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    • R
      ranahashem
      last edited by

      Hello, I am running a Freepbx distro v15 and configure it for chan_sip sms between extensions using the below setup:
      freepbx 15 bistro
      asterisk 17
      centos 7
      …
      sip.config

      accept_outofcall_message = yes
outofcall_message_context = messages
auth_message_requests = no

      …
      extensions_custom.conf
      [astsms]

      ;Deliver to local 3-digit extension
exten => _XXX,1,MessageSend(sip:${EXTEN},"${CALLERID(name)}"${MESSAGE(from)})

      tail -f /var/log/asterisk/full

      [2021-06-18 20:50:23] VERBOSE[3478][C-00000002] pbx.c: Executing [s@func-apply-sipheaders:3] Set("SIP/100-00000001", "TECH=SIP") in new stack
[2021-06-18 20:50:23] VERBOSE[3478][C-00000002] pbx.c: Executing [s@func-apply-sipheaders:4] Set("SIP/100-00000001", "SIPHEADERKEYS=") in new stack
[2021-06-18 20:50:23] VERBOSE[3478][C-00000002] pbx.c: Executing [s@func-apply-sipheaders:5] While("SIP/100-00000001", "0") in new stack
[2021-06-18 20:50:23] VERBOSE[3478][C-00000002] app_while.c: Jumping to priority 13
[2021-06-18 20:50:23] VERBOSE[3478][C-00000002] pbx.c: Executing [s@func-apply-sipheaders:14] Return("SIP/100-00000001", "") in new stack
[2021-06-18 20:50:23] VERBOSE[3478][C-00000002] app_stack.c: Spawn extension (from-internal, 100, 1) exited non-zero on 'SIP/100-00000001'
[2021-06-18 20:50:23] VERBOSE[3478][C-00000002] app_stack.c: SIP/100-00000001 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2021-06-18 20:50:23] VERBOSE[3478][C-00000002] app_dial.c: Called SIP/100
[2021-06-18 20:50:24] VERBOSE[3478][C-00000002] app_dial.c: SIP/100-00000001 is ringing
[2021-06-18 20:50:26] VERBOSE[3478][C-00000002] app_dial.c: SIP/100-00000001 answered SIP/108-00000000
[2021-06-18 20:50:26] VERBOSE[3482][C-00000002] bridge_channel.c: Channel SIP/100-00000001 joined 'simple_bridge' basic-bridge <abc10862-349a-48d1-b359-b5e8bde89e68>
[2021-06-18 20:50:26] VERBOSE[3478][C-00000002] bridge_channel.c: Channel SIP/108-00000000 joined 'simple_bridge' basic-bridge <abc10862-349a-48d1-b359-b5e8bde89e68>
[2021-06-18 20:50:32] VERBOSE[1345][C-00000001] pbx.c: Executing [108@astsms:1] MessageSend("Message/ast_msg_queue", "sip:108,""<sip:[email protected]>") in new stack
[2021-06-18 20:50:32] VERBOSE[1345][C-00000001] pbx.c: Executing [108@astsms:2] Hangup("Message/ast_msg_queue", "") in new stack
[2021-06-18 20:50:32] VERBOSE[1345][C-00000001] pbx.c: Spawn extension (astsms, 108, 2) exited non-zero on 'Message/ast_msg_queue'
[2021-06-18 20:50:35] VERBOSE[1345][C-00000001] pbx.c: Executing [100@astsms:1] MessageSend("Message/ast_msg_queue", "sip:100,""<sip:[email protected]>") in new stack
[2021-06-18 20:50:35] VERBOSE[1345][C-00000001] pbx.c: Executing [100@astsms:2] Hangup("Message/ast_msg_queue", "") in new stack
[2021-06-18 20:50:35] VERBOSE[1345][C-00000001] pbx.c: Spawn extension (astsms, 100, 2) exited non-zero on 'Message/ast_msg_queue'
[2021-06-18 20:50:37] VERBOSE[1345][C-00000001] pbx.c: Executing [100@astsms:1] MessageSend("Message/ast_msg_queue", "sip:100,""<sip:[email protected]>") in new stack
[2021-06-18 20:50:37] VERBOSE[1345][C-00000001] pbx.c: Executing [100@astsms:2] Hangup("Message/ast_msg_queue", "") in new stack
[2021-06-18 20:50:37] VERBOSE[1345][C-00000001] pbx.c: Spawn extension (astsms, 100, 2) exited non-zero on 'Message/ast_msg_queue'
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] bridge_channel.c: Channel SIP/108-00000000 left 'simple_bridge' basic-bridge <abc10862-349a-48d1-b359-b5e8bde89e68>
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] app_macro.c: Spawn extension (macro-dial-one, s, 56) exited non-zero on 'SIP/108-00000000' in macro 'dial-one'
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] app_macro.c: Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/108-00000000' in macro 'exten-vm'
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] pbx.c: Spawn extension (ext-local, 100, 3) exited non-zero on 'SIP/108-00000000'
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] pbx.c: Executing [h@ext-local:1] Macro("SIP/108-00000000", "hangupcall,") in new stack
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/108-00000000", "1?theend") in new stack
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2021-06-18 20:50:41] VERBOSE[3482][C-00000002] bridge_channel.c: Channel SIP/100-00000001 left 'simple_bridge' basic-bridge <abc10862-349a-48d1-b359-b5e8bde89e68>
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/108-00000000", "0?Set(CDR(recordingfile)=)") in new stack
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] pbx.c: Executing [s@macro-hangupcall:4] NoOp("SIP/108-00000000", "SIP/100-00000001 montior file= ") in new stack
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] pbx.c: Executing [s@macro-hangupcall:5] GotoIf("SIP/108-00000000", "1?skipagi") in new stack
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,7)
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] pbx.c: Executing [s@macro-hangupcall:7] Hangup("SIP/108-00000000", "") in new stack
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'SIP/108-00000000' in macro 'hangupcall'
[2021-06-18 20:50:41] VERBOSE[3478][C-00000002] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on 'SIP/108-00000000'
      scottalanmillerS 1 Reply Last reply Reply Quote 0
      • scottalanmillerS
        scottalanmiller @ranahashem
        last edited by

        @ranahashem Have not tested with FreePBX. But I know that on VitalPBX, it works out of the box. It's also super annoying to use, text messages on desk phones is super awkward. But it works.

        R 2 Replies Last reply Reply Quote 0
        • gjacobseG
          gjacobse
          last edited by

          Uh, text on phones? Why not just email?

          That said, e911 will be doing text, some cases are better to communicate that way,..

          R scottalanmillerS 2 Replies Last reply Reply Quote 0
          • R
            ranahashem @scottalanmiller
            last edited by

            @scottalanmiller This is my project I can not switch or change it but all that says this script works well. Can I change the SoftPhone phone type because Linphone on my laptop sends messages to CSIP on the same laptop but does not accept the messages from CSIP or LINEPHONE on your smartphone
            What should I do?
            i show u "sip set debug on" or u can Give me any other secript sip messges

            scottalanmillerS 1 Reply Last reply Reply Quote 0
            • R
              ranahashem @gjacobse
              last edited by

              @gjacobse This is my project I can not switch or change it but all that says this script works well. Can I change the SoftPhone phone type because Linphone on my laptop sends messages to CSIP on the same laptop but does not accept the messages from CSIP or LINEPHONE on your smartphone
              What should I do?
              i show u "sip set debug on" or u can Give me any other secript sip messges

              1 Reply Last reply Reply Quote 0
              • R
                ranahashem @scottalanmiller
                last edited by

                @scottalanmiller

                 Executing [s@macro-dial-one:56] Dial("SIP/108-00000000", "SIP/100,,HhTtrb(func-apply-sipheaders^s^1)") in new stack
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- SIP/100-00000001 Internal Gosub(func-apply-sipheaders,s,1) start
    -- Executing [s@func-apply-sipheaders:1] ExecIf("SIP/100-00000001", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
    -- Executing [s@func-apply-sipheaders:2] NoOp("SIP/100-00000001", "Applying SIP Headers to channel SIP/100-00000001") in new stack
    -- Executing [s@func-apply-sipheaders:3] Set("SIP/100-00000001", "TECH=SIP") in new stack
    -- Executing [s@func-apply-sipheaders:4] Set("SIP/100-00000001", "SIPHEADERKEYS=") in new stack
    -- Executing [s@func-apply-sipheaders:5] While("SIP/100-00000001", "0") in new stack
    -- Jumping to priority 13
    -- Executing [s@func-apply-sipheaders:14] Return("SIP/100-00000001", "") in new stack
  == Spawn extension (from-internal, 100, 1) exited non-zero on 'SIP/100-00000001'
    -- SIP/100-00000001 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
    -- Called SIP/100
    -- SIP/100-00000001 is ringing
       > 0x7f37600408e0 -- Strict RTP learning after remote address set to: 192.168.1.4:7078
    -- SIP/100-00000001 answered SIP/108-00000000
       > 0x7f376c0446a0 -- Strict RTP learning after remote address set to: 192.168.1.4:4000
    -- Channel SIP/100-00000001 joined 'simple_bridge' basic-bridge <337b881c-a182-4e35-8775-658b3ddee0aa>
    -- Channel SIP/108-00000000 joined 'simple_bridge' basic-bridge <337b881c-a182-4e35-8775-658b3ddee0aa>
       > 0x7f376c0446a0 -- Strict RTP switching to RTP target address 192.168.1.4:4000 as source
       > 0x7f37600408e0 -- Strict RTP switching to RTP target address 192.168.1.4:7078 as source
       > 0x7f37600408e0 -- Strict RTP learning complete - Locking on source address 192.168.1.4:7078
       > 0x7f376c0446a0 -- Strict RTP learning complete - Locking on source address 192.168.1.4:4000
    -- Executing [100@astsms:1] MessageSend("Message/ast_msg_queue", "sip:100,""<sip:[email protected]>") in new stack
    -- Executing [100@astsms:2] Hangup("Message/ast_msg_queue", "") in new stack
  == Spawn extension (astsms, 100, 2) exited non-zero on 'Message/ast_msg_queue'
    -- Executing [100@astsms:1] MessageSend("Message/ast_msg_queue", "sip:100,""<sip:[email protected]>") in new stack
    -- Executing [100@astsms:2] Hangup("Message/ast_msg_queue", "") in new stack
  == Spawn extension (astsms, 100, 2) exited non-zero on 'Message/ast_msg_queue'
    -- Executing [108@astsms:1] MessageSend("Message/ast_msg_queue", "sip:108,""<sip:[email protected]>") in new stack
    -- Executing [108@astsms:2] Hangup("Message/ast_msg_queue", "") in new stack
  == Spawn extension (astsms, 108, 2) exited non-zero on 'Message/ast_msg_queue'
    -- Executing [108@astsms:1] MessageSend("Message/ast_msg_queue", "sip:108,""<sip:[email protected]>") in new stack
    -- Executing [108@astsms:2] Hangup("Message/ast_msg_queue", "") in new stack
  == Spawn extension (astsms, 108, 2) exited non-zero on 'Message/ast_msg_queue'
    -- Executing [100@astsms:1] MessageSend("Message/ast_msg_queue", "sip:100,""<sip:[email protected]>") in new stack
    -- Executing [100@astsms:2] Hangup("Message/ast_msg_queue", "") in new stack
  == Spawn extension (astsms, 100, 2) exited non-zero on 'Message/ast_msg_queue'
freepbx*CLI> sip set debug off
``
                R 1 Reply Last reply Reply Quote 0
                • R
                  ranahashem @ranahashem
                  last edited by

                  @ranahashem sip set debug on between linphone and csip on same laptop linphone receive massage but csip not receive !!!!!

