Can't hear voice mail message

Hello

I have my Asterix running on Debian and then use xLite as a client on another computer within the network.

I am running a test for the voicemail feature. But after 20s no response, I can't hear anything ( voice mail menu ) from the xlite, but I can see the text

"Playing 'vm-theperson' (language 'en')" from Asterisk console.

Same thing for the Voicemail manual.

Can anyone tell me how to fix this? I think it may just be a minor config issue.

Thank you very much in advance for you kind help.

Are you using SIP? if so, it could be a NAT problem. Check your NAT settings in the sip.conf. Check the SIP packets for references to IP's in non-routable networks.

Hi

You need to check that you nat settings are correct IE localnet settings and canreinvite is correct (note this has changed in 1.4). Are all on the same lan segment?

Ian

Hi,
I have put my server in the DNZ and in "My SIP Account" page, is shows that my server is not inside NAT. My userid is outraspace.

In my sip.conf, I have

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

register =>outraspace:whatever@voipuser.org/outraspace
nat=no
externip=58.251.75.255
localnet=192.168.1.1/255.255.255.0
canreinvite=no


I have tried turning nat to yes and no,but they don't work.
Any idea how to debug this problem?

Thank you very much for your kind help.

Hi

Ok that looks OK I assume the softphone is also on the same 192.168.1. network.

it may be worth running rtp debug at the asterisk command to see where the packets are going.

Ian

Hi,
When I run rpt debug, it seems like the packet is running in an infinite loop:

-- Executing [2001@my-phones:1] Dial("SIP/2000-081da668", "SIP/2001|20") in new stack
-- Called 2001
-- SIP/2001-081d4df0 is ringing
-- Nobody picked up in 20000 ms
-- Executing [2001@my-phones:2] VoiceMail("SIP/2000-081da668", "2001|u") in new stack
Sent RTP packet to 192.168.1.102:52128 (type 00, seq 050827, ts 000160, len 000160)
-- <SIP/2000-081da668> Playing 'vm-theperson' (language 'en')
Got RTP packet from 192.168.1.102:52128 (type 00, seq 000232, ts 2375600, len 000160)
Got RTP packet from 192.168.1.102:52128 (type 00, seq 000233, ts 2375760, len 000160)
Got RTP packet from 192.168.1.102:52128 (type 00, seq 000234, ts 2375920, len 000160)
Got RTP packet from 192.168.1.102:52128 (type 00, seq 000235, ts 2376080, len 000160)
Got RTP packet from 192.168.1.102:52128 (type 00, seq 000236, ts 2376240, len 000160)


Any idea how to fix it?

Thank you very much for all your help.

I don't understand this cuz my client is running on 192.168.1.100 and my asterisk is running on 192.168.1.101.
how come it is point to 192.168.1.102?

Is there something wrong?

Hi

OK

could you post the sip.conf for the phone.

output of ifconfig and for that matter ipconfig on pc

also what is .102 ?

It seems you have some type of network issue.

Ian

Hi:

My sip.conf is :
[general]
port = 5060
bindaddr = 0.0.0.0
context = others

register =>outraspace:pass@voipuser.org/outraspace
nat=yes
externip=58.251.75.251
localnet=192.168.1.1/255.255.255.0
canreinvite=yes

[2000]
type=friend
context=my-phones
secret=1234
host=dynamic

[2001]
type=friend
context=my-phones
secret=1234
host=dynamic

[voipuser]
type=friend
context=from-voip-provider
username=outraspace
fromuser=outraspace
secret=pass
host=voipuser.org
fromdomain=voipuser.org
qualify=no
insecure=very
nat=yes


My asterisk's ifconfig

debian:/etc# ifconfig
eth0 Link encap:Ethernet HWaddr 00:1D:7D:43:88:F7
inet addr:192.168.1.101 Bcast:255.255.255.255 Mask:255.255.255.0
UP BROADCAST RUNNING MULTICAST MTU:576 Metric:1
RX packets:122469 errors:0 dropped:1835483480 overruns:0 frame:0
TX packets:20236 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:29269175 (27.9 MiB) TX bytes:3011511 (2.8 MiB)
Interrupt:177 Base address:0x2000

lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:1780 errors:0 dropped:0 overruns:0 frame:0
TX packets:1780 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:104473 (102.0 KiB) TX bytes:104473 (102.0 KiB)

My PC(SIP client)'s ipconfig is:

Windows IP Configuration


Ethernet adapter Local Area Connection 2:

Media State . . . . . . . . . . . : Media disconnected
Connection-specific DNS Suffix . :

Wireless LAN adapter Wireless Network Connection:

Connection-specific DNS Suffix . :
Link-local IPv6 Address . . . . . : fe80::7c37:86cf:5075:c430%9
Autoconfiguration IPv4 Address. . : 169.254.196.48
Subnet Mask . . . . . . . . . . . : 255.255.0.0
Default Gateway . . . . . . . . . :

Ethernet adapter Local Area Connection:

Connection-specific DNS Suffix . :
Link-local IPv6 Address . . . . . : fe80::3849:6abc:b15d:a5ee%8
IPv4 Address. . . . . . . . . . . : 192.168.1.102
Subnet Mask . . . . . . . . . . . : 255.255.255.0
Default Gateway . . . . . . . . . : 192.168.1.1

Tunnel adapter Local Area Connection* 6:

Connection-specific DNS Suffix . :
Link-local IPv6 Address . . . . . : fe80::5efe:192.168.1.102%10
Default Gateway . . . . . . . . . :

Tunnel adapter Local Area Connection* 7:

Connection-specific DNS Suffix . :
Link-local IPv6 Address . . . . . : fe80::5efe:169.254.196.48%14
Default Gateway . . . . . . . . . :

Tunnel adapter Local Area Connection* 9:

Media State . . . . . . . . . . . : Media disconnected
Connection-specific DNS Suffix . :

OK so your PC is 102 and not 100 as you thought.

could you change localnet=192.168.1.1/255.255.255.0 to localnet=192.168.1.0/255.255.255.0

and also add some codec information

such as

dissalow=all
allow=ulaw
allow=alaw

and see what you get

Ian

Hi,
I am still getting the error:

[Mar 16 20:21:07] NOTICE[2973]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50)
-- Unregistered SIP '2000'
-- Registered SIP '2000' at 192.168.1.102 port 13962 expires 3600
-- Saved useragent "X-Lite release 1011s stamp 41150" for peer 2000
-- Executing [2999@my-phones:1] VoiceMailMain("SIP/2000-081debe8", "2000|s") in new stack
Sent RTP packet to 192.168.1.102:19092 (type 00, seq 001116, ts 000160, len 000160)
-- <SIP/2000-081debe8> Playing 'vm-youhave' (language 'en')
Got RTP packet from 192.168.1.102:19092 (type 00, seq 003560, ts 139800, len 000160)
Got RTP packet from 192.168.1.102:19092 (type 00, seq 003561, ts 139960, len 000160)
Got RTP packet from 192.168.1.102:19092 (type 00, seq 003562, ts 140120, len 000160)
Got RTP packet from 192.168.1.102:19092 (type 00, seq 003563, ts 140280, len 000160)
Got RTP packet from 192.168.1.102:19092 (type 00, seq 003564, ts 140440, len 000160)


I checked, the asterisk is on .101 and client is on .102.

It seems like my client pc is not responing to the RTP packet? Is it something that I have to do with my router?
Both company is inside the 192 lan and I explicitly put the asterisk server inside the DMZ. Any idea?

I run sniffer on my client, and saw this:

10 9.424368 192.168.1.101 192.168.1.102 RTP PT=ITU-T G.711 PCMU, SSRC=0x5DCD0323, Seq=6787, Time=160

It seems like the RTP packet did get sent to the client pc? Is it true?

I can't figure out why. I run sniffer and can see the RTP packet sent to the client's PC. But I just can't hear the audio from the voicemail. What can be the problem? Please help out.

Pete