VoIP Mailing List Archives

[petedao at gmail.com]

Hi,
I am new to Asterisk and I am having a setup problem that I am trying to
resolved for the last couple days without any success. I am pretty much
desperated on this issue and I don't know why. Can someone please kindly
help me to troubleshoot this? I can't hear any audio from Asterisk when
running Playback or VoiceMail tests.

I have my Asterisk server ( running on Debian, 192.168.1.101 ) and Xlite
(running on Vista, 192.168.1.102) on two different machine within the same
Lan. My network is ADSL ( home-based ) with a dynamic IP.

When I run the
exten=>222,1,Answer()
exten=>222,2,Echo()
exten=>222,3,Hangup()

It works as I am getting RTP packet sent and receied and I can hear the echo
audio.

debian*CLI>
-- Executing [222 at my-phones:1] Answer("SIP/2000-b6d06750", "") in new stack
-- Executing [222 at my-phones:2] Echo("SIP/2000-b6d06750", "") in new stack
Got RTP packet from 192.168.1.102:42406 (type 00, seq 003468, ts 2904300,
len 000160)
Sent RTP packet to 192.168.1.102:42406 (type 00, seq 002928, ts 2904296, len
000160)
Got RTP packet from 192.168.1.102:42406 (type 00, seq 003469, ts 2904460,
len 000160)
Sent RTP packet to 192.168.1.102:42406 (type 00, seq 002929, ts 2904456, len
000160)
Got RTP packet from 192.168.1.102:42406 (type 00, seq 003470, ts 2904620,
len 000160)
Sent RTP packet to 192.168.1.102:42406 (type 00, seq 002930, ts 2904616, len
000160)
Got RTP packet from 192.168.1.102:42406 (type 00, seq 003471, ts 2904780,
len 000160)
Sent RTP packet to 192.168.1.102:42406 (type 00, seq 002931, ts 2904776, le
But if I run this, it does not work and I can't hear any of the playback.
from the console, the packet is not sent to the client.

exten=>333,1,Answer()
exten=>333,2,Playback(vm-goodbye)
exten=>333,3,Hangup()



It does not work and the console output is:

-- Executing [333 at my-phones:1] Answer("SIP/2000-b6d09708", "") in new stack
-- Executing [333 at my-phones:2] Playback("SIP/2000-b6d09708", "vm-goodbye")
in new stack
Sent RTP packet to 192.168.1.102:61588 (type 00, seq 017315, ts 000160, len
000160)
-- <SIP/2000-b6d09708> Playing 'vm-goodbye' (language 'en')
Got RTP packet from 192.168.1.102:61588 (type 00, seq 005474, ts 052000, len
000160)
Got RTP packet from 192.168.1.102:61588 (type 00, seq 005475, ts 052160, len
000160)
Got RTP packet from 192.168.1.102:61588 (type 00, seq 005476, ts 052320, len
000160)
Got RTP packet from 192.168.1.102:61588 (type 00, seq 005477, ts 052480, len
000160)
Got RTP packet from 192.168.1.102:61588 (type 00, seq 005478, ts 052640, len
000160)

My sip.conf is like this:


[general]
port = 5060
bindaddr = 0.0.0.0
context = others

register =>userid:pass at voipuser.org/userid
nat=yes
externip=58.251.75.233
localnet=192.168.1.0/255.255.255.0
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
qualify=yes


Thank you very much for all your kind help.

Regards,
Pete
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[anselm at hoffmeister-...]

Am Montag, den 17.03.2008, 15:08 +0800 schrieb Pete Kay:

[petedao at gmail.com]

Hi,
Thanks for pointing out. I checked the extenip and it is fine. The thing
is that I have already configure gsm as one of the codec in the sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = others

register =>outraspace:whatever at voipuser.org/outraspace
nat=yes
externip=58.251.75.333
localnet=192.168.1.0/255.255.255.0
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
qualify=yes

Any other hints?
On Mon, Mar 17, 2008 at 6:47 PM, Anselm Martin Hoffmeister <
anselm at hoffmeister-online.de> wrote:

[stotaro at totarotechn...]

SIP debug output please.

Thanks,
Steve Totaro

On Mon, Mar 17, 2008 at 7:17 AM, Pete Kay <petedao at gmail.com> wrote:

[james.texter at gmail.com]

Try putting in a wait after you answer. It's possible the message is
playing before the RTP is setup. I would change your dialplan to be

exten => 333,1,Answer()
exten => 333,n,Wait(1)
exten => 333,n,Playback(vm-goodbye)
exten => 333,n,Hangup()

HTH,

James

On Mar 17, 2008, at 5:47 AM, Anselm Martin Hoffmeister wrote:

[stotaro at totarotechn...]

Paste the sip.conf for your softphone.

Thanks,
Steve Totaro

On Mon, Mar 17, 2008 at 9:38 AM, Pete Kay <petedao at gmail.com> wrote:

[petedao at gmail.com]

Hi,
My Sip.conf is like this:

[general]
port = 5060
bindaddr = 0.0.0.0
context = others

register =>outraspace:whatever at voipuser.org/outraspace
nat=yes
externip=58.251.75.251
localnet=192.168.1.0/255.255.255.0
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
qualify=yes
On Mon, Mar 17, 2008 at 9:57 PM, Steve Totaro <
stotaro at totarotechnologies.com> wrote:

[anselm at hoffmeister-...]

Am Montag, den 17.03.2008, 21:38 +0800 schrieb Pete Kay:

[asterisk.org at sedwar...]

On Mon, 17 Mar 2008, Pete Kay wrote:

[petedao at gmail.com]

Hi,

It may seems like my lack of audio problem with PlayBack is due to zaptel
setting.
When I tried to start zaptel, I keep getting errors:

debian:/etc/init.d# ./zaptel start
Zaptel telephony kernel driver: ioctl(ZT_LOADZONE) failed: Invalid argument
Notice: Configuration file is /etc/zaptel.conf
line 223: Unable to register tone zone 'uk'
zaptel.
I changed tone zone to something else and does not work. What is wrong?
Can anyone please give me some hint?

Thank you very much for your help.

Pete
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[stotaro at totarotechn...]

Pete,

You are connecting via a SIP softphone correct? Where is that in your sip.conf?

On Mon, Mar 17, 2008 at 11:42 AM, Pete Kay <petedao at gmail.com> wrote:

[stotaro at totarotechn...]

Did you try US? modprobe zaptel. Do you have any zaptel hardware?
modprobe ztdummy.

On Mon, Mar 17, 2008 at 1:01 PM, Pete Kay <petedao at gmail.com> wrote:

[tzafrir.cohen at xorco...]

On Tue, Mar 18, 2008 at 01:01:02AM +0800, Pete Kay wrote:

[Guest]

I agree, seems odd you didn't have a [peername] section for your
softphone in your sip.conf.

aren't 404 errors a likely symptom of this?

Mojo
Steve Totaro wrote:

[jsmith at digium.com]

On Tue, 2008-03-18 at 01:01 +0800, Pete Kay wrote: