Vonage inbound Help Please


#1

I’ve been banging my head against this for a couple days, and seem to be getting conflicting info from all the stuff ive read. I have a vonage softphone account. I am now running asterisk@home 1.5 (latest).

Ive gotten inbound and outbound calls through Vonage working, but not at the same time. Ive seen several vonage configs that are supposed to work, and ive also read that Vonage wont allow both to work at the same time…

The best i got working was outbound working fine, and when i called into the system i got some indication that asterisk was seeing the call, but the call ended up at vonages voice mail.

I’ve tried sending the call to a digital receptionist, and to a ring group, and an extension.

Below is a log (slightly sanitized) of what happens when i call in from a pots line. I see a couple 404 errors, im assuming that where i am messing up. Since i see my home phone CID in there i am assuming Asterisk is trying to answer the call.

Any ideas how to fix this?

Many thanks.


Reliably Transmitting:
REGISTER sip:sphone.vopr.vonage.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.199:5060;branch=z9hG4bK3d4e5f8f
From: sip:1602xxxxx@sphone.vopr.vonage.net;tag=as6271c279
To: sip:1602xxxxxxx@sphone.vopr.vonage.net
Call-ID: 0ab253ca7a9a894c1e0a52d960c946f3@127.0.0.1
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: sip:s@192.168.0.199
Event: registration
Content-Length: 0

(no NAT) to 216.115.25.198:5060
asterisk1*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.199:5060;branch=z9hG4bK3d4e5f8f
From: sip:1602xxxx@sphone.vopr.vonage.net;tag=as6271c279
To: sip:1602xxxxxx@sphone.vopr.vonage.net
Call-ID: 0ab253ca7a9a894c1e0a52d960c946f3@127.0.0.1
CSeq: 106 REGISTER
Contact: sip:s@192.168.0.199;expires=20
Content-Length: 0

8 headers, 0 lines
Destroying call '0ab253ca7a9a894c1e0a52d960c946f3@127.0.0.1’
asterisk1*CLI>

Sip read:
INVITE sip:1602xxxxxx@68.226.127.204:5060;suppress-features=- SIP/2.0
Via: SIP/2.0/UDP 216.115.25.198:5060
Via: SIP/2.0/UDP 216.115.25.230:5060
Via: SIP/2.0/UDP 63.77.214.15:5060;branch=z9hG4bK1866
Record-Route: sip:1602xxxxxx@216.115.25.198:5060
Record-Route: sip:1602xxxxx@216.115.25.230:5060
From: “ME,HOME” sip:1602xxxxxx@63.77.214.15;tag=572818199
To: sip:1602xxxxxx@inbound3.vonage.net
Call-ID: 44F918A5-158D11DA-8A1BE87E-E013060@63.77.214.15
CSeq: 101 INVITE
Contact: sip:1602xxxxxx@63.77.214.15:5060;rtpupdated=-
Max-Forwards: 13
Content-Type: application/sdp
Content-Length: 358

v=0
o=CiscoSystemsSIP-GW-UserAgent 8208 7991 IN IP4 63.77.214.15
s=SIP Call
c=IN IP4 63.77.214.15
t=0 0
m=audio 20312 RTP/AVP 0 18 2 100 101
c=IN IP4 63.77.214.15
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

14 headers, 15 lines
Using latest request as basis request
Sending to 216.115.25.198 : 5060 (non-NAT)
Found peer '1602xxxxxxxx’
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 63.77.214.15:20312
Found description format PCMU
Found description format G729
Found description format G726-32
Found description format X-NSE
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x114 (ulaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Looking for 1602xxxxx in from-pstn
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 216.115.25.198:5060
Via: SIP/2.0/UDP 216.115.25.230:5060
Via: SIP/2.0/UDP 63.77.214.15:5060;branch=z9hG4bK1866
From: “ME,HOME” sip:1602xxxxxx@63.77.214.15;tag=572818199
To: sip:1602xxxxxx@inbound3.vonage.net;tag=as11c6ab2e
Call-ID: 44F918A5-158D11DA-8A1BE87E-E013060@63.77.214.15
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:1602xxxxxxx@192.168.0.199
Content-Length: 0

