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,
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, only outgoing calls have this problem. I do not
see
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
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 from the old box. Everything comes up as expected and
it
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
-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] One way audio - doesn't seem to be NAT issue
Message-ID: 20150811151044.79872ce9@imp
Content-Type: text/plain; charset=US-ASCII
Given that both of us can make and accept calls and the server is simply
connecting two
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
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 calls, quality is excellent but I cannot talk
with one user. The
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 calls via a trunk work fine with g729
Kind
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,
I've run into a slight issue when using WebRTC and two Asterisk boxes.
I am using SIPml as the test WebRTC client.
My two asterisk boxes, one of them is configured for WebRTC with websockets,
etc (asteriskrtc.local) and the other is just a standard asterisk server
(asteriskgary.local).
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
I'm migrating from Asterisk 1.6.2 to 10.7.0. In 1.6.2, I made a small
patch to allow specifying an address for RTP media. That worked. In
10.7.0, this appears to be built in with media_address, but it doesn't
work for me.
My Asterisk server has multiple addresses, all global address on two
Hi,
On one of our locations, I am having issues with one-way audio when I call
several phones with SIP/Phone_ASIP/Phone_BSIP/Phone_C. When I call the
phones individually, they work fine, so it's not a volume setting on the
phone. Also this setup has worked at other locations.
Any idea's what to
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
We are having a problem when trying to use originate or AMI to make a
call. We have an Asterisk 1.8.5.0 server which uses a SIP provider to
call the PSTN. When dialing from IP phones everything works fine. When
you try making the call with originate, AMI or a call file then the
remote
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
I am having trouble with no return audio on inbound calls. I have been
working on this for 18 hours and even built a fresh server and moved
everything over and I am getting the same results. I need someone that can
help get this resolved tonight. I can provide access to all machines
involved.
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
Hi,
Asterisk is making a call to a peer. In 200 ok, peer is sending its
application server ip in contact field, so asterisk sends ACK to that IP.
RTP starts flowing between endpoints and peer plays an IVR and asks for
destination number. After entering destination number peer's application
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
canreinvite=no[tomfmason]
type=friend
secret=secret
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 am currently facing a dead end and any help will be much appreciated.
I have an a104d installed in an asterisk box, two of which is configured on
ISDN
pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am
getting one way audio when a local on the hipath
Subject: [asterisk-users] One way audio when overlapdial is set to yes
Hi Group,
I am currently facing a dead end and any help will be much appreciated.
I have an a104d installed in an asterisk box, two of which is configured on
ISDN
pri. One is facing pstn and the other one is facing
Hi again
today when i was doing my research on this issue i found that even dialing a
sip user by it's IP also raises this problem. here is what i did,
First I dialed my registered user in normal way like this,
Dial(SIP/XYZ,30,rtT)
and during conversation audio was OK in both ways. Then I
sorry for the typo mistake. the actual dial string that I used is like this
Dial(SIP/x...@192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/x...@192.168.0.12:64290-0966ab80,30,rtT)
it is not
Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT)
it was just a
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
thanks a lot zishan and philipp,
probably that is the problem that is occurring. I am gonna take some
wireshark or etherial trace to further investigate the problem.
i don't wanna stuck into port forwarding issue as it will waste lot of time
and also normal calling is working on my current port
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
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.
here is the scenario..
SIP/x...@192.168.0.20:5060
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
Hello all,
I have a problem where problem with one way audio, and I think it's
related to a=sendonly and a re-invite. Can anyone please assist?
The scenario is as follows
- We send an INVITE to a peer, and it replies with a 100 Trying, and
then a 183 Session Progress message containing
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 the call on hold. During
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
Hello list !
I'm having one way audio on incoming and outgoing calls. Outgoing audio
works fine, incoming audio is not working.
My setup is the following :
incoming calls :
PSTN -- FXOport -- HT503 -- WANport -- Asterisk -- WANport -- HT503 (the
same) -- FXSport -- DECTphone
outgoing calls :
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 Asterisk but cannot get audio to the Asterisk, ie If
I ring the
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
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 perfectly.
I changed the network from loc1 to loc2 but its same.
I tried changing Ethernet Card but no
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
Hi,
I have a couple of users who are having a peculiar problem.
On some outbound numbers where there is a deep IVR tree (3+
selections), and then a live person picks up,
the live person will be unable to hear them on the phone, but they can
hear the live person.
I've done packet traces and it
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
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