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    voip and freepbx / asterisk

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    • kiokomanK
      kiokoman LAYER 8
      last edited by kiokoman

      maybe there is some expert here that can give me some hints
      sorry this has nothing to do with pfSense

      so I have configured a voip trunk to a FreePBX behind pfsense,
      I can make calls but I can't receive

      this voip use the TEL URI scheme so I'm using chan_sip

      if you have any suggestions on where I should look to make it work thanks

      <------------->
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header  0 [ 53]: INVITE sip:0039XXXXXXXX@217.133.80.167:5060 SIP/2.0
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header  1 [ 74]: Via: SIP/2.0/UDP 94.32.130.112:5060;branch=z9hG4bKda2812d8d8cf6f1ecb5ctaN0
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header  2 [ 49]: To: <sip:0039XXXXXXXX@94.32.130.112;user=phone>
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header  3 [ 77]: From: <tel:0039XXXXXXXXX>;tag=ztesip-OuI1Xd1pFvXrhAVYtGwr5x*2-2-20481*cdej.2
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header  4 [ 37]: Call-ID: bM4IWJIsW1O_Hsrq2debc@zteims
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header  5 [ 17]: CSeq: 1000 INVITE
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header  6 [ 16]: Max-Forwards: 63
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header  7 [ 74]: Contact: <sip:0039XXXXXXX@94.32.130.112:5060;zte-did=2-2-20481-3760-12>
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header  8 [ 69]: P-Called-Party-ID: "?"<sip:0039XXXXXXX@ims.tiscali.net;user=phone>
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header  9 [ 23]: Supported: 100rel,timer
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header 10 [ 24]: P-Early-Media: supported
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header 11 [ 37]: P-Asserted-Identity: <tel:373XXXXXX>
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header 12 [ 57]: X-ZTE-Cookie: 7zs4rm21;id=8FD5715F48A9390000@10.253.2.205
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header 13 [ 88]: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,MESSAGE
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header 14 [ 10]: Min-SE: 90
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header 15 [ 35]: Session-Expires: 1800;refresher=uac
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header 16 [ 29]: Content-Type: application/sdp
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header 17 [ 19]: Content-Length: 235
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header 18 [ 28]: Content-Disposition: session
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:  Header 19 [  0]:
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:    Body  0 [  3]: v=0
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:    Body  1 [ 52]: o=UTSTARCOM 894511378 779705889 IN IP4 94.32.130.114
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:    Body  2 [  3]: s=-
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:    Body  3 [ 22]: c=IN IP4 94.32.130.114
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:    Body  4 [  5]: t=0 0
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:    Body  5 [ 30]: m=audio 37904 RTP/AVP 8 18 100
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:    Body  6 [ 10]: a=sendrecv
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:    Body  7 [ 21]: a=rtpmap:18 G729/8000
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:    Body  8 [ 19]: a=fmtp:18 annexb=no
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:    Body  9 [ 33]: a=rtpmap:100 telephone-event/8000
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c:    Body 10 [ 15]: a=fmtp:100 0-15
[2020-09-28 12:22:40] VERBOSE[1901] chan_sip.c: --- (19 headers 11 lines) ---
[2020-09-28 12:22:40] DEBUG[1901] acl.c: For destination '94.32.130.112', our source address is '192.168.50.1'.
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c: Target address 94.32.130.112:5060 is not local, substituting externaddr
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c: Setting AST_TRANSPORT_UDP with address 217.133.80.167:5060
[2020-09-28 12:22:40] VERBOSE[1901] chan_sip.c: Sending to 94.32.130.112:5060 (no NAT)
[2020-09-28 12:22:40] DEBUG[1901] chan_sip.c: Allocating new SIP dialog for bM4IWJIsW1O_Hsrq2debc@zteims - INVITE (No RTP)
[2020-09-28 12:22:40] DEBUG[1901][C-0000001d] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[2020-09-28 12:22:40] DEBUG[1901][C-0000001d] sip/reqresp_parser.c: Begin: parsing SIP "Supported: 100rel,timer"
[2020-09-28 12:22:40] DEBUG[1901][C-0000001d] sip/reqresp_parser.c: Found SIP option: -100rel-
[2020-09-28 12:22:40] DEBUG[1901][C-0000001d] sip/reqresp_parser.c: Matched SIP option: 100rel
[2020-09-28 12:22:40] DEBUG[1901][C-0000001d] sip/reqresp_parser.c: Found SIP option: -timer-
[2020-09-28 12:22:40] DEBUG[1901][C-0000001d] sip/reqresp_parser.c: Matched SIP option: timer
[2020-09-28 12:22:40] VERBOSE[1901][C-0000001d] chan_sip.c: Sending to 94.32.130.112:5060 (no NAT)
[2020-09-28 12:22:40] DEBUG[1901][C-0000001d] chan_sip.c: Initializing initreq for method INVITE - callid bM4IWJIsW1O_Hsrq2debc@zteims
[2020-09-28 12:22:40] VERBOSE[1901][C-0000001d] chan_sip.c: Using INVITE request as basis request - bM4IWJIsW1O_Hsrq2debc@zteims
[2020-09-28 12:22:40] DEBUG[1901][C-0000001d] sip/reqresp_parser.c: No RFC 3966 global number or context found in '0039XXXXXXX'; returning local number anyway
[2020-09-28 12:22:40] NOTICE[1901][C-0000001d] chan_sip.c: From address missing 'sip:', using it anyway
[2020-09-28 12:22:40] ERROR[1901][C-0000001d] chan_sip.c: Empty domain name in FROM header
[2020-09-28 12:22:40] NOTICE[1901][C-0000001d] chan_sip.c: Unexpected error for device <tel:0039XXXXXXXXX>;tag=ztesip-OuI1Xd1pFvXrhAVYtGwr5x*2-2-20481*cdej.2 for INVITE, code = -100
[2020-09-28 12:22:40] VERBOSE[1901][C-0000001d] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 94.32.130.112:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 94.32.130.112:5060;branch=z9hG4bKda2812d8d8cf6f1ecb5ctaN0;received=94.32.130.112

      ̿' ̿'\̵͇̿̿\з=(◕_◕)=ε/̵͇̿̿/'̿'̿ ̿
      Please do not use chat/PM to ask for help
      we must focus on silencing this @guest character. we must make up lies and alter the copyrights !
      Don't forget to Upvote with the 👍 button for any post you find to be helpful.

      1 Reply Last reply Reply Quote 0
      • stephenw10S
        stephenw10 Netgate Administrator
        last edited by

        You see that same 'From address' error when calling from any external client?

        1 Reply Last reply Reply Quote 0
        • kiokomanK
          kiokoman LAYER 8
          last edited by kiokoman

          no error when calling, anyway
          I found the problem in the meantime
          this Isp is not respecting the RFC

          sip/reqresp_parser.c: No RFC 3966 global number or context found

          this is because instead of using +39 it's using 0039.
          the parser expects to find a '+'
          the solution for me was to recompile Asterisk after modifying reqresp_parser.c replacing + with 0
          🙄

          ̿' ̿'\̵͇̿̿\з=(◕_◕)=ε/̵͇̿̿/'̿'̿ ̿
          Please do not use chat/PM to ask for help
          we must focus on silencing this @guest character. we must make up lies and alter the copyrights !
          Don't forget to Upvote with the 👍 button for any post you find to be helpful.

          1 Reply Last reply Reply Quote 1
          • stephenw10S
            stephenw10 Netgate Administrator
            last edited by

            Ah, nice catch. Nice fix! 👍

            ISPs. 🙄

            1 Reply Last reply Reply Quote 0
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