensions configured? If you are in private network, you
might have to disable
-Commercial Discussion
asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Sent: Wednesday, May 21, 2014 3:36:54 PM
Subject: Re: [asterisk-users] One Way Audio with WebRTC (with external
asterisk)
Hi Amit,
ICE/STUN is configured correctly. The extension for the webrtc user
Dear
in normal mode, .call files make a call between the system and who you named
remote person, I don't know where are you?
in natmode=yes, set qualify=yes.
check the negotiated codecs also.
Best
On Sat, Aug 13, 2011 at 1:29 AM, Carlos Chavez cur...@telecomabmex.comwrote:
We are having
I've been having a similar (well exactly the same) problem this last
week and have been bashing my head trying to fix it.
Just one question, are you using RealTime?
Ish
On Wed, 2011-03-09 at 17:40 -0500, Tim King wrote:
I am having trouble with no return audio on inbound calls. I have been
Just fixed our problem with
directmedia=no
but this only applies if your extensions are behind a nat
Ish
On Thu, 2011-03-10 at 09:40 +, Ishfaq Malik wrote:
I've been having a similar (well exactly the same) problem this last
week and have been bashing my head trying to fix it.
Just
On 10 March 2011 11:17, Ishfaq Malik i...@pack-net.co.uk wrote:
Just fixed our problem with
directmedia=no
but this only applies if your extensions are behind a nat
Ish
There are several reasons why directmedia=no might be the correct
configuration.
1) NAT - probably the most common
My message with the configuration attached is awaiting moderator approval. I
will try to paste relevant data here.
*sip.conf*
[general]
context=inbound ;
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
dtmfmode
Also it could be the routing issue as well.
--
Sent from my iPhone
On Mar 9, 2011, at 7:43 PM, Duncan Turnbull dun...@e-simple.co.nz
wrote:
So that suggests audio is flowing as follows in a unidirectional
manner
199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732
It looks like the issue was my provider enforcing a codec translation that
was not working.
On Thu, Mar 10, 2011 at 9:21 AM, Satish Patel satish...@hotmail.com wrote:
Also it could be the routing issue as well.
--
Sent from my iPhone
On Mar 9, 2011, at 7:43 PM, Duncan Turnbull
Still not working now that audio is restored jitter makes it inaudible? I
am ready to move this to commercial if someone can tell me how I need to pay
for support,
Thanks
Tim
On Thu, Mar 10, 2011 at 10:19 AM, Tim King t...@compnetwork.net wrote:
It looks like the issue was my provider
So that suggests audio is flowing as follows in a unidirectional manner
199.173.66.22.53102 74.204.4.5.11732 IP 74.204.4.5.11732
209.216.2.203.60362
Somewhere near the end this pops up which is slightly different, I am guessing
74.204.4.5 is your asterisk box
19:18:36.389548 IP
209.216.2.203 is sip signaling server and 199.173.66.22 is media servers.
BTW Did you try config_1 option. Please send us your configuration and we
will help you configure it properly. Either you can post them here or you
can send them directly to contact-supp...@didforsale.com
Jai
You can use this link too.
http://www.didforsale.com/blog/how-to-setup-your-asterisk-server-with-didforsale
Keep the context as
context=from-trunk.
-Jai
On Wed, Mar 9, 2011 at 5:01 PM, Jai Rangi jpra...@didforsale.com wrote:
209.216.2.203 is sip signaling server and 199.173.66.22 is media
On 09/16/2010 06:58 PM, Thomas Johnson wrote:
I am having a one way audio issue with xlite clients behind NAT. They
can connect to the server and make calls but no audio is heard on the
other end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
the client that is behind nat is
[tomfmason]
type=friend
secret=secret
callerid=Thomas Johnson
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip
do I have to enable nat on all of them?
On Thu, Sep 16, 2010 at 1:36 PM, Sebastian
On 09/16/2010 07:59 PM, Thomas Johnson wrote:
the client that is behind nat is
[tomfmason]
type=friend
secret=secret
callerid=Thomas Johnson
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip
do I have to enable nat on
I have tried doing that with just ulaw and alaw, respectively, and nothing
changed
Also, if I disable the firewall in my router I lose incoming audio and
outgoing audio works.
