[Asterisk-Users] Incoming calls from BudgetPhone.nl

Sun Jul 10 11:04:53 MST 2005

Rene,

I believe you're right, when I disable x-ten's stun server my call isn't
coming through anymore.

But now I don't have a solution but an extra problem I'm afraid!

How to make asterisk run with a stun server? 
Do I have to set one up myself or can I use the x-ten server for example?
Or is there a better way to setup asterisk or my router?

Thanks for your help, hopefully you can help me some more!

Peter Raaijmakers.

-----Oorspronkelijk bericht-----
Van: Rene Kluwen [mailto:rene.kluwen at chimit.nl] 
Verzonden: zondag 10 juli 2005 19:28
Aan: voip at boumakers.nl; Asterisk Users Mailing List - Non-Commercial
Discussion
Onderwerp: Re: [Asterisk-Users] Incoming calls from BudgetPhone.nl

Long short,

Maybe X-Ten has an stun relay setup and Asterisk doesn't?

Rene Kluwen
Chimit

> (this time with subject....)
>
> Hello,
>
> I’m trying to get Asterisk to accept incoming calls from budgetphone.nl.
> When I dial my budgetphone nr on a PSTN KPN line it immediately gives a
> busy
> tone.
> I tried X-lite, which worked perfect, so my modem (with nat) probably is
> not
> the problem.
> I did a sip debug and got the following output.
> Because I’m new to Asterisk I can’t get the error why this is not working.
> To me it all looks fine, no warnings or what so ever

>  
> The settings in sip.conf and extensions.conf are identical to those of
> http://www.voip-info.org/tiki-index.php?page=Talkin2ya
>  
> Does anyone know what I’m doing wrong????
>  
> Thanks,
> Peter.
>  
>  
> -------------------------------
> output of sip debug
> -------------------------------
>  
> 11 headers, 0 lines
> Reliably Transmitting (no NAT) to 81.23.228.150:5060:
> REGISTER sip:budgetphone.nl SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc
> From: <sip:31717110342 at budgetphone.nl>;tag=as5dc83db4
> To: <sip:31717110342 at budgetphone.nl>
> Call-ID: 26dfb15414601a871799536a3de1f776 at 127.0.0.1
> CSeq: 102 REGISTER
> User-Agent: Asterisk PBX
> Expires: 120
> Contact: <sip:31717110342 at 192.168.2.3>
> Event: registration
> Content-Length: 0
>  
>  
> ---
> server*CLI>
> <-- SIP read from 81.23.228.150:5060:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc
> From: <sip:31717110342 at budgetphone.nl>;tag=as5dc83db4
> To:
> <sip:31717110342 at budgetphone.nl>;tag=9b5971f23d18872ff678d4e9dae023f8.247a
> Call-ID: 26dfb15414601a871799536a3de1f776 at 127.0.0.1
> CSeq: 102 REGISTER
> WWW-Authenticate: Digest realm="budgetphone.nl",
> nonce="42d15009299d7652e8da589cee2723af4b6a96ca"
> Server: Sip EXpress router (0.8.14-5 (i386/linux))
> Content-Length: 0
>  
>  
> --- (9 headers 0 lines)---
> Responding to challenge, registration to domain/host name budgetphone.nl
> 12 headers, 0 lines
> Reliably Transmitting (no NAT) to 81.23.228.150:5060:
> REGISTER sip:budgetphone.nl SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e
> From: <sip:31717110342 at budgetphone.nl>;tag=as7e56000d
> To: <sip:31717110342 at budgetphone.nl>
> Call-ID: 26dfb15414601a871799536a3de1f776 at 127.0.0.1
> CSeq: 103 REGISTER
> User-Agent: Asterisk PBX
> Authorization: Digest username="31717110342", realm="budgetphone.nl",
> algorithm=MD5, uri="sip:budgetphone.nl",
> nonce="42d15009299d7652e8da589cee2723af4b6a96ca",
> response="cd69279e6a2512fd48d267ceea3394da", opaque=""
> Expires: 120
> Contact: <sip:31717110342 at 192.168.2.3>
> Event: registration
> Content-Length: 0
>  
>  
> ---
> server*CLI>
> <-- SIP read from 81.23.228.150:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e
> From: <sip:31717110342 at budgetphone.nl>;tag=as7e56000d
> To:
> <sip:31717110342 at budgetphone.nl>;tag=9b5971f23d18872ff678d4e9dae023f8.98b0
> Call-ID: 26dfb15414601a871799536a3de1f776 at 127.0.0.1
> CSeq: 103 REGISTER
> Contact: <sip:31717110342 at 62.131.187.108:5060>;q=0.00;expires=120
> Server: Sip EXpress router (0.8.14-5 (i386/linux))
> Content-Length: 0
>  
>  
> --- (9 headers 0 lines)---
> Jul 10 18:38:04 NOTICE[26004]: chan_sip.c:8266 handle_response: Outbound
> Registration: Expiry for budgetphone.nl is 120 sec (Scheduling
> reregistration in 105000 ms)
> Destroying call '26dfb15414601a871799536a3de1f776 at 127.0.0.1'
> server*CLI>
> <-- SIP read from 81.23.228.150:5060:
> INVITE sip:31717110342 at 62.131.187.108:5060 SIP/2.0
> Max-Forwards: 10
> Record-Route: <sip:31717110342 at 81.23.228.150;ftag=as47419911;lr=on>
> Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0
> Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa
> From: "0031172651375"
> <sip:0031172651375 at voipgw01.budgetphone.nl>;tag=as47419911
> To: <sip:31717110342 at budgetphone.nl>
> Contact: <sip:0031172651375 at 212.203.28.2>
> Call-ID: 3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Sun, 10 Jul 2005 16:37:54 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 345
>  
> v=0
> o=root 26318 26318 IN IP4 212.203.28.2
> s=session
> c=IN IP4 81.23.228.139
> t=0 0
> m=audio 36634 RTP/AVP 3 18 5 0 97 110 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:18 G729/8000
> a=rtpmap:5 DVI4/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:110 speex/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>  
> --- (15 headers 15 lines)---
> Using INVITE request as basis request -
> 3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl
> Sending to 81.23.228.150 : 5060 (NAT)
> Found peer '31717110342'
> Reliably Transmitting (NAT) to 81.23.228.150:5060:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
>
81.23.228.150;branch=z9hG4bK7842.b2602997.0;received=81.23.228.150;rport=506
> 0
> Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa
> From: "0031172651375"
> <sip:0031172651375 at voipgw01.budgetphone.nl>;tag=as47419911
> To: <sip:31717110342 at budgetphone.nl>;tag=as3f35655f
> Call-ID: 3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: <sip:31717110342 at 192.168.2.3>
> Proxy-Authenticate: Digest realm="asterisk", nonce="555b996d"
> Content-Length: 0
>  
>  
> ---
> Scheduling destruction of call
> '3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl' in 15000 ms
> server*CLI>
> <-- SIP read from 81.23.228.150:5060:
> ACK sip:31717110342 at 62.131.187.108:5060 SIP/2.0
> Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0
> From: "0031172651375"
> <sip:0031172651375 at voipgw01.budgetphone.nl>;tag=as47419911
> Call-ID: 3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl
> To: <sip:31717110342 at budgetphone.nl>;tag=as3f35655f
> CSeq: 102 ACK
> User-Agent: Sip EXpress router(0.8.14-5 (i386/linux))
> Content-Length: 0
>  
>  
> --- (8 headers 0 lines)---
> Destroying call '3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl'
> server*CLI>
>  
>  
>
>
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