Issues with Paging in asterisk

#1

I am trying to implement the following dialplan where if someone calls 8XX asterisk will page extension 1XX. The Devices that register for 1XX have been set to auto answer paging/intercom.

exten => _8XX,1,NoOp(Testing calls to autoanswer)
same => n,Set(number=${EXTEN:-2})     
same => n,Page(${PJSIP_DIAL_CONTACTS(1${number})})                    
same => n,Hangup()  

So when I dial 804 it should page the devices registered with endpoint 104 and the device should autoanswer, on contrary if I dial 104 it should dial the PJSIP contacts for 104 and the device should not auto answer as autoanswer is selected for paging only. In my situation, the device won’t auto answer 804.

here is what I get on the cli

 == Setting global variable 'SIPDOMAIN' to '192.168.1.18'
  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
    -- Executing [804@default:1] NoOp("PJSIP/103-00000012", "Testing calls to autoanswer") in new stack
    -- Executing [804@default:2] Set("PJSIP/103-00000012", "number=04") in new stack
    -- Executing [804@default:3] Page("PJSIP/103-00000012", "PJSIP/104/sip:104@192.168.1.237:5062") in new stack
  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
    -- Called 104/sip:104@192.168.1.237:5062
    -- <PJSIP/103-00000012> Playing 'beep.gsm' (language 'en')
    -- PJSIP/104-00000013 is ringing
       > 0x154a160 -- Strict RTP learning after remote address set to: 192.168.1.243:23440
       > 0x154a160 -- Strict RTP switching to RTP target address 192.168.1.243:23440 as source
    -- Channel CBAnn/790859887-00000002;2 joined 'softmix' base-bridge <4c4f905c-a573-41ce-9ea8-e106b1010022>
    -- Channel PJSIP/103-00000012 joined 'softmix' base-bridge <4c4f905c-a573-41ce-9ea8-e106b1010022>
       > 0x154a160 -- Strict RTP learning complete - Locking on source address 192.168.1.243:23440
    -- Channel PJSIP/103-00000012 left 'softmix' base-bridge <4c4f905c-a573-41ce-9ea8-e106b1010022>
    -- Channel CBAnn/790859887-00000002;2 left 'softmix' base-bridge <4c4f905c-a573-41ce-9ea8-e106b1010022>
  == Spawn extension (default, 804, 3) exited non-zero on 'PJSIP/103-00000012'

The softphone from where I dialed starts the call timmer but the phones registered to 104 keeps ringing

No.   Timestamp  (Dir) Address                  SIP Message                        
===== ========== ============================== ===================================
00000 1553170249 * <== 192.168.1.243:19081      INVITE sip:804@192.168.1.18 SIP/2.0
00001 1553170249 * ==> 192.168.1.243:19081      SIP/2.0 401 Unauthorized
00002 1553170249 * <== 192.168.1.243:19081      ACK sip:804@192.168.1.18 SIP/2.0
00003 1553170249 * <== 192.168.1.243:19081      INVITE sip:804@192.168.1.18 SIP/2.0
00004 1553170249 * ==> 192.168.1.243:19081      SIP/2.0 100 Trying
00005 1553170249 * ==> 192.168.1.237:5062       INVITE sip:104@192.168.1.237:5062 SIP/2.0
00006 1553170249 * <== 192.168.1.237:5062       SIP/2.0 100 Trying
00007 1553170249 * <== 192.168.1.237:5062       SIP/2.0 180 Ringing
00008 1553170249 * ==> 192.168.1.243:19081      OPTIONS sip:103@192.168.1.243:19081 SIP/2.0
00009 1553170249 * <== 192.168.1.243:19081      SIP/2.0 200 OK
00010 1553170249 * ==> 192.168.1.243:19081      SIP/2.0 200 OK
00011 1553170249 * <== 192.168.1.243:19081      ACK sip:192.168.1.18:5060 SIP/2.0
00012 1553170260 * <== 192.168.1.243:19081      BYE sip:192.168.1.18:5060 SIP/2.0
00013 1553170260 * ==> 192.168.1.243:19081      SIP/2.0 200 OK
00014 1553170260 * ==> 192.168.1.237:5062       CANCEL sip:104@192.168.1.237:5062 SIP/2.0
00015 1553170261 * <== 192.168.1.237:5062       SIP/2.0 200 OK
00016 1553170261 * <== 192.168.1.237:5062       SIP/2.0 487 Request Terminated
00017 1553170261 * ==> 192.168.1.237:5062       ACK sip:104@192.168.1.237:5062 SIP/2.0


Below is te verbose output of logs on the cli

INVITE sip:104@192.168.1.237:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPj14452f07-629f-4038-b45f-c20cfe46cbca
From: "103" <sip:Zone_3@192.168.1.18>;tag=73263b6d-4a36-49ad-ab1c-3ba2572272eb
To: <sip:104@192.168.1.237>
Contact: <sip:asterisk@192.168.1.18:5060>
Call-ID: 613097ee-8a29-4344-b186-990b01be49b7
CSeq: 9664 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "103" <sip:Zone_3@192.168.1.18>
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Type: application/sdp
Content-Length:   315

