[WXC/VFX] Screwed up DVX transfer

Forums › One New Zealand (including Vodafone, WxC, Farmside) › [WXC/VFX] Screwed up DVX transfer - no incoming calls

Zeon

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#51800 1-Dec-2009 10:38

Hey guys, last week I was in contact with a someone from World Exchange, I had asked him to give me some pricing for switching our Araneo connection over to WxC and also to move to DVX so we could have multiple DDIs on the same trunk for inbound fax services.

Anyway on Friday morning I had the pricing and got my questions answered so I was talking to him on the phone saying - yes I want to go ahead with the DVX transfer and we should schedule it for maybe next Friday evening (in a weeks time) so I can get all the documentation sent through etc. and we can plan the switch.

Well anyway he sends through the order to the provisioning team right there and then so when I'm out at lunch I get a call saying the swap has been done. I say WTF we discussed this happening next week outside of business hours but he assures me that the old liens are still active and all we need to do is switch configuration to the new lines when we were ready. I knew this was BS as how could the calls be routed to both trunks at the same time so anyway I try some of the office numbers and none work.

I race back to the office and find that nothing is working and I can no longer reach him. I call up someone else who seemed to have done the switch and explained the situation. She went "oh ****" and immediately changed everything back.

Problem is that they changed the usernames and secrets so I had to recopy their config back over my original trunk settings. Outbound seems to work but no inbound has worked since. I don't know what it is but I'm guessing the new configs they sent us were different to the old ones.

I have sent these through to WxC but still have had no response and our business has been without phones now for 4 days... Could anyone shed some light?

Example (from a brand new trixbox 2.8 install with a single extension):

== Parsing '/etc/asterisk/chan_dahdi.conf':   == Found
== Parsing '/etc/asterisk/dahdi-channels.conf':   == Found
== Parsing '/etc/asterisk/chan_dahdi_additional.conf':   == Found
== Parsing '/etc/asterisk/users.conf':   == Found
    -- Reloading module 'cdr_odbc.so' (ODBC CDR Backend)
    -- Reloading module 'codec_alaw.so' (A-law Coder/Decoder)
    -- Reloading module 'codec_ulaw.so' (mu-Law Coder/Decoder)
== Manager 'admin' logged off from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
== Manager 'admin' logged on from 127.0.0.1
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
    -- Executing [9950XXXX@from-trunk:1] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "__FROM_DID=9950XXXX") in new stack
    -- Executing [9950XXXX@from-trunk:2] Gosub("SIP/8rpdBXfsuewlfRJaOV-b769c260", "app-blacklist-check,s,1") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Return("SIP/8rpdBXfsuewlfRJaOV-b769c260", "") in new stack
    -- Executing [9950XXXX@from-trunk:3] ExecIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "0 ?Set(CALLERID(name)=9576XXXX)") in new stack
    -- Executing [9950XXXX@from-trunk:4] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [9950XXXX@from-trunk:5] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [9950XXXX@from-trunk:6] Goto("SIP/8rpdBXfsuewlfRJaOV-b769c260", "from-did-direct,901,1") in new stack
    -- Goto (from-did-direct,901,1)
    -- Executing [901@from-did-direct:1] Macro("SIP/8rpdBXfsuewlfRJaOV-b769c260", "exten-vm,novm,901") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/8rpdBXfsuewlfRJaOV-b769c260", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "AMPUSER=9576XXXX") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "1?Set(REALCALLERIDNUM=9576XXXX)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "1?report") in new stack
    -- Goto (macro-user-callerid,s,11)
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:12] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:13] GotoIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("SIP/8rpdBXfsuewlfRJaOV-b769c260", "Using CallerID "Sample USERID" <9576XXXX>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "EXTTOCALL=901") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/8rpdBXfsuewlfRJaOV-b769c260", "record-enable,901,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/8rpdBXfsuewlfRJaOV-b769c260", "recordingcheck,20091201-000440,1259579080.10") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20091201-000440,1259579080.10: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/8rpdBXfsuewlfRJaOV-b769c260", "") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/8rpdBXfsuewlfRJaOV-b769c260", "dial,"",tr,901") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/8rpdBXfsuewlfRJaOV-b769c260", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Manager 'admin' logged on from 127.0.0.1
dialparties.agi: Caller ID name is 'Jonathan Spence' number is '9576XXXX'
dialparties.agi: Methodology of ring is 'none'
    -- dialparties.agi: Added extension 901 to extension map
    -- dialparties.agi: Extension 901 cf is disabled
    -- dialparties.agi: Extension 901 do not disturb is disabled
    -- dialparties.agi: dbset CALLTRACE/901 to 9576XXXX
    -- dialparties.agi: Filtered ARG3: 901
== Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/8rpdBXfsuewlfRJaOV-b769c260", "SIP/901,"",tr") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
    -- Couldn't call 901
== Everyone is busy/congested at this time (0:0/0/0)
    -- Executing [s@macro-dial:8] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-dial:9] GosubIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "0?CHANUNAVAIL,1") in new stack
    -- Executing [s@macro-exten-vm:10] GotoIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "0?exit,return") in new stack
    -- Executing [s@macro-exten-vm:11] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:12] GosubIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "0?docfu,1") in new stack
    -- Executing [s@macro-exten-vm:13] GosubIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "0?docfb,1") in new stack
    -- Executing [s@macro-exten-vm:14] Set("SIP/8rpdBXfsuewlfRJaOV-b769c260", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:15] NoOp("SIP/8rpdBXfsuewlfRJaOV-b769c260", "Voicemail is 'novm'") in new stack
    -- Executing [s@macro-exten-vm:16] GotoIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "1?s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/8rpdBXfsuewlfRJaOV-b769c260", "IVR_RETVM: IVR_CONTEXT: ") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/8rpdBXfsuewlfRJaOV-b769c260", "0?exit,1") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/8rpdBXfsuewlfRJaOV-b769c260", "congestion") in new stack
== Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/8rpdBXfsuewlfRJaOV-b769c260' in macro 'exten-vm'
== Spawn extension (from-did-direct, 901, 1) exited non-zero on 'SIP/8rpdBXfsuewlfRJaOV-b769c260'

[Moderator edit (MF): removed names and bad words]

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Zeon

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#278209 1-Dec-2009 11:35

I have just spoken to WxC who has been quite helpful - its now looking like a codec mismatch. What I think is that the updated configs WxC sent us had g729 which we can't handle so just removing those to see what happens.

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maverick

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#278218 1-Dec-2009 12:01

The issue here was the G729 codec was not licenced on the box but was being allowed in the config so when G729 was offered it was being accepted in the signalling even though it was not supported. All inbound calls would drop after answer due to no G729 support.

We have ammeded the details in the information email you receive to now say this, this should stop any reoccurance of this misconfiguration.

Trunk/Peer Settings:
[General]
port = 5060
dtmfmode = rfc2833
disallow=all
allow=g729 (G729 is a licensed Codec and must be licensed and enabled on your box if adding this line to your config)
allow=ulaw
allow=alaw

Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

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Zeon

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#278225 1-Dec-2009 12:08

Yup that was it. In our original config we didn't have the:

allow=g729
line in the trunk details.

However the new emails that myVFX sends this is included by default. After removing this line everything went perfectly again using g711 as it had previously done. WxC is now going to add a disclaimer where people are warned that they must be licensed for this codec to use it.

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sbiddle

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#278266 1-Dec-2009 13:33

If you are wanting G.729 licences from Digium aren't expensive - US$10 per channel. When you're getting the PBX for free that's not a bad price if you want to think of it as a donation!

There is a freely available G.729 codec for Asterisk that can be downloaded here.

It just comes down to what your bandwidth and voice quality requirements are.

Zeon

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#278318 1-Dec-2009 14:36

To be honest I can't think of too many reasons to use g729? Bandwidth has never been a problem (symmetrical 5mbps internet) and for faxability you need to use g711 anyway? Its just another thing to manage IMO - I'd prefer to give just a straight donation to Trixbox!

