I have a video door phone that uses sip. When someone rings the bell it is supposed to call 601.
The problem that I am facing is the doorbell initiates two calls one for its own protocol with the help of which the indoor monitor can work even without the Asterisk. I was using chan_sip initially and that was not having any issue. After shifting to chan_pjsip I am having a problem that there are two calls placed from the doorphone. One is annoymus annd the other is from the doorphone extension 161.
[default]
exten => 601,n,Dial(PJSIP/101&PJSIP/102&PJSIP/103&PJSIP/104&PJSIP/105&PJSIP/106&PJSIP/107&PJSIP/108&PJSIP/109&PJSIP/110&PJSIP/111&PJSIP/112&PJSIP/113&PJSIP/114&PJSIP/115&PJSIP/116&Local/mobilephones@default)
PJSIP Logging enabled
<--- Received SIP request (497 bytes) from UDP:192.168.1.161:5060 --->
MESSAGE sip:601@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK1515420870
From: <sip:161@asterisk>;tag=1057883902
To: <sip:601@asterisk>
Call-ID: 1942807761
CSeq: 20 MESSAGE
Content-Type: text/plain
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 194
<params>
<app>talk</app>
<event>host2id</event>
<event_url>/talk/host2id</event_url>
<host>161</host>
<building>1</building>
<unit>1</unit>
<floor>0</floor>
<family>1</family>
</params>
<--- Transmitting SIP response (446 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=z9hG4bK1515420870
Call-ID: 1942807761
From: <sip:161@asterisk>;tag=1057883902
To: <sip:601@asterisk>;tag=z9hG4bK1515420870
CSeq: 20 MESSAGE
WWW-Authenticate: Digest realm="asterisk",nonce="1554196790/5def8c5c4b7f994994e73872eb6fecb1",opaque="6409680820158f24",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.2.2
Content-Length: 0
<--- Received SIP request (773 bytes) from UDP:192.168.1.161:5060 --->
INVITE sip:601@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK1565047092
From: <sip:161@asterisk>;tag=69824255
To: <sip:601@asterisk>
Call-ID: 1753293872
CSeq: 20 INVITE
Contact: <sip:161@192.168.1.161:5060>
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 430
v=0
o=dnake 1553834900 1553834900 IN IP4 192.168.1.161
s=dnake
c=IN IP4 192.168.1.161
t=0 0
m=audio 6000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
a=sendrecv
m=video 6200 RTP/AVP 104
a=rtpmap:104 H264/90000
a=fmtp:104 profile-level-id=4D0029; packetization-mode=1
a=ex_fmtp:104 2CIF=1
a=ex_multicast:104 ip=238.9.1.161; port=6300
a=sendrecv
<--- Transmitting SIP response (443 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=z9hG4bK1565047092
Call-ID: 1753293872
From: <sip:161@asterisk>;tag=69824255
To: <sip:601@asterisk>;tag=z9hG4bK1565047092
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1554196790/5def8c5c4b7f994994e73872eb6fecb1",opaque="6cafe9741c002349",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.2.2
Content-Length: 0
<--- Received SIP request (759 bytes) from UDP:192.168.1.161:5060 --->
MESSAGE sip:601@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK1216023852
From: <sip:161@asterisk>;tag=1057883902
To: <sip:601@asterisk>
Call-ID: 1942807761
CSeq: 21 MESSAGE
Authorization: Digest username="161", realm="asterisk", nonce="1554196790/5def8c5c4b7f994994e73872eb6fecb1", uri="sip:601@asterisk", response="364a34d5160cb35eadaf23bab4614f6a", algorithm=MD5, cnonce="0a4f113b", opaque="6409680820158f24", qop=auth, nc=00000001
Content-Type: text/plain
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 194
<params>
<app>talk</app>
<event>host2id</event>
<event_url>/talk/host2id</event_url>
<host>161</host>
<building>1</building>
<unit>1</unit>
<floor>0</floor>
<family>1</family>
</params>
<--- Received SIP request (269 bytes) from UDP:192.168.1.161:5060 --->
ACK sip:601@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK1565047092
Route: <sip:192.168.1.18;lr>
From: <sip:161@asterisk>;tag=69824255
To: <sip:601@asterisk>;tag=z9hG4bK1565047092
Call-ID: 1753293872
CSeq: 20 ACK
Content-Length: 0
<--- Transmitting SIP response (295 bytes) to UDP:192.168.1.161:5060 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.161:5060;rport=5060;received=192.168.1.161;branch=z9hG4bK1216023852
Call-ID: 1942807761
From: <sip:161@asterisk>;tag=1057883902
To: <sip:601@asterisk>;tag=z9hG4bK1216023852
CSeq: 21 MESSAGE
Server: Asterisk PBX 15.2.2
Content-Length: 0
-- Executing [601@default:1] NoOp("Message/ast_msg_queue", "EXECUTING 601 call") in new stack
-- Executing [601@default:2] Dial("Message/ast_msg_queue", "PJSIP/101&PJSIP/102&PJSIP/103&PJSIP/104&PJSIP/105&PJSIP/106&PJSIP/107&PJSIP/108&PJSIP/109&PJSIP/110&PJSIP/111&PJSIP/112&PJSIP/113&PJSIP/114&PJSIP/115&PJSIP/116&Local/mobilephones@default") in new stack
[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '101': Could not create dialog to invalid URI '101'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '101'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