                  1 Reply Last reply Reply Quote 0
                  • scottalanmillerS
                    scottalanmiller @gjacobse
                    last edited by

                    @gjacobse said in chat not working:

                    Uh, text on phones? Why not just email?

                    That said, e911 will be doing text, some cases are better to communicate that way,..

                    SIP on phones generally doesn't leave the PBX. This, we assume from his testing, is extension to extension to replace a LAN texting solution like the 1990s.

                    1 Reply Last reply Reply Quote 0
                    • scottalanmillerS
                      scottalanmiller @ranahashem
                      last edited by

                      @ranahashem said in chat not working:

                      @scottalanmiller This is my project I can not switch or change it but all that says this script works well. Can I change the SoftPhone phone type because Linphone on my laptop sends messages to CSIP on the same laptop but does not accept the messages from CSIP or LINEPHONE on your smartphone
                      What should I do?
                      i show u "sip set debug on" or u can Give me any other secript sip messges

                      Can you test with two laptops and eliminate the extra pieces?

                      It might be all your endpoints, not the PBX, causing issues.

                      R 2 Replies Last reply Reply Quote 0
                      • R
                        ranahashem @scottalanmiller
                        last edited by

                        1.png

                        2.png

                        still on lin phone on laptop i see linphone do not accept any messages from csip messages-sends on smart phone
                        @scottalanmiller

                        1 Reply Last reply Reply Quote 0
                        • R
                          ranahashem @scottalanmiller
                          last edited by

                          @scottalanmiller ```
                          <— Transmitting (no NAT) to 192.168.1.4:5060 —>
                          SIP/2.0 415 Unsupported Media Type
                          Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK.KzEHfsnEU;received=192.168.1.4; rport=5060
                          From: sip:[email protected];tag=YBmt5C-Jz
                          To: sip:[email protected];tag=as10c11416
                          Call-ID: 4n1fgfjS9O
                          CSeq: 20 MESSAGE
                          Server: FPBX-15.0.17.34(17.9.3)
                          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
                          Supported: replaces, timer
                          Content-Length: 0

                          <------------>
                          Scheduling destruction of SIP dialog ‘4n1fgfjS9O’ in 32000 ms (Method: MESSAGE)
                          Retransmitting #2 (no NAT) to 172.23.32.1:21444:
                          OPTIONS sip:[email protected]:21444 SIP/2.0
                          Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK6a2fd815
                          Max-Forwards: 70
                          From: “Unknown” sip:[email protected];tag=as42f31e5a
                          To: sip:[email protected]:21444
                          Contact: sip:[email protected]:5060
                          Call-ID: [email protected]:5060
                          CSeq: 102 OPTIONS
                          User-Agent: FPBX-15.0.17.34(17.9.3)
                          Date: Sat, 19 Jun 2021 12:17:06 GMT
                          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
                          Supported: replaces, timer
                          Content-Length: 0

                          <— SIP read from UDP:192.168.1.4:5060 —>

                          <------------->

                          <— SIP read from UDP:192.168.1.4:5060 —>
                          MESSAGE sip:[email protected] SIP/2.0
                          Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK.-1bJueIwh;rport
                          From: sip:[email protected];tag=iuL6gfJa9
                          To: sip:[email protected]
                          CSeq: 20 MESSAGE
                          Call-ID: jHRSBGXOJY
                          Max-Forwards: 70
                          Supported: replaces, outbound, gruu
                          Date: Sat, 19 Jun 2021 12:17:08 GMT
                          Content-Type: text/plain
                          Content-Length: 3
                          User-Agent: Linphone Desktop/4.2.5 (Windows 10 Version 2009, Qt 5.14.2) Linphone Core/4.4.19

                          yyy
                          <------------->
                          — (12 headers 1 lines) —
                          Sending to 192.168.1.4:5060 (no NAT)
                          Receiving message!
                          Looking for 108 in astsms (domain 192.168.1.6)

