chat not working

ranahashem

@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

ranahashem

@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
``

ranahashem

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

scottalanmiller

@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.

scottalanmiller

@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.

ranahashem

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

ranahashem

@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

ranahashem

@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

ranahashem

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ranahashem

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