v=0
o=- 1851903853 1851903853 IN IP4 192.168.1.18
s=Asterisk
c=IN IP4 192.168.1.18
t=0 0
m=audio 30698 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G    -- Called 104/sip:104@192.168.1.237:5062
    -- <PJSIP/103-00000018> Playing 'beep.gsm' (language 'en')
<--- Received SIP response (488 bytes) from UDP:192.168.1.237:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPj14452f07-629f-4038-b45f-c20cfe46cbca
From: "103" <sip:Zone_3@192.168.1.18>;tag=73263b6d-4a36-49ad-ab1c-3ba2572272eb
To: <sip:104@192.168.1.237>
Call-ID: 613097ee-8a29-4344-b186-990b01be49b7
CSeq: 9664 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (576 bytes) from UDP:192.168.1.237:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPj14452f07-629f-4038-b45f-c20cfe46cbca
From: "103" <sip:Zone_3@192.168.1.18>;tag=73263b6d-4a36-49ad-ab1c-3ba2572272eb
To: <sip:104@192.168.1.237>;tag=1405952063
Call-ID: 613097ee-8a29-4344-b186-990b01be49b7
CSeq: 9664 INVITE
Contact: <sip:104@192.168.1.237:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- PJSIP/104-00000019 is ringing
       > 0x13a5640 -- Strict RTP learning after remote address set to: 192.168.1.243:12824
<--- Transmitting SIP response (862 bytes) to UDP:192.168.1.243:19081 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.243:19081;rport=19081;received=192.168.1.243;branch=z9hG4bK25500860
Call-ID: 255484480-19081-17@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=33030935
To: <sip:804@192.168.1.18>;tag=ff072b28-e1d4-45c1-a153-d3c634f9f9bd
CSeq: 161 INVITE
Server: Asterisk PBX 15.2.2
Contact: <sip:192.168.1.18:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   299

v=0
o=- 8000 8002 IN IP4 192.168.1.18
s=Asterisk
c=IN IP4 192.168.1.18
t=0 0
m=audio 34942 RTP/AVP 0 8 3 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (527 bytes) from UDP:192.168.1.243:19081 --->
ACK sip:192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:19081;branch=z9hG4bK833348070;rport
From: <sip:103@192.168.1.18>;tag=33030935
To: <sip:804@192.168.1.18>;tag=ff072b28-e1d4-45c1-a153-d3c634f9f9bd
Call-ID: 255484480-19081-17@BJC.BGI.B.CED
CSeq: 161 ACK
Contact: <sip:103@192.168.1.243:19081>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.26
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


       > 0x13a5640 -- Strict RTP switching to RTP target address 192.168.1.243:12824 as source
    -- Channel CBAnn/1507844121-00000004;2 joined 'softmix' base-bridge <8e377ad9-d0ba-4ae9-91ec-e740b74445e8>
    -- Channel PJSIP/103-00000018 joined 'softmix' base-bridge <8e377ad9-d0ba-4ae9-91ec-e740b74445e8>
       > 0x13a5640 -- Strict RTP learning complete - Locking on source address 192.168.1.243:12824
<--- Received SIP request (300 bytes) from UDP:192.168.1.161:5060 --->
OPTIONS sip:192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK893451099
From: <sip:161@192.168.1.18>;tag=1807653022
To: <sip:192.168.1.18>
Call-ID: 957209068
CSeq: 20 OPTIONS
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (447 bytes) to UDP:192.168.1.161:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.161:5060;rport=5060;received=192.168.1.161;branch=z9hG4bK893451099
Call-ID: 957209068
From: <sip:161@192.168.1.18>;tag=1807653022
To: <sip:192.168.1.18>;tag=z9hG4bK893451099
CSeq: 20 OPTIONS
WWW-Authenticate: Digest  realm="asterisk",nonce="1553170682/4daf5dfff9255d57821296f3ab0573a6",opaque="3117b4a64f73c1c7",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.2.2
Content-Length:  0


<--- Received SIP request (563 bytes) from UDP:192.168.1.161:5060 --->
OPTIONS sip:192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK1273594464
From: <sip:161@192.168.1.18>;tag=1807653022
To: <sip:192.168.1.18>
Call-ID: 957209068
CSeq: 21 OPTIONS
Authorization: Digest username="161", realm="asterisk", nonce="1553170682/4daf5dfff9255d57821296f3ab0573a6", uri="sip:192.168.1.18", response="cf017a339cd50b2cffb0199b39b0439c", algorithm=MD5, cnonce="0a4f113b", opaque="3117b4a64f73c1c7", qop=auth, nc=00000001
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0


<--- Transmitting SIP response (779 bytes) to UDP:192.168.1.161:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.161:5060;rport=5060;received=192.168.1.161;branch=z9hG4bK1273594464
Call-ID: 957209068
From: <sip:161@192.168.1.18>;tag=1807653022
To: <sip:192.168.1.18>;tag=z9hG4bK1273594464
CSeq: 21 OPTIONS
Accept: application/sdp, application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/pidf+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, message/sipfrag;version=2.0
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: text/plain
Accept-Language: en
Server: Asterisk PBX 15.2.2
Content-Length:  0


