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incoming trunk

kalle
Posts: 64
 Member Since:
2008-06-20
Submitted by kalle on Fri, 06/27/2008 - 5:11am.

hi
please can anyone advice how to set up an incoming trunk just with ip authentication
i just have the ip of the customer and I know that they will be sending SIP trafic and G729 codec

any advice please urgent ?

BR

‹ Working name based "Call Intercept" / "Privacy Manager" Constant GUI Logging Messages ›

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kalle
Posts: 64
 Member Since:
2008-06-20
please any advice ? i need
Tue, 07/01/2008 - 4:02am

please any advice ?
i need to set up an incoming trunk using ip authentication and sip trun


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kalle
Posts: 64
 Member Since:
2008-06-20
hi i realy need help to set
Thu, 07/03/2008 - 1:27pm

hi
i realy need help to set up an incoming trunk with ip authentication
please urgent


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jonnytabpni
Posts: 379
 Member Since:
2007-03-21
Please can you explain this
Thu, 07/03/2008 - 1:48pm

Please can you explain this a bit more.

A sip trunk is a connection between you and a telcome provider to provide a connection to phones on the outside world.

However you are referring to a "customer". Can you explain what you wish to acheive in the end?

Cheers


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kalle
Posts: 64
 Member Since:
2008-06-20
i would like to allow a
Thu, 07/03/2008 - 2:00pm

i would like to allow a client to make calls from my asterisk and calls coming from a sip trunk the only thing i have is the ip of the client and the codec used
the outgoing trunk works fine when i test from a soft phone connected to trixbox
please urgent


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jonnytabpni
Posts: 379
 Member Since:
2007-03-21
okok relax. just set up an
Thu, 07/03/2008 - 2:02pm

okok relax.

just set up an account for him like you would for your test phone.

Then forward the ports on your firewall then get your guy to connect to your server the same way as you would your test phone except use your external IP


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kalle
Posts: 64
 Member Since:
2008-06-20
its not the way i need it i
Thu, 07/03/2008 - 2:09pm

its not the way i need it i need it to be peer-peer and only ip authentication required no username or password

something like this :
type=peer
host=ip of the client
context=i don't know which one to use
insecure= needed ?
qualify=++
canreinvite ??

and soon

br


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kalle
Posts: 64
 Member Since:
2008-06-20
i would like to allow calls
Thu, 07/03/2008 - 2:14pm

i would like to allow calls coming from a specific ip no username/password and then route the calls to the outgoing trunk which is working and the out to the world.
please any advice ?

BR


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kalle
Posts: 64
 Member Since:
2008-06-20
lets say it in different
Thu, 07/03/2008 - 2:25pm

lets say it in different words
how to interconnect a provider for incoming calls and i don't want the calls to be terminated at any extension since the calls are for outgoing just going through my trixbox and they should be terminated out to the world
the only thing i got from provider is the ip and the codec and protocol to be used


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jonnytabpni
Posts: 379
 Member Since:
2007-03-21
www.freeswitch.org
Thu, 07/03/2008 - 2:32pm

www.freeswitch.org


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kalle
Posts: 64
 Member Since:
2008-06-20
sorry what's that for ?
Thu, 07/03/2008 - 2:37pm

sorry what's that for ?


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kalle
Posts: 64
 Member Since:
2008-06-20
well how to receive calls
Thu, 07/03/2008 - 2:38pm

well how to receive calls from my provider and send it out to the world

br


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SkykingOH
Posts: 9538
 Member Since:
2007-12-17
You need to get this working
Thu, 07/03/2008 - 3:07pm

You need to get this working in steps:

  • Turn on 'Allow anonymous SIP' in General Settings
  • Watch the logs and see what the originating DID looks like
  • Add an inbound route for that pattern to an extension for testing
  • Forward that extension to where you want the call to go
  • Either lock down inbound SIP based on the originating IP of the host sendig you the SIP calls or setup a trunk with a permit = directive for that IP
  • You can eliminate the extension forwarding step with some Asterisk code if you wish

    • This should get you pointed in the right direction.

    Scott

    --

    Scott

    aka "Skyking"


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    kalle
    Posts: 64
     Member Since:
    2008-06-20
    well the DID is not the same
    Thu, 07/03/2008 - 3:12pm

    well the DID is not the same the called number will be different since the calls are made to different countries

    any other idea please ?


