No Sound when Accessing VoicemailMain

Asterisk Asterisk Support
1

Hi Everyone,

I am attempting to access voicemailmain to hear messages, but when i do, i cannot hear any sound at all. This procedure works great when using SIP hardphones but when using Linksys Pap2t-NA ATA units it does not function properly, because I cannot hear a thing. I acted as if i heard something and put in my password though i did not hear the vioce response system ask me anything, just to debug if it woudl work at all. How can I make the sound play so that i can hear my messages, for this ATA units i am using dtmfmode=rfc2833 in sip.conf?

Here is the feedback from the console:

-- Executing Answer("SIP/31002-0928d980", "") in new stack
-- Executing VoiceMailMain("SIP/31002-0928d980", "31002") in new stack
-- Playing 'vm-password' (language 'en')
-- Playing 'vm-youhave' (language 'en')
-- Playing 'digits/1' (language 'en')
-- Playing 'vm-INBOX' (language 'en')
-- Playing 'vm-and' (language 'en')
-- Playing 'digits/2' (language 'en')
-- Playing 'vm-Old' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-onefor' (language 'en')
-- Playing 'vm-INBOX' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-helpexit' (language 'en')
-- Playing 'vm-onefor' (language 'en')
-- Playing 'vm-INBOX' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')

== Spawn extension (default, 231002, 2) exited non-zero on ‘SIP/31002-0928d980’

I dial 86231002 to get to voicemailmain and the first two digits are stripped.

Thanks in advance for any help.

2

i use a PAP2 here to provide 2 DECT phones, aside from DTMF tweaking they have worked faultlessly for over a year … even in VM. it’s not a codec issue you’re experiencing here is it ?

3

Hi Everyone,

Thanks for the reply Baconbuttie and I am not quite sure what’s causing the no sound issue. It is only occurring for the PAP2, not any of the sip hardphones. But to give you a little background, I also have openSER in the foreground and I am having someone else analyze the openser.cfg file on another system to ensure the problem does not lie within this file. He, too, uses PAP2’s in his integrated (openser/asterisk) setup and like you, he has not experienced any problems with regard to sound.

If I learn that the file works fine on the other system, then I will continue this thread, but this may not be entirely on Asterisk.

Again, thanks for your response on this issue,

tlofton1000

4

I feel for you brother. I had this same issue using a shitty netcomm V85. Actually, the whole site had them - 30 to be exact. All of them were fine with VoicemailMain except one extension - ext 26.
I never found the problem - It was so odd. The config file was the same for every phone, the phone was factory defaulted to make sure but nada.

There is no happy ending to this story.

They have since moved to Linksys SPA941s which are great, and no VoicemailMain problem. Very bizarre

5

Hi mylo78,

I have come to the conclusoin that this PAP2T-NA device has a bug in it somewhere that is not transmitting signals from Asterisk to the earpiece (speaker) to me nor transmitting signals from the phone to Asterisk as it awaits input from the phone (password, mailbox, etc). It’s as if the signals are possibly splitting and going in some other direction rather than the expected channels/protocols/signal direction, therefore missing and bypassing one another. The strangest thing is, when calling from this ATA via the phone to the PSTN line of Asterisk, I can input *86 to access VoicemailMain, enter the extension/password and it works just fine. I just do not like tying up the PSTN line for ocassions like this.

But again thanks for the info buddy.

tlofton1000

6

check your RTP ports and firewalls. I once had an odd problem where phones could call the server but not each other- turns out only some of the RTP ports were firewall allowed.

Try Answer() before VoiceMailMain()?

make sure the ATA has STUN turned off if on same LAN…

7

Hello IronHelix,

I already had ports open on 5061-5063, 10000-20000, 53, 69, and no firewall. I also already had extensions.conf to perform an Answer() before starting the VoicemailMain priority.