<--- Received SIP request (528 bytes) from UDP:192.168.1.243:19081 --->
BYE sip:192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:19081;branch=z9hG4bK1885072078;rport
From: <sip:103@192.168.1.18>;tag=33030935
To: <sip:804@192.168.1.18>;tag=ff072b28-e1d4-45c1-a153-d3c634f9f9bd
Call-ID: 255484480-19081-17@BJC.BGI.B.CED
CSeq: 162 BYE
Contact: <sip:103@192.168.1.243:19081>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.26
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (335 bytes) to UDP:192.168.1.243:19081 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.243:19081;rport=19081;received=192.168.1.243;branch=z9hG4bK1885072078
Call-ID: 255484480-19081-17@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=33030935
To: <sip:804@192.168.1.18>;tag=ff072b28-e1d4-45c1-a153-d3c634f9f9bd
CSeq: 162 BYE
Server: Asterisk PBX 15.2.2
Content-Length:  0


    -- Channel PJSIP/103-00000018 left 'softmix' base-bridge <8e377ad9-d0ba-4ae9-91ec-e740b74445e8>
    -- Channel CBAnn/1507844121-00000004;2 left 'softmix' base-bridge <8e377ad9-d0ba-4ae9-91ec-e740b74445e8>
<--- Transmitting SIP request (409 bytes) to UDP:192.168.1.237:5062 --->
CANCEL sip:104@192.168.1.237:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPj14452f07-629f-4038-b45f-c20cfe46cbca
From: "103" <sip:Zone_3@192.168.1.18>;tag=73263b6d-4a36-49ad-ab1c-3ba2572272eb
To: <sip:104@192.168.1.237>
Call-ID: 613097ee-8a29-4344-b186-990b01be49b7
CSeq: 9664 CANCEL
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Length:  0


  == Spawn extension (default, 804, 3) exited non-zero on 'PJSIP/103-00000018'
<--- Received SIP response (545 bytes) from UDP:192.168.1.237:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPj14452f07-629f-4038-b45f-c20cfe46cbca
From: "103" <sip:Zone_3@192.168.1.18>;tag=73263b6d-4a36-49ad-ab1c-3ba2572272eb
To: <sip:104@192.168.1.237>;tag=1405952063
Call-ID: 613097ee-8a29-4344-b186-990b01be49b7
CSeq: 9664 CANCEL
Contact: <sip:104@192.168.1.237:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (522 bytes) from UDP:192.168.1.237:5062 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPj14452f07-629f-4038-b45f-c20cfe46cbca
From: "103" <sip:Zone_3@192.168.1.18>;tag=73263b6d-4a36-49ad-ab1c-3ba2572272eb
To: <sip:104@192.168.1.237>;tag=1405952063
Call-ID: 613097ee-8a29-4344-b186-990b01be49b7
CSeq: 9664 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (395 bytes) to UDP:192.168.1.237:5062 --->
ACK sip:104@192.168.1.237:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPj14452f07-629f-4038-b45f-c20cfe46cbca
From: "103" <sip:Zone_3@192.168.1.18>;tag=73263b6d-4a36-49ad-ab1c-3ba2572272eb
To: <sip:104@192.168.1.237>;tag=1405952063
Call-ID: 613097ee-8a29-4344-b186-990b01be49b7
CSeq: 9664 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Length:  0

0 Likes

#3

You don’t seem to have set any SIP headers that might cause the device (not extension) to answer. It didn’t answer.

0 Likes

#4

How to setup them? Would this work?

Set(PJSIP_HEADER(add,answer-after=0)

0 Likes

#5

There isn’t an official standard for such headers and I am not sufficiently familiar with PJSIP to be sure that the way of setting the header is correct, but it probably is.

0 Likes

#6

I tried the following

exten => _8XX,1,NoOp(Testing calls to autoanswer)
same => n,Set(number=${EXTEN:-2})  
same => n,Set(PJSIP_HEADER(add,Call-Info: answer-after=0)   
same => n,Page(PJSIP/102,ib(paging_handler^addheader^1))                    
same => n,Hangup()  


[paging_handler]
exten => addheader,1,Set(PJSIP_HEADER(add,Call-Info)=;Answer-After=0)
same => n,Return()

but am getting this error

WARNING[17666][C-00000003]: app.c:403 ast_app_expand_sub_args: Cannot expand 'Gosub(paging_handler,addheader,1)' arguments.  The app_stack module is not available.

I am using Openwrt and Asterisk 15

0 Likes

#7

The app_stack module does not appear to be loaded, which is needed.

0 Likes

#8

I am able to make the paging call now but I am facing an issue . the audio works one way and the video doesn’t work
.
103 is a sof-tphone and 102 is the IP Phone that has auto answer enabled when there is a paging call.
below is the verbose log of a normal call from 103 to 102.

https://pastebin.com/xN1tGz2r

here is the history

No.   Timestamp  (Dir) Address                  SIP Message                        
===== ========== ============================== ===================================