                          <— Transmitting (no NAT) to 192.168.1.4:5060 —>
                          SIP/2.0 202 Accepted
                          Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK.-1bJueIwh;received=192.168.1.4; rport=5060
                          From: sip:[email protected];tag=iuL6gfJa9
                          To: sip:[email protected];tag=as17281fb4
                          Call-ID: jHRSBGXOJY
                          CSeq: 20 MESSAGE
                          Server: FPBX-15.0.17.34(17.9.3)
                          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
                          Supported: replaces, timer
                          Content-Length: 0

                          <------------>
                          Scheduling destruction of SIP dialog ‘jHRSBGXOJY’ in 32000 ms (Method: MESSAGE)
                          – Executing [108@astsms:1] MessageSend(“Message/ast_msg_queue”, “sip:108,”" sip:[email protected]") in new stack
                          Reliably Transmitting (NAT) to 192.168.1.4:55702:
                          MESSAGE sip:[email protected]:55702;ob SIP/2.0
                          Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK4539e4e2;rport
                          Max-Forwards: 70
                          From: “Unknown” sip:[email protected];tag=as6b575a47
                          To: sip:[email protected]:55702;ob
                          Contact: sip:[email protected]:5060
                          Call-ID: [email protected]:5060
                          CSeq: 102 MESSAGE
                          User-Agent: FPBX-15.0.17.34(17.9.3)
                          Content-Type: text/plain;charset=UTF-8
                          Content-Length: 3

                          yyy
                          Scheduling destruction of SIP dialog ‘[email protected] :5060’ in 6400 ms (Method: MESSAGE)
                          – Executing [108@astsms:2] NoOp(“Message/ast_msg_queue”, “Send status is SU CCESS”) in new stack
                          – Auto fallthrough, channel ‘Message/ast_msg_queue’ status is ‘UNKNOWN’

                          <— SIP read from UDP:192.168.1.4:55702 —>
                          SIP/2.0 200 OK
                          Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=192.168.1.6;branch=z9hG4bK 4539e4e2
                          Call-ID: [email protected]:5060
                          From: “Unknown” sip:[email protected];tag=as6b575a47
                          To: sip:[email protected];ob;tag=z9hG4bK4539e4e2
                          CSeq: 102 MESSAGE
                          Content-Length: 0

                          <------------->
                          — (7 headers 0 lines) —
                          Really destroying SIP dialog ‘[email protected]:5060’ M ethod: MESSAGE
                          Retransmitting #3 (no NAT) to 172.23.32.1:21444:
                          OPTIONS sip:[email protected]:21444 SIP/2.0
                          Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK6a2fd815
                          Max-Forwards: 70
                          From: “Unknown” sip:[email protected];tag=as42f31e5a
                          To: sip:[email protected]:21444
                          Contact: sip:[email protected]:5060
                          Call-ID: [email protected]:5060
                          CSeq: 102 OPTIONS
                          User-Agent: FPBX-15.0.17.34(17.9.3)
                          Date: Sat, 19 Jun 2021 12:17:06 GMT
                          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
                          Supported: replaces, timer
                          Content-Length: 0

                          Retransmitting #4 (no NAT) to 172.23.32.1:21444:
                          OPTIONS sip:[email protected]:21444 SIP/2.0
                          Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK6a2fd815
                          Max-Forwards: 70
                          From: “Unknown” sip:[email protected];tag=as42f31e5a
                          To: sip:[email protected]:21444
                          Contact: sip:[email protected]:5060
                          Call-ID: [email protected]:5060
                          CSeq: 102 OPTIONS
                          User-Agent: FPBX-15.0.17.34(17.9.3)
                          Date: Sat, 19 Jun 2021 12:17:06 GMT
                          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
                          Supported: replaces, timer
                          Content-Length: 0