<--- Received SIP request (1034 bytes) from UDP:192.168.1.161:5060 --->
INVITE sip:601@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.161:5060;rport;branch=z9hG4bK365732804
From: <sip:161@asterisk>;tag=69824255
To: <sip:601@asterisk>
Call-ID: 1753293872
CSeq: 21 INVITE
Contact: <sip:161@192.168.1.161:5060>
Authorization: Digest username="161", realm="asterisk", nonce="1554196790/5def8c5c4b7f994994e73872eb6fecb1", uri="sip:601@asterisk", response="c0a0e198fec2d4826579bc2903cf5b3f", algorithm=MD5, cnonce="0a4f113b", opaque="6cafe9741c002349", qop=auth, nc=00000001
Content-Type: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 430
v=0
o=dnake 1553834900 1553834900 IN IP4 192.168.1.161
s=dnake
c=IN IP4 192.168.1.161
t=0 0
m=audio 6000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
a=sendrecv
m=video 6200 RTP/AVP 104
a=rtpmap:104 H264/90000
a=fmtp:104 profile-level-id[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '104': Could not create dialog to invalid URI '104'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '104'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
== Setting global variable 'SIPDOMAIN' to 'asterisk'
<--- Transmitting SIP response (267 bytes) to UDP:192.168.1.161:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.161:5060;rport=5060;received=192.168.1.161;branch=z9hG4bK365732804
Call-ID: 1753293872
From: <sip:161@asterisk>;tag=69824255
To: <sip:601@asterisk>
CSeq: 21 INVITE
Server: Asterisk PBX 15.2.2
Content-Length: 0
[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '105': Could not create dialog to invalid URI '105'. Is endpoint registered and reachable?
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '105'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
== Using SIP RTP Video TOS bits 136
== Using SIP RTP Video TOS bits 136 in TCLASS field.
[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '106': Could not create dialog to invalid URI '106'. Is endpoint registered and reachable?
-- Executing [601@default:1] NoOp("PJSIP/161-000000ba", "EXECUTING 601 call") in new stack
-- Executing [601@default:2] Dial("PJSIP/161-000000ba", "PJSIP/101&PJSIP/102&PJSIP/103&PJSIP/104&PJSIP/105&PJSIP/106&PJSIP/107&PJSIP/108&PJSIP/109&PJSIP/110&PJSIP/111&PJSIP/112&PJSIP/113&PJSIP/114&PJSIP/115&PJSIP/116&Local/mobilephones@default") in new stack
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '106'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '107': Could not create dialog to invalid URI '107'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '107'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '108': Could not create dialog to invalid URI '108'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '108'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '109': Could not create dialog to invalid URI '109'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '109'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '110': Could not create dialog to invalid URI '110'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '110'
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '101': Could not create dialog to invalid URI '101'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '101'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '111': Could not create dialog to invalid URI '111'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '111'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '112': Could not create dialog to invalid URI '112'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '112'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '113': Could not create dialog to invalid URI '113'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '113'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '114': Could not create dialog to invalid URI '114'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '114'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '115': Could not create dialog to invalid URI '115'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '115'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '116': Could not create dialog to invalid URI '116'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '116'
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[2745][C-00000001]: translate.c:1358 ast_translator_best_choice: Cannot determine best translation path since one capability supports no formats
[Apr 2 09:19:50] WARNING[2745][C-00000001]: channel.c:6120 request_channel: No translator path exists for channel type Local (native (codec2|g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|vp9|red|t140|t38|silk|silk|silk|silk)) to (nothing)
[Apr 2 09:19:50] WARNING[2745][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'Local' (cause 58 - Bearer capability not available)
== 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.
