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Incoming call


Marcel V
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I have a sucessfull outboud call with PBXNSIP 3, but the inboud call does not work

 

The error is:

 

[8] 2008/12/04 18:06:42: Could not find a trunk (2 trunks)

[8] 2008/12/04 18:06:42: Using outbound proxy sip:217.71.123.3:5060;transport=udp because UDP packet source did not match the via header

[9] 2008/12/04 18:06:42: Resolve 69885: udp 217.71.123.3 5060

[6] 2008/12/04 18:06:42: Sending RTP for 1792407011@217.71.123.10#8b444b73ec to 217.71.123.11:14270

[5] 2008/12/04 18:06:42: Received incoming call without trunk information and user has not been found

 

The trunk is setup for in and outboud.

 

What do i have to change in the truck setup to support inbound calls ?

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I have a sucessfull outboud call with PBXNSIP 3, but the inboud call does not work

 

The error is:

 

[8] 2008/12/04 18:06:42: Could not find a trunk (2 trunks)

[8] 2008/12/04 18:06:42: Using outbound proxy sip:217.71.123.3:5060;transport=udp because UDP packet source did not match the via header

[9] 2008/12/04 18:06:42: Resolve 69885: udp 217.71.123.3 5060

[6] 2008/12/04 18:06:42: Sending RTP for 1792407011@217.71.123.10#8b444b73ec to 217.71.123.11:14270

[5] 2008/12/04 18:06:42: Received incoming call without trunk information and user has not been found

 

The trunk is setup for in and outboud.

 

What do i have to change in the truck setup to support inbound calls ?

 

What do you have "Send call to extension:" set to?

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Try adding your phone number as as alias on extension 40.

Hi same case

 

[8] 2008/12/12 17:22:39: Could not find a trunk (1 trunks)

[8] 2008/12/12 17:22:39: Using outbound proxy sip:217.71.123.3:5060;transport=udp because UDP packet source did not match the via header

[9] 2008/12/12 17:22:39: Resolve 23: udp 217.71.123.3 5060

[6] 2008/12/12 17:22:39: Sending RTP for 151170848@217.71.123.10#fab816e05e to 217.71.123.12:12950

[5] 2008/12/12 17:22:39: Received incoming call without trunk information and user has not been found

 

I found the trunk problem, was 2 now 1 as you can see.

 

But i do not understand why it does not sent the incoming calls to extension 40

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Hi same case

 

[8] 2008/12/12 17:22:39: Could not find a trunk (1 trunks)

[8] 2008/12/12 17:22:39: Using outbound proxy sip:217.71.123.3:5060;transport=udp because UDP packet source did not match the via header

[9] 2008/12/12 17:22:39: Resolve 23: udp 217.71.123.3 5060

[6] 2008/12/12 17:22:39: Sending RTP for 151170848@217.71.123.10#fab816e05e to 217.71.123.12:12950

[5] 2008/12/12 17:22:39: Received incoming call without trunk information and user has not been found

 

I found the trunk problem, was 2 now 1 as you can see.

 

But i do not understand why it does not sent the incoming calls to extension 40

Something must be wrong in there, you can call me I am sure I can resolve it, with a quick phone call. 888.IPPBX.US

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Something must be wrong in there, you can call me I am sure I can resolve it, with a quick phone call. 888.IPPBX.US

HI,

 

Thanks for the offer,

 

I changed SIP provider and now it works, BUT the outbounds now does not work.

 

Here is my log:

 

[9] 2008/12/16 15:37:19: UDP: Opening socket on port 49430

[9] 2008/12/16 15:37:19: UDP: Opening socket on port 49431

[8] 2008/12/16 15:37:19: Could not find a trunk (2 trunks)

[8] 2008/12/16 15:37:19: Using outbound proxy sip:195.73.157.235:31268;transport=udp because UDP packet source did not match the via header

[9] 2008/12/16 15:37:19: Resolve 3369: udp 195.73.157.235 31268

[9] 2008/12/16 15:37:19: Resolve 3370: udp 195.73.157.235 31268

[8] 2008/12/16 15:37:20: Tagging request with existing tag

[6] 2008/12/16 15:37:20: Sending RTP for NTg4NDI2MjNkZjUxNTBlYjZlOTc5ZjRiNjQwNGEwNjk.#75d2c00508 to 192.168.3.221:33392

