ranahashem

ranahashem

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

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

@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

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

ranahashem

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

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

@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 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

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'

ranahashem

@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

ranahashem

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

ranahashem

@ranahashem

Best posts made by ranahashem

Latest posts made by ranahashem