Service provider with callcentric; I have no inbound/outbound routes states" call cannot be completed as dialed" or “person you are trying to reach is unavailable” - i have ran sip debug through asterisk getting 404 error on one extension - callcentric port is 5061 and have doubled check everything and I am at a complete loss … can someone please point me in the right direction …
First, confirm that your Callcentric trunk is registered ok. At the Asterisk command prompt, type
sip show registry
or in the GUI, visit Reports -> Asterisk Info and click Registries.
The announcement “your call cannot be completed as dialed” indeed indicates that there is no Outbound Route matching what you dialed, but I don’t understand your question.
You entered an Outbound route but it’s not working? If you used Callcentric’s example at https://www.callcentric.com/support/device/trixbox step 2, that’s for an old-school system where you dial 9 before an outside call. Try dialing e.g. 918004377950. For dialing outside numbers directly, try setting Prepend and Prefix blank and use 1NXXNXXXXXX for the match pattern. You can then dial 1+areacode+number. If you want 7- and/or 10-digit dialing to work, try the ‘Dial pattern wizards’ function in your Outbound Route.
You tried to enter an Outbound Route but get an error or it doesn’t get saved? Please provide details.
You have no idea how to set up an Outbound Route? See https://wiki.freepbx.org/display/FPG/Outbound+Routes+Configuration+Examples .
For incoming, do you have any Inbound Routes set up? If not, make a default route by leaving DID Number and CallerID Number blank; set the Destination to your extension. If it doesn’t work, make a failing test call and then go to Reports -> Asterisk Logfiles and post the relevant entries here.
If you still have trouble please also post general info about your system:
FreePBX version? Cloud or on-site? If cloud, provider? If on-site, server hardware and virtualization platform (if applicable)? Obtained from distro, built from source, other? Endpoint (extensions) make/model or app/version?
freepbx v14
Inbound routes are set up no callerid DID - freepbx v14
this is the error I am receiving on inbound calls:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 204.11.192.170:5080;branch=z9hG4bK-4fef3f5b11f0e690a5653b91428a9428;change=ta;received=204.11.192.170;rport=5080
From: <sip:[email protected]>;tag=3747925149-783405
To: <sip:[email protected]>;tag=as37409886
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Im not sure what Im missing
Callcentric, unlike other high-end providers, has a quirk that requires special handling in Asterisk – they often send calls from an IP address other than the one you registered to. Asterisk’s chan_sip driver does not handle this correctly.
There are three solutions that you might consider: Simplest is to use a pjsip trunk instead. Unfortunately, Callcentric has not yet officially accepted this and provides no documentation. However, if you try it and it works fine it’s an easy fix.
The next option is to define a trunk alias for each of the addresses they use; see https://www.callcentric.com/support/device/asterisk/14 . This FAQ is for plain Asterisk, in FreePBX you would put the code starting with [callcentric1](callcentric);
up to the NOTE into /etc/asterisk/sip_custom_post.conf .
Yet another choice is a custom context that recognizes their IP addresses. See https://www.callcentric.com/support/device/did_trixbox .
Did you get outgoing calls to work?
I cannot say thank you enough!! yes i was able to get them both to work - i seriously have been trying to figure this out for almost 3 days!! thank you so much for your time!!
so now my outbound routes are not working and receive cannot complete call as dialed
peer details:
context=from-pstn-toheader
fromdomain=callcentric.com
fromuser=17778387121
host=callcentric.com
insecure=port,invite
secret=hello1
type=peer
defaultuser=17778387121
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw
extensions_custom.conf:
[incoming]
exten => s,1,Set(Var_FROM_DOMAIN=${CUT(CUT(SIP_HEADER(TO),@,2),>,1)})
exten => s,2,GotoIF($["${Var_FROM_DOMAIN}" = “callcentric.com”]?5:3)
exten => s,3,GotoIF($["${Var_FROM_DOMAIN}" = “ss.callcentric.com”]?5:4)
exten => s,4,GotoIF($["${Var_FROM_DOMAIN}" = “66.193.176.35”]?5:7)
exten => s,5,Set(Var_TO_DID=${CUT(CUT(SIP_HEADER(TO),@,1),:,2)})
exten => s,6,GotoIF($["${Var_TO_DID}" != “”]?ext-did,${Var_TO_DID},1:7)
; Users may edit the lines below to route incoming calls to other locations/contexts.