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    kalle
    Posts: 64
     Member Since:
    2008-06-20
    here is what I'm
    Thu, 07/03/2008 - 3:14pm

    here is what I'm receiving

    --- (9 headers 0 lines) ---
    Really destroying SIP dialog '2e31c9fe24b5209f43ca0ace29e68b6d@91.189.47.52' Method: OPTIONS
    -- Executing [00523336357858@from-sip-external:1] NoOp("SIP/64.71.145.237-b7d09810", "Received incoming SIP connection from unknown peer to 00523336357858") in new stack
    -- Executing [00523336357858@from-sip-external:2] Set("SIP/64.71.145.237-b7d09810", "DID=00523336357858") in new stack
    -- Executing [00523336357858@from-sip-external:3] Goto("SIP/64.71.145.237-b7d09810", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/64.71.145.237-b7d09810", "0?from-trunk|00523336357858|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/64.71.145.237-b7d09810", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2008-07-03 22:13:09 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/64.71.145.237-b7d09810", "") in new stack
    -- Executing [s@from-sip-external:4] Wait("SIP/64.71.145.237-b7d09810", "2") in new stack
    -- Executing [s@from-sip-external:5] Playback("SIP/64.71.145.237-b7d09810", "ss-noservice") in new stack
    -- Playing 'ss-noservice' (language 'en')
    -- Executing [s@from-sip-external:6] PlayTones("SIP/64.71.145.237-b7d09810", "congestion") in new stack
    -- Executing [s@from-sip-external:7] Congestion("SIP/64.71.145.237-b7d09810", "5") in new stack
    == Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/64.71.145.237-b7d09810'
    -- Executing [h@from-sip-external:1] NoOp("SIP/64.71.145.237-b7d09810", "Hangup") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/64.71.145.237-b7d09810", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/64.71.145.237-b7d09810", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/64.71.145.237-b7d09810", "0?from-trunk|s|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/64.71.145.237-b7d09810", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2008-07-03 22:13:20 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/64.71.145.237-b7d09810", "") in new stack
    == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/64.71.145.237-b7d09810'
    Reliably Transmitting (no NAT) to 216.176.182.194:5060:


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    kalle
    Posts: 64
     Member Since:
    2008-06-20
    hi if I allow the anonyms in
    Thu, 07/03/2008 - 3:31pm

    hi
    if I allow the anonyms in general settings the the output as follow:

    --- (9 headers 0 lines) ---
    Really destroying SIP dialog '0eab9d3f2f207fb549cdb64e2bd3c3a9@91.189.47.52' Method: OPTIONS
    -- Executing [00523336804171@from-sip-external:1] NoOp("SIP/64.71.145.237-088cfc58", "Received incoming SIP connection from unknown peer to 00523336804171") in new stack
    -- Executing [00523336804171@from-sip-external:2] Set("SIP/64.71.145.237-088cfc58", "DID=00523336804171") in new stack
    -- Executing [00523336804171@from-sip-external:3] Goto("SIP/64.71.145.237-088cfc58", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/64.71.145.237-088cfc58", "1?from-trunk|00523336804171|1") in new stack
    -- Goto (from-trunk,00523336804171,1)

    and then nothing more


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    SkykingOH
    Posts: 9538
     Member Since:
    2007-12-17
    Why don't you 'set verbose
    Thu, 07/03/2008 - 3:46pm

    Why don't you 'set verbose 0' then 'sip debug' and take a look at the invites and see what field is common you can work with for routing.

    Scott

    --

    Scott

    aka "Skyking"


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    kalle
    Posts: 64
     Member Since:
    2008-06-20
    how to do this ?
    Thu, 07/03/2008 - 3:54pm

    how to do this ?


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    kalle
    Posts: 64
     Member Since:
    2008-06-20
    --- (11 headers 0 lines)
    Thu, 07/03/2008 - 3:58pm

    --- (11 headers 0 lines) ---
    rixbox1*CLI>

    ACK sip:523336958910@91.189.47.52 SIP/2.0
    Via: SIP/2.0/UDP 64.71.145.237;branch=z9hG4bK.68e3;received=64.71.145.237
    From: ;tag=333966323534333
    To: ;tag=as5de76888
    Call-ID: 333966323534333
    CSeq: 1 ACK
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Length: 0

    if you have experience and you can help i can give you access to my box so we can sort this is out maybe

    BR


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    kalle
    Posts: 64
     Member Since:
    2008-06-20
    hi if i have an incoming
    Sat, 07/05/2008 - 11:45am

    hi
    if i have an incoming trunk to receive call on and then i route the call to an outgoing trunk do i need to channels for this call ? 2 G729 licences ?


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