faktortel - what am i doing wrong?

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Zeon

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# 138954 23-Jan-2014 01:41

Hey guys,
Just getting my new 1800 Faktortel number (Aus) setup on our AKL based PBX in a Flash (asterisk). Its all registered but when my friend in Brisbane tried to call she got just beeps and in asterisk -r i see this:

-- Executing [s@from-sip-external:1] GotoIf("SIP/202.43.66.2-00000eed", "0?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,5)
-- Executing [s@from-sip-external:5] Set("SIP/202.43.66.2-00000eed", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2014-01-23 01:35:18.170 NZDT.
-- Executing [s@from-sip-external:6] Answer("SIP/202.43.66.2-00000eed", "") in new stack
-- Executing [s@from-sip-external:7] Wait("SIP/202.43.66.2-00000eed", "2") in new stack
-- Executing [s@from-sip-external:8] Playback("SIP/202.43.66.2-00000eed", "ss-noservice") in new stack
-- <SIP/202.43.66.2-00000eed> Playing 'ss-noservice.gsm' (language 'en')
== Spawn extension (from-sip-external, s, 8) exited non-zero on 'SIP/202.43.66.2-00000eed'
-- Executing [h@from-sip-external:1] Hangup("SIP/202.43.66.2-00000eed", "") in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/202.43.66.2-00000eed'

I thought it was a bad inbound route yet I followed their guide. I then added an inbound router with all DID and CID. No help. The caller just reports beeps.

Any ideas?

sbiddle

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# 972584 23-Jan-2014 07:39

That shows the call coming in as anonymous, ie not matching an inbound route and therefore just hitting the from-sip-external context and being rejected with the default message.

Zeon

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# 980711 5-Feb-2014 14:47

Thanks Sbiddle. I spoke to them about this and it looks like they added a DID number. So I think the DID is 1800177551. I made that the DID on an inbound rotue but still get the same message. Even having a "all did all CID" inbound route still fails. In the trunk settings the context is "context=from-trunk". Unlike VFX they wanted me to add stuff in the context section. Not sure if that gets in the way?

Verbosity is at least 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [1800177551@from-sip-external:1] NoOp("SIP/202.43.66.5-0000109f", "Received incoming SIP connection from unknown peer to 1800177551") in new stack
-- Executing [1800177551@from-sip-external:2] Set("SIP/202.43.66.5-0000109f", "DID=1800177551") in new stack
-- Executing [1800177551@from-sip-external:3] Goto("SIP/202.43.66.5-0000109f", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/202.43.66.5-0000109f", "0?checklang:noanonymous") in new stack
-- Goto (from-sip-external,s,5)
-- Executing [s@from-sip-external:5] Set("SIP/202.43.66.5-0000109f", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2014-02-05 14:39:06.155 NZDT.
-- Executing [s@from-sip-external:6] Answer("SIP/202.43.66.5-0000109f", "") in new stack
-- Executing [s@from-sip-external:7] Wait("SIP/202.43.66.5-0000109f", "2") in new stack
-- Executing [s@from-sip-external:8] Playback("SIP/202.43.66.5-0000109f", "ss-noservice") in new stack
-- Playing 'ss-noservice.gsm' (language 'en')
== Spawn extension (from-sip-external, s, 8) exited non-zero on 'SIP/202.43.66.5-0000109f'
-- Executing [h@from-sip-external:1] Hangup("SIP/202.43.66.5-0000109f", "") in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/202.43.66.5-0000109f'
pbx*CLI>

sbiddle

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# 980864 5-Feb-2014 17:25

It's still an unknown peer so gets treated as an anonymous call. My Default FreePBX blocks anonymous SIP peers for security reasons so it simply treats it as a blocked call and plays the ss-noservice file.

Zeon

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# 980903 5-Feb-2014 18:08

OK so the question is, how do I make it a known peer? This is what asterisk -r says:

VFX/YYYY 58.28.20.150 N 5060 Unmonitored
ftel/XXXX 202.43.66.3 N 5060 OK (67 ms)

I'm not sure why VFX says unmonitored but it works OK. In the faktortel trunk settings it has this too:

type=peer

Do you think Faktortel need to change something?

chevrolux

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# 980955 5-Feb-2014 19:53

I always use type=friend for trunks but not really sure if it would make a difference.

It sounds like Asterisk just can't match any incoming calls to either users or known peers so it is defaulting to the default context which I guess is just flicking it through to the anonymous.

When incoming calls hit the box are they coming from the correct IP for faktortel's server?

Zeon

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# 981551 6-Feb-2014 22:07

Cool thanks for the tips guys. Faktortel support came back and apparently the issue was that the userconext should be the 095 number not the 1800 number. I changed it in the trunk settings and its working now!