Re: [asterisk-users] One way audio on outgoing calls

[Administrator]

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, only outgoing calls have this problem.  I 
do not see anything odd with a packet capture and using PJSIP history 
to check.  The provider says that on outgoing calls the get random 
characters instead of the media port for RTP.


    We are using Asterisk 16.12.0 with PJSIP.  The server is behind 
NAT so we have external_media_address and external_signaling_address 
set to the public IP and all relevant ports are forwarded to the 
Asterisk server.  The other SIP trunks work fine, only this new 
provider has a problem and only for outgoing calls.


    An rtp set debug on shows only outgoing packets to the media 
address but no incoming packets.  Why would there be a difference that 
makes it work on incoming calls but not on outgoing?


We faced this problem and it was a firewall issue on our side. But if 
you say that your provider doesn't get the RTP, I understand that they 
can't return anything. RTP ports ?


Cheers

--
Daniel

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio on new build

[Joshua C. Colp]

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 from the old box. Everything comes up as expected and
> it all seems to work except I have one way audio. I'm still using SIP,
> not pjsip. As soon as I put the old box back the one way audio problem
> is gone. Any suggestions where I should look?
>

Is a firewall running on the new system? Have you examined the SIP traffic
to make sure the right IP addresses are present (sip set debug on)? Which
direction is there no audio?

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!

[D'Arcy J.M. Cain]

On Sat, 15 Aug 2015 12:42:38 -0300
Joshua Colp  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 talking to a mix of setups.
> 
> Most people run without direct media unless they know the network
> topology will allow it 100%.

Perhaps but the default is to run it.  Perhaps the default should be
"no" to prevent these problems.

On the other hand, the documentation seemed to suggest that the default
should have worked anyway.  One leg was public, the other behind a
NAT.  It should recognize the latter and not try to put then in direct
contact.  It's almost like it saw the public one and didn't bother
checking the other.  Or, it checked both with an OR instead of an AND
as I said.  That seems more likely since it didn't matter who started
the call.

I don't really care at this point.  If 1% of the calls go through the
server when they didn't really need to it's no big deal.

Cheers.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!

[Joshua Colp]

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  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 that Asterisk was getting confused when one leg was inside
> NAT and the other was outside.  Perhaps there was an "OR" where there
> should be an "AND".  It makes sense because the other user was the one
> outside NAT and he could hear me and I could not hear him no matter who
> initiated the call.  He could make outside calls because both he and my
> provider were on public IPs.
> 
> 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 talking to a mix of setups.

Most people run without direct media unless they know the network
topology will allow it 100%.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue - SOLVED!

[D'Arcy J.M. Cain]

On Sat, 15 Aug 2015 16:30:39 +0800
Michael Dupree  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 that Asterisk was getting confused when one leg was inside
NAT and the other was outside.  Perhaps there was an "OR" where there
should be an "AND".  It makes sense because the other user was the one
outside NAT and he could hear me and I could not hear him no matter who
initiated the call.  He could make outside calls because both he and my
provider were on public IPs.

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 talking to a mix of setups.

Thanks for that.  I was going nuts trying to figure this out.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue

[Michael Dupree]

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  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 calls, quality is excellent but I cannot talk
> with one user.  The different elements are as follows:
>
> The switch as described above which is in a server room on the Internet
> backbone with a public IP address.
>
> My home system which is behind a bridged modem through a Linksys
> WRT54GS with priority given to my ATA.  The ATA is a Cisco SPA112.  I
> also have an actual SIP phone.  The problem happens with both.
> Obviously I am using NAT but both devices work just fine if I am going
> to the PSTN.
>
> My user who is also going through a bridged modem to a Linksys SPA-2102
> which is doing the PPPOE so it has a public IP address and no NAT
> involved although it serves NAT for the connected computer.
>
> So here is the problem.  While both of us have no problems externally,
> when we call each other we get one way audio and it is always from me
> to him no matter who initiates the call.
>
> A further test, I can call from the SIP phone to the ATA connected
> phone and vice versa just fine.  That involves two devices behind the
> same NAT but since they still need to use the server as an intermediary
> I can't see how that would matter.
>
> Given that both of us can make and accept calls and the server is
> simply connecting two separate channels I can't see where the problem
> might lie.  Can anyone suggest a possible setup issue?
>
> I have tried so many things but I am willing to try them again.  Feel
> free to make any suggestion no matter how silly.  I really need to fix
> this.
>
> Cheers.
>
>
> --
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:da...@vex.net
> VoIP: sip:da...@vex.net
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Michael Dupree jr.
p: +1-248-935-4147
f: +1-866-671-6867
Skype: MichaelDupreeJr
PGP Pub Key: http://www.michaeldupree.net/?page_id=53










This is a private message. This e-mail message, and any attachments
thereto, is for the sole use of the intended recipient(s) and may contain
legally privileged and/or confidential information. Any unauthorized
review, use, disclosure or distribution is strictly prohibited. If you are
not the intended recipient, please contact the sender by reply email and
permanently delete all copies of the original message.
---
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue

[Stefan Viljoen]

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 connection, differed from the IP we were
expecting on that side of the connection and was blocked in our firewall.

Once we perused the SIP traffic we noted this and added the extra IP to the
firewall for RTP traffic.

>> We had slightly different parameters, e. g. that we would have no RTP 
>> at all, but a call that did connect to total silence, dialed from 
>> either side.

>Was NAT involved?

Yes, NAT was being done at both ends, but it turned out that NATing was not
the problem.

>> Also check what RTP port ranges are being used - I have had this 
>> one-directional problem where the port range in /etc/asterisk/rtp.conf 
>> was too broad, and the firewall on my server was only allowing a 
>> smaller subset of RTP ports.

>rtpstart=1
>rtpend=2

>which is exactly what my packet filter allows through.

I assume you have tried turning your packet filter or firewall off
completely (just for a moment) to see if it helped?


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue

[D'Arcy J.M. Cain]

On Thu, 13 Aug 2015 10:41:31 +0200
"Stefan Viljoen"  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?

Both the server and client are outside of NAT so I don't know what this
might mean.  They both have public IPs.

> Not sure how this will relate to your setup, but we had something
> similar here using Asterisk 1.8.11.0 on both sides of the connection,
> via a VOIP service provider in the middle.

This is an Asterisk server talking to an ATA.

> We had slightly different parameters, e. g. that we would have no RTP
> at all, but a call that did connect to total silence, dialed from
> either side.

Was NAT involved?

> Also check what RTP port ranges are being used - I have had this
> one-directional problem where the port range
> in /etc/asterisk/rtp.conf was too broad, and the firewall on my
> server was only allowing a smaller subset of RTP ports.

rtpstart=1
rtpend=2

which is exactly what my packet filter allows through.

> It might require some careful tracing of SIP messages, maybe you can
> try this? Specifically try to determine what RTP port number is being
> negotiated when you have your zero-audio back from the remote party -
> what RTP port and RTP server IP is he using at that moment on his
> side?

I will check that.

Thanks for your suggestions.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio - doesn't seem to be NAT issue

[Joshua Colp]

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 calls, quality is excellent but I cannot talk
> with one user.  The different elements are as follows:



I'd suggest getting a packet capture to see the RTP traffic to see the
actual path of things, not just thinking of what it should be. Media
doesn't just get lost. It's told to go somewhere ultimately and either
that is incorrect for some reason or something is blocking it.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio internal

[Andrew Colin]

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



Kind Regards

Andrew Colin

Converged Data (Pty) Ltd.

Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)





Direct: +27 (0)10 591 4607

Mobile: +27 (0)82 310 3007
Switchboard: +27 (0)10 591 4600
Email: and...@convergedgroup.net

Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). 
Any unauthorized review, use, disclosure or distribution is prohibited. If 
you believe this message has been sent to you in error, please notify the 
sender by replying to this transmission and delete the message without 
disclosing it. Thank you.E-mail including attachments is susceptible to data 
corruption, interception, unauthorized amendment, tampering and viruses, and 
we only send and receive emails on the basis that we are not liable for any 
such corruption, interception, amendment, tampering or viruses or any 
consequences thereof.



From: Mitul Limbani [mailto:mi...@enterux.in]
Sent: 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"  wrote:

I am using the free g729







Kind Regards

Andrew Colin

Converged Data (Pty) Ltd.

Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)





Direct: +27 (0)10 591 4607 

Mobile: +27 (0)82 310 3007 
Switchboard: +27 (0)10 591 4600 
Email: and...@convergedgroup.net

Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). 
Any unauthorized review, use, disclosure or distribution is prohibited. If 
you believe this message has been sent to you in error, please notify the 
sender by replying to this transmission and delete the message without 
disclosing it. Thank you.E-mail including attachments is susceptible to data 
corruption, interception, unauthorized amendment, tampering and viruses, and 
we only send and receive emails on the basis that we are not liable for any 
such corruption, interception, amendment, tampering or viruses or any 
consequences thereof.



From: Mitul Limbani [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 Stiles"  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 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

Unless you have serious bandwidth issues, just forget about g.729 and change
to a-law throughout.  A-law is what the PSTN  (in civilised countries)  uses
anyway, so you won't need to transcode  (which chews up processor resources
and risks compromising quality)  for calls to and from the outside world.

If you really need to use g.729 and are outside the USA  (therefore, beyond
the reach of software patents),  there is a free version that you can use --
and this one, better than Digium's offering, comes with the Source Code so 
you
can be sure it isn't doing anything nasty behind the scenes.

But to be honest, you probably are better off just sticking with a-law.

--
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio internal

[A J Stiles]

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).

Did it ever work properly?  If your kernel, C library or some other 
fundamental system component has been updated since you installed g.729, then 
it might have been broken by the upgrade.  Navigating to the folder with the 
Source Code and re-running `make` followed by `make install` ought to fix it.


But why are you using g.729 anyway?  What special reason have you for doing it 
differently than the rest of the world?


-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio internal

[Andrew Colin]

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 Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio internal

[Mitul Limbani]

Then something to do with your codec selection priority.
On 21-Nov-2014 4:26 PM, "Andrew Colin"  wrote:

> I am using the free g729
>
>
>
>
>
>
>
> Kind Regards
>
> Andrew Colin
>
> *Converged Data (Pty) Ltd.*
>
> *Licensed Telecoms Operator :* (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
>
>
>
> Direct: +27 (0)10 591 4607
>
> Mobile: +27 (0)82 310 3007
> Switchboard: +27 (0)10 591 4600
> Email: and...@convergedgroup.net
>
> Web: http://www.convergedgroup.net
> 75 Witkoppen Road, Northriding, Johannesburg, 2169
> P O Box 7246, Weltevredenpark, 1715
> This communication is confidential and intended solely for the
> addressee(s). Any unauthorized review, use, disclosure or distribution is
> prohibited. If you believe this message has been sent to you in error,
> please notify the sender by replying to this transmission and delete the
> message without disclosing it. Thank you.E-mail including attachments is
> susceptible to data corruption, interception, unauthorized amendment,
> tampering and viruses, and we only send and receive emails on the basis
> that we are not liable for any such corruption, interception, amendment,
> tampering or viruses or any consequences thereof.
>
>
>
> *From:* Mitul Limbani [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 Stiles" 
> 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 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
>
> Unless you have serious bandwidth issues, just forget about g.729 and
> change
> to a-law throughout.  A-law is what the PSTN  (in civilised countries)
> uses
> anyway, so you won't need to transcode  (which chews up processor resources
> and risks compromising quality)  for calls to and from the outside world.
>
> If you really need to use g.729 and are outside the USA  (therefore, beyond
> the reach of software patents),  there is a free version that you can use
> --
> and this one, better than Digium's offering, comes with the Source Code so
> you
> can be sure it isn't doing anything nasty behind the scenes.
>
> But to be honest, you probably are better off just sticking with a-law.
>
> --
> AJS
>
> Note:  Originating address only accepts e-mail from list!  If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio internal

[Andrew Colin]

I am using the free g729







Kind Regards

Andrew Colin

Converged Data (Pty) Ltd.

Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)





Direct: +27 (0)10 591 4607

Mobile: +27 (0)82 310 3007
Switchboard: +27 (0)10 591 4600
Email: and...@convergedgroup.net

Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). 
Any unauthorized review, use, disclosure or distribution is prohibited. If 
you believe this message has been sent to you in error, please notify the 
sender by replying to this transmission and delete the message without 
disclosing it. Thank you.E-mail including attachments is susceptible to data 
corruption, interception, unauthorized amendment, tampering and viruses, and 
we only send and receive emails on the basis that we are not liable for any 
such corruption, interception, amendment, tampering or viruses or any 
consequences thereof.



From: Mitul Limbani [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 Stiles"  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 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

Unless you have serious bandwidth issues, just forget about g.729 and change
to a-law throughout.  A-law is what the PSTN  (in civilised countries)  uses
anyway, so you won't need to transcode  (which chews up processor resources
and risks compromising quality)  for calls to and from the outside world.

If you really need to use g.729 and are outside the USA  (therefore, beyond
the reach of software patents),  there is a free version that you can use --
and this one, better than Digium's offering, comes with the Source Code so 
you
can be sure it isn't doing anything nasty behind the scenes.