; If you don’t know what this means then you should likely skip the lines below and
; allow the script to run unmodified
exten => s,7,GoTo(from-pstn,s,1)
exten => h,8,Playback(ss-noservice)
exten => h,9,Macro(hangupcall)
[100]
context=to-callcentric
type=friend
defaultuser=100
secret=PASSWORD
host=dynamic
[101]
context=to-callcentric
type=friend
defaultuser=101
secret=PASSWORD
host=dynamic
[to-callcentric]
exten => _X.,1,Dial(SIP/${EXTEN}@callcentric)
any help would be greatly appreciated
You’re not going to get the FROM Domain by cutting the TO header. You need to do this to the FROM header.
unfortunately that was not the answer - still no outbound calls
Show us an actual debug of a failed call.
asterisk -rvvvvvvv
sip set debug on (if chan_sip)
pjsip set logger on (if chan_pjsip)
make the call
post the output here
Here is what is happening now- my outbound calls work in the morning and stop working at a certain time (noticed between 11am -12pm) but inbound calls work all the time. There are time conditions set however this is happening during business hours… Im so confused because configs still stay the same; rebooted asus router that didn’t work… whats next?
Exactly as I said before. You need to get a live SIP trace of the calls when they are failing. If they are failing everyday around the same time and only during those times this actually maybe a provider issue. Providers have to account for what is called “busy hours” which is the busiest part of the day where the most calls are happening. This is usually between the late morning-early afternoon because that is generally the time (at least in the US/Canada) where clients across all time zones will be active. Afternoon in the day for the East cost, middle-ish of the day for the Central/Mountain part and mornings for the West coast.
So it is totally possible your calls are being rejected because they are out of capacity. That rejection will be visible in a full SIP trace.
This isn’t a full trace. You need to grab the entire call from the console buffer after you’ve run the commands I gave you. We need to see the actual call happening.
asterisk -rvvv (is saying its not a command) ?
SSH to the PBX
asterisk -rvvvvvv <-- puts you in the Asterisk console.
If you are already in the Asterisk console, you don’t need that command. You can run the pjsip set logger on command right away.
No. No screenshots. I didn’t say take screenshots I said copy and paste. We need to see it ALL. From the moment you see it say “pjsip logger enabled” to the end of the call, all that needs to be copied and pasted here for us to look at.
This is two screenshots of the same thing and doesn’t show anything vital of the actual call.
– <SIP/4000-00000184> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
– Executing [h@from-internal:1] Macro(“SIP/4000-00000184”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/4000-00000184”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/4000-00000184”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“SIP/4000-00000184”, " monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“SIP/4000-00000184”, “attendedtransfer-rec-restart.php,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
<— SIP read from UDP:192.168.1.47:5060 —>
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK3438808599
From: “4000” sip:[email protected]:5061;tag=95768332
To: sip:[email protected]:5061;tag=as7bd2af73
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0
<------------->
— (7 headers 0 lines) —
– <SIP/4000-00000184>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“SIP/4000-00000184”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/4000-00000184’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/4000-00000184’
Really destroying SIP dialog ‘[email protected]’ Method: ACK
[2018-10-10 00:34:08] NOTICE[6933]: chan_sip.c:15716 sip_reregister: – Re-registration for [email protected]
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 204.11.192.23:5080:
REGISTER sip:callcentric.com:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK5ddcfbfc;rport
Max-Forwards: 70
From: sip:[email protected];tag=as25708ff1
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 104 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.19(13.19.1)
Authorization: Digest username=“17778387121”, realm=“callcentric.com”, algorithm=MD5, uri=“sip:sip:callcentric.com”, nonce=“5ff36d3721370c805e143e082af1e4c3”, response=“22f37ac7b3d1e362ddf74eb7f5c70201”
Expires: 120
Contact: sip:[email protected]:5061
Content-Length: 0
<— SIP read from UDP:204.11.192.23:5080 —>
SIP/2.0 200 Ok
v: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK5ddcfbfc;rport=5061;received=24.234.182.117
f: sip:[email protected];tag=as25708ff1
t: sip:[email protected]
i: [email protected]
CSeq: 104 REGISTER
m: sip:[email protected]:5061;expires=61
l: 0
<------------->
— (8 headers 0 lines) —
