Hi,
Asterisk is making a call to a peer. In 200 ok, peer is sending its
application server ip in contact field, so asterisk sends ACK to that IP.
RTP starts flowing between endpoints and peer plays an IVR and asks for
destination number. After entering destination number peer's application
Hello all,
I have a problem where problem with one way audio, and I think it's
related to a=sendonly and a re-invite. Can anyone please assist?
The scenario is as follows
- We send an INVITE to a peer, and it replies with a 100 Trying, and
then a 183 Session Progress message containing
On Thu, 16 Oct 2008, GNUbie wrote:
Hello,
On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
A packet trace will probably show exactly what is happening. Try:
tcpdump -nlXs 8192 -i eth0 port 5060
You should be able to see the SIP information going back
GNUbie wrote:
What particular configs are you looking for? Below is my current setup
and scenario:
[snom] ==LAN== [asterisk] ==FXO/POTS == [analog_telephone/mobile_phone]
SNOM is using the 192.168.101.102 IP address
Asterisk is using 192.168.101.1 IP address for its eth1 interface
FXO port
On Thu, Oct 16, 2008 at 09:22:01AM +0800, GNUbie wrote:
Hello Daniel,
On Tue, Oct 14, 2008 at 12:12 AM, Daniel Hazelbaker
[EMAIL PROTECTED] wrote:
Might be a stretch, but does the Asterisk log show that the call was
answered? I had this problem when interfacing * with an NEC system to
Hi,
Am Donnerstag, den 16.10.2008, 09:37 +0800 schrieb GNUbie:
Hello Karsten,
On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote:
Please post Your sip.conf.
Which IP-Address do You configure in the snom for Your asterisk? (eth0
or eth1)?
The SNOM 300 is
Hello,
On Tue, Oct 14, 2008 at 12:07 AM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
A packet trace will probably show exactly what is happening. Try:
tcpdump -nlXs 8192 -i eth0 port 5060
You should be able to see the SIP information going back and forth and
will probably show you that your
Hello Daniel,
On Tue, Oct 14, 2008 at 12:12 AM, Daniel Hazelbaker
[EMAIL PROTECTED] wrote:
Might be a stretch, but does the Asterisk log show that the call was
answered? I had this problem when interfacing * with an NEC system to
do call parking pickup. The NEC would never give a dialtone
Hello Karsten,
On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote:
Please post Your sip.conf.
Which IP-Address do You configure in the snom for Your asterisk? (eth0
or eth1)?
The SNOM 300 is using the NET interface beside the DC 5V port to
connect to the LAN.
The
Did you try it the magic number of times, three?
On Sun, Oct 12, 2008 at 9:57 PM, GNUbie [EMAIL PROTECTED] wrote:
Hello Tzafrir,
On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
This means Zaptel gets silence from Asterisk.
What codecs are used? What do you see on
Change all canreinvites to no.
On Wed, Oct 15, 2008 at 9:37 PM, GNUbie [EMAIL PROTECTED] wrote:
Hello Karsten,
On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer [EMAIL PROTECTED] wrote:
Please post Your sip.conf.
Which IP-Address do You configure in the snom for Your asterisk? (eth0
or
canreinvite defaults to yes, whether specified or not.
http://www.voip-info.org/wiki/view/tips
If you follow these directions adapting to your particular
circumstances and it doesn't work, post your whole sip.conf
Start asterisk with verbose set to 3 or so and turn on sip debugging.
I get
Hello Steve,
On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Did you try it the magic number of times, three?
I'm sorry. What do you mean?
Regards,
GNUbie
___
-- Bandwidth and Colocation Provided by
Maybe I have my threads confused but I thought you got one way audio
when three calls were made, you only mentioned one call.
On Thu, Oct 16, 2008 at 12:44 AM, GNUbie [EMAIL PROTECTED] wrote:
Hello Steve,
On Thu, Oct 16, 2008 at 12:04 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Did you try it
Sorry, wrong thread, time for bed. I thought this was the thread
where the guy was having issues with one way audio on his third call,
and his Asterisk server was behind NAT.
Good night everyone and have pleasant dreams of 700 point drops in the DOW!
OT, did you know if the government took the
Hello Steve,
On Thu, Oct 16, 2008 at 12:42 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
canreinvite defaults to yes, whether specified or not.
http://www.voip-info.org/wiki/view/tips
If you follow these directions adapting to your particular
circumstances and it doesn't work, post your whole
On Mon, Oct 13, 2008 at 09:57:33AM +0800, GNUbie wrote:
Hello Tzafrir,
On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
This means Zaptel gets silence from Asterisk.
What codecs are used? What do you see on 'sip show channels'?
I am using the following codecs:
Hello Tzafrir,
On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
So the call is not established yet, right?
