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309 posts

Ultimate Geek
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Topic # 21343 23-Apr-2008 18:10
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Hi,

I've just registered a second fusion account and trying to get the second phone number working properly, can someone confirm that if I'm using asterisk I don't have to have the sip entry as [vfx] ?

I.e. I have my initial one:
[vfx]
type=peer
fromdomain=as.wxcnz.net
fromuser=9440XXXX
host=as.wxcnz.net
insecure=invite,port
canreinvite=no
nat=no
secret=XYZ
username=XYZ
disallow=all
allow=g729
allow=ulaw
allow=alaw
qualify=yes
context=home

Then a second line

[mumhome]
type=peer
fromdomain=as.wxcnz.net
fromuser=9950XXXX
host=as.wxcnz.net
insecure=invite,port
canreinvite=no
nat=no
secret=XYZ
username=XYZ
disallow=all
allow=g729
allow=ulaw
allow=alaw
qualify=yes
context=local

I don't appear to be receiving calls to the second number, it was supposidly activated yesterday.

cheers
B





Barry Murphy
ISPMap - New Zealand ISP map
Vibe Communications LTD - Business ISP and Wholesale Carrier



Any comments made by myself don't reflect the views of my employer, they are mine and mine alone

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205 posts

Master Geek
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WorldxChange

  Reply # 126028 24-Apr-2008 08:12
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Hi Barry,

It looks like your asterisk box is sending back an address incomplete message:

----------------------------------------
Timestamp     : 21:48:15.773 2008-04-23
Direction     : RX
Remote IP/Port: x.x.x.x/5060
Local IP/Port : 58.28.20.150/5060
Transport     : UDP
----------------------------------------
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-7e2a6-480f05df-8ed97039-3c959778;received=58.28.20.150
From: <sip:027xxxxxx@as.wxcnz.net>;tag=96141c3a-13c4-480f05df-8ed97039-3b3f90db
To: "Bxxx"<sip:995xxxxx@xport.co.nz>;tag=as4de06f1e
Call-ID: BW214815798230408-660382737@10.251.1.11
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

This is being sent back from your asterisk box, I would guess that either the extension or number is not setup correctly on the box.  Try running some debugs on your asterisk box to see if you can find where the call is stopping.  On the command line you should be able to run #asterisk -vvvvvvr

place a call and it should output all of the debug messages, hopefully something in the debug can point you in the right direction.

Cheers

Josh






309 posts

Ultimate Geek
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  Reply # 126048 24-Apr-2008 09:36
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No different to my other setup which is weird:

The problem is my PBX wont accept the call, I see it being routed to me... I have the following in

[local]
exten => 9950XXXX,1,Dial(SIP/1020,120)
exten => 9950XXXX,2,Hangup


Then Debug:

<-- SIP read from 58.28.20.150:5060:
INVITE sip:9950XXXX@60.234.68.5;transport=UDP SIP/2.0
From: <sip:09337XXXX@as.wxcnz.net>;tag=96141c3a-13c4-480fab2f-915ef8d7-7cae38bf
To: "Barry Murphy"<sip:9950XXXX@xport.co.nz>
Call-ID: BW093335790240408-1724830681@10.251.1.11
CSeq: 1 INVITE
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-5873-480fab2f-915ef8d7-e328b8c
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Accept: multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards: 9
Contact: <sip:09337XXXX@as.wxcnz.net:5060;maddr=58.28.20.150;transport=UDP>
Content-Type: application/sdp
Content-Length: 335

v=0
o=BroadWorks 5280590 1 IN IP4 58.28.20.150
s=-
c=IN IP4 58.28.20.150
t=0 0
m=audio 35616 RTP/AVP 18 8 0 100 101
a=rtpmap:101 telephone-event/8000
a=rtpmap:100 X-NSE/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:101 0-16
a=fmtp:100 192-194
a=fmtp:18 annexb=yes
a=bsoft:1 image udptl t38

--- (12 headers 15 lines) ---
Using INVITE request as basis request - BW093335790240408-1724830681@10.251.1.11
Sending to 58.28.20.150 : 5060 (non-NAT)
Found peer 'vfx'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 58.28.20.150:35616
Found description format telephone-event
Found description format X-NSE
Found description format PCMU
Found description format PCMA
Found description format G729
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 9950XXXX in home (domain 60.234.68.5;transport=UDP)
Reliably Transmitting (no NAT) to 58.28.20.150:5060:
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-5873-480fab2f-915ef8d7-e328b8c;received=58.28.20.150
From: <sip:09337XXXX@as.wxcnz.net>;tag=96141c3a-13c4-480fab2f-915ef8d7-7cae38bf
To: "Barry Murphy"<sip:9950XXXX@xport.co.nz>;tag=as2dab3007
Call-ID: BW093335790240408-1724830681@10.251.1.11
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
voip*CLI> sip
<-- SIP read from 58.28.20.150:5060:
ACK sip:9950XXXX@60.234.68.5;transport=UDP SIP/2.0
From: <sip:09337XXXX@as.wxcnz.net>;tag=96141c3a-13c4-480fab2f-915ef8d7-7cae38bf
To: "Barry Murphy"<sip:9950XXXX@xport.co.nz>;tag=as2dab3007
Call-ID: BW093335790240408-1724830681@10.251.1.11
CSeq: 1 ACK
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-5873-480fab2f-915ef8d7-e328b8c
Max-Forwards: 69
Contact: <sip:09337XXXX@as.wxcnz.net:5060;maddr=58.28.20.150;transport=UDP>
Content-Length: 0


--- (9 headers 0 lines) ---
Destroying call 'BW093335790240408-1724830681@10.251.1.11'





Barry Murphy
ISPMap - New Zealand ISP map
Vibe Communications LTD - Business ISP and Wholesale Carrier



Any comments made by myself don't reflect the views of my employer, they are mine and mine alone

 
 
 
 




309 posts

Ultimate Geek
+1 received by user: 5

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  Reply # 126049 24-Apr-2008 09:40
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Found the problem to which I first thought, it was picking up peer as.xxxx as peer VFX so looking at context HOME rather than the next option LOCAL. I've now setup a new context called [incomming] and placed all numbers in here rather.

thanks
Barry




Barry Murphy
ISPMap - New Zealand ISP map
Vibe Communications LTD - Business ISP and Wholesale Carrier



Any comments made by myself don't reflect the views of my employer, they are mine and mine alone

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