[2018-10-10 00:34:08] NOTICE[6933]: chan_sip.c:24551 handle_response_register: Outbound Registration: Expiry for callcentric.com is 61 sec (Scheduling reregistration in 46 s)
Really destroying SIP dialog ‘[email protected]’ Method: REGISTER
<— SIP read from UDP:192.168.1.167:5060 —>
SUBSCRIBE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK3192592796
From: “4202” sip:[email protected]:5061;tag=1207640251
To: sip:[email protected]:5061
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Contact: sip:[email protected]:5060
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.83.0.43
Expires: 3600
Event: message-summary
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.167:5060 (NAT)
Creating new subscription
Sending to 192.168.1.167:5060 (NAT)
sip_route_dump: route/path hop: sip:[email protected]:5060
Found peer ‘4202’ for ‘4202’ from 192.168.1.167:5060
<— Transmitting (NAT) to 192.168.1.167:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK3192592796;received=192.168.1.167;rport=5060
From: “4202” sip:[email protected]:5061;tag=1207640251
To: sip:[email protected]:5061;tag=as5e449a9b
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“1a2ddf39”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: SUBSCRIBE)
<— SIP read from UDP:192.168.1.167:5060 —>
SUBSCRIBE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK135735588
From: “4202” sip:[email protected]:5061;tag=1207640251
To: sip:[email protected]:5061
Call-ID: [email protected]
CSeq: 2 SUBSCRIBE
Contact: sip:[email protected]:5060
Authorization: Digest username=“4202”, realm=“asterisk”, nonce=“1a2ddf39”, uri=“sip:[email protected]:5061”, response=“fe8b573121cfe18696e30ae62e773b34”, algorithm=MD5
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.83.0.43
Expires: 3600
Event: message-summary
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to 192.168.1.167:5060 (NAT)
Found peer ‘4202’ for ‘4202’ from 192.168.1.167:5060
<— Transmitting (NAT) to 192.168.1.167:5060 —>
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK135735588;received=192.168.1.167;rport=5060
From: “4202” sip:[email protected]:5061;tag=1207640251
To: sip:[email protected]:5061;tag=as5e449a9b
Call-ID: [email protected]
CSeq: 2 SUBSCRIBE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
[2018-10-10 00:34:11] NOTICE[6933]: chan_sip.c:28418 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 4202
Really destroying SIP dialog ‘[email protected]’ Method: SUBSCRIBE
Reliably Transmitting (NAT) to 192.168.1.107:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK3d7d5b3e;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected]:5061;tag=as73e5d908
To: sip:[email protected]:5060
Contact: sip:[email protected]:5061
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.3.19(13.19.1)
Date: Wed, 10 Oct 2018 00:34:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Reliably Transmitting (NAT) to 192.168.1.167:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK3e930b86;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected]:5061;tag=as531d0fce
To: sip:[email protected]:5060
Contact: sip:[email protected]:5061
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.3.19(13.19.1)
Date: Wed, 10 Oct 2018 00:34:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:192.168.1.107:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK3d7d5b3e;rport=5061
From: “Unknown” sip:[email protected]:5061;tag=as73e5d908
To: sip:[email protected]:5060;tag=672522817
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.80.0.61
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5061’ Method: OPTIONS
<— SIP read from UDP:192.168.1.167:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK3e930b86;rport=5061
From: “Unknown” sip:[email protected]:5061;tag=as531d0fce
To: sip:[email protected]:5060;tag=1145193198
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.83.0.43
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5061’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.1.47:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK5981af6f;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected]:5061;tag=as388d6822
To: sip:[email protected]:5060
Contact: sip:[email protected]:5061
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.3.19(13.19.1)
Date: Wed, 10 Oct 2018 00:34:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:192.168.1.47:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK5981af6f;rport=5061
From: “Unknown” sip:[email protected]:5061;tag=as388d6822
To: sip:[email protected]:5060;tag=1050772521
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.80.0.70
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5061’ Method: OPTIONS
<— SIP read from UDP:192.168.1.107:5060 —>
<------------->
<— SIP read from UDP:192.168.1.47:5060 —>
@Neromarketing18 What you posted is not that. Still the end of the call. You need to make sure the scrollback buffer is a decent size. Not a single piece of information about the call was in there.