Thanks for your response and your time. I do not know what to do about this issue, other than consider and accept the unit/device as defective.

tlofton1000

8

do sip debug peer (itspeername) and call voicemailmain from it

post the result.

also try rtp debug while it’s connected, that will generate a handful of output, only need about 20 lines of it…

9

Hi IronHelix,

I started the debug process for ext 31002 and then called VoicemailMain by dialing 861 from the phone itself connected to the ATA unit in question. I did not put in password, because I cannot hear the request to input password. This was result of the entire dialogue until I hung up: (domain name and static ip address altered for posting)

*CLI> sip debug peer 31002

<-- SIP read from 24.xx.44.xx:5060:
INVITE sip:861@voip.com SIP/2.0
Record-Route: sip:24.xx.44.xx;nat=yes;ftag=70f9d3c26530ef74o0;lr=on
Via: SIP/2.0/UDP 24.xx.44.xx;branch=z9hG4bK3f8c.51eecbf6.0
Via: SIP/2.0/UDP 192.168.1.100:5061;received=68.234.55.244;branch=z9hG4bK-a1019f24
From: 31002 sip:31002@voip.com;tag=70f9d3c26530ef74o0
To: sip:861@voip.com
Call-ID: bbeecc5a-dfe28fbc@192.168.1.100
CSeq: 101 INVITE
Max-Forwards: 69
Contact: 31002 sip:31002@68.234.55.244:5061
Expires: 240
User-Agent: Linksys/PAP2T-3.1.10(LSc)
Content-Length: 444
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 675279 675279 IN IP4 192.168.1.100
s=-
c=IN IP4 192.168.1.100
t=0 0
m=audio 16408 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
— (16 headers 20 lines)—
Using INVITE request as basis request - bbeecc5a-dfe28fbc@192.168.1.100
Sending to 24.xx.44.xx : 5060 (non-NAT)
Found user '31002’
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.100:16408
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x105 (g723|ulaw|g729), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/v
ideo=0x0 (nothing), combined - 0x105 (g723|ulaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x
1 (telephone-event)
Looking for 861 in default (domain voip.com)
list_route: hop: sip:24.xx.44.xx;nat=yes;ftag=70f9d3c26530ef74o0;lr=on
Transmitting (no NAT) to 24.xx.44.xx:5060:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 24.xx.44.xx;branch=z9hG4bK3f8c.51eecbf6.0;received=24.xx.44.xx
Via: SIP/2.0/UDP 192.168.1.100:5061;received=68.234.55.244;branch=z9hG4bK-a1019f24
From: 31002 sip:31002@voip.com;tag=70f9d3c26530ef74o0
To: sip:861@voip.com
Call-ID: bbeecc5a-dfe28fbc@192.168.1.100
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:861@24.xx.44.xx:5065
Content-Length: 0

-- Executing Answer("SIP/31002-0a0991c0", "") in new stack

We’re at 24.xx.44.xx port 15068
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 24.xx.44.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 24.xx.44.xx;branch=z9hG4bK3f8c.51eecbf6.0;received=24.xx.44.xx
Via: SIP/2.0/UDP 192.168.1.100:5061;received=68.234.55.244;branch=z9hG4bK-a1019f24
Record-Route: sip:24.xx.44.xx;nat=yes;ftag=70f9d3c26530ef74o0;lr=on
From: 31002 sip:31002@voip.com;tag=70f9d3c26530ef74o0
To: sip:861@voip.com;tag=as27697ea2
Call-ID: bbeecc5a-dfe28fbc@192.168.1.100
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:861@24.xx.44.xx:5065
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 12566 12566 IN IP4 24.xx.44.xx
s=session
c=IN IP4 24.xx.44.xx
t=0 0
m=audio 15068 RTP/AVP 0 4 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

-- Executing VoiceMailMain("SIP/31002-0a0991c0", "31002") in new stack
-- Playing 'vm-password' (language 'en')

