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Topic # 138954 23-Jan-2014 01:41 Send private message

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?





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  Reply # 972584 23-Jan-2014 07:39 Send private message

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.








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  Reply # 980711 5-Feb-2014 14:47 Send private message

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>






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  Reply # 980864 5-Feb-2014 17:25 Send private message

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.



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  Reply # 980903 5-Feb-2014 18:08 Send private message

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?





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  Reply # 980955 5-Feb-2014 19:53 Send private message

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?



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  Reply # 981551 6-Feb-2014 22:07 Send private message

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!





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