Connected to Asterisk 13.19.1 currently running on freepbx (pid = 6857)
[2018-10-10 01:34:23] NOTICE[6933]: chan_sip.c:15716 sip_reregister: – Re-registration fo r [email protected]
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 204.11.192.23:5080:
REGISTER sip:callcentric.com:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK147c3ed4;rport
Max-Forwards: 70
From: sip:[email protected];tag=as12b032f6
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 110 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.19(13.19.1)
Authorization: Digest username=“17778387121”, realm=“callcentric.com”, algorithm=MD5, uri=“si p:sip:callcentric.com”, nonce=“a86c1356c60ae0e23861a5e669d975c1”, response=“82cdaf434e0a5312d cb0df9559d493a8”
Expires: 120
Contact: sip:[email protected]:5061
Content-Length: 0
<— SIP read from UDP:204.11.192.23:5080 —>
SIP/2.0 200 Ok
v: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK147c3ed4;rport=5061;received=24.234.182.117
f: sip:[email protected];tag=as12b032f6
t: sip:[email protected]
i: [email protected]
CSeq: 110 REGISTER
m: sip:[email protected]:5061;expires=63
l: 0
<------------->
— (8 headers 0 lines) —
[2018-10-10 01:34:23] NOTICE[6933]: chan_sip.c:24551 handle_response_register: Outbound Regis tration: Expiry for callcentric.com is 63 sec (Scheduling reregistration in 48 s)
Really destroying SIP dialog ‘[email protected]’ Method: REGISTER
<— SIP read from UDP:192.168.1.167:5060 —>
SUBSCRIBE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK1085573563
From: “4202” sip:[email protected]:5061;tag=2212402902
To: sip:[email protected]:5061
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Contact: sip:[email protected]:5060
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.83.0.43
Expires: 3600
Event: message-summary
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.167:5060 (NAT)
Creating new subscription
Sending to 192.168.1.167:5060 (NAT)
sip_route_dump: route/path hop: sip:[email protected]:5060
Found peer ‘4202’ for ‘4202’ from 192.168.1.167:5060
<— Transmitting (NAT) to 192.168.1.167:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK1085573563;received=192.168.1.167;rport=506 0
From: “4202” sip:[email protected]:5061;tag=2212402902
To: sip:[email protected]:5061;tag=as52516cfc
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“405ef9b4”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: SUBSCRI BE)
<— SIP read from UDP:192.168.1.167:5060 —>
SUBSCRIBE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK3585533801
From: “4202” sip:[email protected]:5061;tag=2212402902
To: sip:[email protected]:5061
Call-ID: [email protected]
CSeq: 2 SUBSCRIBE
Contact: sip:[email protected]:5060
Authorization: Digest username=“4202”, realm=“asterisk”, nonce=“405ef9b4”, uri=“sip:4202@192. 168.1.84:5061”, response=“26ad87397cffda5cde0da61d6ca7323c”, algorithm=MD5
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.83.0.43
Expires: 3600
Event: message-summary
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to 192.168.1.167:5060 (NAT)
Found peer ‘4202’ for ‘4202’ from 192.168.1.167:5060
<— Transmitting (NAT) to 192.168.1.167:5060 —>
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK3585533801;received=192.168.1.167;rport=506 0
From: “4202” sip:[email protected]:5061;tag=2212402902
To: sip:[email protected]:5061;tag=as52516cfc
Call-ID: [email protected]
CSeq: 2 SUBSCRIBE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
[2018-10-10 01:34:24] NOTICE[6933]: chan_sip.c:28418 handle_request_subscribe: Received SIP s ubscribe for peer without mailbox: 4202
Really destroying SIP dialog ‘[email protected]’ Method: SUBSCRIBE
<— SIP read from UDP:192.168.1.107:5060 —>
<------------->
<— SIP read from UDP:192.168.1.47:5060 —>
<------------->
<— SIP read from UDP:192.168.1.167:5060 —>
<------------->
freepbx*CLI> sip set debug on
SIP Debugging re-enabled
<— SIP read from UDP:192.168.1.47:5060 —>
INVITE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK2816011902
From: “4000” sip:[email protected]:5061;tag=1339102870
To: sip:[email protected]:5061
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PU BLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.70
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 306
v=0
o=- 20101 20101 IN IP4 192.168.1.47
s=SDP data
c=IN IP4 192.168.1.47
t=0 0
m=audio 12010 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (14 headers 15 lines) —
Sending to 192.168.1.47:5060 (NAT)
Sending to 192.168.1.47:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘4000’ for ‘4000’ from 192.168.1.47:5060
<— Reliably Transmitting (NAT) to 192.168.1.47:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK2816011902;received=192.168.1.47;rport=5060
From: “4000” sip:[email protected]:5061;tag=1339102870
To: sip:[email protected]:5061;tag=as7f3ac243
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“76b7eccd”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)
<— SIP read from UDP:192.168.1.47:5060 —>
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK2816011902
From: “4000” sip:[email protected]:5061;tag=1339102870
To: sip:[email protected]:5061;tag=as7f3ac243
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:192.168.1.47:5060 —>
INVITE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK2974185349
From: “4000” sip:[email protected]:5061;tag=1339102870
To: sip:[email protected]:5061
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: sip:[email protected]:5060
Authorization: Digest username=“4000”, realm=“asterisk”, nonce=“76b7eccd”, uri=“sip:161980757 [email protected]:5061”, response=“07151d7ac406eb042f4cf64130554dd9”, algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PU BLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.80.0.70
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 306
v=0
o=- 20101 20101 IN IP4 192.168.1.47
s=SDP data
c=IN IP4 192.168.1.47
t=0 0
m=audio 12010 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (15 headers 15 lines) —
Sending to 192.168.1.47:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘4000’ for ‘4000’ from 192.168.1.47:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|alaw|g722|g729)/video=(nothi ng)/text=(nothing), combined - (ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), co mbined - 0x1 (telephone-event|)
> 0x3f05120 – Strict RTP learning after remote address set to: 192.168.1.47:12010
Peer audio RTP is at port 192.168.1.47:12010
Looking for 16198075779 in from-internal (domain 192.168.1.84)
sip_route_dump: route/path hop: sip:[email protected]:5060
<— Transmitting (NAT) to 192.168.1.47:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK2974185349;received=192.168.1.47;rport=5060
From: “4000” sip:[email protected]:5061;tag=1339102870
To: sip:[email protected]:5061
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5061
Content-Length: 0
<------------>
– Executing [16198075779@from-internal:1] ResetCDR(“SIP/4000-0000019e”, “”) in new stack
– Executing [16198075779@from-internal:2] NoCDR(“SIP/4000-0000019e”, “”) in new stack
– Executing [16198075779@from-internal:3] Progress(“SIP/4000-0000019e”, “”) in new stack
Audio is at 17430
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (NAT) to 192.168.1.47:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK2974185349;received=192.168.1.47;rport=5060
From: “4000” sip:[email protected]:5061;tag=1339102870
To: sip:[email protected]:5061;tag=as17ff2912
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5061
Content-Type: application/sdp
Content-Length: 300
v=0
o=root 1969946581 1969946581 IN IP4 192.168.1.84
s=Asterisk PBX 13.19.1
c=IN IP4 192.168.1.84
t=0 0
m=audio 17430 RTP/AVP 0 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
– Executing [16198075779@from-internal:4] Wait(“SIP/4000-0000019e”, “1”) in new stack
> 0x3f05120 – Strict RTP switching to RTP target address 192.168.1.47:12010 as source
– Executing [16198075779@from-internal:5] Playback(“SIP/4000-0000019e”, “silence/1&canno t-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <SIP/4000-0000019e> Playing ‘silence/1.ulaw’ (language ‘en’)
– <SIP/4000-0000019e> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
Reliably Transmitting (NAT) to 192.168.1.107:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK08be2c2a;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected]:5061;tag=as2806ea1a
To: sip:[email protected]:5060
Contact: sip:[email protected]:5061
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.3.19(13.19.1)
Date: Wed, 10 Oct 2018 01:34:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:192.168.1.107:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK08be2c2a;rport=5061
From: “Unknown” sip:[email protected]:5061;tag=as2806ea1a
To: sip:[email protected]:5060;tag=3332368030
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.80.0.61
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5061’ Method: OPT IONS
Reliably Transmitting (NAT) to 192.168.1.167:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK43742a4b;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected]:5061;tag=as3f721cf5
