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    • 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

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

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                        • R
                          ranahashem @ranahashem
                          last edited by

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