But to be honest, you probably are better off just sticking with a-law.

--
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio internal

[Mitul Limbani]

You probably do not have enough g729 channels license.
On 21-Nov-2014 4:17 PM, "A J Stiles"  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 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
>
> Unless you have serious bandwidth issues, just forget about g.729 and
> change
> to a-law throughout.  A-law is what the PSTN  (in civilised countries)
> uses
> anyway, so you won't need to transcode  (which chews up processor resources
> and risks compromising quality)  for calls to and from the outside world.
>
> If you really need to use g.729 and are outside the USA  (therefore, beyond
> the reach of software patents),  there is a free version that you can use
> --
> and this one, better than Digium's offering, comes with the Source Code so
> you
> can be sure it isn't doing anything nasty behind the scenes.
>
> But to be honest, you probably are better off just sticking with a-law.
>
> --
> AJS
>
> Note:  Originating address only accepts e-mail from list!  If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio internal

[A J Stiles]

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 calls via a trunk work fine with g729

Unless you have serious bandwidth issues, just forget about g.729 and change 
to a-law throughout.  A-law is what the PSTN  (in civilised countries)  uses 
anyway, so you won't need to transcode  (which chews up processor resources 
and risks compromising quality)  for calls to and from the outside world.  

If you really need to use g.729 and are outside the USA  (therefore, beyond 
the reach of software patents),  there is a free version that you can use -- 
and this one, better than Digium's offering, comes with the Source Code so you 
can be sure it isn't doing anything nasty behind the scenes.

But to be honest, you probably are better off just sticking with a-law.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

[Gary Shergill]

Hi Amit,

My rtp.conf has the stunaddr listed and icesupport set to yes.

It looks like the issue is that the media isn't being sent from 192.168.3.150 
to 192.168.3.131 (chrome browser to asteriskrtc.local). 

When using asteriskrtc.local to originate the call (make a call directly from 
sipml client to another number on asteriskrtc.local or to a number on another 
asterisk server) audio flows both ways with no issue, it's just when 
asteriskgary.local is originating the call that there is no audio flowing from 
chrome to asteriskrtc.local.

I should probably rephrase the above though to say that on tshark I can 
actually see the packets flowing (tshark host 192.168.3.150):

  2.384874 192.168.3.150 -> 192.168.3.131 STUN 174 Binding Request user: 
15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik
  2.384925 192.168.3.150 -> 192.168.3.131 STUN 174 Binding Request user: 
15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik
  2.385060 192.168.3.131 -> 192.168.3.150 STUN 122 Binding Success Response 
XOR-MAPPED-ADDRESS: 192.168.3.150:60175
  2.385256 192.168.3.131 -> 192.168.3.150 STUN 122 Binding Success Response 
XOR-MAPPED-ADDRESS: 192.168.3.150:65021
  2.394891 192.168.3.131 -> 192.168.3.150 UDP 224 Source port: 16514  
Destination port: 65021
  2.415195 192.168.3.131 -> 192.168.3.150 UDP 224 Source port: 16514  
Destination port: 65021
  2.434063 192.168.3.150 -> 192.168.3.131 STUN 174 Binding Request user: 
15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik
  2.434121 192.168.3.150 -> 192.168.3.131 STUN 174 Binding Request user: 
15bd74963e5dcabb5eda052841431514:EnntMcdgY8Rm17Ik
  2.434296 192.168.3.131 -> 192.168.3.150 STUN 122 Binding Success Response 
XOR-MAPPED-ADDRESS: 192.168.3.150:60175
  2.434462 192.168.3.131 -> 192.168.3.150 STUN 122 Binding Success Response 
XOR-MAPPED-ADDRESS: 192.168.3.150:65021
  2.435083 192.168.3.131 -> 192.168.3.150 UDP 224 Source port: 16514  
Destination port: 65021
  2.455310 192.168.3.131 -> 192.168.3.150 UDP 224 Source port: 16514  
Destination port: 65021
  2.475009 192.168.3.131 -> 192.168.3.150 UDP 224 Source port: 16514  
Destination port: 65021

Thanks again for your time!

Kind Regards,

Gary Shergill


- Original Message -
From: "Amit Patkar" 
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)



Please check rtp.conf 

Look for stunaddr setting. You can try with google STUN server 
stunaddr = stun.l.google.com:19302 





Thanks & Regards, 
Amit Patkar 
On 5/21/2014 9:13 PM, Gary Shergill wrote: 


Hi again,

Just noticed this is being sent to the wrong thread... first time using a 
mailing list and I just replied to the mail sent by the mailing list for Amit's 
reply. Hope this time it works...

Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side 
(I tested using the SIPml demo site and it worked, then realised I was missing 
a setting).

However, the issue still remains where 1000 can not always hear 6901. As 
mentioned before, this works only SOMETIMES, and when it does work 
asteriskgary.local sees RTP packets coming FROM 192.168.3.131 
(asteriskrtc.local).

Unsure what would be causing this, because it does work sometimes and doesn't 
at others, with no obvious reason either way.

Thanks again.

Kind Regards,

Gary Shergill


- Original Message -
From: "Gary Shergill"  To: "Asterisk Users Mailing List - 
Non-Commercial Discussion"  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. The other user is defined in 
Elastix.

I have "directmedia=no" set for the user on asteriskrtc.local.

My exact setup/scenario is below:
- asteriskgary.local has a route to dial extensions on my Elastix server.
- asteriskgary.local has a route to dial extensions on asteriskrtc.local server.
- The call is being originated from asteriskgary.local. The first party is an 
extension on asteriskgary.local, the destination party is an extension on my 
Elastix server.

What's happening is as follows (this is a reverse of the previous case as 6901 
is now dialling 1000):
- User on asteriskgary.local places a call to 1000, his number is 6901
- 6901 answers on the web browser and begins to dial 1000
- 1000 answers and the call is established correctly
- SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...)
- 6901 can NEVER hear 1000

key:
192.168.3.127 - asteriskgary.local
192.168.3.131 - asteriskrtc.local
192.168.3.150 - machine running chrome browser where 6901 is logged on
192.168.3.100 - phone where 1000 is logged on

(1000 can hear 6901) RTP TRACE ON asteriskrtc.local

Got  RTP packet from192.1

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

[bhavik patel]

Hi,

I am also trying to integrate sipml5 demo.For that i made some
configuration.
Call works fine using chrome browser but facing "One way audio issue".
And firefox browser not able to originate call.

Here is the my configuration: http://pastebin.com/EtVzK2T2

let me know if i miss something.




On Wed, May 21, 2014 at 9:25 PM, Amit Patkar  wrote:

>  Please check rtp.conf
>
> Look for stunaddr setting. You can try with google STUN server
> stunaddr = stun.l.google.com:19302
>
>   *Thanks & Regards,*
> Amit Patkar
>   On 5/21/2014 9:13 PM, Gary Shergill wrote:
>
> Hi again,
>
> Just noticed this is being sent to the wrong thread... first time using a 
> mailing list and I just replied to the mail sent by the mailing list for 
> Amit's reply. Hope this time it works...
>
> Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side 
> (I tested using the SIPml demo site and it worked, then realised I was 
> missing a setting).
>
> However, the issue still remains where 1000 can not always hear 6901. As 
> mentioned before, this works only SOMETIMES, and when it does work 
> asteriskgary.local sees RTP packets coming FROM 192.168.3.131 
> (asteriskrtc.local).
>
> Unsure what would be causing this, because it does work sometimes and doesn't 
> at others, with no obvious reason either way.
>
> Thanks again.
>
> Kind Regards,
>
> Gary Shergill
>
>
> - Original Message -
> From: "Gary Shergill"  
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
>  
> 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. The other user is 
> defined in Elastix.
>
> I have "directmedia=no" set for the user on asteriskrtc.local.
>
> My exact setup/scenario is below:
> - asteriskgary.local has a route to dial extensions on my Elastix server.
> - asteriskgary.local has a route to dial extensions on asteriskrtc.local 
> server.
> - The call is being originated from asteriskgary.local. The first party is an 
> extension on asteriskgary.local, the destination party is an extension on my 
> Elastix server.
>
> What's happening is as follows (this is a reverse of the previous case as 
> 6901 is now dialling 1000):
> - User on asteriskgary.local places a call to 1000, his number is 6901
> - 6901 answers on the web browser and begins to dial 1000
> - 1000 answers and the call is established correctly
> - SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...)
> - 6901 can NEVER hear 1000
>
> key:
> 192.168.3.127 - asteriskgary.local
> 192.168.3.131 - asteriskrtc.local
> 192.168.3.150 - machine running chrome browser where 6901 is logged on
> 192.168.3.100 - phone where 1000 is logged on
>
> (1000 can hear 6901) RTP TRACE ON asteriskrtc.local
> 
> Got  RTP packet from192.168.3.150:55148 (type 00, seq 014308, ts 
> 2304496631, len 000160)
> Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054709, ts 
> 2304496624, len 000160)
>> 0x7fe73c021740 -- Probation passed - setting RTP source address to 
> 192.168.3.127:15942
> Got  RTP packet from192.168.3.127:15942 (type 00, seq 003763, ts 000160, 
> len 000160)
> Sent RTP packet to  192.168.3.150:55148 (via ICE) (type 00, seq 047008, 
> ts 000160, len 4294967284)
> Got  RTP packet from192.168.3.150:55148 (type 00, seq 014309, ts 
> 2304496791, len 000160)
> Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054710, ts 
> 2304496784, len 000160)
> Got  RTP packet from192.168.3.150:55148 (type 00, seq 014310, ts 
> 2304496951, len 000160)
> Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054711, ts 
> 2304496944, len 000160)
> Got  RTP packet from192.168.3.127:15942 (type 00, seq 003764, ts 000320, 
> len 000160)
> Sent RTP packet to  192.168.3.150:55148 (via ICE) (type 00, seq 047009, 
> ts 000320, len 4294967284)
> Got  RTP packet from192.168.3.150:55148 (type 00, seq 014311, ts 
> 2304497111, len 000160)
> Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054712, ts 
> 2304497104, len 000160)
> Got  RTP packet from192.168.3.127:15942 (type 00, seq 003765, ts 000480, 
> len 000160)
> Sent RTP packet to  192.168.3.150:55148 (via ICE) (type 00, seq 047010, 
> ts 000480, len 4294967284)
> 
>
> (1000 can hear 6901) RTP TRACE ON asteriskgary.local
> ...
> Got  RTP packet from192.168.3.131:17836 (type 00, seq 055375, ts 
> 2304603184, len 000160

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

[Amit Patkar]

Please check rtp.conf

Look for stunaddr setting. You can try with google STUN server
stunaddr = stun.l.google.com:19302

*Thanks & Regards,*
Amit Patkar

On 5/21/2014 9:13 PM, Gary Shergill wrote:

Hi again,

Just noticed this is being sent to the wrong thread... first time using a 
mailing list and I just replied to the mail sent by the mailing list for Amit's 
reply. Hope this time it works...

Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side 
(I tested using the SIPml demo site and it worked, then realised I was missing 
a setting).

However, the issue still remains where 1000 can not always hear 6901. As 
mentioned before, this works only SOMETIMES, and when it does work 
asteriskgary.local sees RTP packets coming FROM 192.168.3.131 
(asteriskrtc.local).

Unsure what would be causing this, because it does work sometimes and doesn't 
at others, with no obvious reason either way.

Thanks again.

Kind Regards,

Gary Shergill


- Original Message -
From: "Gary Shergill" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

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. The other user is defined in 
Elastix.

I have "directmedia=no" set for the user on asteriskrtc.local.

My exact setup/scenario is below:
- asteriskgary.local has a route to dial extensions on my Elastix server.
- asteriskgary.local has a route to dial extensions on asteriskrtc.local server.
- The call is being originated from asteriskgary.local. The first party is an 
extension on asteriskgary.local, the destination party is an extension on my 
Elastix server.

