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970 posts

Ultimate Geek

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Topic # 107679 16-Aug-2012 19:44 Send private message

I ported my number about 2 weeks ago from 2Talk to VFX. Got nothing bad to say about 2Talk just wanted to give VFX a go.
Anyway, porting went through sweet as and the Xnet guys are so so so good to deal with. Was really impressed with the whole process.
Chucked in the registration details in to Asterisk and assumed all is good. In my stupidity though I forgot to change the outbound route to pick the VFX trunk instead of the 2Talk trunk which still had an 028 number attached so I could make calls out no problem. The problem arose when Mum tried ringing me lol. she couldn't get through. I looked at Asterisk and realised what I had done. So have deleted the entire 2Talk trunk out of there. However, now I am just having no joy with the VFX trunk.
Below is what I have put in to the trunk settings. They are pretty much as supplied by Xnet but I must be missing something...

Trunk name: VFX

type=peer
fromuser=63564340
host=pan.wxnz.net
insecure=invite,port
canreinvite=no
nat=yes
secret=*****
username=******
context=from-trunk

User context: **username**

Reg string: as supplied

Any thoughts as to what I might be doing?

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16699 posts

Uber Geek

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Biddle Corp
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  Reply # 673496 16-Aug-2012 21:35 Send private message

The registration string is important but since you haven't listed it we don't know if you're using the correct details.

Do you have an inbound route set? Is the call hitting your PBX or are you hearing Neil Waka?




*Need help configuring your Linksys ATA or IP Phones for New Zealand? Check my blog post



970 posts

Ultimate Geek

Subscriber

  Reply # 673508 16-Aug-2012 22:05 Send private message

So the registration string is:
63564340:secret:username@pan.wxnz.net/63564340

I have an inbound route for 063564340 set to go to my front end IVR. When I ring from a cell phone the call just drops and I don't see anything show in the asterisk debug logs. Actually as a side note, this is what I see in the log from the registration..

[2012-08-16 21:53:05] NOTICE[2647] chan_sip.c: -- Registration for '63564340@pan.wxnz.net' timed out, trying again (Attempt #261)
[2012-08-16 21:53:25] WARNING[2647] chan_sip.c: Maximum retries exceeded on transmission 1b6cf21e0f2f85c75bae9d1760d90306@127.0.0.1 for seqno 363 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Just checked on my firewall too and nothing is getting blocked as far as I can see. Added a pass rule just for good luck for 58.28.20.150. There is also no port forwarding for in place for 5060.

2367 posts

Uber Geek

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  Reply # 673544 17-Aug-2012 01:05 Send private message

chevrolux: So the registration string is:
63564340:secret:username@pan.wxnz.net/63564340

I have an inbound route for 063564340 set to go to my front end IVR. When I ring from a cell phone the call just drops and I don't see anything show in the asterisk debug logs. Actually as a side note, this is what I see in the log from the registration..

[2012-08-16 21:53:05] NOTICE[2647] chan_sip.c: -- Registration for '63564340@pan.wxnz.net' timed out, trying again (Attempt #261)
[2012-08-16 21:53:25] WARNING[2647] chan_sip.c: Maximum retries exceeded on transmission 1b6cf21e0f2f85c75bae9d1760d90306@127.0.0.1 for seqno 363 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Just checked on my firewall too and nothing is getting blocked as far as I can see. Added a pass rule just for good luck for 58.28.20.150. There is also no port forwarding for in place for 5060.


Your issue is probably with the CID not being picked up. I believe you need to edit the SIP file slightly for asterisk to get correct CID from VFX. Try an any/any CID/DID rule (f you are using FreePBX) and it should work.





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Biddle Corp
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  Reply # 673552 17-Aug-2012 06:52 Send private message

chevrolux: So the registration string is:
63564340:secret:username@pan.wxnz.net/63564340

I have an inbound route for 063564340 set to go to my front end IVR. When I ring from a cell phone the call just drops and I don't see anything show in the asterisk debug logs. Actually as a side note, this is what I see in the log from the registration..

[2012-08-16 21:53:05] NOTICE[2647] chan_sip.c: -- Registration for '63564340@pan.wxnz.net' timed out, trying again (Attempt #261)
[2012-08-16 21:53:25] WARNING[2647] chan_sip.c: Maximum retries exceeded on transmission 1b6cf21e0f2f85c75bae9d1760d90306@127.0.0.1 for seqno 363 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Just checked on my firewall too and nothing is getting blocked as far as I can see. Added a pass rule just for good luck for 58.28.20.150. There is also no port forwarding for in place for 5060.


By the looks of that your box isn't even registering, what does a show peers tell you?

You also have your DID wrong for the incoming route.




*Need help configuring your Linksys ATA or IP Phones for New Zealand? Check my blog post



970 posts

Ultimate Geek

Subscriber

  Reply # 673618 17-Aug-2012 10:23 Send private message

show peers gives me:
VFX/username 58.28.20.150 N 5060 Unmonitored

Do I need to add register=*something* in to the trunk settings to make it register?

Does the DID from VFX not have the leading zero or something like that?

EDIT:
Just added register=*string* in to the trunk settings and I can make calls!! Originally the only place I had the register string was in the dedicated field of FreePBX.

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