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93 posts

Master Geek
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Topic # 137961 14-Dec-2013 20:48
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Hello, I've got my SIP trunk registered correctly (according to asterisk), but when I make a call TO my number from my cellphone, I get a message saying "The party you are trying to reach is not available" and asterisk -r shows this:

 
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5


When I make an outgoing call from my soft phone registered to my asterisk I get "Your call cannot be completed as dialled, please check the number & dial again." asterisk -r shows this:    


== Using SIP VIDEO TOS bits 136

  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [0800000000@from-internal:1] ResetCDR("SIP/001-00000016", "") in new stack
    -- Executing [0800000000@from-internal:2] NoCDR("SIP/001-00000016", "") in new stack
    -- Executing [0800000000@from-internal:3] Progress("SIP/001-00000016", "") in new stack
    -- Executing [0800000000@from-internal:4] Wait("SIP/001-00000016", "1") in new stack
    -- Executing [0800000000@from-internal:5] Progress("SIP/001-00000016", "") in new stack
    -- Executing [0800000000@from-internal:6] Playback("SIP/001-00000016", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
    -- <SIP/001-00000016> Playing 'silence/1.gsm' (language 'en')
    -- <SIP/001-00000016> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
    -- <SIP/001-00000016> Playing 'check-number-dial-again.gsm' (language 'en')
    -- Executing [0800000000@from-internal:7] Wait("SIP/001-00000016", "1") in new stack
    -- Executing [0800000000@from-internal:8] Congestion("SIP/001-00000016", "20") in new stack
  == Spawn extension (from-internal, 0800000000, 8) exited non-zero on 'SIP/001-00000016'
    -- Executing [h@from-internal:1] Hangup("SIP/001-00000016", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/001-00000016'



Any ideas why this isn't working? My knowledge level with Asterisk/PBX in general is "knows enough to be dangerous" so step by step insturctions on how to nail down what's wrong with it would be appreciated.

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25591 posts

Uber Geek
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  Reply # 952357 14-Dec-2013 21:24
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SIP registration is only used for inbound calls. It's not possible to tell what's happening from what debug - that's jut Asterisk rejecting the outbound trunk.

You either have your trunk settings wrong or the call isn't being correctly routed via the VFX trunk.

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Uber Geek
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  Reply # 952376 14-Dec-2013 22:53
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Post up your config and reg string....




.....just don't give us user and password for obvious reasons

 
 
 
 




93 posts

Master Geek
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  Reply # 952390 15-Dec-2013 00:03
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Hi,

I am quite positive its registering correctly because when I log into WxC portal and go into registered devices I can see my freePBX there, and when I go to Reports > Asterisk Info > SIP Info, it's showing as registered. My network consists of a Linksys AM300 (slingshot ADSL) bridged with a WRT54G running Tomato. I've forwarded port 5060 to my PBX which is NAT'ed behind my ADSL.

Here are my registration details as configured in the "trunk" section:

Outbound CallerID: 99501111
CID Options: Allow any CID
Maximum Channels: 1

PEER Details
######################################################
username= REDACTED
type=peer
secret= REDACTED
qualify=yes
nat=yes
insecure=invite,port
host=pan.wxnz.net
fromuser=99501111; this is your VFX Number without the leading 0 i.e. 9950XXXX
dtmfmode=rfc2833
disallow=all
context=from-trunk-dvx
canreinvite=no
allow=ulaw&alaw&g722
######################################################

Registration String:
######################################################
99501111:REDACTED:REDACTED@pan.wxnz.net/99501111
######################################################

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Uber Geek
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  Reply # 952426 15-Dec-2013 07:57
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Do you have a from-trunk-dvx context defined?



93 posts

Master Geek
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  Reply # 952446 15-Dec-2013 10:02
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I have a "context=from-trunk-dvx " line in my peer config -- so I suppose yes?

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  Reply # 952449 15-Dec-2013 10:08
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No.

You're referring to a context, but to do that the context 'from-trunk-dvx' needs to exist. One presumes you haven't created this.

What is specific about your setup that you're wanting to use a custom context rather than just the usual from-trunk or from-pstn? If you're wanting to route DDI's my blog post of contexts on VFX describes how to set this up correctly.




93 posts

Master Geek
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  Reply # 952474 15-Dec-2013 10:43
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Hi Steve,

I'm not trying to do anything fancy. Just want to get it to work. Obviously I've got something configured wrong. I managed to grab a look at another VFX PiAF PBX which is working correctly, and changed around my config a bit, I think it's correct now:

#####
port=5060
dtmfmode=rfc2833
disallow=all
allow=ulaw&alaw&g722
registertimeout=20
regseconds=180
type=peer
fromuser=REDACTED; this is your VFX Number without the leading 0 i.e. 9950XXXX
host=pan.wxnz.net
insecure=invite,port
canreinvite=no
nat=yes
secret=REDACTED
username=REDACTED
#####

I now have inbound calling working, no cigar on outbound calls. I've created a 'default' outbound route (no restrictions -- allow all) in PiAF and selected my trunk (config of which is above) as the source, but it's still not working. The debug shows the same as what I have in my first post.

Is anything else other than setting up the SIP trunk and then creating outbound routes required to get outbound calling working?

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Uber Geek
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  Reply # 952505 15-Dec-2013 11:51
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Yea he copied from another PBX which uses the custom context to pass DDI information in DVX hence why that line was in there (and I configured that based on your blog post Steve).





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Uber Geek
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  Reply # 952564 15-Dec-2013 14:46
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You still need a 'context=' line in there.
If you just have the one registration with a single DDI just use 'context=from-trunk'.

If you have multiple DDI's you can us a custom context but you need to tell Asterisk what this custom context actually is.



93 posts

Master Geek
+1 received by user: 3


  Reply # 952614 15-Dec-2013 17:27
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Alright everything is working now. I see I was missing a pattern for outbound calling.

http://i.snag.gy/mpxAv.jpg

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