00006 1554804978 * ==> 192.168.1.243:54711      OPTIONS sip:103@192.168.1.243:54711 SIP/2.0
00007 1554804978 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00008 1554804983 * <== 192.168.1.243:54711      INVITE sip:102@192.168.1.18 SIP/2.0
00009 1554804983 * ==> 192.168.1.243:54711      SIP/2.0 401 Unauthorized
00010 1554804983 * <== 192.168.1.243:54711      ACK sip:102@192.168.1.18 SIP/2.0
00011 1554804983 * <== 192.168.1.243:54711      INVITE sip:102@192.168.1.18 SIP/2.0
00012 1554804983 * ==> 192.168.1.243:54711      SIP/2.0 100 Trying
00013 1554804983 * ==> 192.168.1.236:5064       INVITE sip:102@192.168.1.236:5064 SIP/2.0
00014 1554804983 * <== 192.168.1.236:5064       SIP/2.0 100 Trying
00015 1554804983 * <== 192.168.1.236:5064       SIP/2.0 180 Ringing
00016 1554804983 * ==> 192.168.1.243:54711      SIP/2.0 180 Ringing
00017 1554804987 * <== 192.168.1.236:5064       SIP/2.0 200 OK
00018 1554804987 * ==> 192.168.1.236:5064       ACK sip:102@192.168.1.236:5064 SIP/2.0
00019 1554804987 * ==> 192.168.1.243:54711      SIP/2.0 200 OK
00020 1554804987 * <== 192.168.1.243:54711      ACK sip:192.168.1.18:5060 SIP/2.0
00021 1554804987 * <== 192.168.1.243:54711      INFO sip:192.168.1.18:5060 SIP/2.0
00022 1554804987 * ==> 192.168.1.243:54711      SIP/2.0 200 OK
00023 1554804987 * ==> 192.168.1.236:5064       INFO sip:102@192.168.1.236:5064 SIP/2.0
00024 1554804987 * <== 192.168.1.236:5064       SIP/2.0 200 OK
00025 1554804990 * <== 192.168.1.236:5064       BYE sip:asterisk@192.168.1.18:5060 SIP/2.0
00026 1554804990 * ==> 192.168.1.236:5064       SIP/2.0 200 OK
00027 1554804990 * ==> 192.168.1.243:54711      BYE sip:103@192.168.1.243:54711 SIP/2.0
00028 1554804990 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00029 1554804992 * ==> 192.168.1.236:5064       OPTIONS sip:102@192.168.1.236:5064 SIP/2.0
00030 1554804992 * <== 192.168.1.236:5064       SIP/2.0 200 OK
00031 1554805011 * ==> 192.168.1.249:5060       OPTIONS sip:151@192.168.1.249:5060;line=952faf1027056a3 SIP/2.0
00032 1554805012 * <== 192.168.1.249:5060       SIP/2.0 200 OK
00033 1554805038 * ==> 192.168.1.243:54711      OPTIONS sip:103@192.168.1.243:54711 SIP/2.0
00034 1554805038 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00035 1554805052 * ==> 192.168.1.236:5064       OPTIONS sip:102@192.168.1.236:5064 SIP/2.0
00036 1554805052 * <== 192.168.1.236:5064       SIP/2.0 200 OK
00037 1554805071 * ==> 192.168.1.249:5060       OPTIONS sip:151@192.168.1.249:5060;line=952faf1027056a3 SIP/2.0
00038 1554805071 * <== 192.168.1.249:5060       SIP/2.0 200 OK
00039 1554805098 * ==> 192.168.1.243:54711      OPTIONS sip:103@192.168.1.243:54711 SIP/2.0
00040 1554805099 * ==> 192.168.1.243:54711      OPTIONS sip:103@192.168.1.243:54711 SIP/2.0
00041 1554805099 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00042 1554805099 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00043 1554805112 * ==> 192.168.1.236:5064       OPTIONS sip:102@192.168.1.236:5064 SIP/2.0
00044 1554805112 * <== 192.168.1.236:5064       SIP/2.0 200 OK
00045 1554805131 * ==> 192.168.1.249:5060       OPTIONS sip:151@192.168.1.249:5060;line=952faf1027056a3 SIP/2.0
00046 1554805131 * <== 192.168.1.249:5060       SIP/2.0 200 OK
00047 1554805158 * ==> 192.168.1.243:54711      OPTIONS sip:103@192.168.1.243:54711 SIP/2.0
00048 1554805158 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00049 1554805172 * ==> 192.168.1.236:5064       OPTIONS sip:102@192.168.1.236:5064 SIP/2.0
00050 1554805172 * <== 192.168.1.236:5064       SIP/2.0 200 OK
00051 1554805180 * <== 192.168.1.249:5060       REGISTER sip:192.168.1.18 SIP/2.0
00052 1554805180 * ==> 192.168.1.249:5060       SIP/2.0 401 Unauthorized
00053 1554805180 * <== 192.168.1.249:5060       REGISTER sip:192.168.1.18 SIP/2.0
00054 1554805180 * ==> 192.168.1.249:5060       SIP/2.0 200 OK
00055 1554805191 * ==> 192.168.1.249:5060       OPTIONS sip:151@192.168.1.249:5060;line=952faf1027056a3 SIP/2.0
00056 1554805191 * <== 192.168.1.249:5060       SIP/2.0 200 OK
00057 1554805218 * ==> 192.168.1.243:54711      OPTIONS sip:103@192.168.1.243:54711 SIP/2.0
00058 1554805219 * ==> 192.168.1.243:54711      OPTIONS sip:103@192.168.1.243:54711 SIP/2.0
00059 1554805219 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00060 1554805219 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00061 1554805232 * ==> 192.168.1.236:5064       OPTIONS sip:102@192.168.1.236:5064 SIP/2.0
00062 1554805232 * <== 192.168.1.236:5064       SIP/2.0 200 OK
00063 1554805251 * ==> 192.168.1.249:5060       OPTIONS sip:151@192.168.1.249:5060;line=952faf1027056a3 SIP/2.0
00064 1554805251 * <== 192.168.1.249:5060       SIP/2.0 200 OK

