Problem Receiving Calls from VoIP Provider

nachogomez

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#1
Hi list,

I have problems receiving calls from my SIP VoIP Provider (Netuno in Venezuela), I can call any number using this trunk, but when someone tries to call me, the call is rejected.

If I do a TCPdump I can see a response from my PBX saying:


18:07:44.424368 IP (tos 0x10, ttl 58, id 4085, offset 0, flags [DF], proto: UDP (17), length: 1427) IP_OF_MY_PROVIDER.sip > MY_IP_ADDRESS.sip: SIP, length: 1399
INVITE sip:MY_NUMBER@MY_IP SIP/2.0

18:07:44.424983 IP (tos 0x60, ttl 64, id 14573, offset 0, flags [none], proto: UDP (17), length: 597) MY_IP_ADDRESS.sip > IP_OF_MY_PROVIDER.sip: SIP, length: 569
SIP/2.0 407 Proxy Authentication Required


I can receive calls with my TDM Card just fine.

The support person of my provider have said to me that my config is OK.

What can be happening here???
 

jgutierrez

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#2
Una pregunta, sí has definido correctamente las rutas entrantes para tu sip trunk?
 

nachogomez

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#3
Well, I have defined my Inbound Routes to accept "Any DID / Any CallerID". I can receive calls made to my PSTN lines OK, so I'm a little lost right now...
 

nachogomez

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#4
Someone has had luck setting up Elastix (or any other flavor of Asterisk) with Netuno??? (Venezuela).

My peer config is working OK since I can place call using it, but I think the "user" part of the setting needs an options that is missing, these are my configurations:


[netuno]
username=MyPhoneNumber
type=peer
secret=UltraSecret
qualify=no
port=5060
nat=never
host=IP_OF_SIP_PROVIDER
fromuser=MyPhoneNumber
dtmfmode=rfc2833
context=from-internal
canreinvite=no
disallow=all
allow=g729

register=MyPhoneNumber:UltraSecret@IP_OF_SIP_PROVIDER/MyPhoneNumber

[MyPhoneNumber]
username=MyPhoneNumber
type=user
secret=UltraSecret
rtptimeout=30
qualify=no
port=5060
nat=never
host=MyPublicIP
dtmfmode=rfc2833
context=from-sip-external
canreinvite=no
authuser=MyPhoneNumber
disallow=all
allow=g729

Note: MyPhoneNumber is set without the leading 0
 

jgutierrez

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#5
Ok,

If you can receive your phone calls setting the Inbound Routes to "Any DID / Any CallerID", you will need to know what DID are you receiving from your SIP trunk provider.

The are some providers that doens't pass all DID digits, only the three last of them, or the fourth last of them. To know exactly what you are receiving, Ill suggest you to do the following:

On the cli, capture the logs of asterisk when you receive a phone call, and then paste that info in here.
 

nachogomez

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#6
I can only receive calls made to my PSTN (Analog TDM) lines, I still can't receive any calls made to my SIP line :(, even with the "Any DID / Any CallerID" in the Inbound Route
 

nachogomez

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#7
Well, today was a very productive day working with my Elastix PBX, I've resolve the problem playing the envelope of the voicemail (thanks commandLineUruguay), I have configured successfully the BLF of my GXP-2000 (thanks nano) and now the most problematic thing: "Problem Receiving Calls from VoIP Provider".

The problem was basically an include in sip.conf (called sip_general_additional.conf) which only allow calls with ulaw or alaw codec, since my provider only accept g729 and doesn't use the user part of the trunk, it wasn't able to receive the incoming call.

I've just commented the include and pasted its content to sip.conf, made several reloads in order to confirm that the changes stays and now is working perfect!!!

Thanks to all who post (danardf, jgutierrez, etc).

Nacho
 

jgutierrez

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#8
Oh, excellent, those are good news ! B)

Just make sure that it doesn't get overwriiten when you do a "submit" + "Apply changes" on the web interface
 

nachogomez

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#9
BTW, how can I change this settings via FreePBX??? I cannot find it in the web gui...
 

nachogomez

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#10
Nevermind, I've just added allow=g729 to sip_general_custom.conf which is fine because this file doesn't get overwrite by FreePBX
 

jgutierrez

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#11
Yep, that is correct
 

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