To: sip:[email protected]:5060
Contact: sip:[email protected]:5061
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.3.19(13.19.1)
Date: Wed, 10 Oct 2018 01:34:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:192.168.1.167:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK43742a4b;rport=5061
From: “Unknown” sip:[email protected]:5061;tag=as3f721cf5
To: sip:[email protected]:5060;tag=2896093555
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.83.0.43
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5061’ Method: OPT IONS
Reliably Transmitting (NAT) to 192.168.1.47:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK3fa4131a;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected]:5061;tag=as7c7144af
To: sip:[email protected]:5060
Contact: sip:[email protected]:5061
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: FPBX-14.0.3.19(13.19.1)
Date: Wed, 10 Oct 2018 01:34:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:192.168.1.47:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK3fa4131a;rport=5061
From: “Unknown” sip:[email protected]:5061;tag=as7c7144af
To: sip:[email protected]:5060;tag=829016234
Call-ID: [email protected]:5061
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T21P_E2 52.80.0.70
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5061’ Method: OPT IONS
– <SIP/4000-0000019e> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
> 0x3f05120 – Strict RTP learning complete - Locking on source address 192.168.1.47:1 2010
– Executing [16198075779@from-internal:6] Wait(“SIP/4000-0000019e”, “1”) in new stack
– Executing [16198075779@from-internal:7] Congestion(“SIP/4000-0000019e”, “20”) in new s tack
<— Reliably Transmitting (NAT) to 192.168.1.47:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK2974185349;received=192.168.1.47;rport=5060
From: “4000” sip:[email protected]:5061;tag=1339102870
To: sip:[email protected]:5061;tag=as17ff2912
Call-ID: [email protected]
CSeq: 2 INVITE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
[2018-10-10 01:34:50] WARNING[26952][C-0000012d]: channel.c:5005 ast_prod: Prodding channel ’ SIP/4000-0000019e’ failed
== Spawn extension (from-internal, 16198075779, 7) exited non-zero on ‘SIP/4000-0000019e’
– Executing [h@from-internal:1] Macro(“SIP/4000-0000019e”, “hangupcall”) in new stack
<— SIP read from UDP:192.168.1.47:5060 —>
ACK sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.47:5060;branch=z9hG4bK2974185349
From: “4000” sip:[email protected]:5061;tag=1339102870
To: sip:[email protected]:5061;tag=as17ff2912
Call-ID: [email protected]
CSeq: 2 ACK
Content-Length: 0
<------------->
— (7 headers 0 lines) —
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/4000-0000019e”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/4000-0000019e”, “0?Set(CDR(recordingfile) =)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“SIP/4000-0000019e”, " monior file= ") in new st ack
– Executing [s@macro-hangupcall:5] AGI(“SIP/4000-0000019e”, “attendedtransfer-rec-restar t.php,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <SIP/4000-0000019e>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“SIP/4000-0000019e”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘SIP/4000-0000019e’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/4000-0000019e’
Really destroying SIP dialog ‘[email protected]’ Method: ACK
<— SIP read from UDP:192.168.1.167:5060 —>
SUBSCRIBE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK2788958441
From: “4202” sip:[email protected]:5061;tag=3620450120
To: sip:[email protected]:5061
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Contact: sip:[email protected]:5060
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.83.0.43
Expires: 3600
Event: message-summary
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.167:5060 (NAT)
Creating new subscription
Sending to 192.168.1.167:5060 (NAT)
sip_route_dump: route/path hop: sip:[email protected]:5060
Found peer ‘4202’ for ‘4202’ from 192.168.1.167:5060
<— Transmitting (NAT) to 192.168.1.167:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK2788958441;received=192.168.1.167;rport=506 0
From: “4202” sip:[email protected]:5061;tag=3620450120
To: sip:[email protected]:5061;tag=as73342ce0
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“3a88743d”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: SUBSCRI BE)
<— SIP read from UDP:192.168.1.167:5060 —>