<-- SIP read from 24.xx.44.xx:5060:
ACK sip:861@24.xx.44.xx:5065 SIP/2.0
Record-Route: sip:24.xx.44.xx;ftag=70f9d3c26530ef74o0;lr=on
Via: SIP/2.0/UDP 24.xx.44.xx;branch=0
Via: SIP/2.0/UDP 192.168.1.100:5061;received=68.234.55.244;branch=z9hG4bK-9603cb83
From: 31002 sip:31002@voip.com;tag=70f9d3c26530ef74o0
To: sip:861@voip.com;tag=as27697ea2
Call-ID: bbeecc5a-dfe28fbc@192.168.1.100
CSeq: 101 ACK
Max-Forwards: 69
Contact: 31002 sip:31002@192.168.1.100:5061
User-Agent: Linksys/PAP2T-3.1.10(LSc)
Content-Length: 0

— (12 headers 0 lines)—
– Incorrect password ‘’ for user ‘31002’ (context = default)
– Playing ‘vm-incorrect’ (language ‘en’)
– Playing ‘vm-password’ (language ‘en’)
– Incorrect password ‘’ for user ‘31002’ (context = default)
– Playing ‘vm-incorrect’ (language ‘en’)

<-- SIP read from 24.xx.44.xx:5060:
BYE sip:861@24.xx.44.xx:5065 SIP/2.0
Record-Route: sip:24.xx.44.xx;ftag=70f9d3c26530ef74o0;lr=on
Via: SIP/2.0/UDP 24.xx.44.xx;branch=z9hG4bK0f8c.95befe72.0
Via: SIP/2.0/UDP 192.168.1.100:5061;received=68.234.55.244;branch=z9hG4bK-afb4fc07
From: 31002 sip:31002@voip.com;tag=70f9d3c26530ef74o0
To: sip:861@voip.com;tag=as27697ea2
Call-ID: bbeecc5a-dfe28fbc@192.168.1.100
CSeq: 102 BYE
Max-Forwards: 69
Route: sip:24.xx.44.xx;nat=yes;ftag=70f9d3c26530ef74o0;lr=on
User-Agent: Linksys/PAP2T-3.1.10(LSc)
Content-Length: 0

— (12 headers 0 lines)—
Sending to 24.xx.44.xx : 5060 (non-NAT)
Transmitting (no NAT) to 24.xx.44.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 24.xx.44.xx;branch=z9hG4bK0f8c.95befe72.0;received=24.xx.44.xx
Via: SIP/2.0/UDP 192.168.1.100:5061;received=68.234.55.244;branch=z9hG4bK-afb4fc07
Record-Route: sip:24.xx.44.xx;ftag=70f9d3c26530ef74o0;lr=on
From: 31002 sip:31002@voipm.com;tag=70f9d3c26530ef74o0
To: sip:861@voip.com;tag=as27697ea2
Call-ID: bbeecc5a-dfe28fbc@192.168.1.100
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:861@24.xx.44.xx:5065
Content-Length: 0

Earlier while the phone was connected with Asterisk, I ran RTP debug and this was the result:

RTP Debugging Enabled
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1357, ts 61107886, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1358, ts 61108126, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1359, ts 61108366, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1360, ts 61108606, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1361, ts 61108846, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1362, ts 61109086, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1363, ts 61109326, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1364, ts 61109566, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1365, ts 61109806, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1366, ts 61110046, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1367, ts 61110286, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1368, ts 61110526, len 240)
Got RTP packet from 68.234.55.244:16402 (type 0, seq 1369, ts 61110766, len 240)

I am not at all familiar with this rtp data I have never used this feature. What do you make of this data?

I appreciate your help.

tlofton1000

10

Hi All,

It looks like it is selecting g723 first rather than ulaw which is what i have listed first in sip.conf. Could it be a codec issue afterall?

tlofton1000

11

I removed the choices in sip.conf and only now only allow ulaw and it still responds the same.

Hum…

12

Hi all,

I apologize for the troubles to everyone and also thank all for the time given to this post. I added a setting in sip.conf to nat=yes and everything now works.

tlofton1000