On Thu, Sep 16, 2010 at 2:50 PM, Sebastian s...@open-t.co.uk wrote:
On 09/16/2010 07:59 PM, Thomas Johnson wrote:
On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson tomfma...@gmail.com wrote:
Also, if I disable the firewall in my router I lose incoming audio and
outgoing audio works.
http://www.aocomputing.net/?p=3
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC:
I already have that covered
[tomfmason]
type=friend
secret=secret
callerid=Thomas Johnson
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip
The server is not behind NAT only the client above is
On Thu, Sep 16, 2010 at 4:59 PM, Paul
On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson tomfma...@gmail.com wrote:
The server is not behind NAT only the client above is
Sounds like a phone (not asterisk) issue then, make sure you have
setup your NAT and port forwarding properly on the client side.
--
Paul Belanger | dCAP
Polybeacon
-A FORWARD
-p UDP --dport 5060 -j ACCEPT
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Date: Thu, 16 Sep 2010 18:45:38 -0400
From: paul.belan...@polybeacon.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] one way audio for xlite
Hi Group,
I was able to resolve the problem by disabling the echo cancellation in a104d
and using the same dahdi config.
Thanks...
- Original Message
From: leonimar cape leo_mac...@yahoo.com
To: asterisk-users@lists.digium.com
Sent: Wednesday, September 15, 2010 6:12:35 PM
Hi Nasir,
Please don't send me direct emails, unless you want to secure my paid
consultancy services or want to do some other business.
For setting up the RTP, you need to do it on your firewall, which is
receiving RTP traffic from these particular IP address. I can't guess how to
do it on your
I am sure you can't achieve what you are trying to achieve here. Simply use
two different extensions instead of one.
Considering how SIP communication works, I believe SIP doesn't allow
multiple registrations like this. Maybe somebody can correct me here if I am
wrong.
Zeeshan A Zakaria
--
Based on the info you provided (though wireshark analysis will tell more
about it), I am sure what is happening is that rtp coming back from the
called doesn't know which ip to go to, because asterisk knows two ip
addressses for the same extension due to the way you dialed it, i.e. in
ringgroup
One-way audio is mostly firewall problem.
Are you behind firewall ?
You can check the audio-ports that are being used in the SDP-message by
doing a /sip debug/.
Maybe you do not have enough UDP-ports open for the audio ?
Jonas.
On 07/15/2010 04:38 PM, Nasir Javaid wrote:
Hi,
I am
Hi!
I am working on calling 2 registrations of same user on 2 different ip or
ports. It works fine and both phones ring simultaneously. the problem is
that there is one way audio, calling party can hear me but i can't hear
calling party.
You need to make sure that these two phones use
Brent Torrenga wrote:
I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the
localnet, and a trunk to Sipphone/Gizmo/Google Voice. The externhost
and localnet parameters are all set correctly in sip.conf. An inbound
call from Sipphone works great until the local channel places
On Tue, Jul 7, 2009 at 9:55 PM, Paul Edgarp...@tabs.co.nz wrote:
I have a problem with one way audio on Sip and I guess it may be a NAT
issue, in the example below 204 is rung by 208 (xlite external)
I dial perfectly but when I get to the answering of the Asterisk, I can hear
audio from the
a
netstat -an during each call and see what is different.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David @ULC
Sent: Monday, April 06, 2009 6:04 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] One way AUDIO
Can it be that any Port got blocked ?
On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote:
I have a server with 2 Lan Cards.
Now, when I am trying to make calls using Local Lan, its One way Audio
which means customer cant hear me but if I use Static IP with Wan
Connection,
How tcpdump on interface show??
2009/4/6 David @ULC ucoms2...@gmail.com:
Can it be that any Port got blocked ?
On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote:
I have a server with 2 Lan Cards.
Now, when I am trying to make calls using Local Lan, its One way Audio
Few Running figures !!
On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote:
I have a server with 2 Lan Cards.