to 216.115.25.198:5060
asterisk1*CLI>

Sip read:
ACK sip:1602xxxxxx@inbound3.vonage.net SIP/2.0
Via: SIP/2.0/UDP 216.115.25.198:5060
From: “ME,HOME” sip:1602xxxxxxx@63.77.214.15;tag=572818199
To: sip:1602xxxxxx@inbound3.vonage.net;tag=as11c6ab2e
Call-ID: 44F918A5-158D11DA-8A1BE87E-E013060@63.77.214.15
CSeq: 101 ACK
Content-Length: 0

7 headers, 0 lines
Destroying call '44F918A5-158D11DA-8A1BE87E-E013060@63.77.214.15’
asterisk1*CLI>

Sip read:
OPTIONS sip:192.168.0.199:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.217;rport;branch=z9hG4bKc0a800d900000010430f509d0000257c000000df
Content-Length: 0
Call-ID: 2152686C-A96B-4150-A19F-65C26BB6E215@192.168.0.217
CSeq: 55 OPTIONS
From: sip:202@192.168.0.199;tag=515261403707
Max-Forwards: 70
To: sip:192.168.0.199:5060

8 headers, 0 lines
Looking for 192.168.0.199:5060 in from-pstn
Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.217;branch=z9hG4bKc0a800d900000010430f509d0000257c000000df
From: sip:202@192.168.0.199;tag=515261403707
To: sip:192.168.0.199:5060;tag=as6f82b4c2
Call-ID: 2152686C-A96B-4150-A19F-65C26BB6E215@192.168.0.217
CSeq: 55 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:192.168.0.199
Accept: application/sdp
Content-Length: 0

to 192.168.0.217:5060
Destroying call '2152686C-A96B-4150-A19F-65C26BB6E215@192.168.0.217’
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sphone.vopr.vonage.net SIP/2.0
Via: SIP/2.0/UDP 192.168.0.199:5060;branch=z9hG4bK2e90f25b
From: sip:1602xxxxxx@sphone.vopr.vonage.net;tag=as0706c7c7
To: sip:1602xxxxxx@sphone.vopr.vonage.net
Call-ID: 0ab253ca7a9a894c1e0a52d960c946f3@127.0.0.1
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: sip:s@192.168.0.199
Event: registration
Content-Length: 0

(no NAT) to 216.115.25.198:5060
asterisk1*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.199:5060;branch=z9hG4bK2e90f25b
From: sip:1602xxxxxx@sphone.vopr.vonage.net;tag=as0706c7c7
To: sip:1602xxxxxxxx@sphone.vopr.vonage.net
Call-ID: 0ab253ca7a9a894c1e0a52d960c946f3@127.0.0.1
CSeq: 107 REGISTER
Contact: sip:s@192.168.0.199;expires=20
Content-Length: 0

8 headers, 0 lines
Destroying call ‘0ab253ca7a9a894c1e0a52d960c946f3@127.0.0.1’


#2

My Guess would be you have one vonage account? That means only one call at a time either inbound or out bound. I think what you are seeing is maybe callwaitng coming in to the asterisk box from vonage


#3

Thanks for the reply,

yes i have one account. I commented out my outgoing sip config and left the incoming one. I dont have any softphone clients logged into vonage, or asterisk at the moment.

No other live extensions on asterisk either. Just trying to get a call to come in and be picked up by the system.

any other ideas?

thanks


#4

Guess I missed understood. I as thinking you had it all working but were trying to make an outbound call and recieve a call at the same time.
I do not use vonage so I am afraid I cant be much help


#5

DId you ever get this figured out, I am having similar problems.

TIA


#6

Is there any answer to this? I am having this problem too.