In the above call all is fine

For Paging call

<--- Received SIP request (1269 bytes) from UDP:192.168.1.243:54711 --->
INVITE sip:802@192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK732960916;rport
From: <sip:103@192.168.1.18>;tag=1427670926
To: <sip:802@192.168.1.18>
Call-ID: 1871616531-54711-7@BJC.BGI.B.CED
CSeq: 60 INVITE
Contact: <sip:103@192.168.1.243:54711>
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.27
Privacy: none
P-Preferred-Identity: <sip:103@192.168.1.18>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   632

v=0
o=103 8000 8000 IN IP4 192.168.1.243
s=SIP Call
c=IN IP4 192.168.1.243
t=0 0
m=audio 16156 RTP/AVP 0 8 9 123 2 97 3 101
a=sendrecv
a=rtcp:16157 IN IP4 192.168.1.243
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:123 opus/48000/2
<--- Transmitting SIP response (475 bytes) to UDP:192.168.1.243:54711 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.243:54711;rport=54711;received=192.168.1.243;branch=z9hG4bK732960916
Call-ID: 1871616531-54711-7@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=1427670926
To: <sip:802@192.168.1.18>;tag=z9hG4bK732960916
CSeq: 60 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1554806400/81aafbd64d0c258c7e562b2d550b631b",opaque="7f305a7d08aaf711",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.2.2
Content-Length:  0


<--- Received SIP request (274 bytes) from UDP:192.168.1.243:54711 --->
ACK sip:802@192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK732960916;rport
From: <sip:103@192.168.1.18>;tag=1427670926
To: <sip:802@192.168.1.18>;tag=z9hG4bK732960916
Call-ID: 1871616531-54711-7@BJC.BGI.B.CED
CSeq: 60 ACK
Content-Length: 0


<--- Received SIP request (1535 bytes) from UDP:192.168.1.243:54711 --->
INVITE sip:802@192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK585985027;rport
From: <sip:103@192.168.1.18>;tag=1427670926
To: <sip:802@192.168.1.18>
Call-ID: 1871616531-54711-7@BJC.BGI.B.CED
CSeq: 61 INVITE
Contact: <sip:103@192.168.1.243:54711>
Authorization: Digest username="103", realm="asterisk", nonce="1554806400/81aafbd64d0c258c7e562b2d550b631b", uri="sip:802@192.168.1.18", response="9436c0ca2243d9a26ed9254d62733ab2", algorithm=md5, cnonce="00977457", opaque="7f305a7d08aaf711", qop=auth, nc=00000008
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.27
Privacy: none
P-Preferred-Identity: <sip:103@192.168.1.18>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   632

v=0
o=103 8000 8000 IN IP  == Setting global variable 'SIPDOMAIN' to '192.168.1.18'
<--- Transmitting SIP response (301 bytes) to UDP:192.168.1.243:54711 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.243:54711;rport=54711;received=192.168.1.243;branch=z9hG4bK585985027
Call-ID: 1871616531-54711-7@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=1427670926
To: <sip:802@192.168.1.18>
CSeq: 61 INVITE
Server: Asterisk PBX 15.2.2
Content-Length:  0


  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Video TOS bits 136
  == Using SIP RTP Video TOS bits 136 in TCLASS field.
    -- Executing [802@default:1] NoOp("PJSIP/103-0000000f", "Testing calls to autoanswer") in new stack
    -- Executing [802@default:2] Set("PJSIP/103-0000000f", "number=02") in new stack
    -- Executing [802@default:3] Set("PJSIP/103-0000000f", "PJSIP_HEADER(add,Call-Info: answer-after=0") in new stack
[Apr  9 10:40:00] WARNING[26297][C-0000000c]: pbx_variables.c:1099 pbx_builtin_setvar: Please avoid unnecessary spaces on variables as it may lead to unexpected results ('PJSIP_HEADER(add,Call-Info: answer-after' set to '0').
    -- Executing [802@default:4] Page("PJSIP/103-0000000f", "PJSIP/102,ib(paging_handler^addheader^1)") in new stack
    -- PJSIP/102-00000010 Internal Gosub(paging_handler,addheader,1) start
    -- Executing [addheader@paging_handler:1] Set("PJSIP/102-00000010", "PJSIP_HEADER(add,Call-Info)=answer-after=0") in new stack
    -- Executing [addheader@paging_handler:2] Return("PJSIP/102-00000010", "") in new stack
  == Spawn extension (default, s, 1) exited non-zero on 'PJSIP/102-00000010'
    -- PJSIP/102-00000010 Internal Gosub(paging_handler,addheader,1) complete GOSUB_RETVAL=
    -- Called 102
  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
    -- <PJSIP/103-0000000f> Playing 'beep.gsm' (language 'en')
  == Using SIP RTP Video TOS bits 136
  == Using SIP RTP Video TOS bits 136 in TCLASS field.
<--- Transmitting SIP request (1113 bytes) to UDP:192.168.1.236:5064 --->
INVITE sip:102@192.168.1.236:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPjc9b427f6-e592-4c71-9915-5a2a82704ef9
From: "103" <sip:Zone_3@192.168.1.18>;tag=3ccb7bb0-0111-47d7-88db-ff78d7e6c7c1
To: <sip:102@192.168.1.236>
Contact: <sip:asterisk@192.168.1.18:5060>
Call-ID: cf37fdab-490a-42e4-8ae0-3b37fa998de7
CSeq: 2848 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "103" <sip:Zone_3@192.168.1.18>
Call-Info: answer-after=0
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Type: application/sdp
Content-Length:   375

