Deleting VoiceMail Prompts...

[g00fy]

Hi Martin,

Tried with X-lite:
When placing to calls at the same time (because x-lite support two calls at once) and calling my DID-number with a cellphone, with:
VM off: after 20-30 sec a message appear on the cell that the call can not get connected.
VM on: unavailable message.

Only difference with the zyxel is that when the VM is off, the call is terminated at once. With X-lite it is after 20-30 secs.

But, it is not as it's supposed to be. What to do next?

Edit: Tried a few times more and with X-lite most of the time call is terminated at once also.
When calling from another VoIP connection I saw I was getting back an error 480 when I was calling my Voxalot account while the phone was off the hook. Shouldn't that be error 486?
It could be though that I pushed the DND in my X-lite client, which I thought would be the same is indicating I was busy. Either way the symptoms were the same.

[martin]

Originally Posted by g00fy

When calling from another VoIP connection I saw I was getting back an error 480 when I was calling my Voxalot account while the phone was off the hook. Shouldn't that be error 486?

Yes it should be a 486. If the device is sending back a 480 then this explains the "unavailable" message. The only way Voxalot will handle a busy is if it receives a 486 "BUSY".

I'm not familiar with the zyxel.

Perhaps another member that uses a zyxel can comment.

Anyone?

[g00fy]

Hi Martin,

In fact this was when I was using the X-lite softphone. I will try tonight what I'll get when only using my zyxel (not zytel) modem. BTW: it's a zyxel prestige 2602-63c.

[g00fy]

Tried the following. Called my voxalot account with anothet voip number I got from my local provider. At the same time initiated a call to my voxalot account with my cellphone. The cellphone terminated the call directly and after I ended the call from the voxalot account I see a code 487 in my logs.

The other way around (calling my local providers voip number) I see a code 486 as it is supposed to be. All is initiated from the same modem and sip accounts configured the same.

Voxalot:
Jan 27 17:40:24 mikmak adsl-modem: src="0.0.0.0" dst="0.0.0.0" msg="VoIP Call Start Ph[1] -> 0786xxxxxx" note="Traffic Log" devID="1CF396" cat="FSM"
Jan 27 17:40:25 mikmak adsl-modem: src="0.0.0.0" dst="0.0.0.0" msg="VoIP Call Start from SIP[1]" note="Traffic Log" devID="1CF396" cat="FSM"
Jan 27 17:40:25 mikmak adsl-modem: src="0.0.0.0" dst="0.0.0.0" msg="VoIP Call Established Ph[1] -> 0786xxxxxx" note="Traffic Log" devID="1CF396" cat="FSM"
Jan 27 17:40:48 mikmak adsl-modem: src="0.0.0.0" dst="0.0.0.0" msg="VoIP Call End Phone[1]" note="Traffic Log" devID="1CF396" cat="FSM"
Jan 27 17:40:48 mikmak adsl-modem: msg="Get SIP error response code 487" note="SIP[2]" devID="1CF396" cat="VOIP_SIP"
Jan 27 17:40:49 mikmak adsl-modem: src="0.0.0.0" dst="0.0.0.0" msg="VoIP Call End Phone[2]" note="Traffic Log" devID="1CF396" cat="FSM"


Local provider:
Jan 27 17:36:18 mikmak adsl-modem: src="0.0.0.0" dst="0.0.0.0" msg="VoIP Call Start from SIP[2]" note="Traffic Log" devID="1CF396" cat="FSM"
Jan 27 17:36:28 mikmak adsl-modem: src="0.0.0.0" dst="0.0.0.0" msg="VoIP Call Start Ph[1] -> 0787xxxxxx" note="Traffic Log" devID="1CF396" cat="FSM"
Jan 27 17:36:29 mikmak adsl-modem: src="0.0.0.0" dst="0.0.0.0" msg="VoIP Call Established Ph[1] -> 0787xxxxxx" note="Traffic Log" devID="1CF396" cat="FSM"
Jan 27 17:36:32 mikmak adsl-modem: msg="Get SIP error response code 486" note="SIP[1]" devID="1CF396" cat="VOIP_SIP"
Jan 27 17:36:32 mikmak adsl-modem: src="0.0.0.0" dst="0.0.0.0" msg="VoIP Call End Phone[1]" note="Traffic Log" devID="1CF396" cat="FSM"
Jan 27 17:36:36 mikmak adsl-modem: src="0.0.0.0" dst="0.0.0.0" msg="VoIP Call End Phone[2]" note="Traffic Log" devID="1CF396" cat="FSM"

Hope this helps to solve the problem....

