ML
    • Recent
    • Categories
    • Tags
    • Popular
    • Users
    • Groups
    • Register
    • Login
    1. Topics
    2. ranahashem
    3. Posts
    R
    • Profile
    • Following 4
    • Followers 0
    • Topics 4
    • Posts 20
    • Best 1
    • Controversial 0
    • Groups 0

    Posts made by ranahashem

    • RE: chat not working

      @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

      posted in IT Discussion
      R
      ranahashem
    • RE: chat not working

      @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

      posted in IT Discussion
      R
      ranahashem
    • RE: chat not working

      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

      posted in IT Discussion
      R
      ranahashem
    • RE: chat not working

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

      posted in IT Discussion
      R
      ranahashem
    • RE: chat not working

      @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
``
      posted in IT Discussion
      R
      ranahashem
    • RE: chat not working

      @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

      posted in IT Discussion
      R
      ranahashem
    • RE: 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

      posted in IT Discussion
      R
      ranahashem
    • chat not working

      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'
      posted in IT Discussion
      R
      ranahashem
    • RE: Sms chat not working on freepbx with tow linephone softphone

      @ranahashem

      sip_general_custom.conf
      accept_outofcall_message=yes
      outofcall_message_context=dialplan_name
      auth_message_requests=yes
      …

      sip_general_additional.conf
      accept_outofcall_message=yes
      auth_message_requests=no
      outofcall_message_context=dpma_message_context
      faxdetect=no
      vmexten=*97
      useragent=FPBX-15.0.17.34(17.9.3)
      language=en
      disallow=all
      allow=ulaw
      allow=alaw
      allow=gsm
      allow=g726
      allow=g722
      context=from-sip-external
      callerid=Unknown
      notifyringing=yes
      notifyhold=yes
      tos_sip=cs3
      tos_audio=ef
      tos_video=af41
      alwaysauthreject=yes
      limitonpeers=yes
      accept_outofcall_message=yes
      outofcall_message_context=astsms
      auth_message_requests=yes
      context=from-sip-external
      callerid=Unknown
      tcpenable=no
      callevents=yes
      jbenable=no
      checkmwi=10
      maxexpiry=3600
      minexpiry=60
      srvlookup=no
      tlsenable=no
      allowguest=yes
      notifyhold=yes
      rtptimeout=30
      canreinvite=no
      tlsbindaddr=[::]:5161
      rtpkeepalive=0
      videosupport=no
      defaultexpiry=120
      notifyringing=yes
      maxcallbitrate=384
      rtpholdtimeout=300
      g726nonstandard=no
      registertimeout=20
      tlsclientmethod=tlsv1
      registerattempts=0
      nat=force_rport,comedia
      ALLOW_SIP_ANON=no
      udpbindaddr=0.0.0.0:5060
      tlscafile=/etc/pki/tls/certs/ca-bundle.crt
      externip=104.145.12.182
      localnet=192.168.1.6/24
      …

      sip_additional.conf
      [100]
      deny=0.0.0.0/0.0.0.0
      secret=12345
      dtmfmode=rfc2833
      canreinvite=no
      context=from-internal
      host=dynamic
      defaultuser=
      trustrpid=yes
      user_eq_phone=no
      sendrpid=pai
      type=friend
      session-timers=accept
      nat=force_rport,comedia
      port=5060
      qualify=yes
      qualifyfreq=60
      transport=udp
      avpf=no
      force_avp=no
      icesupport=no
      rtcp_mux=no
      encryption=no
      namedcallgroup=
      namedpickupgroup=
      dial=SIP/100
      accountcode=
      permit=0.0.0.0/0.0.0.0
      callerid=Omer RIT <100>
      recordonfeature=apprecord
      recordofffeature=apprecord
      callcounter=yes
      faxdetect=no

      [101]
      deny=0.0.0.0/0.0.0.0
      secret=123
      dtmfmode=rfc2833
      canreinvite=no
      context=from-internal
      host=dynamic
      defaultuser=
      trustrpid=yes
      user_eq_phone=no
      sendrpid=pai
      type=friend
      session-timers=accept
      nat=force_rport,comedia
      port=5060
      qualify=yes
      qualifyfreq=60
      transport=udp
      avpf=no
      force_avp=no
      icesupport=no
      rtcp_mux=no
      encryption=no
      namedcallgroup=
      namedpickupgroup=
      dial=SIP/101
      accountcode=
      permit=0.0.0.0/0.0.0.0
      callerid=Ahmed RIT <101>
      recordonfeature=apprecord
      recordofffeature=apprecord
      callcounter=yes
      faxdetect=no

      posted in IT Discussion
      R
      ranahashem
    • Sms chat not working on freepbx with tow linephone softphone

      i am using freepbx 15 distro

      Other SIP Settings
      accept_outofcall_message = yes
      outofcall_message_contex = astsms
      auth_message_requests = yes
      Write below lines in extensions_custom.conf file. This is dialplan to send IM.
      [astsms]
      ;Deliver to local 3-digit extension
      exten => _XXX,1,MessageSend(sip:${EXTEN},${MESSAGE(from)})
      same => n,Hangup()

      res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
      [2021-06-16 22:34:12] ERROR[22726][C-00000005]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
      [2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 83105471079628703765225689606832e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
      Packet timed out after 6401ms with no response
      [2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 10635857318019912468722712362a17e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
      Packet timed out after 6400ms with no response
      [2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 177607474807695456872236a3608968e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
      Packet timed out after 6402ms with no response
      [2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 163608105081901038768722bc289057e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
      Packet timed out after 6400ms with no response
      [2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 38370651581339966687222681d658b7e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
      Packet timed out after 6401ms with no response
      [2021-06-16 22:34:19] WARNING[3301]: chan_sip.c:4142 retrans_pkt: Retransmission timeout reached on transmission 92561586182471080867122238561512e7 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
      Packet timed out after 6400ms with no response

      posted in IT Discussion
      R
      ranahashem
    • RE: linephone softphone file sharing

      @scottalanmiller
      I didn't understand why you can't be pampered, is it because you will be restricted to using the linephone as the only soft phone, or is there another reason for that?

      posted in IT Discussion
      R
      ranahashem
    • linephone softphone file sharing

      hi every one
      how I can enable file sharing on my softphone access to FreePBX distro 15 asterisk 17
      thx

      i cant access.png

      posted in IT Discussion
      R
      ranahashem
    • RE: linphone: remove/hide “default identity”

      @scottalanmiller said in linphone: remove/hide “default identity”:

      Of course not. But TLS isn't what we are discussing, either. You brought TLS into the conversation and it has nothing to do with anything.
      LTS means Long Term Support. It is LTS that Jared told you about.

      u mean easier , I can download Freepbx Distro on my a real server,

      posted in IT Discussion
      R
      ranahashem
    • RE: linphone: remove/hide “default identity”

      @JaredBusch said in linphone: remove/hide “default identity”:

      I think you have no idea what you are talking about Student or not. Asterisk has zero licensing.
      If you are going to use FreePBX, then you use the distribution, which includes the OS. You don't install it piecemeal.
      If you are going to install only Asterisk, then you start by building the OS and dependencies required to compile it yourself.

      You mean, TLS doesn't need a license to activate on freepbx

      posted in IT Discussion
      R
      ranahashem
    • RE: linphone: remove/hide “default identity”

      @JaredBusch
      I am still a student. I cannot use the TLS protocol because it is neeeded to a license .. do you think that there is a possibility that there is a problem in the issuance of the asterisk ..

      I do not think so because all kinds of free soft phones have successfully registered with the system .. Do I download it again

      (l https://technologyrss.com/how-to-install-freepbx-14-and-asterisk-14-on-centos-7/)

      posted in IT Discussion
      R
      ranahashem
    • RE: linphone: remove/hide “default identity”

      @scottalanmiller Honestly, I have a student and a beginner in this system .. So for this I choose this version because I did not face difficult problems in it when downloading it, this is in experiments, and I know that I can upgrade it. As for Soft Phone, it registered successfully in Aasterisk but no contact with zoiper

      posted in IT Discussion
      R
      ranahashem
    • RE: linphone: remove/hide “default identity”

      @scottalanmiller If the linphone cannot connect with any softphone on the same network and it gives me unavailable in my local network what is the solution in your opinion
      I am freepbx 14
      asterisk 14 install manual
      sip port 5060
      rtp port 10000-30000
      context=internal
      disallow=all ; only the sensible codecs
      allow=ulaw
      allow=alaw
      qualify=no ;
      firewall disable

      posted in IT Discussion
      R
      ranahashem
    • linphone: remove/hide “default identity”

      I'm using linphone 4.3. with freepbx14 /centos7 systems.

      I have only one question i can't resolve on my own. The option "My current identity" some times starts in the working option and other times this start in the default identity. I can't delete the option who has

      The problem is i can't make a call when Linphone starts using this profile. I need to switch to the other identity and then all works ok.

      Is there any way to force linphone to only use my SIP account? I cant delete the "Default identity".

      Best regards.

      Rana

      OWPvQ.png

      posted in IT Discussion
      R
      ranahashem
    • 1 / 1