[WXC/VFX] PBX in a Flash + VFX -- can't make/receive calls

Forums › Vodafone New Zealand (including Vodafone, WxC, Farmside) › [WXC/VFX] PBX in a Flash + VFX -- can't make/receive calls

breakaway

105 posts

Master Geek

#137961 14-Dec-2013 20:48

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.

sbiddle

30853 posts

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Biddle Corp

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#952357 14-Dec-2013 21:24

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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chevrolux

4962 posts

Uber Geek
Inactive user

#952376 14-Dec-2013 22:53

Post up your config and reg string....

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

breakaway

105 posts

Master Geek

#952390 15-Dec-2013 00:03

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
######################################################

sbiddle

30853 posts

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#952426 15-Dec-2013 07:57

Do you have a from-trunk-dvx context defined?

breakaway

105 posts

Master Geek

#952446 15-Dec-2013 10:02

I have a "context=from-trunk-dvx " line in my peer config -- so I suppose yes?

sbiddle

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#952449 15-Dec-2013 10:08

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.

breakaway

105 posts

Master Geek

#952474 15-Dec-2013 10:43

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?

Zeon

3861 posts

Uber Geek

Trusted

#952505 15-Dec-2013 11:51

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).

Speedtest 2019-10-14

chevrolux

4962 posts

Uber Geek
Inactive user

#952564 15-Dec-2013 14:46

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.

breakaway

105 posts

Master Geek

#952614 15-Dec-2013 17:27

Alright everything is working now. I see I was missing a pattern for outbound calling.

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