What's happening is as follows (this is a reverse of the previous case as 6901 
is now dialling 1000):
- User on asteriskgary.local places a call to 1000, his number is 6901
- 6901 answers on the web browser and begins to dial 1000
- 1000 answers and the call is established correctly
- SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...)
- 6901 can NEVER hear 1000

key:
192.168.3.127 - asteriskgary.local
192.168.3.131 - asteriskrtc.local
192.168.3.150 - machine running chrome browser where 6901 is logged on
192.168.3.100 - phone where 1000 is logged on

(1000 can hear 6901) RTP TRACE ON asteriskrtc.local

Got  RTP packet from192.168.3.150:55148 (type 00, seq 014308, ts 
2304496631, len 000160)
Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054709, ts 
2304496624, len 000160)
> 0x7fe73c021740 -- Probation passed - setting RTP source address to 
192.168.3.127:15942
Got  RTP packet from192.168.3.127:15942 (type 00, seq 003763, ts 000160, 
len 000160)
Sent RTP packet to  192.168.3.150:55148 (via ICE) (type 00, seq 047008, ts 
000160, len 4294967284)
Got  RTP packet from192.168.3.150:55148 (type 00, seq 014309, ts 
2304496791, len 000160)
Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054710, ts 
2304496784, len 000160)
Got  RTP packet from192.168.3.150:55148 (type 00, seq 014310, ts 
2304496951, len 000160)
Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054711, ts 
2304496944, len 000160)
Got  RTP packet from192.168.3.127:15942 (type 00, seq 003764, ts 000320, 
len 000160)
Sent RTP packet to  192.168.3.150:55148 (via ICE) (type 00, seq 047009, ts 
000320, len 4294967284)
Got  RTP packet from192.168.3.150:55148 (type 00, seq 014311, ts 
2304497111, len 000160)
Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054712, ts 
2304497104, len 000160)
Got  RTP packet from192.168.3.127:15942 (type 00, seq 003765, ts 000480, 
len 000160)
Sent RTP packet to  192.168.3.150:55148 (via ICE) (type 00, seq 047010, ts 
000480, len 4294967284)


(1000 can hear 6901) RTP TRACE ON asteriskgary.local
...
Got  RTP packet from192.168.3.131:17836 (type 00, seq 055375, ts 
2304603184, len 000160)
Sent RTP packet to  192.168.3.131:17836 (type 00, seq 004428, ts 106560, 
len 000160)
Got  RTP packet from192.168.3.131:17836 (type 00, seq 055376, ts 
2304603344, len 000160)
Sent RTP packet to  192.168.3.131:17836 (type 00, seq 004429, ts 106720, 
len 000160)
Got  RTP packet from192.168.3.131:17836 (type 00, seq 055377, ts 
2304603504, len 000160)
Sent RTP packet to  192.168.3.131:17836 (type 00, seq 004430, ts 106880, 
len 000160)
Got  RTP packet from192.168.3.131:17836 (type 00, seq 055378, ts 
2304603664, len 000160)
Sent RTP packet to  192.168.3.131:17836 (type 00, seq 004431, ts 107040, 
len 000160)
...

(no audio) RTP TRACE ON asteriskrtc.local

Got  RTP packet from192.168.3.127:17796 (type 00, seq 035016, ts 000640, 
len 000160)
Sent RTP packet to  192.168.3.150:53684 (via ICE) (type 00, seq 060981, ts 
000640, len 4294967284)
Got  RTP pac

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

[Gary Shergill]

Hi again,

Just noticed this is being sent to the wrong thread... first time using a 
mailing list and I just replied to the mail sent by the mailing list for Amit's 
reply. Hope this time it works...

Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side 
(I tested using the SIPml demo site and it worked, then realised I was missing 
a setting).

However, the issue still remains where 1000 can not always hear 6901. As 
mentioned before, this works only SOMETIMES, and when it does work 
asteriskgary.local sees RTP packets coming FROM 192.168.3.131 
(asteriskrtc.local).

Unsure what would be causing this, because it does work sometimes and doesn't 
at others, with no obvious reason either way.

Thanks again.

Kind Regards,

Gary Shergill


- Original Message -
From: "Gary Shergill" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

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. The other user is defined in 
Elastix.

I have "directmedia=no" set for the user on asteriskrtc.local.

My exact setup/scenario is below:
- asteriskgary.local has a route to dial extensions on my Elastix server.
- asteriskgary.local has a route to dial extensions on asteriskrtc.local server.
- The call is being originated from asteriskgary.local. The first party is an 
extension on asteriskgary.local, the destination party is an extension on my 
Elastix server.

What's happening is as follows (this is a reverse of the previous case as 6901 
is now dialling 1000):
- User on asteriskgary.local places a call to 1000, his number is 6901
- 6901 answers on the web browser and begins to dial 1000
- 1000 answers and the call is established correctly
- SOMETIMES 1000 can hear 6901. Other times he can not (seems to be random...)
- 6901 can NEVER hear 1000

key:
192.168.3.127 - asteriskgary.local
192.168.3.131 - asteriskrtc.local
192.168.3.150 - machine running chrome browser where 6901 is logged on
192.168.3.100 - phone where 1000 is logged on

(1000 can hear 6901) RTP TRACE ON asteriskrtc.local

Got  RTP packet from192.168.3.150:55148 (type 00, seq 014308, ts 
2304496631, len 000160)
Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054709, ts 
2304496624, len 000160)
   > 0x7fe73c021740 -- Probation passed - setting RTP source address to 
192.168.3.127:15942
Got  RTP packet from192.168.3.127:15942 (type 00, seq 003763, ts 000160, 
len 000160)
Sent RTP packet to  192.168.3.150:55148 (via ICE) (type 00, seq 047008, ts 
000160, len 4294967284)
Got  RTP packet from192.168.3.150:55148 (type 00, seq 014309, ts 
2304496791, len 000160)
Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054710, ts 
2304496784, len 000160)
Got  RTP packet from192.168.3.150:55148 (type 00, seq 014310, ts 
2304496951, len 000160)
Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054711, ts 
2304496944, len 000160)
Got  RTP packet from192.168.3.127:15942 (type 00, seq 003764, ts 000320, 
len 000160)
Sent RTP packet to  192.168.3.150:55148 (via ICE) (type 00, seq 047009, ts 
000320, len 4294967284)
Got  RTP packet from192.168.3.150:55148 (type 00, seq 014311, ts 
2304497111, len 000160)
Sent RTP packet to  192.168.3.127:15942 (type 00, seq 054712, ts 
2304497104, len 000160)
Got  RTP packet from192.168.3.127:15942 (type 00, seq 003765, ts 000480, 
len 000160)
Sent RTP packet to  192.168.3.150:55148 (via ICE) (type 00, seq 047010, ts 
000480, len 4294967284)


(1000 can hear 6901) RTP TRACE ON asteriskgary.local
...
Got  RTP packet from192.168.3.131:17836 (type 00, seq 055375, ts 
2304603184, len 000160)
Sent RTP packet to  192.168.3.131:17836 (type 00, seq 004428, ts 106560, 
len 000160)
Got  RTP packet from192.168.3.131:17836 (type 00, seq 055376, ts 
2304603344, len 000160)
Sent RTP packet to  192.168.3.131:17836 (type 00, seq 004429, ts 106720, 
len 000160)
Got  RTP packet from192.168.3.131:17836 (type 00, seq 055377, ts 
2304603504, len 000160)
Sent RTP packet to  192.168.3.131:17836 (type 00, seq 004430, ts 106880, 
len 000160)
Got  RTP packet from192.168.3.131:17836 (type 00, seq 055378, ts 
2304603664, len 000160)
Sent RTP packet to  192.168.3.131:17836 (type 00, seq 004431, ts 107040, 
len 000160)
...

(no audio) RTP TRACE ON asteriskrtc.local

Got  RTP packet from192.168.3.127:17796 (type 00, seq 035016, ts 000640, 
len 000160)
Sent RTP packet to  192.168.3.150:53684 (via ICE) (type 00, seq 060981, ts 
000640, len 4294967284)
Got  RTP packet from192.168.3.127:17796 (type 00, seq 035017, ts 000800, 
len 000160)
Sent RTP packet to  192.168.3.150:53684 (via ICE) (type 00, seq 060982, ts 
000800, len 4294967284)
Got  RTP packet from

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

[Gary Shergill]

ualify=yes
context=webrtc
hasiax = no
hassip = yes
encryption = yes
avpf = yes
icesupport = yes
videosupport=no
directmedia=no
canreinvite=no

You can see from the trace packets that sometimes asteriskgary.local sees no 
packets from asteriskrtc.local, and at the same time the packets on 
asteriskrtc.local show half the number of records (there is no "Probation 
passed - setting RTP source address to 192.168.3.127:15942 which causes twice 
the number of packets, no idea if this is relevant though).

Please ask if you need anything else. I'm totally stumped with this issue... 
Note that on asteriskgary.local ICE is not configured, I wouldn't have though 
it would need it as it isn't talking with the webrtc client itself, it is just 
talking to an Asterisk server (and that asterisk server is the one which talks 
to the webrtc client).

Thank you.

Kind Regards,

Gary Shergill


- Original Message -
From: "Amit Patkar" 
To: asterisk-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 DirectMedia / reInvite for calls going between 2 
asterisk boxes.
I hope this helps to resolve your issue.

*Thanks & Regards,*
Amit Patkar


On 5/21/2014 2:26 PM, Gary Shergill wrote:
> 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).
>
> Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to 
> log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC 
> user, and vice versa, and all the media flows.
>
> When I try making a call from the other asterisk server (asteriskgary.local) 
> to asteriskrtc.local (all routes are set up) I am seeing the following 
> behaviour:
>
> - asteriskgary.local user, 1000, dials asteriskrtc.local number, 6901
> - 6901 sees the call and has the option to answer
> - 6901 answers the call
> - 6901 can hear 1000 talking
> - 1000 can not hear 6901
>
> The weird thing is, sometimes it works, sometimes it doesn't...
>
> I think it has something to do with the port destination changing when the 
> call is answered but I'm not sure (wireshark suggests that, as it says "Port 
> Unreachable").
>
> Has anyone tried this before and seen this issue? Or knows why it is and how 
> to debug it? I can provide any logs required, I have some logs from when it 
> works and doesn't.
>
> Thank you for your help.
>
> Kind Regards,
>
> Gary Shergill
>

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio with WebRTC (with external asterisk)

[Amit Patkar]

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 5/21/2014 2:26 PM, Gary Shergill wrote:

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).

Dealing with just the WebRTC asterisk server, asteriskrtc.local, I am able to 
log in to the SIPml webpage and make a call from a SIP Phone to that WebRTC 
user, and vice versa, and all the media flows.

When I try making a call from the other asterisk server (asteriskgary.local) to 
asteriskrtc.local (all routes are set up) I am seeing the following behaviour:

- asteriskgary.local user, 1000, dials asteriskrtc.local number, 6901
- 6901 sees the call and has the option to answer
- 6901 answers the call
- 6901 can hear 1000 talking
- 1000 can not hear 6901

The weird thing is, sometimes it works, sometimes it doesn't...

I think it has something to do with the port destination changing when the call is 
answered but I'm not sure (wireshark suggests that, as it says "Port 
Unreachable").

Has anyone tried this before and seen this issue? Or knows why it is and how to 
debug it? I can provide any logs required, I have some logs from when it works 
and doesn't.

Thank you for your help.

Kind Regards,

Gary Shergill




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio when using originate...

[Pezhman Lali]

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 wrote:

>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 person can hear you but you cannot hear them.  Why would it
> behave differently when dialing from a phone?
>
>The server is behind NAT and uses externaddr to set the external IP
> (static).  Anyone had any experience with this?
>
> Here is my (edited) sip.conf entry:
>
> [libre-8793]
> defaultuser=123456789
> secret=X
> fromuser=123456789
> trustrpid=yes
> sendrpid=yes
> type=peer
> fromdomain=i2next.com.mx
> host=i2next.com.mx
> nat=yes
> qualify=no
> insecure=port,invite
> directmedia=no
> disallow=all
> allow=g729
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Pezhman Lali
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio

[Tim King]

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  wrote:

> 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 wrote:
>
>> Also it could be the routing issue as well.
>>
>> --
>> Sent from my iPhone
>>
>> On Mar 9, 2011, at 7:43 PM, Duncan Turnbull 
>> 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 >
>> 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 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length
>> 172
>>
>> I am not sure why this is happening or if its still part of the same
>> conversation
>>
>> Overall it looks a bit like the asterisk box thinks it needs to send rtp
>> to a different location than perhaps its meant to i.e. its asymmetric - you
>> can check the sdp in the sip invite to see where media is expected to be
>> sent to
>>
>> There is no rtp coming back from 209.216.2.203 so possibly this is device
>> that isn't meant to be part of the conversation and either doesn't exist or
>> is not expecting anything and thus not responding
>>
>> What are the addresses of the devices in this conversation? so that you
>> can match the traffic to device
>>
>> Cheers Duncan
>>
>> On 10/03/2011, at 1:20 PM, Tim King wrote:
>>
>> It looks like this:
>> 19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.129590 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.141878 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.149709 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.161886 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.169561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.181879 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.189710 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.201965 IP 199.173.66.22.53103 > 74.204.4.5.11733:

Re: [asterisk-users] One Way Audio

[Tim King]

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  wrote:

> Also it could be the routing issue as well.
>
> --
> Sent from my iPhone
>
> On Mar 9, 2011, at 7:43 PM, Duncan Turnbull  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 >
> 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 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length
> 172
>
> I am not sure why this is happening or if its still part of the same
> conversation
>
> Overall it looks a bit like the asterisk box thinks it needs to send rtp to
> a different location than perhaps its meant to i.e. its asymmetric - you can
> check the sdp in the sip invite to see where media is expected to be sent to
>
> There is no rtp coming back from 209.216.2.203 so possibly this is device
> that isn't meant to be part of the conversation and either doesn't exist or
> is not expecting anything and thus not responding
>
> What are the addresses of the devices in this conversation? so that you can
> match the traffic to device
>
> Cheers Duncan
>
> On 10/03/2011, at 1:20 PM, Tim King wrote:
>
> It looks like this:
> 19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.129590 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.141878 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.149709 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.161886 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.169561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.181879 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.189710 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.201965 IP 199.173.66.22.53103 > 74.204.4.5.11733: UDP, length 60
> 19:18:35.201974 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.209552 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.221898 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.229625 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 1

Re: [asterisk-users] One Way Audio

[Satish Patel]

Also it could be the routing issue as well.