[martin]

Originally Posted by g00fy

Hope this helps to solve the problem....

It would be better of you could provide a message dump of the status codes that your modem is initiating (not receiving).

Is this possible?

[g00fy]

Originally Posted by martin

It would be better of you could provide a message dump of the status codes that your modem is initiating (not receiving).

Is this possible?

I wouldn't know how to do that. May be someone else can give me a hint on how to establish this?

But you agree that it is strange that there is no code 486 coming when connected to my Voxalot account? Can you see in the Voxalot logs what is happening at the moment of the tests?

[g00fy]

I sniffed the network while calling out with an x-lite softphone (both lines busy) and then calling the busy line with my cellphone. As always the call terminated immediately.

Here is the log from what goes over port 5060 at that time:

INVITE sip:<VOXALOT_NO>@212.xxx.xxx.xxx:15353;rinstance=f 628ff98fe21d44f;transport=TCP SIP/2.0
Record-Route: <sip:85.17.19.194;transport=tcp;r2=on;lr=on;ftag=a s61669e84>
Record-Route: <sip:85.17.19.194;r2=on;lr=on;ftag=as61669e84>
Via: SIP/2.0/TCP 85.17.19.194;branch=z9hG4bKe3ec.b7eda456.1
Via: SIP/2.0/UDP 85.17.19.194:5061;branch=z9hG4bK5ff61925;rport=506 1
From: "065xxxxxxx" <sip:065xxxxxxx@85.17.19.194:5061>;tag=as61669e8 4
To: <sip:*010<VOXALOT-NO>@proxy01.eu1.voxalot.com>
Contact: <sip:065xxxxxxx@85.17.19.194:5061>
Call-ID: 4a709c6a3443a6c4754e273f6c3454de@85.17.19.194
CSeq: 102 INVITE
User-Agent: voxaLot
Max-Forwards: 69
Date: Wed, 28 Jan 2009 22:55:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 391
P-hint: usrloc applied

v=0
o=root 15029 15029 IN IP4 85.17.19.194
s=session
c=IN IP4 85.17.19.194
t=0 0
m=audio 17548 RTP/AVP 0 8 18 111 97 3 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

SIP/2.0 486 Busy Here
Via: SIP/2.0/TCP 85.17.19.194;branch=z9hG4bKe3ec.b7eda456.1
Via: SIP/2.0/UDP 85.17.19.194:5061;branch=z9hG4bK5ff61925;rport=506 1
To: <sip:*010<VOXALOT-NO>@proxy01.eu1.voxalot.com>;tag=e7795e64
From: "065xxxxxxx"<sip:065xxxxxxx@85.17.19.194:5061>;tag =as61669e84
Call-ID: 4a709c6a3443a6c4754e273f6c3454de@85.17.19.194
CSeq: 102 INVITE
User-Agent: X-Lite release 1100l stamp 47546
Warning: 399 devnull "User reject"
Content-Length: 0

ACK sip:<VOXALOT-NO>@212.xxx.xxx.xxx:15353;rinstance=f628ff98fe21d4 4f;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 85.17.19.194;branch=z9hG4bKe3ec.b7eda456.1
From: "065xxxxxxx" <sip:065xxxxxxx@85.17.19.194:5061>;tag=as61669e8 4
Call-ID: 4a709c6a3443a6c4754e273f6c3454de@85.17.19.194
To: <sip:*010<VOXALOT-NO@proxy01.eu1.voxalot.com>;tag=e7795e64
CSeq: 102 ACK
User-Agent: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0

[martin]

Thanks for the SIP logs. To be honest I am unsure why the symptoms you describe are occurring.

I will chat with the other developers and see what they think.

I'll try and set aside some time over the next few days to debug on our test server and attempt to reproduce.

I'll keep you posted.

[g00fy]

Thanks Martin for looking in to this.

I did one final test today, unregistered my DID-number from budgetphone in Voxalot and configured it in my ATA. I waited to let the Voxalot registration expire at budgetphones server and setup a call with the budgetphone account from my ATA.
Then a call with my cellphone to that number gives a busy tone as supposed to. The people at budgetphone monitored for me and they could see my ATA is sending a 486 response at that moment.
When I do the same with budgetphone registered only in my Voxalot account, they see a 480 response. So, it seems the Voxalot system is converting the 486 my ATA is sending to a 480 to budgetphone?

Hopefully you'll manage to solve this issue.

[g00fy]

Hi Martin, just curious, did you allready manage to take a look into this problem?