Can't hear voice mail message

Is Xlite setup correctly, Might be worth trying a different softphone. sjphone for example.

Ian

Is Xlite setup correctly, Might be worth trying a different softphone. sjphone for example.

Ian

I have both wengo and xlite tried. Same problem, no audio from Asterix. I am desperatedly looking for a solution...

Hi

Have you made all the changes with respect to codecs localhost etc.
Have you set up a echotest extension? and do extension to extension calls work ?

Can you show what rtp debug shows ? as your original one shows no traffic to the softphone just from it.


Ian

I did the echo test, but did not hear anything back. The CLI output is:
*CLI> rtp debug
RTP Debugging Enabled
*CLI> [Mar 17 08:43:08] -- Executing [#8108449863246@my-phones:1] Dial("SIP/2000-081e0ca0", "SIP/08449863246@voipuser|60") in new stack
[Mar 17 08:43:08] -- Called 08449863246@voipuser
[Mar 17 08:43:09] -- SIP/voipuser-081e4c18 answered SIP/2000-081e0ca0
[Mar 17 08:43:09] Got RTP packet from 192.168.1.102:2920 (type 00, seq 004378, ts 1300600, len 000160)
[Mar 17 08:43:09] Sent RTP packet to 87.238.75.71:37532 (type 08, seq 040159, ts 1300600, len 000160)
[Mar 17 08:43:09] Got RTP packet from 192.168.1.102:2920 (type 00, seq 004379, ts 1300760, len 000160)
[Mar 17 08:43:09] Sent RTP packet to 87.238.75.71:37532 (type 08, seq 040160, ts 1300760, len 000160)
[Mar 17 08:43:09] Got RTP packet from 192.168.1.102:2920 (type 00, seq 004380, ts 1300920, len 000160)
[Mar 17 08:43:09] Sent RTP packet to 87.238.75.71:37532 (type 08, seq 040161, ts 1300920, len 000160)
[Mar 17 08:43:09] Got RTP packet from 192.168.1.102:2920 (type 00, seq 004381, ts 1301080, len 000160)
[Mar 17 08:43:09] Sent RTP packet to 87.238.75.71:37532 (type 08, seq 040162, ts 1301080, len 000160)


In my sip.conf, I have:

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

register =>outraspace:whatever@voipuser.org/outraspace
nat=yes
externip=58.251.75.251
localnet=192.168.1.1/255.255.255.0
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
qualify=yes

If I call a landline, the landline can hear my voice, but I can't hear the landline from the SIP client.

Here is another log for the voicemail test:

[Mar 17 08:49:45] == Spawn extension (my-phones, #81008613828755048, 1) exited non-zero on 'SIP/2000-081df4c8'
[Mar 17 08:49:45] WARNING[11322]: chan_sip.c:12735 handle_response: Remote host can't match request BYE to call '4361ec096073b4ea6561c4d31b93be7d@voipuser.org'. Giving up.
[Mar 17 08:49:57] -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org, port 5060
[Mar 17 08:50:04] NOTICE[11326]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50)
[Mar 17 08:50:06] -- Executing [2999@my-phones:1] VoiceMailMain("SIP/2000-081db3e8", "2000|s") in new stack[Mar 17 08:50:06] Sent RTP packet to 192.168.1.102:2372 (type 00, seq 058336, ts 000160, len 000160)
[Mar 17 08:50:06] -- <SIP/2000-081db3e8> Playing 'vm-youhave' (language 'en')
[Mar 17 08:50:07] Got RTP packet from 192.168.1.102:2372 (type 00, seq 000781, ts 587500, len 000160)
[Mar 17 08:50:07] Got RTP packet from 192.168.1.102:2372 (type 00, seq 000782, ts 587660, len 000160)
[Mar 17 08:50:07] Got RTP packet from 192.168.1.102:2372 (type 00, seq 000783, ts 587820, len 000160)
[Mar 17 08:50:07] Got RTP packet from 192.168.1.102:2372 (type 00, seq 000784, ts 587980, len 000160)
[Mar 17 08:50:07] Got RTP packet from 192.168.1.102:2372 (type 00, seq 000785, ts 588140, len 000160)
[Mar 17 08:50:07] Got RTP packet from 192.168.1.102:2372 (type 00, seq 000786, ts 588300, len 000160)
[Mar 17 08:50:07] Got RTP packet from 192.168.1.102:2372 (type 00, seq 000787, ts 588460, len 000160)
[Mar 17 08:50:07] Got RTP packet from 192.168.1.102:2372 (type 00, seq 000788, ts 588620, len 000160)
[Mar 17 08:50:07] Got RTP packet from 192.168.1.102:2372 (type 00, seq 000789, ts 588780, len 000160)
[Mar 17 08:50:07] Got RTP packet from 192.168.1.102:2372 (type 00, seq 000790, ts 588940, len 000160)
[Mar 17 08:50:07] Got RTP packet from 192.168.1.102:2372 (type 00, seq 000791, ts 589100, len 000160)

From the log, I can see the "Sent Packet...".