It is already. The CALLER hears the CALLEE's voice but the CALLEE
cannot hear the CALLER's voices.
This is not a temporary state?
What do you mean?
Regards,
On Mon, Oct 13, 2008 at 9:04 AM, GNUbie [EMAIL PROTECTED] wrote:
Hello Tzafrir,
On Mon, Oct 13, 2008 at 3:33 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
So the call is not established yet, right?
It is already. The CALLER hears the CALLEE's voice but the CALLEE
cannot hear the CALLER's
Hello Steve,
On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
If you are going to dismiss (the most probable) problem (NAT) without
posting configs, I am not sure how much help you will get, you will
probably be dismissed as well.
What particular configs are you looking
On Mon, Oct 13, 2008 at 10:49 AM, GNUbie [EMAIL PROTECTED] wrote:
Hello Steve,
On Mon, Oct 13, 2008 at 9:52 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
If you are going to dismiss (the most probable) problem (NAT) without
posting configs, I am not sure how much help you will get, you will
On Oct 13, 2008, at 9:29 AM, [EMAIL PROTECTED]
wrote:
IME: One-way audio problems are almost always casued by NAT gateways
and/or incorrect NAT settings in sip.conf and/or incorrect IP
address or
extenal proxy settings in the SIP phone.
And reinvite issues in particular. Those have been
Hello Norman,
On Mon, Oct 13, 2008 at 11:02 PM, Norman Franke [EMAIL PROTECTED] wrote:
And reinvite issues in particular. Those have been the only one-way
audio problems I've experienced. Setting reinvite=no fixed everything
for me.
You mean, canreinvite=no? I already have done line on my
Hello Steve,
On Mon, Oct 13, 2008 at 10:59 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
First, drop firewall/iptables/selinux and try again.
I already turned off the firewall and I don't have SELinux on my
system and the problem is still there.
Regards,
GNUbie
A packet trace will probably show exactly what is happening. Try:
tcpdump -nlXs 8192 -i eth0 port 5060
You should be able to see the SIP information going back and forth and
will probably show you that your NAT rules are applying when they
shouldn't. I agree with first turning off your
Might be a stretch, but does the Asterisk log show that the call was
answered? I had this problem when interfacing * with an NEC system to
do call parking pickup. The NEC would never give a dialtone (nor did
it give answer supervision) so * never knew the call got picked up so
audio only
Hi,
Am Montag, den 13.10.2008, 10:00 +0800 schrieb GNUbie:
Hello Gordon,
On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
You mention the SIP phone being inside the LAN. Where is the Asterisk box?
It is the main gateway of the IP phones and my laptop to the
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
On Sun, Oct 12, 2008 at 11:53:18PM +0800, GNUbie wrote:
Hello all,
I've been lobbying for some time at the #asterisk IRC channel. Until
now, I still can't find a solution to my one way audio problem. I
rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my
Debian Etch. I got
On Sun, 12 Oct 2008, GNUbie wrote:
Hello all,
I've been lobbying for some time at the #asterisk IRC channel. Until
now, I still can't find a solution to my one way audio problem. I
rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my
Debian Etch. I got a Digium TDM400P
Hello Tzafrir,
On Mon, Oct 13, 2008 at 2:12 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
This means Zaptel gets silence from Asterisk.
What codecs are used? What do you see on 'sip show channels'?
I am using the following codecs:
# asterisk -rx 'sip show settings' | grep Codecs
Codecs:
Hello Gordon,
On Mon, Oct 13, 2008 at 2:22 AM, Gordon Henderson
[EMAIL PROTECTED] wrote:
You mention the SIP phone being inside the LAN. Where is the Asterisk box?
It is the main gateway of the IP phones and my laptop to the Internet.
In this case, the eth1 of the Asterisk box is connected to
Did you solved this Problem?
I have the same problem, and i can't solve it, did you know anything
about?
Thanks
Nico
On Thu, 14 Sep 2006, Kai Militzer wrote:
Hello everyone,
since some weeks I experience strange problems on my gateways to the
PSTN. The gateways use chan_ss7 and SIP. My
Hi,
I have similar symptoms (usually one-way audio like you, but sometimes
echoed, distorded, or low volume sound), in a simpler configuration,
using just SIP with a few phones and a TDM400 card with two FXOs:
Asterisk -- PSTN
I have kernel 2.6.18-XEN and using Asterisk 1.4
François.
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
Hi Kai,
we had a similar problem with a PBX which had PSTN lines and SIP phones:
sometimes some phones had one way calls...the caller couldn't hear. We
hadn't tried to restart but we reduced the number of RTP ports (rtp.conf
if memory helps!) to a range of 200 (it depends from the number of
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