Now, when I am trying to make calls using Local Lan, its One way Audio
which means customer cant hear me but if I use Static IP with Wan
Connection, it works
On Thu, 16 Oct 2008, GNUbie wrote:
Hello,
On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
A packet trace will probably show exactly what is happening. Try:
tcpdump -nlXs 8192 -i eth0 port 5060
You should be able to see the SIP information going back
GNUbie wrote:
What particular configs are you looking for? Below is my current setup
and scenario:
[snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone]
SNOM is using the 192.168.101.102 IP address
Asterisk is using 192.168.101.1 IP address for its eth1 interface
FXO port
On Thu, Oct 16, 2008 at 09:22:01AM +0800, GNUbie wrote:
Hello Daniel,
On Tue, Oct 14, 2008 at 12:12 AM, Daniel Hazelbaker
[EMAIL PROTECTED] wrote:
Might be a stretch, but does the Asterisk log show that the call was
answered? I had this problem when interfacing * with an NEC system to
Hi,
Am Donnerstag, den 16.10.2008, 09:37 +0800 schrieb GNUbie:
Hello Karsten,
On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote:
Please post Your sip.conf.
Which IP-Address do You configure in the snom for Your asterisk? (eth0
or eth1)?
The SNOM 300 is
Hello,
On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
A packet trace will probably show exactly what is happening. Try:
tcpdump -nlXs 8192 -i eth0 port 5060
You should be able to see the SIP information going back and forth and
will probably show you that your
Hello Daniel,
On Tue, Oct 14, 2008 at 12:12 AM, Daniel Hazelbaker
[EMAIL PROTECTED] wrote:
Might be a stretch, but does the Asterisk log show that the call was
answered? I had this problem when interfacing * with an NEC system to
do call parking pickup. The NEC would never give a dialtone
Hello Karsten,
On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote:
Please post Your sip.conf.
Which IP-Address do You configure in the snom for Your asterisk? (eth0
or eth1)?
The SNOM 300 is using the NET interface beside the DC 5V port to
connect to the LAN.
The
Did you try it the magic number of times, three?
On Sun, Oct 12, 2008 at 9:57 PM, GNUbie [EMAIL PROTECTED] wrote:
Hello Tzafrir,
On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
This means Zaptel gets silence from Asterisk.
What codecs are used? What do you see on
Change all canreinvites to no.
On Wed, Oct 15, 2008 at 9:37 PM, GNUbie [EMAIL PROTECTED] wrote:
Hello Karsten,
On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote:
Please post Your sip.conf.
Which IP-Address do You configure in the snom for Your asterisk? (eth0
or
canreinvite defaults to yes, whether specified or not.
http://www.voip-info.org/wiki/view/tips
If you follow these directions adapting to your particular
circumstances and it doesn't work, post your whole sip.conf
Start asterisk with verbose set to 3 or so and turn on sip debugging.
I get
Hello Steve,
On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Did you try it the magic number of times, three?
I'm sorry. What do you mean?
Regards,
GNUbie
___
-- Bandwidth and Colocation Provided by
Maybe I have my threads confused but I thought you got one way audio
when three calls were made, you only mentioned one call.
On Thu, Oct 16, 2008 at 12:44 AM, GNUbie [EMAIL PROTECTED] wrote:
Hello Steve,
On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Did you try it
Sorry, wrong thread, time for bed. I thought this was the thread
where the guy was having issues with one way audio on his third call,
and his Asterisk server was behind NAT.
Good night everyone and have pleasant dreams of 700 point drops in the DOW!
OT, did you know if the government took the
Hello Steve,
On Thu, Oct 16, 2008 at 12:42 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
canreinvite defaults to yes, whether specified or not.
http://www.voip-info.org/wiki/view/tips
If you follow these directions adapting to your particular
circumstances and it doesn't work, post your whole
On Mon, Oct 13, 2008 at 09:57:33AM +0800, GNUbie wrote:
Hello Tzafrir,
On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
This means Zaptel gets silence from Asterisk.
What codecs are used? What do you see on 'sip show channels'?
I am using the following codecs:
Hello Tzafrir,
On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
So the call is not established yet, right?
It is already. The CALLER hears the CALLEE's voice but the CALLEE
cannot hear the CALLER's voices.
This is not a temporary state?
What do you mean?
Regards,
On Mon, Oct 13, 2008 at 9:04 AM, GNUbie [EMAIL PROTECTED] wrote:
Hello Tzafrir,
On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
So the call is not established yet, right?
