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3 posts

Wannabe Geek


Topic # 87974 11-Aug-2011 22:31 Send private message

Hi All,

I have been using VFX with Asterisk 1.6 for several months now and recently decided to add a second phone line (SIP) and ported my home phone number from another provider.

I now have 2 SIP Trunks happily registered and can make calls on either line.

Host                           dnsmgr Username       Refresh State                Reg.Time
pan.wxnz.net:5060              N      xxxxxx17           285 Registered           Sun, 07 Aug 2011 16:14:22
pan.wxnz.net:5060              N      xxxxxx77           285 Registered           Sun, 07 Aug 2011 16:14:22
2 SIP registrations.

I have a very basic setup with just 3 extensions and now 2 SIP Trunks

Name/username              Host            Dyn Nat ACL Port     Status
6000/6000                  192.168.x.x      D   N      5060     Unmonitored
6001/6001                  192.168.x.x      D   N      5061     Unmonitored
6002/6002                  (Unspecified)    D   N      5060     Unmonitored
VFX_LINE1/xxxxxx77         58.28.20.150         N      5060     OK (42 ms)
VFX_LINE2/xxxxxx17         58.28.20.150         N      5060     OK (67 ms)
5 sip peers [Monitored: 2 online, 0 offline Unmonitored: 3 online, 0 offline]


When I make a call to Line 1 I can happily answer the call and it is showing me that it came in on Line 1

Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry
58.28.20.150 xxxxxx77         xxxxxxxxxxxxxxx  0x100 (g729)     No       Rx: ACK
192.168.x.x      6001             xxxxxxxxxxxxxxx  0x100 (g729)     No       Tx: ACK
2 active SIP dialogs
 
However, when I make a call to Line 2, this is also coming in on Line 1.

Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry
58.28.20.150 xxxxxx77         xxxxxxxxxxxxxxx  0x100 (g729)     No       Rx: ACK
192.168.x.x      6000             xxxxxxxxxxxxxxx  0x100 (g729)     No       Tx: ACK
2 active SIP dialogs

If I unregister Line 1 so that I only have Line 2 active, and then make a call to Line 1, this gets correctly diverted to my cell phone as I have set that up in VFX Portal.

However, when I now make a call to Line 2 I also get diverted to my cell phone.

It's as if the phone number for Line 2 is somehow pointing at the SIP Trunk for Line 1.

Does anyone have any suggestions of what I might have missed during setting up of my second Line?

Regards

Ron

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3551 posts

Uber Geek
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WorldxChange

  Reply # 505294 12-Aug-2011 06:00 Send private message

Hi Ron,

I see only one incoming call to both of those numbers and both where answered, you have call forwards setup for not reachable , busy and no answer and no problems there, but I do not see any calls for yesterday being forwarded off to your mobile 027...xxxx33

If you can do this again and send me the time I will take a look




Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well Wink

             

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  Reply # 505297 12-Aug-2011 06:18 Send private message

Try registering both VFX registrations on different ports.



3 posts

Wannabe Geek


  Reply # 505312 12-Aug-2011 07:57 Send private message

Let's have another go.

called to xxxxxxx77 at 7:29 am
Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry
192.168.x.xxx    6000             xxxxxxxxxxxxxxx  0x100 (g729)     No       Tx: ACK
58.28.20.150     xxxxxx77         xxxxxxxxxxxxxxx  0x100 (g729)     No       Rx: ACK
2 active SIP dialogs

The above is correct and as expected


Call to xxxxxx17 at 7:30 am
Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry
192.168.x.xxx    6000             xxxxxxxxxxxxxxx  0x100 (g729)     No       Tx: ACK
58.28.20.150     xxxxxx77         xxxxxxxxxxxxxxx  0x100 (g729)     No       Rx: ACK
2 active SIP dialogs

This should of come in on Line 2 xxxxxx17, however indicates Line 1 xxxxxx77


Now disabled Line 1 so only Line 2 active
Host                           dnsmgr Username       Refresh State                Reg.Time
pan.wxnz.net:5060              N      xxxxxx17           285 Registered           Fri, 12 Aug 2011 07:35:32
1 SIP registrations.

Call to xxxxxx77 at 7:36 am
Diverted to my cell, this is correct and expected

Call to xxxxxx17 at 7:39 am
Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry
192.168.x.xxx    6000             xxxxxxxxxxxxxxx  0x100 (g729)     No       Tx: ACK
58.28.20.150     xxxxxx77         xxxxxxxxxxxxxxx  0x100 (g729)     No       Rx: ACK
2 active SIP dialogs

Hmmm, I received the call this time, however it is indicating that it came in on Line 1 xxxxx77.
I have been trying al sorts to resolve so may have sorted the diversion part?

I have definately got something wronge with my Asterisk setup as it is not recognising the correct Line internally so that I can route calls to appropriate extensions based on the Line they come in on.

Note: I can answer any call on any extension as all extensions are in ring groups for both lines, this is temporay as I currently cannot route Line 2 calls to a specific extension.

I'll pull out my Asterisk configs and post those later as I'm out of time this morning.
 
Cheers



3 posts

Wannabe Geek


Reply # 506243 14-Aug-2011 13:35 Send private message

Here are my config settings for Asterisk 1.6.2.11
Note: I'm using AsteriskNOW VM Pre-Build.

entries in users.conf

Note: I have not used the exact seetings as supplied by WxC as they do not register the SIP Trunc for me.

[general]
; VFX WorldxChange settings
port = 5060
dtfmmode = rfc2833
disallow = all
allow = g729
allow = ulaw
allow = alaw
registertimeout = 20
regseconds = 180

[VFX_LINE1]
type=peer
fromuser=xxxxxx77 ; this is your VFX Number without the leading 0 i.e. 9950XXXX
host=pan.wxnz.net
insecure=invite,port
canreinvite=no
nat=yes
secret={WxC_Secret}
username=xxxxxx77
;username={WxC_Username}
auth = {WxC_UserName}:{WxC_Secret}@xport.co.nz
;register=xxxxxx77:{WxC_Secret}:{WxC_Username}@pan.wxnz.net/xxxxxx77
context = DID_VFX_LINE1
registersip = yes


[VFX_LINE2]
type=peer
fromuser=xxxxxx17 ; this is your VFX Number without the leading 0 i.e. 9950XXXX
host=pan.wxnz.net
insecure=invite,port
canreinvite=no
nat=yes
secret={WxC_Secret}
username=xxxxxx17
;username={WxC_Username}
auth = {WxC_UserName}:{WxC_Secret}@xport.co.nz
;register=xxxxxx17:{WxC_Secret}:{WxC_Username}@pan.wxnz.net/xxxxxx17
context = DID_VFX_LINE2
registersip = yes


;Plus my phone extensions, now reduced to 2
[6000]
username = 6000
transfer = yes
mailbox = 6000
type = friend
host = dynamic
context = DLPN_Line1
hasvoicemail = yes
vmsecret = XXXXXXXX
hassip = yes
secret = XXXXXXXX
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = port,invite
callcounter = yes
disallow = all
allow = g729,ulaw,alaw

[6001]
username = 6001
transfer = yes
mailbox = 6001
type = friend
host = dynamic
context = DLPN_Line2
hasvoicemail = yes
vmsecret = XXXXXXXXXX
hassip = yes
secret = XXXXXXXXXX
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = port,invite
callcounter = yes
disallow = all
allow = g729,ulaw,alaw


entries in extensions.conf

[globals]
VFX_LINE1 = SIP/VFX_LINE1
VFX_LINE2 = SIP/VFX_LINE2

[DID_VFX_LINE1]
exten = s,1,Dial(SIP/6000,15,${DIALOPTIONS}i)

[DID_VFX_LINE2]
exten = s,1,Dial(SIP/6001,15,${DIALOPTIONS}i)

[CallingRule_OutboundCalls_Line1]
exten = _XXXXXX!,1,Verbose(2,Trunk Dial VFX Line 1 Dial ${VFX_LINE1}/${EXTEN:1})
exten = _XXXXXX!,n,Macro(trunkdial-failover-0.3,${VFX_LINE1}/${EXTEN},,,,)

[CallingRule_OutboundCalls_Line2]
exten = _XXXXXX!,1,Verbose(2,Trunk Dial VFX Line 2 Dial ${VFX_LINE2}/${EXTEN:1})
exten = _XXXXXX!,n,Macro(trunkdial-failover-0.3,${VFX_LINE2}/${EXTEN},,,,)

[default_line1]
exten = _6XXX,1,Set(MBOX=${EXTEN:1}@default)
exten = o,1,Goto(default,6000,1)
exten = _#6XXX,1,Set(MBOX=${EXTEN:1}@default)
exten = _#6XXX,n,VoiceMail(${MBOX})
exten = a,1,VoicemailMain(${MBOX})
exten = 97,1,VoiceMailMain(${CALLERID(num)}@default)

[default_line2]
exten = _6XXX,1,Set(MBOX=${EXTEN:1}@default)
exten = o,1,Goto(default,6001,1)
exten = _#6XXX,1,Set(MBOX=${EXTEN:1}@default)
exten = _#6XXX,n,VoiceMail(${MBOX})
exten = a,1,VoicemailMain(${MBOX})
exten = 97,1,VoiceMailMain(${CALLERID(num)}@default)

[DLPN_Line1]
include = CallingRule_OutboundCalls_Line1
include = default_line1

[DLPN_Line2]
include = CallingRule_OutboundCalls_Line2
include = default_line2


Basically all outgoing calls are fine, however incomming calls are all coming in on Line 1, or at least that's what it looks like.

If I make a call to Line 1 and answer it the channel details show it came in on Line 1.
If I make a call to Line 2 and answer it the channel details show it came in on Line 1.
If I make a call to Line 1 and answer it and leave the phone off the hook, then make a second call to Line 2, I get an engaged signal.

Any suggestions would be great


 

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  Reply # 506249 14-Aug-2011 13:52 Send private message

You presumably have you tried registering them on different ports yet? Use 5060 and 8060.

Context based routing isn't going to work how you're trying to make it work. You need to match on ANI if you've got multiple calls coming from different numbers using the same registration and port.

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