--
Sent from my iPhone

On Mar 9, 2011, at 7:43 PM, Duncan Turnbull   
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 > 
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 74.204.4.5.11732 > 174.133.195.194.18364: UDP,  
length 172


I am not sure why this is happening or if its still part of the same  
conversation


Overall it looks a bit like the asterisk box thinks it needs to send  
rtp to a different location than perhaps its meant to i.e. its  
asymmetric - you can check the sdp in the sip invite to see where  
media is expected to be sent to


There is no rtp coming back from 209.216.2.203 so possibly this is  
device that isn't meant to be part of the conversation and either  
doesn't exist or is not expecting anything and thus not responding


What are the addresses of the devices in this conversation? so that  
you can match the traffic to device


Cheers Duncan

On 10/03/2011, at 1:20 PM, Tim King wrote:


It looks like this:
19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.129590 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.141878 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.149709 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.161886 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.169561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.181879 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.189710 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.201965 IP 199.173.66.22.53103 > 74.204.4.5.11733: UDP,  
length 60
19:18:35.201974 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.209552 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.221898 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.229625 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.241894 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP,  
length 172
19:18:35.249566 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP,  
length 172
19:18:35.261999 IP 199.17

Re: [asterisk-users] One Way Audio

[Tim King]

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 = rfc2833
directmedia=no

[castlewire]
type=user
host=74.204.4.206
context=outb2
dtmfmode=rfc2833
username=castlewire
secret=1234
quallify=yes
canreinvite=no

[equity]
type=friend
host=dynamic
context=outb2
dtmfmode=rfc2833
username=equity
secret=1234
quallify=yes
canreinvite=no

[3000]
type=friend
host=dynamic
nat=yes
context=inbound
dtmfmode=rfc2833
username=3000
secret=1234
quallify=yes
canreinvite=no

[6168182996]
type=friend
host=dynamic
nat=yes
context=outb2
dtmfmode=rfc2833
username=6168182996
secret=1234
quallify=yes
canreinvite=no

[VITELITY]
type=friend
host=64.2.142.93
port=5060
dtmfmode=auto
context=inbound

[QWEST_OUT]
type=friend
host=67.135.79.80
port=5060
dtmfmode=inband

[QWEST8XX_IN]
type=friend
host=67.135.79.199
port=5060
context=qwest800

[DIDX1]
type=peer
host=67.15.128.14
context=inbound
canreinvite=no

[DIDX2]
type=peer
host=67.15.128.18
context=inbound
canreinvite=no

[DIDX3]
type=peer
host=208.44.220.237
context=inbound
canreinvite=no

[DIDX4]
type=peer
host=208.44.220.234
context=inbound
canreinvite=no

[DIDX5]
type=peer
host=209.62.66.242
context=inbound
canreinvite=no

[DIDX6]
type=peer
host=64.246.22.119
context=inbound
canreinvite=no

[DIDX7]
type=peer
host=70.84.58.18
context=inbound
canreinvite=no

[DIDX8]
type=peer
host=174.133.195.194
context=inbound
canreinvite=no

*iax.conf*

[general]
bandwidth=low
disallow=all
allow=ulaw
allow=alaw
jitterbuffer=no
forcejitterbuffer=no
autokill=yes

register=equity_out:1234@74.204.4.166
;register => IAX2/castlewire_trix:1234@74.204.4.206

[CASTLEWIRE]
type=friend
;host=74.204.4.206
host=dynamic
trunk=yes
auth=md5,plaintext,rsa
secret=1234
username=CASTLEWIRE
qualify=yes
context=outb2

[castlewire_trix]
type=friend
;host=74.204.4.206
host=dynamic
trunk=yes
auth=md5,plaintext,rsa
secret=1234
username=castlewire_trix
qualify=yes
context=outb2
requirecalltoken=no

[equity]
type=friend
host=dynamic
context=equity-fix
secret=1234
username=default
channels=10
trunk=yes
timezone=America/Detroit
qualify=yes
requirecalltoken=no

[equity_out]
type=friend
host=dynamic
context=outb2
secret=1234
username=equity_out
channels=10
trunk=yes
timezone=America/Detroit
qualify=yes
requirecalltoken=no

*extensions.conf*

[inbound]
;Equity Logistics
;exten => 6168182400,1,Dial(IAX2/equity/${EXTEN})
;exten => 6168182400,n,Hangup()
;exten => 8182400,1,Dial(IAX2/equity/${EXTEN})
;exten => 8182400,n,Hangup()

exten => 6168182400,1,Dial(SIP/equity/${EXTEN})
exten => 6168182400,n,Hangup()

exten => 6168182996,1,Dial(SIP/${EXTEN})
exten => 6168182996,n,Hangup()
;exten => 6168182996,1,Answer()
;exten => 6168182996,n,Milliwatt()

exten => 3000,1,Dial(SIP/${EXTEN})
exten => 3000,n,Hangup()

;CASTELWIRE NUMBERS
exten => 6168182000,1,Dial(IAX2/castlewire_trix/${EXTEN})
exten => 6168182000,n,Hangup()

;exten => 6168182000,1,Dial(SIP/4403712250@12.194.10.18)
;exten => 6168182000,n,Hangup()


exten => 6168182999,1,Set(portnum=${CALLERID(rdnis)})
exten => 6168182999,n,Set(cutNum=${CUT(portnum|\-|6)})
exten => 6168182999,n,Dial(SIP/${cutNum})
exten => 6168182999,n,Hangup()
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio

[Steve Davies]

On 10 March 2011 11:17, Ishfaq Malik  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 reason
2) Routing - Sometimes devices cannot route to each other directly
3) ITSP calls. Many SIP providers will not accept a redirect

and I am sure there are many more...

Cheers,
Steve

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio

[Ishfaq Malik]

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 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
> > 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.
> > 
> > Please email me at tim.compnetw...@gmail.com if you can help.
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >http://www.asterisk.org/hello
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio

[Ishfaq Malik]

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
> 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.
> 
> Please email me at tim.compnetw...@gmail.com if you can help.
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio

[Jai Rangi]

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  wrote:

>
> 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
> www.didforsale.com.
>
> On Wed, Mar 9, 2011 at 4:43 PM, Duncan Turnbull 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 >
>> 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 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length
>> 172
>>
>> I am not sure why this is happening or if its still part of the same
>> conversation
>>
>> Overall it looks a bit like the asterisk box thinks it needs to send rtp
>> to a different location than perhaps its meant to i.e. its asymmetric - you
>> can check the sdp in the sip invite to see where media is expected to be
>> sent to
>>
>> There is no rtp coming back from  <209.216.2.203>209.216.2.203 so
>> possibly this is device that isn't meant to be part of the conversation and
>> either doesn't exist or is not expecting anything and thus not responding
>>
>> What are the addresses of the devices in this conversation? so that you
>> can match the traffic to device
>>
>> Cheers Duncan
>>
>> On 10/03/2011, at 1:20 PM, Tim King wrote:
>>
>> It looks like this:
>> 19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.129590 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.141878 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.149709 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.161886 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.169561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
>> 19:18:35.181879 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
>> 19:18:35.189710 IP 74.204.4.5.11732 > 209.216.2.203.60362: UD

Re: [asterisk-users] One Way Audio

[Jai Rangi]

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
www.didforsale.com.

On Wed, Mar 9, 2011 at 4:43 PM, Duncan Turnbull 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 >
> 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 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length
> 172
>
> I am not sure why this is happening or if its still part of the same
> conversation
>
> Overall it looks a bit like the asterisk box thinks it needs to send rtp to
> a different location than perhaps its meant to i.e. its asymmetric - you can
> check the sdp in the sip invite to see where media is expected to be sent to
>
> There is no rtp coming back from 209.216.2.203 so possibly this is device
> that isn't meant to be part of the conversation and either doesn't exist or
> is not expecting anything and thus not responding
>
> What are the addresses of the devices in this conversation? so that you can
> match the traffic to device
>
> Cheers Duncan
>
> On 10/03/2011, at 1:20 PM, Tim King wrote:
>
> It looks like this:
> 19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.129590 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.141878 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.149709 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.161886 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.169561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.181879 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.189710 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.201965 IP 199.173.66.22.53103 > 74.204.4.5.11733: UDP, length 60
> 19:18:35.201974 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.209552 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.221898 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172

Re: [asterisk-users] One Way Audio

[Duncan Turnbull]

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 74.204.4.5.11732 > 174.133.195.194.18364: UDP, length 172

I am not sure why this is happening or if its still part of the same 
conversation

Overall it looks a bit like the asterisk box thinks it needs to send rtp to a 
different location than perhaps its meant to i.e. its asymmetric - you can 
check the sdp in the sip invite to see where media is expected to be sent to

There is no rtp coming back from 209.216.2.203 so possibly this is device that 
isn't meant to be part of the conversation and either doesn't exist or is not 
expecting anything and thus not responding

What are the addresses of the devices in this conversation? so that you can 
match the traffic to device

Cheers Duncan

On 10/03/2011, at 1:20 PM, Tim King wrote:

> It looks like this:
> 19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.129590 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.141878 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.149709 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.161886 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.169561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.181879 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.189710 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.201965 IP 199.173.66.22.53103 > 74.204.4.5.11733: UDP, length 60
> 19:18:35.201974 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.209552 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.221898 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.229625 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.241894 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.249566 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.261999 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
> 19:18:35.269701 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
> 19:18:35.2

Re: [asterisk-users] One Way Audio

[Tim King]

It looks like this:
19:18:34.782016 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:34.789527 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:34.802064 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:34.809757 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:34.821855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:34.829598 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:34.842015 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:34.849764 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:34.861902 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:34.869568 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:34.881882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:34.889739 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:34.901882 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:34.909612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:34.921984 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:34.929664 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:34.941855 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:34.949589 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:34.962003 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:34.969592 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:34.981851 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:34.989543 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.002006 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.009973 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.022008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.029539 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.042071 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.049561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.062008 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.069612 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.081986 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.089519 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.101918 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.109722 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.122021 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.129590 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.141878 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.149709 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.161886 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.169561 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.181879 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.189710 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.201965 IP 199.173.66.22.53103 > 74.204.4.5.11733: UDP, length 60
19:18:35.201974 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.209552 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.221898 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.229625 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.241894 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.249566 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.261999 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.269701 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.281873 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.289521 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.301898 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.309599 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.322057 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.329595 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.341889 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.349555 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.361905 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.369599 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.381961 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.389534 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.401947 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.409753 IP 74.204.4.5.11732 > 209.216.2.203.60362: UDP, length 172
19:18:35.421901 IP 199.173.66.22.53102 > 74.204.4.5.11732: UDP, length 172
19:18:35.429605 IP 74.204.4.5.

Re: [asterisk-users] One Way Audio

[Duncan Turnbull]

Can you do a tcpdump to look at the rtp streams on your box and check they are 
both generating and aiming at the right places

IAX will have no issue with NAT/firewall but SIP / RTP can. 

tcpdump -nn udp and portrange 1-2 
(pick your portrange if its operating on something else)

Should show you mad lines of rtp going backwards and forwards (like below) when 
there is a conversation in place. If you can see it being sent from the 
asterisk box but not heard by the client then either try a different client, or 
something is blocking the return leg to your client

13:00:21.309139 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length 172
13:00:21.328703 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length 172
13:00:21.348572 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length 172
13:00:21.369096 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length 172
13:00:21.388572 IP 192.168.63.83.16450 > 192.168.63.30.13902: UDP, length 172

Cheers Duncan

On 10/03/2011, at 12:26 PM, Tim King wrote:

> Thank you I have also tried those settings. The main thing is coming from my 
> voip provider all I am doing is bridging the calls to two other devices (1 
> trixbox and 1 digium aa50) via IAX trunks. Both devices are answering with an 
> IVR and when I call in I can not hear the IVR. However if I call directly to 
> a SIP client the person answering the SIP phone can hear me but I can not 
> hear them at all.  Its definately not a NAT issue which is what makes it even 
> more confusing. When the call is in place a sip show channels shows me both 
> lefs of the call and they are both using either alaw or ulaw so it should not 
> be a codec translation issue either.
> 
> On Wed, Mar 9, 2011 at 6:19 PM, Satish Patel  wrote:
> 

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio

[Tim King]

Thank you I have also tried those settings. The main thing is coming from my
voip provider all I am doing is bridging the calls to two other devices (1
trixbox and 1 digium aa50) via IAX trunks. Both devices are answering with
an IVR and when I call in I can not hear the IVR. However if I call directly
to a SIP client the person answering the SIP phone can hear me but I can not
hear them at all.  Its definately not a NAT issue which is what makes it
even more confusing. When the call is in place a sip show channels shows me
both lefs of the call and they are both using either alaw or ulaw so it
should not be a codec translation issue either.

On Wed, Mar 9, 2011 at 6:19 PM, Satish Patel  wrote:

> What about your  sip clients? Are they on public network?
>
> Try on sip.conf
>
> Nat=no/yes
>
> conreinvite=yes/no
>
> --
> Sent from my iPhone
>
> On Mar 9, 2011, at 6:11 PM, Tim King  wrote:
>
> IPTBALES is off and I have all firewalls disabled. All network elements
> currently involved have public IP's assigned to them. My main asterisk box
> has a public IP. I have multiple trunks to voip peers for inbound and
> outbound calls which are also all public IP's. My two clients are trunked
> via IAX and one is a Trixbox and the other is a digium AA50 which both also
> have public IP's assigned to them.
>
> On Wed, Mar 9, 2011 at 6:04 PM, Adolphe Cher-Aime < 
> achera...@gmail.com> wrote:
>
>> How is your network is organized? Is your server behind a firewal, about
>> iptables ?
>>
>>
>>
>>
>> On Wed, Mar 9, 2011 at 5:40 PM, Tim King < 
>> t...@compnetwork.net> wrote:
>>
>>> 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.
>>>
>>> Please email me at tim.compnetwork@gmail.comif 
>>> you can help.
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by 
>>> http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   
>>> http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> --
>> *Adolphe CHER-AIME
>> Network / VoIP  Engineer
>> CCNA, CCNA VOICE, Global VSAT Forum Certified
>> (509) 3449-4280*
>>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio

[Satish Patel]

What about your  sip clients? Are they on public network?

Try on sip.conf

Nat=no/yes

conreinvite=yes/no

--
Sent from my iPhone

On Mar 9, 2011, at 6:11 PM, Tim King  wrote:

IPTBALES is off and I have all firewalls disabled. All network  
elements currently involved have public IP's assigned to them. My  
main asterisk box has a public IP. I have multiple trunks to voip  
peers for inbound and outbound calls which are also all public IP's.  
My two clients are trunked via IAX and one is a Trixbox and the  
other is a digium AA50 which both also have public IP's assigned to  
them.


On Wed, Mar 9, 2011 at 6:04 PM, Adolphe Cher-Aime  
 wrote:
How is your network is organized? Is your server behind a firewal,  
about  iptables ?





On Wed, Mar 9, 2011 at 5:40 PM, Tim King  wrote:
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.


Please email me at tim.compnetw...@gmail.com if you can help.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



--
Adolphe CHER-AIME
Network / VoIP  Engineer
CCNA, CCNA VOICE, Global VSAT Forum Certified
(509) 3449-4280

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio

[Tim King]

IPTBALES is off and I have all firewalls disabled. All network elements
currently involved have public IP's assigned to them. My main asterisk box
has a public IP. I have multiple trunks to voip peers for inbound and
outbound calls which are also all public IP's. My two clients are trunked
via IAX and one is a Trixbox and the other is a digium AA50 which both also
have public IP's assigned to them.

On Wed, Mar 9, 2011 at 6:04 PM, Adolphe Cher-Aime wrote:

> How is your network is organized? Is your server behind a firewal, about
> iptables ?
>
>
>
>
> On Wed, Mar 9, 2011 at 5:40 PM, Tim King  wrote:
>
>> 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.
>>
>> Please email me at tim.compnetw...@gmail.com if you can help.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> *Adolphe CHER-AIME
> Network / VoIP  Engineer
> CCNA, CCNA VOICE, Global VSAT Forum Certified
> (509) 3449-4280*
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio

[Adolphe Cher-Aime]

How is your network is organized? Is your server behind a firewal, about
iptables ?




On Wed, Mar 9, 2011 at 5:40 PM, Tim King  wrote:

> 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.
>
> Please email me at tim.compnetw...@gmail.com if you can help.
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
*Adolphe CHER-AIME
Network / VoIP  Engineer
CCNA, CCNA VOICE, Global VSAT Forum Certified
(509) 3449-4280*
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio

[Paul Belanger]

On 11-03-09 05:40 PM, Tim King wrote:
> 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.
> 
> Please email me at tim.compnetw...@gmail.com if you can help.
> 
One way audio is almost always a NAT issue. Have you setup your RTP
ports properly on your firewall?

-- 
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one way audio for xlite clients behind NAT

[Flavio Miranda]

If you are using linux firewall, try this, it was very usefull to me:


iptables -t nat -A PREROUTING -i ethx -p tcp --dport 5060 -j DNAT --to 
ip_phoneiptables -t nat -A PREROUTING -i ethx -p udp --dport 5060 -j DNAT --to 
iip_phoneiptables -A FORWARD -p TCP --dport 5060 -j ACCEPTiptables -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 clients behind NAT
> 
> On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson  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 | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
  -- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one way audio for xlite clients behind NAT

[Paul Belanger]

On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson  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 | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one way audio for xlite clients behind NAT

[Thomas Johnson]

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 Belanger  wrote:

> On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson 
> 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: pabelanger (Freenode)
> blog.polybeacon.com
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one way audio for xlite clients behind NAT

[Paul Belanger]

On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson  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: pabelanger (Freenode)
blog.polybeacon.com

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one way audio for xlite clients behind NAT

[Thomas Johnson]

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  wrote:

>
>
> 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 all of them?
>
> I don't think so. It's just that you didn't specify which client is which.
>
> My next guess is that it is a codec problem. I've had similar problems -
> and upon checking Asterisk logs - I discovered that the client and
> Asterisk weren't agreeing correctly on codecs. Can you double-check your
> X-lite configuration - and maybe try to ulaw or alaw as the only codec
> at both ends?
>
> Sebastian
>
> > On Thu, Sep 16, 2010 at 1:36 PM, Sebastian  > > wrote:
> >
> >
> >
> > 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
> >  > canreinvite=no
> >  >
> >  > [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
> >  >
> >  > [1001];Work
> >  > type=peer
> >  > dtmfmode=rfc2833
> >  > context=sip
> >  > insecure=very
> >  > host=sip.domain.com  <
> http://sip.domain.com>
> >  > nat=no
> >  >
> >  > [1000];IPKall
> >  > type=peer
> >  > dtmfmode=rfc2833
> >  > context=sip
> >  > insecure=very
> >  > host=voiper.ipkall.com 
> > 
> >  > nat=no
> >
> > You seem to be using nat=no
> >
> > shouldn't that be nat=yes?
> >
> >  >
> >  >
> >  >
> >  > I pasted the log here -> http://pastie.org/1163238
> >  >
> >  >
> >  > I have tried connecting both of the clients to another sip
> > service(sip2sip.info  )
> > and did not have the same problems.
> >  >
> >  >
> >  > Any suggestions would be great.
> >  >
> >  > Thanks,
> >  >
> >  > Tom
> >  >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com--
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> > http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one way audio for xlite clients behind NAT

[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 all of them?

I don't think so. It's just that you didn't specify which client is which.

My next guess is that it is a codec problem. I've had similar problems - 
and upon checking Asterisk logs - I discovered that the client and 
Asterisk weren't agreeing correctly on codecs. Can you double-check your 
X-lite configuration - and maybe try to ulaw or alaw as the only codec 
at both ends?

Sebastian

> On Thu, Sep 16, 2010 at 1:36 PM, Sebastian  > wrote:
>
>
>
> 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
>  > canreinvite=no
>  >
>  > [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
>  >
>  > [1001];Work
>  > type=peer
>  > dtmfmode=rfc2833
>  > context=sip
>  > insecure=very
>  > host=sip.domain.com  
>  > nat=no
>  >
>  > [1000];IPKall
>  > type=peer
>  > dtmfmode=rfc2833
>  > context=sip
>  > insecure=very
>  > host=voiper.ipkall.com 
> 
>  > nat=no
>
> You seem to be using nat=no
>
> shouldn't that be nat=yes?
>
>  >
>  >
>  >
>  > I pasted the log here -> http://pastie.org/1163238
>  >
>  >
>  > I have tried connecting both of the clients to another sip
> service(sip2sip.info  )
> and did not have the same problems.
>  >
>  >
>  > Any suggestions would be great.
>  >
>  > Thanks,
>  >
>  > Tom
>  >
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one way audio for xlite clients behind NAT

[Thomas Johnson]

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  wrote:

>
>
> 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
> > canreinvite=no
> >
> > [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
> >
> > [1001];Work
> > type=peer
> > dtmfmode=rfc2833
> > context=sip
> > insecure=very
> > host=sip.domain.com  
> > nat=no
> >
> > [1000];IPKall
> > type=peer
> > dtmfmode=rfc2833
> > context=sip
> > insecure=very
> > host=voiper.ipkall.com  
> > nat=no
>
> You seem to be using nat=no
>
> shouldn't that be nat=yes?
>
> >
> >
> >
> > I pasted the log here ->  http://pastie.org/1163238
> >
> >
> > I have tried connecting both of the clients to another sip service(
> sip2sip.info  ) and did not have the same problems.
> >
> >
> > Any suggestions would be great.
> >
> > Thanks,
> >
> > Tom
> >
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one way audio for xlite clients behind NAT

[Sebastian]

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
> canreinvite=no
>
> [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
>
> [1001];Work
> type=peer
> dtmfmode=rfc2833
> context=sip
> insecure=very
> host=sip.domain.com  
> nat=no
>
> [1000];IPKall
> type=peer
> dtmfmode=rfc2833
> context=sip
> insecure=very
> host=voiper.ipkall.com  
> nat=no

You seem to be using nat=no

shouldn't that be nat=yes?

>
>
>
> I pasted the log here ->  http://pastie.org/1163238
>
>
> I have tried connecting both of the clients to another sip 
> service(sip2sip.info  ) and did not have the same 
> problems.
>
>
> Any suggestions would be great.
>
> Thanks,
>
> Tom
>

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio when overlapdial is set to yes

[leonimar cape]

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 
> To: asterisk-users@lists.digium.com
> Sent: Wednesday, September 15, 2010 6:12:35 PM
> 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 a  hipath 300e Siemens. I 
>am 
>
> getting one way audio when a local on the hipath  tries to make a pstn call 
> but 
>
> no issue on incoming calls from pstn going to  the hipath locals.
> 
> local ---> hipath 300 - isdn pri   asterisk -- isdn 
> pri 
>
> - telco--> dest.
> 
> Here is my  dahdi  config
> 
> [channels]
> context=default
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> overlapdial=yes
> autofalltrought=yes
> echocancelwhenbridged=yes
> relaxdtmf=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> 
> ;  Span 2: WPE1/1 "wanpipe2 card 1" HDB3/CCS/CRC4  RED
> group=1,12
> context=from-internal
> switchtype =  euroisdn
> ;overlapdial = outgoing
> priindication = inband
> signalling =  pri_net
> channel => 32-46,48-62
> context = default
> group =  63
> 
> 
> 
>  Span 4: WPE1/3 "wanpipe4 card 3" HDB3/CCS/CRC4 
> group=4,14
> context=outrt-001-PSTN_E1
> switchtype=qsig
> signalling=pri_cpe
> ;facilityenable=yes
> ;callprogress=yes
> pridialplan=unknown
> prilocaldialplan=unknown
> ;priindication  = outofband
> ;overlapdial = incoming
> ;priexclusive = yes
> ;pritimer  => t200,1000
> ;pritimer => t313,4000
> ;immediate=yes
> channel =>  94-108,110-124
> context = default
> group = 63
> 
> 
> Any suggestion will  be much appreciated.
> 
> Regards,
> 
> Mac
> 
> 
> 
> -- 
> _
> --  Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for  a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To  UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio when dialing multiple registrations

[Zeeshan Zakaria]

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

--
www.ilovetovoip.com

On 2010-07-19 12:28 PM, "Nasir Javaid"  wrote:

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 forwarding.

what i am currently trying to grab the channel name along with it's unique
id and dial it directly like simple Dial(SIP/xyz ) dialing

for example

Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT)
   ^
|
|
 |
but problem is that asterisk assigns random unique-id for every call. and
also it is available only when dialing...
what are my options?

your help will be highly appreciated.

regards,


Naisr Javaid

-

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 fashion

I have had this problem once and I never tried registering same extension
from two different places after that.

Try Phillip's suggestion, maybe it'll work for you.

Zeeshan A Zakaria

--
www.ilovetovoip.com
On 2010-07-15 11:42 AM, "Philipp von Klitzing" <
klitz...@xx> wrote:

Hi!

> I am working on calling 2 registrations of same user on 2 different ip or
> ports. It works f...
You need to make sure that these two phones use *different* RTP ports,
and that this is handled correctly in your router/NAT device (by port
forwarding or other methods).

Philipp

---
Hi Zeeshan,

I saw many of your posts on forum. i also put my problem on forum but did
not get any satisfying answer. I wish if you could help me out. below is my
post.

==
==
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...@119.68.0.90:5060

SIP/x...@202.16.34.10:5678

i dial using following dial string

Dial( SIP/x...@119.68.0.90:5060& SIP/x...@202.16.34.10:5678,30,tTog)

both destinations ring at the same time and one that is answered starts
conversations. but audio is one sided as i mentioned above.

But simply dialing  single registration of XYZ like
Dial(SIP/XYZ,30,tTog)   works fine and audio is fine at both ends.

have any idea what is going wrong??

any help will be highly appreciated

regards,

Nasir Javaid
==

thanks in advance ...

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio when dialing multiple registrations

[Zeeshan Zakaria]

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 router/firewall. And it may still not solve your problem. I
would suggest using separate extensions for separate IP addresses.

For wireshark sniffing, my following blog might be helpful:

http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/



Zeeshan
--
www.ilovetovoip.com
www.trashinternetexplorer.com



On Fri, Jul 16, 2010 at 12:21 PM, Zeeshan Zakaria wrote:

> 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 fashion
>
> I have had this problem once and I never tried registering same extension
> from two different places after that.
>
> Try Phillip's suggestion, maybe it'll work for you.
>
> Zeeshan A Zakaria
>
> --
> www.ilovetovoip.com
>
> On 2010-07-15 11:42 AM, "Philipp von Klitzing" <
> klitz...@pool.informatik.rwth-aachen.de> wrote:
>
> Hi!
>
> > I am working on calling 2 registrations of same user on 2 different ip or
> > ports. It works f...
>
> You need to make sure that these two phones use *different* RTP ports,
> and that this is handled correctly in your router/NAT device (by port
> forwarding or other methods).
>
> Philipp
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>


-- 
Zeeshan A Zakaria
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio when dialing multiple registrations

[Zeeshan 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 fashion

I have had this problem once and I never tried registering same extension
from two different places after that.

Try Phillip's suggestion, maybe it'll work for you.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-07-15 11:42 AM, "Philipp von Klitzing" <
klitz...@pool.informatik.rwth-aachen.de> wrote:

Hi!

> I am working on calling 2 registrations of same user on 2 different ip or
> ports. It works f...
You need to make sure that these two phones use *different* RTP ports,
and that this is handled correctly in your router/NAT device (by port
forwarding or other methods).

Philipp


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio when dialing multiple registrations

[Philipp von Klitzing]

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 *different* RTP ports, 
and that this is handled correctly in your router/NAT device (by port 
forwarding or other methods).

Philipp


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio when dialing multiple registrations

[Jonas Kellens]

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 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 
SIP/x...@192.168.0.10:5678 

i dial using following dial string

Dial(SIP/x...@192.168.0.20:5060&SIP/x...@192.168.0.10:5678 
,30,tTog)


both destinations ring at the same time and one that is answered 
starts conversations. but audio is one sided as i mentioned above.


But simply dialing  single registration of XYZ like
Dial(SIP/XYZ,30,tTog)   works fine and audio is fine at both ends.


have any idea what is going wrong??

any help will be highly appreciated

regards,

Nasir Javaid




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One-Way Audio after Hold

[Mike Diehl]

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 the call 
on hold.  During hold, the Sipphone user cannot hear music, only 
silence.  The silence continues after the hold, though the local phone 
can hear the Sipphone user.


 
Every possible combination of nat=yes, no, maybe, possibly or never 
gives the same result.  Further, canreinvite=yes/no/nonat has no 
result.  I suspect a possible reinvite issue with Asterisk being out 
of the RTP stream, so I have tried all the usual variables in the 
DialI() command as well to no avail. 
Any thoughts on how to fix one-way-audio after a hold?


I have the same problem, only my customers report that it only happens 
occasionally.  Most of the time, they can transfer calls just fine.  
They can also put calls on hold and retrieve them as expected.  However, 
sometimes, about once a day, they try to recover a call and the caller 
can't hear them, but they can hear the caller.


I've seen this happen once, but I've been unable to reproduce it reliably.

Any ideas?

Mike Diehl.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio from External Sip Soft & Hard Phone

[Steve Totaro]

On Tue, Jul 7, 2009 at 9:55 PM, Paul Edgar 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 Asterisk but cannot get audio to the Asterisk, ie If I ring
> the voice mail , Asterisk answers and then cannot hear my password…
>
>
>
> I have put the Ports Forward etc…5004-5080 & 1-2
>
>
>
> Any ideas – even what to test next would be good…
>
>
>
>
>
> -- Executing [...@macro-stdexten:13] Dial("SIP/208-00a10004", "SIP/204") in
> new stack
>
>
>
>     -- Called 204
>
>
>
>     -- SIP/204-00a11584 is ringing
>
>
>
>     -- SIP/204-00a11584 answered SIP/208-00a10004
>
>
>
> [Jul  7 16:15:08] WARNING[834]: chan_sip.c:1993 retrans_pkt: Maximum retries
> exceeded on transmission NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. for
> seqno 2 (Critical Response)
>
> [Jul  7 16:15:08] WARNING[834]: chan_sip.c:2017 retrans_pkt: Hanging up call
> NjQ1NTA3Mzg3YzI5ZTNlYzI3MWU2NzE4YWE3ZTI2MDE. - no reply to our critical
> packet.
>
>
>
>   == Spawn extension (macro-stdexten, s, 13) exited non-zero on
> 'SIP/208-00a10004' in macro 'stdexten'
>
>   == Spawn extension (macro-stdexten, s, 13) exited non-zero on
> 'SIP/208-00a10004'
>
>

Where is the NAT or is it on both sides?

Answer that and turn on SIP debugging and post the output and I am
sure someone can help you.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way AUDIO

[Danny Nicholas]

Here's my .02 - local lan is probably behind a firewall meaning that the
5060 gets out ok to send your audio, but the 1-2 range that the
other side comes in on is blocked.  You don't have the problem with static
WAN because it is not behind the firewall or has more ports open.  Do 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

 

Few Running figures !!

On Tue, Apr 7, 2009 at 3:41 AM, David @ULC  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 perfectly. 

I changed the network from loc1 to loc2 but its same. 

I tried changing Ethernet Card but no use. 

What could be the Issue ?

 

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way AUDIO

[David @ULC]

Few Running figures !!

On Tue, Apr 7, 2009 at 3:41 AM, David @ULC  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 perfectly.
>
> I changed the network from loc1 to loc2 but its same.
>
> I tried changing Ethernet Card but no use.
>
> What could be the Issue ?
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way AUDIO

[Giancarlo Rubio]

How  tcpdump on interface show??

2009/4/6 David @ULC :
>
> Can it be that any Port got blocked ?
>
> On Tue, Apr 7, 2009 at 3:41 AM, David @ULC  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 perfectly.
>>
>> I changed the network from loc1 to loc2 but its same.
>>
>> I tried changing Ethernet Card but no use.
>>
>> What could be the Issue ?
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Giancarlo Rubio

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way AUDIO

[David @ULC]

Can it be that any Port got blocked ?

On Tue, Apr 7, 2009 at 3:41 AM, David @ULC  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 perfectly.
>
> I changed the network from loc1 to loc2 but its same.
>
> I tried changing Ethernet Card but no use.
>
> What could be the Issue ?
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Brent Davidson]

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 is connected to the POTS
> SNOM doesn't need to go out to the Internet in this scenario, AFAIK.
>
> Below is my current NAT rules:
>
> # iptables -L -v -t nat
> Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes)
>  pkts bytes target prot opt in out source
> destination
> 11460  760K RETURN 0--  anyany 192.168.101.0/24
> !192.168.101.0/24
>
> Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes)
>  pkts bytes target prot opt in out source
> destination
> 11408  757K MASQUERADE  0--  anyeth0192.168.101.0/24
> anywhere
>
> Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes)
>  pkts bytes target prot opt in out source   
> destination
>
> Please advice if you need more information from me.
>
> Regards,
>
> GNUbie
Having had many years of experience working with iptables I can tell you 
that when IP Forwarding is enabled on a Linux machine things can get a 
bit tricky. In my experience using a Masquerade rule can cause some 
major weirdness.  Try doing this:

Instead of the Masquerade rule use:

iptables -t nat -A POSTROUTING -i eth1 -o eth0 -j SNAT --to-source 


Also, in the general section of your sip.conf make sure you have:

bindaddr=192.168.101.1

to make sure asterisk is not sending sip packets using the public IP 
then effectively trying to communicate with the phone by Masquerading 
the packets coming in over the eth1 to eth0.  This is more than likely 
what is happening. (It's normlly bindaddr=0.0.0.0)

Good luck,
Brent


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Jeff LaCoursiere]

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 and forth and
> > will probably show you that your NAT rules are applying when they
> > shouldn't.  I agree with first turning off your firewall and testing...
> > but if that actually solves the problem you need to know why.  This should
> > tell why.
>
> Why eth0 when in fact it is not being used AFAIK? My Asterisk box is
> connected to the LAN via its eth1 interface and the SIP phone is
> calling from the LAN to the analog telephone via FXO/POTS. Again,
> below is the call scenario diagram:
>
> [SNOM] ==LAN==> eth1 [ASTERISK] fxo ==POTS==> [ANALOG_TELEPHONE]
> eth0
>   ||
> INTERNET

You should try on both interfaces.  If you see packets on eth0 then your
NAT rules are leaking!  Try on eth1 to see the SIP headers and tell if
your NAT rules are doing what you expect.

This is always my first attack...

j

>
> Please advice.  Thank you in advance.
>
> Regards,
>
> GNUbie
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> *** Handled by Will's new toy ***
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Karsten Wemheuer]

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 using the NET interface beside the DC 5V port to
> connect to the LAN.
> 
> The Asterisk box is using the eth1 to connect to the LAN.
> 
> As per your instruction, below is my /etc/asterisk/sip.conf :
> 
> - - - < s n i p > - - -
> 
> [general]
> realm=pbx.domain.com
> bindport=5060
> bindaddr=0.0.0.0
> rtptimeout=60
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> externip=pbx.domain.com
> localnet=192.168.101.0/255.255.255.0
> jbforce=yes
> allowtransfers=yes
> maxexpiry=3600
> minexpiry=1800
> videosupport=no
> 
> [internal-phones](!)
> type=friend
> host=dynamic
> context=family
> dtmfmode=rfc2833
> insecure=port,invite
> canreinvite=no
> nat=no
> qualify=yes
> port=5060
> 
> [102](internal-phones)
> username=102
> secret=102
> callerid="GNUbie"<102>
> [EMAIL PROTECTED]
> 
> - - - < s n i p > - - -

Thanks for the information. In an earlier post You told us, that the
local phones talk to asterisk on eth1 using 192.168.101.0 network. Could
You please double check, that the phone did not try to register on
another IP? The asterisk is IIRC on a dual homed machine. Is Your phone
using a DNS lookup to find the asterisk? To which address is that lookup
resolved?
Another hint: Is Your SNOM using some sort of STUN to lookup an public
address? (Just to eliminate some things).

HTH,
Karsten



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Tzafrir Cohen]

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
> > 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 worked one way.  I ended up rigging * to force the line to
> > be considered answered with a patch.
> 
> Yes, the call has been answered as per Asterisk logs. The CALLER (SNOM
> SIP Phone) can hear clearly the voice of the target CALLEE (POTS
> analog telephone) but it is the CALLEE that cannot hear the CALLER's
> voice.

And yet in the output that you showed us, the channels were not in a
state of "Up". That is: not in a state of "finished dialing and stuff
and now part of a call". Could you plese double check that?

What is the output of 'core show channels' at the time of a call?

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[GNUbie]

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 sip.conf
>
> Start asterisk with verbose set to 3 or so and turn on sip debugging.
> I get somewhere in the debug, you will see local NAT IPs that don't
> belong there, or it will just work.

My /etc/asterisk/sip.conf is at
http://lists.digium.com/pipermail/asterisk-users/2008-October/220256.html
and my SIP phone is located within the LAN where the Asterisk box is
also part of it.

Regards,

GNUbie

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Steve Totaro]

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 $700+ billion dollars and
did not bail out the greedy banks, we could have immediate relief
since for the most part, we could suspend Federal Income tax for
everyone.  A $300 rebate check, give me a break, how about some real
stimulus, a rebate (or lack of theft because there is no law that we
as individuals have to pay Federal Income tax, and I dare anyone to
point it out, a real law, not something the IRS made up, I don't think
they are part of the Legislative branch) weekly or bi-weekly depending
on how you get paid.

It would be immediate and give more money to the people who need it.
All your Fed Income tax pays for anyways is the national debt, the
clock just maxed out at $10 trillion.  Rather than paying it down
below the max and keeping it that way, they are building another one
with additional digits.

Sorry for a TOTALLY OFF topic post.  I screwed up so I thought I might
as well rant a little.

Apologies in sheer exhaustion,
Steve Totaro

Thanks,
Steve Totaro

On Thu, Oct 16, 2008 at 12:46 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> 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 the magic number of times, three?
>>
>> I'm sorry. What do you mean?
>>
>> Regards,
>>
>> GNUbie
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>



-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Steve Totaro]

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 the magic number of times, three?
>
> I'm sorry. What do you mean?
>
> Regards,
>
> GNUbie
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[GNUbie]

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 http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Steve Totaro]

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 somewhere in the debug, you will see local NAT IPs that don't
belong there, or it will just work.