SUBSCRIBE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK2383541220
From: “4202” sip:[email protected]:5061;tag=3620450120
To: sip:[email protected]:5061
Call-ID: [email protected]
CSeq: 2 SUBSCRIBE
Contact: sip:[email protected]:5060
Authorization: Digest username=“4202”, realm=“asterisk”, nonce=“3a88743d”, uri=“sip:4202@192. 168.1.84:5061”, response=“5cb80a3381b258b4de26b7d94b3487dc”, algorithm=MD5
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.83.0.43
Expires: 3600
Event: message-summary
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to 192.168.1.167:5060 (NAT)
Found peer ‘4202’ for ‘4202’ from 192.168.1.167:5060
<— Transmitting (NAT) to 192.168.1.167:5060 —>
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK2383541220;received=192.168.1.167;rport=506 0
From: “4202” sip:[email protected]:5061;tag=3620450120
To: sip:[email protected]:5061;tag=as73342ce0
Call-ID: [email protected]
CSeq: 2 SUBSCRIBE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
[2018-10-10 01:34:54] NOTICE[6933]: chan_sip.c:28418 handle_request_subscribe: Received SIP s ubscribe for peer without mailbox: 4202
Really destroying SIP dialog ‘[email protected]’ Method: SUBSCRIBE
<— SIP read from UDP:192.168.1.107:5060 —>
<------------->
<— SIP read from UDP:192.168.1.47:5060 —>
<------------->
<— SIP read from UDP:192.168.1.167:5060 —>
<------------->
[2018-10-10 01:35:11] NOTICE[6933]: chan_sip.c:15716 sip_reregister: – Re-registration for [email protected]
REGISTER 12 headers, 0 lines
Reliably Transmitting (NAT) to 204.11.192.23:5080:
REGISTER sip:callcentric.com:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK7ad32462;rport
Max-Forwards: 70
From: sip:[email protected];tag=as12b032f6
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 111 REGISTER
Supported: replaces, timer
User-Agent: FPBX-14.0.3.19(13.19.1)
Authorization: Digest username=“17778387121”, realm=“callcentric.com”, algorithm=MD5, uri=“sip:sip:callcentric.com”, nonce=“a86c1356c60ae0e23861a5e669d975c1”, response=“82cdaf434e0a5312dcb0df9559d493a8”
Expires: 120
Contact: sip:[email protected]:5061
Content-Length: 0
<— SIP read from UDP:204.11.192.23:5080 —>
SIP/2.0 200 Ok
v: SIP/2.0/UDP 192.168.1.84:5061;branch=z9hG4bK7ad32462;rport=5061;received=24.234.182.117
f: sip:[email protected];tag=as12b032f6
t: sip:[email protected]
i: [email protected]
CSeq: 111 REGISTER
m: sip:[email protected]:5061;expires=64
l: 0
<------------->
— (8 headers 0 lines) —
[2018-10-10 01:35:11] NOTICE[6933]: chan_sip.c:24551 handle_response_register: Outbound Registration: Expiry for callcentric.com is 64 sec (Scheduling reregistration in 49 s)
Really destroying SIP dialog ‘[email protected]’ Method: REGISTER
<— SIP read from UDP:192.168.1.167:5060 —>
SUBSCRIBE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK4199058373
From: “4202” sip:[email protected]:5061;tag=4290401819
To: sip:[email protected]:5061
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Contact: sip:[email protected]:5060
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.83.0.43
Expires: 3600
Event: message-summary
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.167:5060 (NAT)
Creating new subscription
Sending to 192.168.1.167:5060 (NAT)
sip_route_dump: route/path hop: sip:[email protected]:5060
Found peer ‘4202’ for ‘4202’ from 192.168.1.167:5060
<— Transmitting (NAT) to 192.168.1.167:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK4199058373;received=192.168.1.167;rport=5060
From: “4202” sip:[email protected]:5061;tag=4290401819
To: sip:[email protected]:5061;tag=as14d2f22e
Call-ID: [email protected]
CSeq: 1 SUBSCRIBE
Server: FPBX-14.0.3.19(13.19.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“1ac2a637”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: SUBSCRIBE)
<— SIP read from UDP:192.168.1.167:5060 —>
SUBSCRIBE sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.167:5060;branch=z9hG4bK2285184995
From: “4202” sip:[email protected]:5061;tag=4290401819
To: sip:[email protected]:5061
Call-ID: [email protected]
CSeq: 2 SUBSCRIBE
Contact: sip:[email protected]:5060
Authorization: Digest username=“4202”, realm=“asterisk”, nonce=“1ac2a637”, uri=“sip:[email protected]:5061”, response=“dca3136711a270cd7379ca85c76887c2”, algorithm=MD5
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T21P_E2 52.83.0.43
Expires: 3600
Event: message-summary
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to 192.168.1.167:5060 (NAT)
Found peer ‘4202’ for ‘4202’ from 192.168.1.167:5060