Thank you so much for all your help.

Hi

I did mention changing
localnet=192.168.1.1/255.255.255.0 to localnet=192.168.1.0/255.255.255.0

also I can see no rtp packets being sent to the softphone only from it.

If you have put the server in a defined DMZ remove it. NAT and if required port forwarding will deal with it, but at the moment that isnt required..

Ian

Hi,

I removed my sever from DMZ.
My sip.conf is now changed to:

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

register =>outraspace:whatever@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
qualify=yes


When running the VoiceMail test, I get :



*CLI> [Mar 17 09:30:15] -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host sip.voipuser.org, port 5060
[Mar 17 09:30:16] -- Executing [2999@my-phones:1] VoiceMailMain("SIP/2000-081e0960", "2000|s") in new stack[Mar 17 09:30:16] Sent RTP packet to 192.168.1.102:29088 (type 00, seq 054509, ts 000160, len 000160)
[Mar 17 09:30:16] -- <SIP/2000-081e0960> Playing 'vm-youhave' (language 'en')
[Mar 17 09:30:17] Got RTP packet from 192.168.1.102:29088 (type 00, seq 007573, ts 2281400, len 000160)
[Mar 17 09:30:17] Got RTP packet from 192.168.1.102:29088 (type 00, seq 007574, ts 2281560, len 000160)
[Mar 17 09:30:17] Got RTP packet from 192.168.1.102:29088 (type 00, seq 007575, ts 2281720, len 000160)
[Mar 17 09:30:17] Got RTP packet from 192.168.1.102:29088 (type 00, seq 007576, ts 2281880, len 000160)
[Mar 17 09:30:17] Got RTP packet from 192.168.1.102:29088 (type 00, seq 007577, ts 2282040, len 000160)


I highlighted the sent packet in red but I don't know if this is the audio packet or something else.

Pete

Hi
Thats one packet! asterisk is sending a packet every 20ms your screen will be flooded by them.

I have a feeling there is something fundamentally wrong here as its not even showing packets sent to the wrong destination.

can you setup a echotest number on your server and test to that.

Ian

Sorry. I am a newbie here. Can you kindly explain what you mean by setting up the echotest on server?

Hi,
I run a simple echotest with extensions.conf:

exten=>222,1,Answer()
exten=>222,2,Echo()
exten=>222,3,Hangup()

It seems to work:

debian*CLI>
-- Executing [222@my-phones:1] Answer("SIP/2000-b6d06750", "") in new stack
-- Executing [222@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:

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@my-phones:1] Answer("SIP/2000-b6d09708", "") in new stack
-- Executing [333@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)

Hi,
I think the problem may not be with the network as I tried running another SIP client inside the Debian box running asterisk server. It stills does not sent the packet:

-- Executing [2999@my-phones:1] VoiceMailMain("SIP/2002-b6d09c08", "2002|s") in new stack
Sent RTP packet to 192.168.1.101:8000 (type 00, seq 060593, ts 000160, len 000160)
-- <SIP/2002-b6d09c08> Playing 'vm-login' (language 'en')
Got RTP packet from 192.168.1.101:8000 (type 00, seq 005890, ts 2437337283, len 000160)
Got RTP packet from 192.168.1.101:8000 (type 00, seq 005891, ts 2437337443, len 000160)
Got RTP packet from 192.168.1.101:8000 (type 00, seq 005892, ts 2437337603, len 000160)
Got RTP packet from 192.168.1.101:8000 (type 00, seq 005893, ts 2437337763, len 000160)
Got RTP packet from 192.168.1.101:8000 (type 00, seq 005894, ts 2437337923, len 000160)
Got RTP packet from 192.168.1.101:8000 (type 00, seq 005895, ts 2437338083, len 000160)
Got RTP packet from 192.168.1.101:8000 (type 00, seq 005921, ts 2437342243, len 000160)
Got RTP packet from 192.168.1.101:8000 (type 00, seq 005922, ts 2437342403, len 000160)
[Mar 17 23:33:58] WARNING[15735]: app_voicemail.c:6304 vm_authenticate: Couldn't read username

Hi

I have seen this before, and echotest has proved it, It seems to be down to misconfigured zaptel.

Check you configuration for zaptel, If you have no hardware make sure you are loadingthe ztdummy driver.

Ian

Hi,
In CLI, I tried
*CLI> zap show status
Description Alarms IRQ bpviol CRC4
ZTDUMMY/1 (source: RTC) 1 UNCONFIGUR 0 0 0


It seems like Ztdummy is loaded. Is this a correct result? I am pretty sure zaptel is installed. Is this a configuration issue? If so, how to fix it? Is there anyway to test whether the ztdummy is functioning properly?

If your problem is just of hearing Prompts at X-LITE then,
Just Give Stun under X-Lite as stun.counterpath.net.

You'll find this under Topology tab of sip account properties window.

Cheers,
Rahul