Thanks,
Steve Totaro

On Thu, Oct 16, 2008 at 12:12 AM, Steve Totaro
<[EMAIL PROTECTED]> wrote:
> 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 eth1)?
>>
>> The SNOM 300 is using the NET interface beside the DC 5V port to
>> connect to the LAN.
>>
>> The Asterisk box is using the eth1 to connect to the LAN.
>>
>> As per your instruction, below is my /etc/asterisk/sip.conf :
>>
>> - - - < s n i p > - - -
>>
>> [general]
>> realm=pbx.domain.com
>> bindport=5060
>> bindaddr=0.0.0.0
>> rtptimeout=60
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> allow=gsm
>> externip=pbx.domain.com
>> localnet=192.168.101.0/255.255.255.0
>> jbforce=yes
>> allowtransfers=yes
>> maxexpiry=3600
>> minexpiry=1800
>> videosupport=no
>>
>> [internal-phones](!)
>> type=friend
>> host=dynamic
>> context=family
>> dtmfmode=rfc2833
>> insecure=port,invite
>> canreinvite=no
>> nat=no
>> qualify=yes
>> port=5060
>>
>> [102](internal-phones)
>> username=102
>> secret=102
>> callerid="GNUbie"<102>
>> [EMAIL PROTECTED]
>>
>> - - - < s n i p > - - -
>>
>> Thank you in advance.
>>
>> Regards,
>>
>> GNUbie
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>



-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Steve Totaro]

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 eth1)?
>
> The SNOM 300 is using the NET interface beside the DC 5V port to
> connect to the LAN.
>
> The Asterisk box is using the eth1 to connect to the LAN.
>
> As per your instruction, below is my /etc/asterisk/sip.conf :
>
> - - - < s n i p > - - -
>
> [general]
> realm=pbx.domain.com
> bindport=5060
> bindaddr=0.0.0.0
> rtptimeout=60
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> externip=pbx.domain.com
> localnet=192.168.101.0/255.255.255.0
> jbforce=yes
> allowtransfers=yes
> maxexpiry=3600
> minexpiry=1800
> videosupport=no
>
> [internal-phones](!)
> type=friend
> host=dynamic
> context=family
> dtmfmode=rfc2833
> insecure=port,invite
> canreinvite=no
> nat=no
> qualify=yes
> port=5060
>
> [102](internal-phones)
> username=102
> secret=102
> callerid="GNUbie"<102>
> [EMAIL PROTECTED]
>
> - - - < s n i p > - - -
>
> Thank you in advance.
>
> Regards,
>
> GNUbie
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Steve Totaro]

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 'sip show channels'?
>
> I am using the following codecs:
>
> # asterisk -rx 'sip show settings' | grep Codecs
>  Codecs: 0xe (gsm|ulaw|alaw)
>
> Below is the CLI output:
>
>-- Executing [EMAIL PROTECTED]:1] Dial("SIP/102-081d11d0",
> "Zap/4/1234567") in new stack
>-- Called 4/1234567
>
> *CLI> sip show channels
> Peer User/ANRCall ID  Seq (Tx/Rx)  Format
>  Hold Last Message
> 192.168.101.102  102 3c27a6824ba  00101/2  0x4 (ulaw)
>  No   Rx: INVITE
> 1 active SIP channel
>
> *CLI> core show channels
> Channel  Location State   Application(Data)
> Zap/4-1  [EMAIL PROTECTED] Dialing AppDial((Outgoing Line))
> SIP/102-081d11d0 [EMAIL PROTECTED]:1   RingDial(Zap/4/1234567)
> 2 active channels
> 1 active call
>
>> Can you call from the FXO to Asterisk? (e.g.: to echo test)
>
> There is no problem with an inbound calls. I just tried to call the
> echo test extension number from my mobile phone via FXO/POTS and it
> works fine. I can hear my own voice.
>
> Thank you.
>
> Regards,
>
> GNUbie
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[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 using the NET interface beside the DC 5V port to
connect to the LAN.

The Asterisk box is using the eth1 to connect to the LAN.

As per your instruction, below is my /etc/asterisk/sip.conf :

- - - < s n i p > - - -

[general]
realm=pbx.domain.com
bindport=5060
bindaddr=0.0.0.0
rtptimeout=60
disallow=all
allow=ulaw
allow=alaw
allow=gsm
externip=pbx.domain.com
localnet=192.168.101.0/255.255.255.0
jbforce=yes
allowtransfers=yes
maxexpiry=3600
minexpiry=1800
videosupport=no

[internal-phones](!)
type=friend
host=dynamic
context=family
dtmfmode=rfc2833
insecure=port,invite
canreinvite=no
nat=no
qualify=yes
port=5060

[102](internal-phones)
username=102
secret=102
callerid="GNUbie"<102>
[EMAIL PROTECTED]

- - - < s n i p > - - -

Thank you in advance.

Regards,

GNUbie

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[GNUbie]

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 (nor did
> it give answer supervision) so * never knew the call got picked up so
> audio only worked one way.  I ended up rigging * to force the line to
> be considered answered with a patch.

Yes, the call has been answered as per Asterisk logs. The CALLER (SNOM
SIP Phone) can hear clearly the voice of the target CALLEE (POTS
analog telephone) but it is the CALLEE that cannot hear the CALLER's
voice.

Regards,

GNUbie

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[GNUbie]

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 NAT rules are applying when they
> shouldn't.  I agree with first turning off your firewall and testing...
> but if that actually solves the problem you need to know why.  This should
> tell why.

Why eth0 when in fact it is not being used AFAIK? My Asterisk box is
connected to the LAN via its eth1 interface and the SIP phone is
calling from the LAN to the analog telephone via FXO/POTS. Again,
below is the call scenario diagram:

[SNOM] ==LAN==> eth1 [ASTERISK] fxo ==POTS==> [ANALOG_TELEPHONE]
eth0
  ||
INTERNET

Please advice.  Thank you in advance.

Regards,

GNUbie

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Karsten Wemheuer]

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 Internet.
> In this case, the eth1 of the Asterisk box is connected to the LAN and
> eth0 is connected to the Internet.
> 
> > 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.
> 
> I don't think NAT is involve on this one way audio problem.

Please post Your sip.conf.
Which IP-Address do You configure in the snom for Your asterisk? (eth0
or eth1)?

Regards,
Karsten


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Daniel Hazelbaker]

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 worked one way.  I ended up rigging * to force the line to  
be considered answered with a patch.

Daniel

On Oct 13, 2008, at 8:57 AM, GNUbie wrote:

> 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
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Jeff LaCoursiere]

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 firewall and testing...
but if that actually solves the problem you need to know why.  This should
tell why.


j

On Mon, 13 Oct 2008, GNUbie wrote:

> 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 sip.conf.
>
> Thanks.
>
> Regards,
>
> GNUbie
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> *** Handled by Will's new toy ***
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[GNUbie]

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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[GNUbie]

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 sip.conf.

Thanks.

Regards,

GNUbie

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Norman Franke]

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 the only one-way  
audio problems I've experienced. Setting reinvite=no fixed everything  
for me.

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Steve Totaro]

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
> > probably be dismissed as well.
>
> 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 is connected to the POTS
> SNOM doesn't need to go out to the Internet in this scenario, AFAIK.
>
> Below is my current NAT rules:
>
> # iptables -L -v -t nat
> Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes)
>  pkts bytes target prot opt in out source
> destination
> 11460  760K RETURN 0--  anyany 192.168.101.0/24
> !192.168.101.0/24
>
> Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes)
>  pkts bytes target prot opt in out source
> destination
> 11408  757K MASQUERADE  0--  anyeth0192.168.101.0/24
> anywhere
>
> Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes)
>  pkts bytes target prot opt in out source
> destination
>
> Please advice if you need more information from me.
>
> Regards,
>
> GNUbie
>

First, drop firewall/iptables/selinux and try again.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[GNUbie]

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 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 is connected to the POTS
SNOM doesn't need to go out to the Internet in this scenario, AFAIK.

Below is my current NAT rules:

# iptables -L -v -t nat
Chain PREROUTING (policy ACCEPT 63795 packets, 7162K bytes)
 pkts bytes target prot opt in out source
destination
11460  760K RETURN 0--  anyany 192.168.101.0/24
!192.168.101.0/24

Chain POSTROUTING (policy ACCEPT 570 packets, 41836 bytes)
 pkts bytes target prot opt in out source
destination
11408  757K MASQUERADE  0--  anyeth0192.168.101.0/24
anywhere

Chain OUTPUT (policy ACCEPT 570 packets, 41836 bytes)
 pkts bytes target prot opt in out source   destination

Please advice if you need more information from me.

Regards,

GNUbie

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Steve Totaro]

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 voices.
>
>> This is not a temporary state?
>
> What do you mean?
>
> Regards,
>
> GNUbie
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[GNUbie]

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,

GNUbie

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Tzafrir Cohen]

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:
> 
> # asterisk -rx 'sip show settings' | grep Codecs
>   Codecs: 0xe (gsm|ulaw|alaw)
> 
> Below is the CLI output:
> 
> -- Executing [EMAIL PROTECTED]:1] Dial("SIP/102-081d11d0",
> "Zap/4/1234567") in new stack
> -- Called 4/1234567
> 
> *CLI> sip show channels
> Peer User/ANRCall ID  Seq (Tx/Rx)  Format
>  Hold Last Message
> 192.168.101.102  102 3c27a6824ba  00101/2  0x4 (ulaw)
>  No   Rx: INVITE
> 1 active SIP channel
> 
> *CLI> core show channels
> Channel  Location State   Application(Data)
> Zap/4-1  [EMAIL PROTECTED] Dialing AppDial((Outgoing Line))
> SIP/102-081d11d0 [EMAIL PROTECTED]:1   RingDial(Zap/4/1234567)
> 2 active channels
> 1 active call

So the call is not established yet, right?

This is not a temporary state?

> 
> > Can you call from the FXO to Asterisk? (e.g.: to echo test)
> 
> There is no problem with an inbound calls. I just tried to call the
> echo test extension number from my mobile phone via FXO/POTS and it
> works fine. I can hear my own voice.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[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 Internet.
In this case, the eth1 of the Asterisk box is connected to the LAN and
eth0 is connected to the Internet.

> 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.

I don't think NAT is involve on this one way audio problem.

Thank you.

Regards,

GNUbie

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[GNUbie]

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: 0xe (gsm|ulaw|alaw)

Below is the CLI output:

-- Executing [EMAIL PROTECTED]:1] Dial("SIP/102-081d11d0",
"Zap/4/1234567") in new stack
-- Called 4/1234567

*CLI> sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Format
 Hold Last Message
192.168.101.102  102 3c27a6824ba  00101/2  0x4 (ulaw)
 No   Rx: INVITE
1 active SIP channel

*CLI> core show channels
Channel  Location State   Application(Data)
Zap/4-1  [EMAIL PROTECTED] Dialing AppDial((Outgoing Line))
SIP/102-081d11d0 [EMAIL PROTECTED]:1   RingDial(Zap/4/1234567)
2 active channels
1 active call

> Can you call from the FXO to Asterisk? (e.g.: to echo test)

There is no problem with an inbound calls. I just tried to call the
echo test extension number from my mobile phone via FXO/POTS and it
works fine. I can hear my own voice.

Thank you.

Regards,

GNUbie

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Gordon Henderson]

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 with 1 FXO (channel 4) and 1 FXS
> (channel 1). My SIP extension phone located inside the LAN is a SNOM
> 300 IP phone.

You mention the SIP phone being inside the LAN. Where is the Asterisk box?

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.

Gordon

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One Way Audio Problem

[Tzafrir Cohen]

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 a Digium TDM400P with 1 FXO (channel 4) and 1 FXS
> (channel 1). My SIP extension phone located inside the LAN is a SNOM
> 300 IP phone.
> 
> This one way audio problem only happens when the SIP extension phone
> (let's call it the CALLER) places an outbound call to a mobile phone
> or analog telephone (let's call it the CALLEE) via FXO/POTS. The
> CALLER can hear the CALLEE's voice but the CALLEE cannot hear the
> CALLER's voice. I used this command "ztmonitor 4 -vv -f /tmp/test.raw"
> to monitor the RX/TX but the TX is totally zero. Below is a sample
> output of the ztmonitor command:
> 
> - - - < s n i p > - - -
> 
> # ztmonitor 4 -vv -f /tmp/test.raw
> Output to /tmp/test.raw
> Run e.g., 'sox -r 8000 -s -w -c 1 /tmp/test.raw /tmp/test.raw.wav' to convert.
> 
> Visual Audio Levels.
> 
>  Use zapata.conf file to adjust the gains if needed.
> 
> ( # = Audio Level  * = Max Audio Hit )
> <(RX <(TX
>  ###*
> Rx:   718 (  718) Tx: 0 (0)

This means Zaptel gets silence from Asterisk.

What codecs are used? What do you see on 'sip show channels'?

Can you call from the FXO to Asterisk? (e.g.: to echo test)

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio...