It is already. The CALLER hears the CALLEE's voice but the CALLEE
cannot hear the CALLER's
Hello Steve,
On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
If you are going to dismiss (the most probable) problem (NAT) without
posting configs, I am not sure how much help you will get, you will
probably be dismissed as well.
What particular configs are you looking
On Mon, Oct 13, 2008 at 10:49 AM, GNUbie [EMAIL PROTECTED] wrote:
Hello Steve,
On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
If you are going to dismiss (the most probable) problem (NAT) without
posting configs, I am not sure how much help you will get, you will
On Oct 13, 2008, at 9:29 AM, [EMAIL PROTECTED]
wrote:
IME: One-way audio problems are almost always casued by NAT gateways
and/or incorrect NAT settings in sip.conf and/or incorrect IP
address or
extenal proxy settings in the SIP phone.
And reinvite issues in particular. Those have been
Hello Norman,
On Mon, Oct 13, 2008 at 11:02 PM, Norman Franke [EMAIL PROTECTED] wrote:
And reinvite issues in particular. Those have been the only one-way
audio problems I've experienced. Setting reinvite=no fixed everything
for me.
You mean, canreinvite=no? I already have done line on my
Hello Steve,
On Mon, Oct 13, 2008 at 10:59 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
First, drop firewall/iptables/selinux and try again.
I already turned off the firewall and I don't have SELinux on my
system and the problem is still there.
Regards,
GNUbie
A packet trace will probably show exactly what is happening. Try:
tcpdump -nlXs 8192 -i eth0 port 5060
You should be able to see the SIP information going back and forth and
will probably show you that your NAT rules are applying when they
shouldn't. I agree with first turning off your
Might be a stretch, but does the Asterisk log show that the call was
answered? I had this problem when interfacing * with an NEC system to
do call parking pickup. The NEC would never give a dialtone (nor did
it give answer supervision) so * never knew the call got picked up so
audio only
Hi,
Am Montag, den 13.10.2008, 10:00 +0800 schrieb GNUbie:
Hello Gordon,
On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
You mention the SIP phone being inside the LAN. Where is the Asterisk box?
It is the main gateway of the IP phones and my laptop to the
On Sun, Oct 12, 2008 at 11:53:18PM +0800, GNUbie wrote:
Hello all,
I've been lobbying for some time at the #asterisk IRC channel. Until
now, I still can't find a solution to my one way audio problem. I
rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my
Debian Etch. I got
On Sun, 12 Oct 2008, GNUbie wrote:
Hello all,
I've been lobbying for some time at the #asterisk IRC channel. Until
now, I still can't find a solution to my one way audio problem. I
rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my
Debian Etch. I got a Digium TDM400P
Hello Tzafrir,
On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
This means Zaptel gets silence from Asterisk.
What codecs are used? What do you see on 'sip show channels'?
I am using the following codecs:
# asterisk -rx 'sip show settings' | grep Codecs
Codecs:
Hello Gordon,
On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
You mention the SIP phone being inside the LAN. Where is the Asterisk box?
It is the main gateway of the IP phones and my laptop to the Internet.
In this case, the eth1 of the Asterisk box is connected to
Two things you could consider trying:
1) In sip.conf, set the externip and localnet parameters correctly.
2) Also in sip.conf, try the following on the PAP2's sections:
disallow=all
allow=alaw:10
In case that fails, try also
disallow=all
allow=alaw:20
Att
Vinícius Fontes
Desenvolvimento
I have narrowed the problem to a parameter called Symmetric RTP on
the SPA3102. If I disable that I will get the same one way audio
problem as the PAP2T. Unfortunately it seems that the Symmetric RTP
parameter is only available on the SPA3102 and not on the PAP2T. I got
this definition
After many days of testing I finally found the problem. It turns out
that Asterisk was ignoring the externip setting in sip.conf. Today I
decided to enable externhost with the FQDN of the server and magically
the PAP2T started working!
On Thu, 2008-05-08 at 16:38 -0300, Vinícius Fontes
Do you mean the problem is solved using asterisk 1.4.18? Are you
using the setting as mine?
Below is my setting. One way audio is a result after A B connected.