v=0
o=- 645679766 645679766 IN IP4 192.168.1.18
s=Asterisk
c=IN IP4 192.168.1.18
t=0 0
m=audio 31782 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3<--- Received SIP response (488 bytes) from UDP:192.168.1.236:5064 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPjc9b427f6-e592-4c71-9915-5a2a82704ef9
From: "103" <sip:Zone_3@192.168.1.18>;tag=3ccb7bb0-0111-47d7-88db-ff78d7e6c7c1
To: <sip:102@192.168.1.236>
Call-ID: cf37fdab-490a-42e4-8ae0-3b37fa998de7
CSeq: 2848 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (576 bytes) from UDP:192.168.1.236:5064 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPjc9b427f6-e592-4c71-9915-5a2a82704ef9
From: "103" <sip:Zone_3@192.168.1.18>;tag=3ccb7bb0-0111-47d7-88db-ff78d7e6c7c1
To: <sip:102@192.168.1.236>;tag=1318596784
Call-ID: cf37fdab-490a-42e4-8ae0-3b37fa998de7
CSeq: 2848 INVITE
Contact: <sip:102@192.168.1.236:5064>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- PJSIP/102-00000010 is ringing
<--- Received SIP response (1061 bytes) from UDP:192.168.1.236:5064 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPjc9b427f6-e592-4c71-9915-5a2a82704ef9
From: "103" <sip:Zone_3@192.168.1.18>;tag=3ccb7bb0-0111-47d7-88db-ff78d7e6c7c1
To: <sip:102@192.168.1.236>;tag=1318596784
Call-ID: cf37fdab-490a-42e4-8ae0-3b37fa998de7
CSeq: 2848 INVITE
Contact: <sip:102@192.168.1.236:5064>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length:   428

v=0
o=102 8000 8000 IN IP4 192.168.1.236
s=SIP Call
c=IN IP4 192.168.1.236
t=0 0
m=audio 5004 RTP/AVP 0 8 101
a=sendrecv
a=rtcp:5005 IN IP4 192.168.1.236
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 5006 RTP/       > 0x184e780 -- Strict RTP learning after remote address set to: 192.168.1.236:5004
       > 0x1851200 -- Strict RTP learning after remote address set to: 192.168.1.236:5006
<--- Transmitting SIP request (395 bytes) to UDP:192.168.1.236:5064 --->
ACK sip:102@192.168.1.236:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPj930b74f3-7b0f-4a91-8797-93cc7143c81d
From: "103" <sip:Zone_3@192.168.1.18>;tag=3ccb7bb0-0111-47d7-88db-ff78d7e6c7c1
To: <sip:102@192.168.1.236>;tag=1318596784
Call-ID: cf37fdab-490a-42e4-8ae0-3b37fa998de7
CSeq: 2848 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Length:  0


    -- PJSIP/102-00000010 answered
    -- Channel PJSIP/102-00000010 joined 'softmix' base-bridge <890631c6-b5db-467b-a7d5-4fa18132eb24>
    -- Channel CBAnn/1874744698-00000001;2 joined 'softmix' base-bridge <890631c6-b5db-467b-a7d5-4fa18132eb24>
       > 0x1849280 -- Strict RTP learning after remote address set to: 192.168.1.243:16156
       > 0x184bd00 -- Strict RTP learning after remote address set to: 192.168.1.243:37094
<--- Transmitting SIP response (928 bytes) to UDP:192.168.1.243:54711 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.243:54711;rport=54711;received=192.168.1.243;branch=z9hG4bK585985027
Call-ID: 1871616531-54711-7@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=1427670926
To: <sip:802@192.168.1.18>;tag=b983c71e-6a28-4ec5-be09-b34d0483363b
CSeq: 61 INVITE
Server: Asterisk PBX 15.2.2
Contact: <sip:192.168.1.18:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   363

v=0
o=- 8000 8002 IN IP4 192.168.1.18
s=Asterisk
c=IN IP4 192.168.1.18
t=0 0
m=audio 37554 RTP/AVP 0 8 3 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 35222 RTP/AVP 105
a=rtpmap:105 H264/90000
<--- Received SIP request (529 bytes) from UDP:192.168.1.243:54711 --->
ACK sip:192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK1028103670;rport
From: <sip:103@192.168.1.18>;tag=1427670926
To: <sip:802@192.168.1.18>;tag=b983c71e-6a28-4ec5-be09-b34d0483363b
Call-ID: 1871616531-54711-7@BJC.BGI.B.CED
CSeq: 61 ACK
Contact: <sip:103@192.168.1.243:54711>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP request (743 bytes) from UDP:192.168.1.243:54711 --->
INFO sip:192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK406165872;rport
From: <sip:103@192.168.1.18>;tag=1427670926
To: <sip:802@192.168.1.18>;tag=b983c71e-6a28-4ec5-be09-b34d0483363b
Call-ID: 1871616531-54711-7@BJC.BGI.B.CED
CSeq: 62 INFO
Contact: <sip:103@192.168.1.243:54711>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/media_control+xml
Content-Length:   164

