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Unread 09-17-2007, 11:25 AM   #1
Lanh
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Default Problem with incoming DID

Hi,

Yes I am back with this problem again. It would seem that the provider of the DID is unable to contact my Voxalot account and this is why the problem is occuring, according to them anyway.

To me personally it's strange that other providers can connect/forward to Voxalot without any issues whatsoever but they can't, because that to me would suggest a problem with their service and not Voxalot, but they have given me a log to pass along regardless and are laying the blame at Voxalot's door from the tones of their emails.

Here is the log...

Code:
OK, this is the log. Not sure if it would help them. Looks like they
return trying and then temporary unavailble.

We're at 81.187.81.110 port 19944
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 64.34.173.199:5060:
INVITE sip:xxxxxx@voxalot.com SIP/2.0
Via: SIP/2.0/UDP 81.187.81.110:5060;branch=z9hG4bK2c243b8d;rport
From: "RevK-E61i" <sip:03333400007@81.187.81.110>;tag=as7f4e120d
To: <sip:xxxxxx@voxalot.com>
Contact: <sip:03333400007@81.187.81.110>
Call-ID: 73b408a41452df160a502dc000b13db3@81.187.81.110
CSeq: 102 INVITE
User-Agent: A&A
Max-Forwards: 70
Date: Mon, 17 Sep 2007 09:19:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 19962 19962 IN IP4 81.187.81.110
s=session
c=IN IP4 81.187.81.110
t=0 0
m=audio 19944 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

- ---
   -- Called xxxxxx@voxalot.com
a*CLI>
<-- SIP read from 64.34.173.199:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.187.81.110:5060;branch=z9hG4bK2c243b8d;rport=5060
From: "RevK-E61i" <sip:03333400007@81.187.81.110>;tag=as7f4e120d
To: <sip:xxxxxx@voxalot.com>
Call-ID: 73b408a41452df160a502dc000b13db3@81.187.81.110
CSeq: 102 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 64.34.173.199:5060 "Noisy feedback tells:  pid=29067
req_src_ip=81.187.81.110 req_src_port=5060 in_uri=sip:xxxxxx@voxalot.com
out_uri=sip:xxxxxx@voxalot.com via_cnt==1"


- --- (9 headers 0 lines) ---
a*CLI>
<-- SIP read from 64.34.173.199:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 81.187.81.110:5060;branch=z9hG4bK2c243b8d;rport=5060
From: "RevK-E61i" <sip:03333400007@81.187.81.110>;tag=as7f4e120d
To: <sip:xxxxxx@voxalot.com>
Call-ID: 73b408a41452df160a502dc000b13db3@81.187.81.110
CSeq: 102 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 64.34.173.199:5060 "Noisy feedback tells:  pid=29067
req_src_ip=81.187.81.110 req_src_port=5060 in_uri=sip:xxxxxx@voxalot.com
out_uri=sip:*010319145@64.34.163.35 via_cnt==1"


- --- (9 headers 0 lines) ---
a*CLI>
<-- SIP read from 64.34.173.199:5060:
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 81.187.81.110:5060;branch=z9hG4bK2c243b8d;rport=5060
From: "RevK-E61i" <sip:03333400007@81.187.81.110>;tag=as7f4e120d
To: <sip:xxxxxx@voxalot.com>;tag=54333a515cb8c1bd95fa9f9db754f4d6-9ec8
Call-ID: 73b408a41452df160a502dc000b13db3@81.187.81.110
CSeq: 102 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 64.34.163.35:5060 "Noisy feedback tells:  pid=26847
req_src_ip=64.34.173.199 req_src_port=5060
in_uri=sip:*010319145@64.34.163.35 out_uri=sip:xxxxxx@voxalot.com
via_cnt==3"


- --- (9 headers 0 lines) ---
   -- Got SIP response 480 "Temporarily Unavailable" back from
64.34.173.199
Transmitting (no NAT) to 64.34.173.199:5060:
ACK sip:xxxxxx@voxalot.com SIP/2.0
Via: SIP/2.0/UDP 81.187.81.110:5060;branch=z9hG4bK2c243b8d;rport
From: "RevK-E61i" <sip:03333400007@81.187.81.110>;tag=as7f4e120d
To: <sip:xxxxxx@voxalot.com>;tag=54333a515cb8c1bd95fa9f9db754f4d6-9ec8
Contact: <sip:03333400007@81.187.81.110>
Call-ID: 73b408a41452df160a502dc000b13db3@81.187.81.110
CSeq: 102 ACK
User-Agent: A&A
Max-Forwards: 70
Content-Length: 0
Advice would be appreciated.

PS Is there no ticket service for Premium members?
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Unread 09-17-2007, 03:59 PM   #2
emoci
 
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Well try to point to any of these and see if you have results:

1. 123456@voxalot.com

2. 123456@xx.voxalot.com (xx is us, au, or eu depending on which cluster you are using, see main page when logged into VoXalot Account on the web)

3. *010123456@sipbroker.com


Naturally replace 123456 with your VoXalot Number. If all that fails, it's possible this provider is having issues with forwarding....

To check that their problem is not VoXalot Specific, get yourself an FWD account and forward there. If that works, you can then have FWD forward to VoXalot (I can confirm FWD to VoXalot does work)

For the time being VoXalot does not have a Help Ticket system (although word is that one is in the plans)
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Unread 09-17-2007, 04:11 PM   #3
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Default

Yep, done steps 1-3, with the Sipbroker step I get a message "We're sorry, the SipBroker code you have dialed is not correct", which I am presuming means that they don't peer with SipBroker or allow access to it.

With regard to the FWD service, I have a FWD account which is forwarded to my Voxalot account and works flawlessly other than with this number/forwarding provider.

It's not a codec issue because I can receive calls when I have it registered as a provider in my account and it works flawlessly, I just don't want to have it as a registered provider.
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Unread 09-17-2007, 09:53 PM   #4
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Quote:
Originally Posted by Carl. View Post
It's not a codec issue because I can receive calls when I have it registered as a provider in my account and it works flawlessly, I just don't want to have it as a registered provider.
I actually think it is a Codec issue. When registered as a provider Voxalot is likely to be transcoding the audio.

By the looks of it, your provider only handles g711a (alaw)

Quote:
Adding codec 0x8 (alaw) to SDP
As Voxalot is only forwarding in this scenario (ie provider not registered) there is no ability to transcode.

By the looks of things, your device is unable to handle alaw, failing and i'm guessing you have voicemail turned off which results in a 480 being returned.

Hope this helps.
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Unread 09-17-2007, 11:13 PM   #5
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Default

Codec wise on my ATA I have:-

G.711_PCMU
G.711_PCMA
G.729
G.726
G.723

I am presuming the G.711 codecs (or one of them at least), are the ones which you're referring to?
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Unread 09-17-2007, 11:48 PM   #6
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Default

I have dropped the number and the provider as it is all more hassle than it's worth and it's just taking their time and my time and getting nowhere fast unfortunately.

Shame because it was such a memorable number, really easy to remember, but seeing as I have upgraded my Voxalot membership I can now register another provider where I have three incoming DID's which makes things a lot easier.
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