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    • S
      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
      • G
        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 S 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

          S 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:108@192.168.1.6>") 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:108@192.168.1.6>") 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:100@192.168.1.4>") 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:100@192.168.1.4>") 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:108@192.168.1.6>") 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
                • S
                  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
                  • S
                    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:100@192.168.1.6;tag=YBmt5C-Jz
                        To: sip:108@192.168.1.6;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:104@172.23.32.1:21444 SIP/2.0
                        Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK6a2fd815
                        Max-Forwards: 70
                        From: “Unknown” sip:Unknown@192.168.1.6;tag=as42f31e5a
                        To: sip:104@172.23.32.1:21444
                        Contact: sip:Unknown@192.168.1.6:5060
                        Call-ID: 151fa20a357cebe80b4e970c23c3d683@192.168.1.6:5060
                        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:108@192.168.1.6 SIP/2.0
                        Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK.-1bJueIwh;rport
                        From: sip:100@192.168.1.6;tag=iuL6gfJa9
                        To: sip:108@192.168.1.6
                        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:100@192.168.1.6;tag=iuL6gfJa9
                        To: sip:108@192.168.1.6;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:100@192.168.1.6") in new stack
                        Reliably Transmitting (NAT) to 192.168.1.4:55702:
                        MESSAGE sip:108@192.168.1.4:55702;ob SIP/2.0
                        Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK4539e4e2;rport
                        Max-Forwards: 70
                        From: “Unknown” sip:100@192.168.1.6;tag=as6b575a47
                        To: sip:108@192.168.1.4:55702;ob
                        Contact: sip:100@192.168.1.6:5060
                        Call-ID: 72d5a8261bc62f314a6c9c606e74336c@127.0.0.1:5060
                        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 ‘72d5a8261bc62f314a6c9c606e74336c@127.0.0.1 :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: 72d5a8261bc62f314a6c9c606e74336c@127.0.0.1:5060
                        From: “Unknown” sip:100@192.168.1.6;tag=as6b575a47
                        To: sip:108@192.168.1.4;ob;tag=z9hG4bK4539e4e2
                        CSeq: 102 MESSAGE
                        Content-Length: 0

                        <------------->
                        — (7 headers 0 lines) —
                        Really destroying SIP dialog ‘72d5a8261bc62f314a6c9c606e74336c@127.0.0.1:5060’ M ethod: MESSAGE
                        Retransmitting #3 (no NAT) to 172.23.32.1:21444:
                        OPTIONS sip:104@172.23.32.1:21444 SIP/2.0
                        Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK6a2fd815
                        Max-Forwards: 70
                        From: “Unknown” sip:Unknown@192.168.1.6;tag=as42f31e5a
                        To: sip:104@172.23.32.1:21444
                        Contact: sip:Unknown@192.168.1.6:5060
                        Call-ID: 151fa20a357cebe80b4e970c23c3d683@192.168.1.6:5060
                        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:104@172.23.32.1:21444 SIP/2.0
                        Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK6a2fd815
                        Max-Forwards: 70
                        From: “Unknown” sip:Unknown@192.168.1.6;tag=as42f31e5a
                        To: sip:104@172.23.32.1:21444
                        Contact: sip:Unknown@192.168.1.6:5060
                        Call-ID: 151fa20a357cebe80b4e970c23c3d683@192.168.1.6:5060
                        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 ‘151fa20a357cebe80b4e970c23c3d683@192.168.1.6:5060’ Method: OPTIONS

                        <— SIP read from UDP:192.168.1.4:55702 —>
                        SUBSCRIBE sip:101@192.168.1.6:5060 SIP/2.0
                        Via: SIP/2.0/UDP 192.168.1.4:55702;rport;branch=z9hG4bKPj190dffbab9cb4d8f9dab72d ebfddc7b6
                        Max-Forwards: 70
                        From: sip:108@192.168.1.6;tag=48c287b1b6414c2183cd5ce3671d4ae0
                        To: sip:101@192.168.1.6;tag=as26020ce9
                        Contact: sip:108@192.168.1.4:55702;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:108@192.168.1.6;tag=48c287b1b6414c2183cd5ce3671d4ae0
                        To: sip:101@192.168.1.6;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:101@192.168.1.6:5060 SIP/2.0
                        Via: SIP/2.0/UDP 192.168.1.4:55702;rport;branch=z9hG4bKPj81bd0c4026d5430da9d76c7 4b36b25c7
                        Max-Forwards: 70
                        From: sip:108@192.168.1.6;tag=8c3eda753e0845bf8ebfa94c8884d64d
                        To: sip:101@192.168.1.6
                        Contact: sip:108@192.168.1.4:55702;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:108@192.168.1.4:55702;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:108@192.168.1.6;tag=8c3eda753e0845bf8ebfa94c8884d64d
                        To: sip:101@192.168.1.6;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:101@192.168.1.6:5060 SIP/2.0
                          Via: SIP/2.0/UDP 192.168.1.4:55702;rport;branch=z9hG4bKPj127792c779874087b9e3965 ce1e390c2
                          Max-Forwards: 70
                          From: sip:108@192.168.1.6;tag=8c3eda753e0845bf8ebfa94c8884d64d
                          To: sip:101@192.168.1.6
                          Contact: sip:108@192.168.1.4:55702;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:101@192.168.1.6:5060”, 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:108@192.168.1.6;tag=8c3eda753e0845bf8ebfa94c8884d64d
                          To: sip:101@192.168.1.6;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:101@192.168.1.6:5060;expires=600
                          Content-Length: 0

                          <------------>
                          Reliably Transmitting (NAT) to 192.168.1.4:55702:
                          NOTIFY sip:108@192.168.1.4:55702;ob SIP/2.0
                          Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK12c62c7c;rport
                          Max-Forwards: 70
                          From: sip:101@192.168.1.6;tag=as47f06dc0
                          To: sip:108@192.168.1.6;tag=8c3eda753e0845bf8ebfa94c8884d64d
                          Contact: sip:101@192.168.1.6:5060
                          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:101@192.168.1.6
                          closed

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                            ranahashem @ranahashem
                            last edited by

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                              ranahashem @ranahashem
                              last edited by

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