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sbiddle

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#278337 1-Dec-2009 15:08

If you have a good internet connection with plenty of bandwidth and reasonable cap then there is no need to move away from G.711 to G.729 or any other codec.

Most people move to G.729 if they're constrainted by bandwidth or want to take advantage of the better jitter correction and packet loss on poor quality connections.

For faxing you need G.711 or ideally hardware that supports T.38

richms

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#278341 1-Dec-2009 15:11

I have tried both the paidfor and the free g729 and the paidfor one is a lot better IMO - the free one was having massive breakup on playing short speach files to some callers. Its the only way to get usable calls on a crappy adsl connection with no incoming QOS, but if you have a proper connection there is no way that I would be using it.

Richard rich.ms

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bender

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#278393 1-Dec-2009 17:02

G.729 sounds horrible anyway, I usually disable it.

Also in future you would have been able to see the problem immediately if you do sip set debug ip 58.28.20.150 (or 58.28.20.101 on direct connect). You'll find the codec list each side supports at the bottom of the packets, and you'd see the 488 messages dropping the calls due to unacceptable codecs.

-Scott

Zeon

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#278401 1-Dec-2009 17:13

bender: G.729 sounds horrible anyway, I usually disable it.

Also in future you would have been able to see the problem immediately if you do sip set debug ip 58.28.20.150 (or 58.28.20.101 on direct connect). You'll find the codec list each side supports at the bottom of the packets, and you'd see the 488 messages dropping the calls due to unacceptable codecs.

Ahh right - I didn't know that command. Yea luckily WxC did it on their end hence how we found that it wasn't accepting g729. Good to know for the future though!

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coffeebaron

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#278406 1-Dec-2009 17:21

Zeon: To be honest I can't think of too many reasons to use g729? Bandwidth has never been a problem (symmetrical 5mbps internet) and for faxability you need to use g711 anyway? Its just another thing to manage IMO - I'd prefer to give just a straight donation to Trixbox!

Good for remote extensions where the broadband at other end could be questionable.

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Zeon

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#278414 1-Dec-2009 17:33

Never had a situation with questionable broadband TBH. Even in our old building where the wiring was absolutely atrocious (IE got cut because birds were nesting in the roof and put their claws through the cable) G711 still ran no problems?

Dial Up?

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coffeebaron

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#278422 1-Dec-2009 17:45

Zeon: Never had a situation with questionable broadband TBH. Even in our old building where the wiring was absolutely atrocious (IE got cut because birds were nesting in the roof and put their claws through the cable) G711 still ran no problems?

Dial Up?

EG. using a softphone in a hotel in China or somewhere to dial out via NZ office. Also, using over mobile 3G connection.

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maverick

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#278578 2-Dec-2009 05:52

bender: G.729 sounds horrible anyway, I usually disable it.

Also in future you would have been able to see the problem immediately if you do sip set debug ip 58.28.20.150 (or 58.28.20.101 on direct connect). You'll find the codec list each side supports at the bottom of the packets, and you'd see the 488 messages dropping the calls due to unacceptable codecs.

-Scott

That is a good command to use for debugging, however in this particular case mesaging is fine and would not show the problem , you do not see a SIP 488 at all, The issue appears only as a drop inbound call

The G729 cdoec is offered by the incoming call (Invite) and accepted in the 200 okay from the Asterisk end, but because no g729 is actually available from the Asterisk end the call will drop with a bye from the Asterisk end,

Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

https://www.facebook.com/wxccommunications

Zeon

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#279089 3-Dec-2009 17:37

OK, the DVX switch is all done and working fine. Yay!

Anyway, how does one differentiate between the different DIDs that Xnet deliver? At the moment I have set up incoming routes in Trixbox for 2 of our DDIs - 9950330X and 9951044X. The actual trunk is setup with the number 9951044X but when I try the 9950330X number it still comes up with the DID of the trunk. This is what Asterisk is saying:

Executing [9951044X@from-trunk:1] Set("SIP/UmSR8eRHcqRFcibCq8-b7710a38", "__FROM_DID=9951044X") in new stack

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