                          Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

                          <— SIP read from UDP:192.168.1.4:55702 —>
                          SUBSCRIBE sip:[email protected]:5060 SIP/2.0
                          Via: SIP/2.0/UDP 192.168.1.4:55702;rport;branch=z9hG4bKPj190dffbab9cb4d8f9dab72d ebfddc7b6
                          Max-Forwards: 70
                          From: sip:[email protected];tag=48c287b1b6414c2183cd5ce3671d4ae0
                          To: sip:[email protected];tag=as26020ce9
                          Contact: sip:[email protected]:55702;ob
                          Call-ID: 11940a87d16849fdb0b55cc715939098
                          CSeq: 3992 SUBSCRIBE
                          Event: presence
                          Expires: 600
                          Supported: replaces, 100rel, timer, norefersub
                          Accept: application/pidf+xml, application/xpidf+xml
                          Allow-Events: presence, message-summary, refer
                          Content-Length: 0

                          <------------->
                          — (14 headers 0 lines) —
                          Sending to 192.168.1.4:55702 (no NAT)

                          <— Transmitting (no NAT) to 192.168.1.4:55702 —>
                          SIP/2.0 481 Call/Transaction Does Not Exist
                          Via: SIP/2.0/UDP 192.168.1.4:55702;branch=z9hG4bKPj190dffbab9cb4d8f9dab72debfddc 7b6;received=192.168.1.4;rport=55702
                          From: sip:[email protected];tag=48c287b1b6414c2183cd5ce3671d4ae0
                          To: sip:[email protected];tag=as26020ce9
                          Call-ID: 11940a87d16849fdb0b55cc715939098
                          CSeq: 3992 SUBSCRIBE
                          Server: FPBX-15.0.17.34(17.9.3)
                          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
                          Supported: replaces, timer
                          Content-Length: 0

                          <------------>
                          Scheduling destruction of SIP dialog ‘11940a87d16849fdb0b55cc715939098’ in 32000 ms (Method: SUBSCRIBE)

                          <— SIP read from UDP:192.168.1.4:55702 —>
                          SUBSCRIBE sip:[email protected]:5060 SIP/2.0
                          Via: SIP/2.0/UDP 192.168.1.4:55702;rport;branch=z9hG4bKPj81bd0c4026d5430da9d76c7 4b36b25c7
                          Max-Forwards: 70
                          From: sip:[email protected];tag=8c3eda753e0845bf8ebfa94c8884d64d
                          To: sip:[email protected]
                          Contact: sip:[email protected]:55702;ob
                          Call-ID: 401fcf1687ec4601a3a3278d6227db07
                          CSeq: 12287 SUBSCRIBE
                          Event: presence
                          Expires: 600
                          Supported: replaces, 100rel, timer, norefersub
                          Accept: application/pidf+xml, application/xpidf+xml
                          Allow-Events: presence, message-summary, refer
                          User-Agent: MicroSIP/3.20.6
                          Content-Length: 0

                          <------------->
                          — (15 headers 0 lines) —
                          Sending to 192.168.1.4:55702 (no NAT)
                          Creating new subscription
                          Sending to 192.168.1.4:55702 (no NAT)
                          sip_route_dump: route/path hop: sip:[email protected]:55702;ob
                          Found peer ‘108’ for ‘108’ from 192.168.1.4:55702

                          <— Transmitting (NAT) to 192.168.1.4:55702 —>
                          SIP/2.0 401 Unauthorized
                          Via: SIP/2.0/UDP 192.168.1.4:55702;branch=z9hG4bKPj81bd0c4026d5430da9d76c74b36b2 5c7;received=192.168.1.4;rport=55702
                          From: sip:[email protected];tag=8c3eda753e0845bf8ebfa94c8884d64d
                          To: sip:[email protected];tag=as47f06dc0
                          Call-ID: 401fcf1687ec4601a3a3278d6227db07
                          CSeq: 12287 SUBSCRIBE
                          Server: FPBX-15.0.17.34(17.9.3)
                          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
                          Supported: replaces, timer
                          WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“3f0d59a1”
                          Content-Length: 0