<?xml version="1.0" encoding="utf-8" ?><media_control>  <vc_primitive>    <to_encoder>      <picture_fast_update/>    </to_encoder>  </vc_primitive></media_control>
<--- Transmitting SIP response (336 bytes) to UDP:192.168.1.243:54711 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.243:54711;rport=54711;received=192.168.1.243;branch=z9hG4bK406165872
Call-ID: 1871616531-54711-7@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=1427670926
To: <sip:802@192.168.1.18>;tag=b983c71e-6a28-4ec5-be09-b34d0483363b
CSeq: 62 INFO
Server: Asterisk PBX 15.2.2
Content-Length:  0


       > 0x1849280 -- Strict RTP switching to RTP target address 192.168.1.243:16156 as source
    -- Channel PJSIP/103-0000000f joined 'softmix' base-bridge <890631c6-b5db-467b-a7d5-4fa18132eb24>
<--- Transmitting SIP request (623 bytes) to UDP:192.168.1.236:5064 --->
INFO sip:102@192.168.1.236:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPj99d75dd4-cf91-495b-9746-7ba7338aeb28
From: "103" <sip:Zone_3@192.168.1.18>;tag=3ccb7bb0-0111-47d7-88db-ff78d7e6c7c1
To: <sip:102@192.168.1.236>;tag=1318596784
Call-ID: cf37fdab-490a-42e4-8ae0-3b37fa998de7
CSeq: 2849 INFO
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>

<--- Transmitting SIP request (611 bytes) to UDP:192.168.1.243:54711 --->
INFO sip:103@192.168.1.243:54711 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPjba09b1c9-f42a-4a90-b5bf-1807b4e99296
From: <sip:802@192.168.1.18>;tag=b983c71e-6a28-4ec5-be09-b34d0483363b
To: <sip:103@192.168.1.18>;tag=1427670926
Call-ID: 1871616531-54711-7@BJC.BGI.B.CED
CSeq: 22353 INFO
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>

<--- Received SIP response (543 bytes) from UDP:192.168.1.236:5064 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPj99d75dd4-cf91-495b-9746-7ba7338aeb28
From: "103" <sip:Zone_3@192.168.1.18>;tag=3ccb7bb0-0111-47d7-88db-ff78d7e6c7c1
To: <sip:102@192.168.1.236>;tag=1318596784
Call-ID: cf37fdab-490a-42e4-8ae0-3b37fa998de7
CSeq: 2849 INFO
Contact: <sip:102@192.168.1.236:5064>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (527 bytes) from UDP:192.168.1.243:54711 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPjba09b1c9-f42a-4a90-b5bf-1807b4e99296
From: <sip:802@192.168.1.18>;tag=b983c71e-6a28-4ec5-be09-b34d0483363b
To: <sip:103@192.168.1.18>;tag=1427670926
Call-ID: 1871616531-54711-7@BJC.BGI.B.CED
CSeq: 22353 INFO
Contact: <sip:103@192.168.1.243:54711>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (418 bytes) to UDP:192.168.1.236:5064 --->
OPTIONS sip:102@192.168.1.236:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPj2cb38a40-2737-4d8c-abc1-220057f5a740
From: <sip:102@192.168.1.18>;tag=6a6a78ef-6b6a-4a61-8b26-7a7459a5d5b2
To: <sip:102@192.168.1.236>
Contact: <sip:102@192.168.1.18:5060>
Call-ID: e043ae2c-2fa6-480b-b5e2-353d3dff3600
CSeq: 24309 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Length:  0


<--- Received SIP response (492 bytes) from UDP:192.168.1.236:5064 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPj2cb38a40-2737-4d8c-abc1-220057f5a740
From: <sip:102@192.168.1.18>;tag=6a6a78ef-6b6a-4a61-8b26-7a7459a5d5b2
To: <sip:102@192.168.1.236>;tag=1481941891
Call-ID: e043ae2c-2fa6-480b-b5e2-353d3dff3600
CSeq: 24309 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


       > 0x184e780 -- Strict RTP switching to RTP target address 192.168.1.236:5004 as source
       > 0x184bd00 -- Strict RTP switching to RTP target address 192.168.1.243:37094 as source
       > 0x1851200 -- Strict RTP switching to RTP target address 192.168.1.236:5006 as source
       > 0x1851200 -- Strict RTP learning complete - Locking on source address 192.168.1.236:5006
       > 0x184e780 -- Strict RTP learning complete - Locking on source address 192.168.1.236:5004
       > 0x1849280 -- Strict RTP learning complete - Locking on source address 192.168.1.243:16156
       > 0x184bd00 -- Strict RTP learning complete - Locking on source address 192.168.1.243:37094
<--- Received SIP request (556 bytes) from UDP:192.168.1.236:5064 --->
BYE sip:asterisk@192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.236:5064;branch=z9hG4bK1644138885;rport
From: <sip:102@192.168.1.236>;tag=1318596784
To: "103" <sip:Zone_3@192.168.1.18>;tag=3ccb7bb0-0111-47d7-88db-ff78d7e6c7c1
Call-ID: cf37fdab-490a-42e4-8ae0-3b37fa998de7
CSeq: 2849 BYE
Contact: <sip:102@192.168.1.236:5064>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (350 bytes) to UDP:192.168.1.236:5064 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.236:5064;rport=5064;received=192.168.1.236;branch=z9hG4bK1644138885
Call-ID: cf37fdab-490a-42e4-8ae0-3b37fa998de7
From: <sip:102@192.168.1.236>;tag=1318596784
To: "103" <sip:Zone_3@192.168.1.18>;tag=3ccb7bb0-0111-47d7-88db-ff78d7e6c7c1
CSeq: 2849 BYE
Server: Asterisk PBX 15.2.2
Content-Length:  0