PSTN (A)--1200P-- Asterisk -- GXP2000 --blind transfer -- Extension B
You can see that involve many parties in the blind transfer
I had a similar issue in 1.2 after transfer and we were using SIP only
but an upgrade cured it
We are now on 1.4.18 still without issues
Cheers Duncan
Rilawich Ango wrote:
Hi all,
Recently, I experienced one way audio after call transfer.
incalling call (PSTN) A -- GXP2000 thro' zap
Did you solved this Problem?
I have the same problem, and i can't solve it, did you know anything
about?
Thanks
Nico
On Thu, 14 Sep 2006, Kai Militzer wrote:
Hello everyone,
since some weeks I experience strange problems on my gateways to the
PSTN. The gateways use chan_ss7 and SIP. My
Hi,
I have similar symptoms (usually one-way audio like you, but sometimes
echoed, distorded, or low volume sound), in a simpler configuration,
using just SIP with a few phones and a TDM400 card with two FXOs:
Asterisk -- PSTN
I have kernel 2.6.18-XEN and using Asterisk 1.4
François.
More info: I've noticed that Asterisk CPU utilization has spiked to
100% for a period of 10-20 seconds.
Michael Welter wrote:
The commonality between all sites is Asterisk/Zaptel 1.4.0. The TDM04B
site started reporting this problem after the upgrade to 1.4.0.
So no one has any solution to this, huh? We can't be the only two
people having this problem.
On 10/24/06, Matt [EMAIL PROTECTED] wrote:
Just as a follow up.. on the OTHER server that is connected I'm seeing:
chan_iax2.c: Received VNAK: resending outstanding frames
On 10/24/06, Matt
I am getting the following on my server when the problem happens:
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
within window 209-209
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
within window 209-210
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received
Just as a follow up.. on the OTHER server that is connected I'm seeing:
chan_iax2.c: Received VNAK: resending outstanding frames
On 10/24/06, Matt [EMAIL PROTECTED] wrote:
I am getting the following on my server when the problem happens:
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received
I have same problem, but with 1.4 branch, after several minutes,
asterisk stops sending packets resulting one way audio,
this problem appears especialy when bigger jitter appears (300ms) on
one connection (I have jitterbuffer enabled on IAX),
bigger jitter resulting in bigger one way audio
Have you tried disabling the jitterbuffer? Maybe it is a bug in the
jitterbuffer code, then?
On 10/23/06, Pavel Jezek [EMAIL PROTECTED] wrote:
I have same problem, but with 1.4 branch, after several minutes,
asterisk stops sending packets resulting one way audio,
this problem appears especialy
From our experience, chan_jabber doesnt work behind nat. We tried to
patch it (in a similar way as nat=yes in chan_sip) but quickly bumped
into other problems.
(problems explained on mantis).
Zoa.
Gustavo Hernandez Baratta wrote:
Hi!
I´m trying with 1.4b2, chan_jabber and chan_gtalk.
Hi Zoa:
Thanks for your answer. Let me explain: Asterisk
are not behind a NAT, google talk user are. Do
you think that is the same problem?
Thanks a lot!
gus
At 10:28 a.m. 17/10/2006, you wrote:
From our experience, chan_jabber doesnt work
behind nat. We tried to patch it (in a similar
Yes, its the same as what we tried.
Gustavo Hernandez Baratta wrote:
Hi Zoa:
Thanks for your answer. Let me explain: Asterisk are not behind a NAT,
google talk user are. Do you think that is the same problem?
Thanks a lot!
gus
At 10:28 a.m. 17/10/2006, you wrote:
From our experience,
Hi Kai,
we had a similar problem with a PBX which had PSTN lines and SIP phones:
sometimes some phones had one way calls...the caller couldn't hear. We
hadn't tried to restart but we reduced the number of RTP ports (rtp.conf
if memory helps!) to a range of 200 (it depends from the number of
I would say NAT somewhere misconfigured.
On 6/28/06, leonimar cape [EMAIL PROTECTED] wrote:
Hi Group,
Just want to asked if some of you have experience 1
way audio. Currently I am using two asterisk box. One
handles the prepaid platform, and the other one is for
media gateway connection. I am
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