                          R 1 Reply Last reply Reply Quote 0
                          • R
                            ranahashem @ranahashem
                            last edited by

                            @ranahashem

                            Scheduling destruction of SIP dialog ‘401fcf1687ec4601a3a3278d6227db07’ in 6400 ms (Method: SUBSCRIBE)

                            <— SIP read from UDP:192.168.1.4:55702 —>
                            SUBSCRIBE sip:[email protected]:5060 SIP/2.0
                            Via: SIP/2.0/UDP 192.168.1.4:55702;rport;branch=z9hG4bKPj127792c779874087b9e3965 ce1e390c2
                            Max-Forwards: 70
                            From: sip:[email protected];tag=8c3eda753e0845bf8ebfa94c8884d64d
                            To: sip:[email protected]
                            Contact: sip:[email protected]:55702;ob
                            Call-ID: 401fcf1687ec4601a3a3278d6227db07
                            CSeq: 12288 SUBSCRIBE
                            Event: presence
                            Expires: 600
                            Supported: replaces, 100rel, timer, norefersub
                            Accept: application/pidf+xml, application/xpidf+xml
                            Allow-Events: presence, message-summary, refer
                            User-Agent: MicroSIP/3.20.6
                            Authorization: Digest username=“108”, realm=“asterisk”, nonce=“3f0d59a1”, uri=“s ip:[email protected]:5060”, response=“003e71376f0034f4bcbbc47bd83a0e8a”, algorithm =MD5
                            Content-Length: 0

                            <------------->
                            — (16 headers 0 lines) —
                            Creating new subscription
                            Sending to 192.168.1.4:55702 (NAT)
                            Found peer ‘108’ for ‘108’ from 192.168.1.4:55702
                            Looking for 101 in from-internal (domain 192.168.1.6)
                            Scheduling destruction of SIP dialog ‘401fcf1687ec4601a3a3278d6227db07’ in 61000 0 ms (Method: SUBSCRIBE)

                            <— Transmitting (NAT) to 192.168.1.4:55702 —>
                            SIP/2.0 200 OK
                            Via: SIP/2.0/UDP 192.168.1.4:55702;branch=z9hG4bKPj127792c779874087b9e3965ce1e39 0c2;received=192.168.1.4;rport=55702
                            From: sip:[email protected];tag=8c3eda753e0845bf8ebfa94c8884d64d
                            To: sip:[email protected];tag=as47f06dc0
                            Call-ID: 401fcf1687ec4601a3a3278d6227db07
                            CSeq: 12288 SUBSCRIBE
                            Server: FPBX-15.0.17.34(17.9.3)
                            Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
                            Supported: replaces, timer
                            Expires: 600
                            Contact: sip:[email protected]:5060;expires=600
                            Content-Length: 0

                            <------------>
                            Reliably Transmitting (NAT) to 192.168.1.4:55702:
                            NOTIFY sip:[email protected]:55702;ob SIP/2.0
                            Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK12c62c7c;rport
                            Max-Forwards: 70
                            From: sip:[email protected];tag=as47f06dc0
                            To: sip:[email protected];tag=8c3eda753e0845bf8ebfa94c8884d64d
                            Contact: sip:[email protected]:5060
                            Call-ID: 401fcf1687ec4601a3a3278d6227db07
                            CSeq: 102 NOTIFY
                            User-Agent: FPBX-15.0.17.34(17.9.3)
                            Subscription-State: active
                            Event: presence
                            Content-Type: application/pidf+xml
                            Content-Length: 524

                            <?xml version="1.0" encoding="ISO-8859-1"?>

                            pp:person
                            ep:activitiesep:away/</ep:activities>
                            </pp:person>
                            Unavailable

                            sip:[email protected]
                            closed

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                            • R
                              ranahashem @ranahashem
                              last edited by

                              This post is deleted!
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                              • R
                                ranahashem @ranahashem
                                last edited by

                                This post is deleted!
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