[9] 2008/12/16 15:37:20: Resolve 3371: udp 195.73.157.235 31268

[9] 2008/12/16 15:37:20: Dialplan: Evaluating !^0([0-9]*)@.*!sip:0031\1@\r;user=phone!i against 0345549745@195.73.157.237

[5] 2008/12/16 15:37:20: Dialplan Standaard: Match 0345549745@195.73.157.237 to <sip:0031345549745@sip.goandcall.com;user=phone> on trunk Xeloq

[8] 2008/12/16 15:37:20: Play audio_moh/noise.wav

[9] 2008/12/16 15:37:20: UDP: Opening socket on port 60242

[9] 2008/12/16 15:37:20: UDP: Opening socket on port 60243

[9] 2008/12/16 15:37:20: Resolve 3372: url sip:sip.goandcall.com

[9] 2008/12/16 15:37:20: Resolve 3372: naptr sip.goandcall.com

[9] 2008/12/16 15:37:20: Resolve 3372: srv tls _sips._tcp.sip.goandcall.com

[9] 2008/12/16 15:37:20: Resolve 3372: srv tcp _sip._tcp.sip.goandcall.com

[9] 2008/12/16 15:37:20: Resolve 3372: srv udp _sip._udp.sip.goandcall.com

[9] 2008/12/16 15:37:20: Resolve 3372: a udp sip.goandcall.com 5060

[9] 2008/12/16 15:37:20: Resolve 3372: udp 81.26.212.150 5060

[9] 2008/12/16 15:37:20: Resolve 3373: udp 195.73.157.235 31268

[8] 2008/12/16 15:37:20: Answer challenge with username 725319

[9] 2008/12/16 15:37:20: Resolve 3374: udp 81.26.212.150 5060 udp:1

[9] 2008/12/16 15:37:20: Resolve 3375: udp 81.26.212.150 5060 udp:1

[9] 2008/12/16 15:37:20: Message repetition, packet dropped

[6] 2008/12/16 15:37:20: Sending RTP for NTg4NDI2MjNkZjUxNTBlYjZlOTc5ZjRiNjQwNGEwNjk.#75d2c00508 to 195.73.157.235:33392

[7] 2008/12/16 15:37:20: Call 6b86f62f@pbx#3710: Clear last INVITE

[9] 2008/12/16 15:37:20: Resolve 3376: url sip:sip.goandcall.com

[9] 2008/12/16 15:37:20: Resolve 3376: naptr sip.goandcall.com

[9] 2008/12/16 15:37:20: Resolve 3376: srv tls _sips._tcp.sip.goandcall.com

[9] 2008/12/16 15:37:20: Resolve 3376: srv tcp _sip._tcp.sip.goandcall.com

[9] 2008/12/16 15:37:20: Resolve 3376: srv udp _sip._udp.sip.goandcall.com

[9] 2008/12/16 15:37:20: Resolve 3376: a udp sip.goandcall.com 5060

[9] 2008/12/16 15:37:20: Resolve 3376: udp 81.26.212.150 5060

[5] 2008/12/16 15:37:20: INVITE Response: Terminate 6b86f62f@pbx

[7] 2008/12/16 15:37:20: Other Ports: 1

[7] 2008/12/16 15:37:20: Call Port: NTg4NDI2MjNkZjUxNTBlYjZlOTc5ZjRiNjQwNGEwNjk.#75d2c00508

[9] 2008/12/16 15:37:20: Resolve 3377: udp 195.73.157.235 31268

[9] 2008/12/16 15:37:23: Resolve 3378: udp 195.73.157.235 61639

[9] 2008/12/16 15:37:23: Message repetition, packet dropped

 

Appreciate the feedback,

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Can you turn SIP packet logging on? You should see a packet with the header "INVITE sip:xxx SIP/2.0" (see http://wiki.pbxnsip.com/index.php/Log_Setup on how to do that). Probably the SIP provider rejects the call.

 

I have xlite connected to PBXNSIP to call and changed this xlite client to connect directly to the sip provider instead of the PBXNSIP and this works !!

 

??

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I have attached the file with the log, i have tried to dial out twice. I have a dialplan attached as picture and my trunks.

 

The "GNC SS5 Proxy" replies with 603 Declined. I assume that is because the authentication information is in the P-Asserted-Identity header. Try using "Remote Party/Privacy Indication" in the trunk "No indication"

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