    -- Channel PJSIP/102-00000010 left 'softmix' base-bridge <890631c6-b5db-467b-a7d5-4fa18132eb24>

History

No.   Timestamp  (Dir) Address                  SIP Message                        
===== ========== ============================== ===================================
00000 1554806383 * ==> 192.168.1.243:54711      OPTIONS sip:103@192.168.1.243:54711 SIP/2.0
00001 1554806383 * ==> 192.168.1.243:54711      OPTIONS sip:103@192.168.1.243:54711 SIP/2.0
00002 1554806384 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00003 1554806384 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00004 1554806400 * <== 192.168.1.243:54711      INVITE sip:802@192.168.1.18 SIP/2.0
00005 1554806400 * ==> 192.168.1.243:54711      SIP/2.0 401 Unauthorized
00006 1554806400 * <== 192.168.1.243:54711      ACK sip:802@192.168.1.18 SIP/2.0
00007 1554806400 * <== 192.168.1.243:54711      INVITE sip:802@192.168.1.18 SIP/2.0
00008 1554806400 * ==> 192.168.1.243:54711      SIP/2.0 100 Trying
00009 1554806400 * ==> 192.168.1.236:5064       INVITE sip:102@192.168.1.236:5064 SIP/2.0
00010 1554806400 * <== 192.168.1.236:5064       SIP/2.0 100 Trying
00011 1554806400 * <== 192.168.1.236:5064       SIP/2.0 180 Ringing
00012 1554806401 * <== 192.168.1.236:5064       SIP/2.0 200 OK
00013 1554806401 * ==> 192.168.1.236:5064       ACK sip:102@192.168.1.236:5064 SIP/2.0
00014 1554806401 * ==> 192.168.1.243:54711      SIP/2.0 200 OK
00015 1554806401 * <== 192.168.1.243:54711      ACK sip:192.168.1.18:5060 SIP/2.0
00016 1554806401 * <== 192.168.1.243:54711      INFO sip:192.168.1.18:5060 SIP/2.0
00017 1554806401 * ==> 192.168.1.243:54711      SIP/2.0 200 OK
00018 1554806401 * ==> 192.168.1.236:5064       INFO sip:102@192.168.1.236:5064 SIP/2.0
00019 1554806401 * ==> 192.168.1.243:54711      INFO sip:103@192.168.1.243:54711 SIP/2.0
00020 1554806401 * <== 192.168.1.236:5064       SIP/2.0 200 OK
00021 1554806401 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00022 1554806402 * ==> 192.168.1.236:5064       OPTIONS sip:102@192.168.1.236:5064 SIP/2.0
00023 1554806402 * <== 192.168.1.236:5064       SIP/2.0 200 OK
00024 1554806430 * <== 192.168.1.236:5064       BYE sip:asterisk@192.168.1.18:5060 SIP/2.0
00025 1554806430 * ==> 192.168.1.236:5064       SIP/2.0 200 OK
00026 1554806437 * ==> 192.168.1.249:5060       OPTIONS sip:151@192.168.1.249:5060;line=952faf1027056a3 SIP/2.0
00027 1554806437 * <== 192.168.1.249:5060       SIP/2.0 200 OK
00028 1554806443 * ==> 192.168.1.243:54711      OPTIONS sip:103@192.168.1.243:54711 SIP/2.0
00029 1554806443 * <== 192.168.1.243:54711      SIP/2.0 200 OK
00030 1554806462 * ==> 192.168.1.236:5064       OPTIONS sip:102@192.168.1.236:5064 SIP/2.0
00031 1554806462 * <== 192.168.1.236:5064       SIP/2.0 200 OK

in the above case the in paging there is one way audio and no video

0 Likes

#9

It’s on purpose that audio is one way, as for video I doubt anyone has ever tested paging with video. I kind of doubt it would work. It was strictly done for audio purposes.

0 Likes

#10

Is there a way to make two way audio? Indeed I need to make the phone behave as an video intercom.

0 Likes

#11

The documentation for Page[1] shows a full duplex option. As for video, there’s no options or anything for it. It’s just not something anyone has done with Page.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Page

1 Like

#12

I think you are confusing Paging with the proprietary auto answer headers supported by some phones. The phone cannot distinguish between Page and Dial,so you can force an autoanswer, where supported, for Dial, and get full point to point call media capabilities.

Just remember that anyone else who can access the phone can do this, as well.

1 Like

#13

Changing Page to Dial in the below results in Normal call and the Phone doesn’t auto answer

0 Likes

#14

Both Dial and Page operate the same way fundamentally, if it’s not working then it’s likely your configuration (for example Dial doesn’t have the same argument format as Page).

1 Like

#15

Although it is relatively new, both Page and Dial have b optiions. DId you include the b option on Dial?

1 Like

#16

As @jcolp said the Dial argument format is different.

same => n,Dial(PJSIP/1${number},,ib(paging_handler^addheader^1))

0 Likes