[Carlos Chavez]

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 wrote:
> 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
> Canall Tecnologia em Comunicações Ltda.
> 
> - "Carlos Chavez" <[EMAIL PROTECTED]> escreveu:
> 
> > I am still having a very frustrating problem win an Avaya-Asterisk
> > system.  I have written about this before but I am expanding the
> > description of the problem just in case someone can give me some
> > insight.
> > 
> > This installation is an Asterisk 1.4.19.1 server connected to an
> > Avaya
> > PBX using a PRI E1.  Integration works great and we can dial from any
> > extension to any extension on both sides.  The problem happens when
> > we
> > connect a Linksys PAP2T outside the network.  If I dial an extension
> > on
> > the Avaya from that PAP2T I get one way audio (I can hear them but
> > they
> > cannot hear me).  This only happens when I dial an extension on the
> > Avaya.  If I dial to the voicemail extension I can get my messages. 
> > I
> > can speak to any SIP extension connected to the Asterisk server.
> > 
> > Here is the strangest part: If they dial the PAP2T from an Ayava
> > extension everything works great, audio both ways.  In this
> > installation
> > there are 45 PAP2T and 45 SPA3102 external extensions.  All the
> > SPA3102
> > extensions do NOT have the problem the PAP2T does.  I always get two
> > way
> > audio with the SPA3102.  When I do an "rtp debug" I can see that
> > incoming RTP packets stop the moment the Avaya extension picks up. 
> > If
> > the PAP2T is connected on the same internal network as the Asterisk
> > then
> > everything works, only when the PAP2T is outside the network do we
> > get
> > one way audio.
> > 
> > The only difference I can find between the configuration of the
> > SPA3102
> > and the PAP2T is a parameter called "Symmetric RTP" which is enabled
> > on
> > the SPA but does not exist on the PAP2T.  I do not know if this has
> > anything to do with the problem but there is nothing else I can find.
> > 
> > Any recommendations on how to tackle this problem?  Right now the
> > only
> > solution I can see is to replace all PAP2T with SPA3102 but obviously
> > I
> > would like to avoid the expense.
> > 
> > -- 
> > Telecomunicaciones Abiertas de México S.A. de C.V.
> > Carlos Chávez Prats
> > Director de Tecnología
> > +52-55-91169161 ext 2001
> > 
> > 
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


signature.asc
Description: This is a digitally signed message part
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio...

[Carlos Chavez]

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 from the web:

(SPA3102 only) Enable symmetric RTP operation. If enabled, the SPA3102
sends RTP packets to the source address and port of the last received
valid inbound RTP packet. If disabled (or before the first RTP packet
arrives) the SPA3102 sends RTP to the destination as indicated in the
inbound SDP.

So I am guessing that the problem is that the inbound SDP is not set
correctly by Asterisk when the call is bridged to the Avaya.

On Thu, 2008-05-08 at 16:38 -0300, Vinícius Fontes wrote:
> 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
> Canall Tecnologia em Comunicações Ltda.
> 
> - "Carlos Chavez" <[EMAIL PROTECTED]> escreveu:
> 
> > I am still having a very frustrating problem win an Avaya-Asterisk
> > system.  I have written about this before but I am expanding the
> > description of the problem just in case someone can give me some
> > insight.
> > 
> > This installation is an Asterisk 1.4.19.1 server connected to an
> > Avaya
> > PBX using a PRI E1.  Integration works great and we can dial from any
> > extension to any extension on both sides.  The problem happens when
> > we
> > connect a Linksys PAP2T outside the network.  If I dial an extension
> > on
> > the Avaya from that PAP2T I get one way audio (I can hear them but
> > they
> > cannot hear me).  This only happens when I dial an extension on the
> > Avaya.  If I dial to the voicemail extension I can get my messages. 
> > I
> > can speak to any SIP extension connected to the Asterisk server.
> > 
> > Here is the strangest part: If they dial the PAP2T from an Ayava
> > extension everything works great, audio both ways.  In this
> > installation
> > there are 45 PAP2T and 45 SPA3102 external extensions.  All the
> > SPA3102
> > extensions do NOT have the problem the PAP2T does.  I always get two
> > way
> > audio with the SPA3102.  When I do an "rtp debug" I can see that
> > incoming RTP packets stop the moment the Avaya extension picks up. 
> > If
> > the PAP2T is connected on the same internal network as the Asterisk
> > then
> > everything works, only when the PAP2T is outside the network do we
> > get
> > one way audio.
> > 
> > The only difference I can find between the configuration of the
> > SPA3102
> > and the PAP2T is a parameter called "Symmetric RTP" which is enabled
> > on
> > the SPA but does not exist on the PAP2T.  I do not know if this has
> > anything to do with the problem but there is nothing else I can find.
> > 
> > Any recommendations on how to tackle this problem?  Right now the
> > only
> > solution I can see is to replace all PAP2T with SPA3102 but obviously
> > I
> > would like to avoid the expense.
> > 
> > -- 
> > Telecomunicaciones Abiertas de México S.A. de C.V.
> > Carlos Chávez Prats
> > Director de Tecnología
> > +52-55-91169161 ext 2001
> > 
> > 
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


signature.asc
Description: This is a digitally signed message part
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio...

[Vinícius Fontes]

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
Canall Tecnologia em Comunicações Ltda.

- "Carlos Chavez" <[EMAIL PROTECTED]> escreveu:

> I am still having a very frustrating problem win an Avaya-Asterisk
> system.  I have written about this before but I am expanding the
> description of the problem just in case someone can give me some
> insight.
> 
>   This installation is an Asterisk 1.4.19.1 server connected to an
> Avaya
> PBX using a PRI E1.  Integration works great and we can dial from any
> extension to any extension on both sides.  The problem happens when
> we
> connect a Linksys PAP2T outside the network.  If I dial an extension
> on
> the Avaya from that PAP2T I get one way audio (I can hear them but
> they
> cannot hear me).  This only happens when I dial an extension on the
> Avaya.  If I dial to the voicemail extension I can get my messages. 
> I
> can speak to any SIP extension connected to the Asterisk server.
> 
>   Here is the strangest part: If they dial the PAP2T from an Ayava
> extension everything works great, audio both ways.  In this
> installation
> there are 45 PAP2T and 45 SPA3102 external extensions.  All the
> SPA3102
> extensions do NOT have the problem the PAP2T does.  I always get two
> way
> audio with the SPA3102.  When I do an "rtp debug" I can see that
> incoming RTP packets stop the moment the Avaya extension picks up. 
> If
> the PAP2T is connected on the same internal network as the Asterisk
> then
> everything works, only when the PAP2T is outside the network do we
> get
> one way audio.
> 
>   The only difference I can find between the configuration of the
> SPA3102
> and the PAP2T is a parameter called "Symmetric RTP" which is enabled
> on
> the SPA but does not exist on the PAP2T.  I do not know if this has
> anything to do with the problem but there is nothing else I can find.
> 
>   Any recommendations on how to tackle this problem?  Right now the
> only
> solution I can see is to replace all PAP2T with SPA3102 but obviously
> I
> would like to avoid the expense.
> 
> -- 
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one way audio after call transfer

[Rilawich Ango]

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 operation.
 I am not sure the problem is related to 1200P, Asterisk or GXP2000.
That's why I seeking the solution from any person who touch the same
problem before.

asterisk version:
asterisk 1.4.15
zaptel 1.4.7
asterisk addons 1.4.5

On Thu, May 1, 2008 at 4:49 AM, Duncan Turnbull <[EMAIL PROTECTED]> wrote:
> 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
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] one way audio after call transfer

[Duncan Turnbull]

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 --blind transfer--> destination 
>B
>Finally A and B reach each others, but there is only one way audio.
>Anyone get the same experience before?  How to solve the problem?
>
>Asterisk vesion:
>Asterisk 1.4.15
>zaptel 1.4.7
>asteriks-addon 1.4.5
>
>___
>-- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
>asterisk-users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio problem on gateway to PSTN after some time, no NAT involved

[François Delawarde]

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.



[EMAIL PROTECTED] wrote:


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 setup is roughly like that

SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN

What happens is, that after a while (uptime was a least two days) the
gateway starts to not transmit audio to the PSTN on outgoing calls, but
the caller can still hear the called party. There is no NAT involved and
firewall rules allow the RTP ports defined in rtp.conf on both asterisk
(A and B) machines. The SIP packages look good, no errors messages from
asterisk or anything else, so I have really no idea what causes it and I
cannot reproduce it except by waiting till it happens again. :(

Now the strange thing is, that if I restart the asterisk all works fine
again. A reload does not help, only a restart. Until now I came across
this phenomenon two times on different machines and it all started about
three weeks ago. Before that I ran asterisk 1.2.10 on the machines and
then updated to 1.2.11. I looked through the Changelog but coulnd't find
anything that seems related, but I guess it's a bug that was introduced
somewhere between 1.2.10 and 1.2.11 ...

Does anyone else have similar problems?

Regards,
Kai

--
Kai Militzer WESTEND GmbH  |  Internet-Business-Provider
Technik  CISCO Systems Partner - Authorized Reseller
Lütticher Straße 10  Tel 0241/701333-14
[EMAIL PROTECTED]   D-52064 Aachen  Fax 0241/911879

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
  

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio problem on gateway to PSTN after some time, no NAT involved

[asterisk]

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 setup is roughly like that

SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN

What happens is, that after a while (uptime was a least two days) the
gateway starts to not transmit audio to the PSTN on outgoing calls, but
the caller can still hear the called party. There is no NAT involved and
firewall rules allow the RTP ports defined in rtp.conf on both asterisk
(A and B) machines. The SIP packages look good, no errors messages from
asterisk or anything else, so I have really no idea what causes it and I
cannot reproduce it except by waiting till it happens again. :(

Now the strange thing is, that if I restart the asterisk all works fine
again. A reload does not help, only a restart. Until now I came across
this phenomenon two times on different machines and it all started about
three weeks ago. Before that I ran asterisk 1.2.10 on the machines and
then updated to 1.2.11. I looked through the Changelog but coulnd't find
anything that seems related, but I guess it's a bug that was introduced
somewhere between 1.2.10 and 1.2.11 ...

Does anyone else have similar problems?

Regards,
Kai

--
Kai Militzer WESTEND GmbH  |  Internet-Business-Provider
Technik  CISCO Systems Partner - Authorized Reseller
Lütticher Straße 10  Tel 0241/701333-14
[EMAIL PROTECTED]   D-52064 Aachen  Fax 0241/911879

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One-way audio after several minutes 1.4.0

[Michael Welter]

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.



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio half way through call

[Matt]

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 <[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 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 iseqno 207 not
> within window 209->211
> Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
> within window 209->211
> Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
> within window 209->211
> Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 208 not
> within window 209->212
>
> Any idea what this means?  To me it looks like it just is missing a
> packet, but why does it not continue?
>
> On 10/23/06, Matt <[EMAIL PROTECTED]> wrote:
> > 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 when bigger jitter appears (>300ms) on
> > > one connection (I have jitterbuffer enabled on IAX),
> > > bigger jitter resulting in bigger "one way audio" probability in my 
case...
> > > PJ
> > >
> > >
> > > Matt wrote:
> > > > Hi,
> > > > I have asterisk 1.2.12 running on my server.   Everything seems to be
> > > > working fine on it.  It has an IAX connection to the
> > > > terminator/orignator.   Again, everything seems to be fine.. calls
> > > > come in and go out.  However, it seems that after a call has been up
> > > > for several minutes audio will go one-way.   That is, we can hear the
> > > > other person, but they can not hear us.
> > > >
> > > > Any thoughts?
> > > > ___
> > > > --Bandwidth and Colocation provided by Easynews.com --
> > > >
> > > > asterisk-users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > > ___
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
>


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio half way through call

[Matt]

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 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 iseqno 207 not
within window 209->211
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
within window 209->211
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
within window 209->211
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 208 not
within window 209->212

Any idea what this means?  To me it looks like it just is missing a
packet, but why does it not continue?

On 10/23/06, Matt <[EMAIL PROTECTED]> wrote:
> 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 when bigger jitter appears (>300ms) on
> > one connection (I have jitterbuffer enabled on IAX),
> > bigger jitter resulting in bigger "one way audio" probability in my case...
> > PJ
> >
> >
> > Matt wrote:
> > > Hi,
> > > I have asterisk 1.2.12 running on my server.   Everything seems to be
> > > working fine on it.  It has an IAX connection to the
> > > terminator/orignator.   Again, everything seems to be fine.. calls
> > > come in and go out.  However, it seems that after a call has been up
> > > for several minutes audio will go one-way.   That is, we can hear the
> > > other person, but they can not hear us.
> > >
> > > Any thoughts?
> > > ___
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] One way audio half way through call

[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 iseqno 207 not
within window 209->211
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
within window 209->211
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
within window 209->211
Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 208 not
within window 209->212

Any idea what this means?  To me it looks like it just is missing a
packet, but why does it not continue?

On 10/23/06, Matt <[EMAIL PROTECTED]> wrote:

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 when bigger jitter appears (>300ms) on
> one connection (I have jitterbuffer enabled on IAX),
> bigger jitter resulting in bigger "one way audio" probability in my case...
> PJ
>
>
> Matt wrote:
> > Hi,
> > I have asterisk 1.2.12 running on my server.   Everything seems to be
> > working fine on it.  It has an IAX connection to the
> > terminator/orignator.   Again, everything seems to be fine.. calls
> > come in and go out.  However, it seems that after a call has been up
> > for several minutes audio will go one-way.   That is, we can hear the
> > other person, but they can not hear us.
> >
> > Any thoughts?
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users