-- Called PJSIP/102
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
-- Called PJSIP/103
== Using SIP RTP Video TOS bits 136
== Using SIP RTP Video TOS bits 136 in TCLASS field.
-- PJSIP/103-000000b9 connected line has changed. Saving it until answer for Message/ast_msg_queue
<--- Transmitting SIP request (1142 bytes) to UDP:192.168.1.243:54583 --->
INVITE sip:103@192.168.1.243:54583 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPjc26dff16-83a8-486f-9c90-a18b4dd78c78
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=d628cfe8-e045-4f5e-a262-8cbec0ec7489
To: <sip:103@192.168.1.243>
Contact: <sip:asterisk@192.168.1.18:5060>
Call-ID: ed974515-9563-4a8c-9889-f5b520e02ba8
CSeq: 30497 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
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Type: application/sdp
Content-Length: 469
v=0
o=- 1048728448 1048728448 IN IP4 192.168.1.18
s=Asterisk
c=IN IP4 192.168.1.18
t=0 0
m=audio 35104 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 G726-32/8000
a=rtpmap:101 telephone-e<--- Transmitting SIP request (1139 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=z9hG4bKPjb083ae9a-700a-437d-8d52-b6e0c0d98d7d
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9003d673-da48-41f5-b4f9-e3d92a8a045f
To: <sip:102@192.168.1.236>
Contact: <sip:asterisk@192.168.1.18:5060>
Call-ID: 13decb8d-401d-4e1c-b6d1-6e436492a180
CSeq: 24199 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
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Type: application/sdp
Content-Length: 467
v=0
o=- 482859899 482859899 IN IP4 192.168.1.18
s=Asterisk
c=IN IP4 192.168.1.18
t=0 0
m=audio 34878 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 G726-32/8000
a=rtpmap:101 telephone-event -- PJSIP/102-000000b8 connected line has changed. Saving it until answer for Message/ast_msg_queue
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '104': Could not create dialog to invalid URI '104'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '104'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '105': Could not create dialog to invalid URI '105'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '105'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '106': Could not create dialog to invalid URI '106'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '106'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '107': Could not create dialog to invalid URI '107'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '107'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[24758]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '108': Could not create dialog to invalid URI '108'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[24758]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '108'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[24758]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '109': Could not create dialog to invalid URI '109'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[24758]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '109'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[24758]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '110': Could not create dialog to invalid URI '110'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[24758]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '110'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[24758]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '111': Could not create dialog to invalid URI '111'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[24758]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '111'
<--- Received SIP response (503 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=z9hG4bKPjb083ae9a-700a-437d-8d52-b6e0c0d98d7d
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9003d673-da48-41f5-b4f9-e3d92a8a045f
To: <sip:102@192.168.1.236>
Call-ID: 13decb8d-401d-4e1c-b6d1-6e436492a180
CSeq: 24199 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
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '112': Could not create dialog to invalid URI '112'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '112'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '113': Could not create dialog to invalid URI '113'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '113'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[12748]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '114': Could not create dialog to invalid URI '114'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[12748]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '114'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '115': Could not create dialog to invalid URI '115'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '115'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
[Apr 2 09:19:50] ERROR[23109]: res_pjsip.c:3163 ast_sip_create_dialog_uac: Endpoint '116': Could not create dialog to invalid URI '116'. Is endpoint registered and reachable?
[Apr 2 09:19:50] ERROR[23109]: chan_pjsip.c:2475 request: Failed to create outgoing session to endpoint '116'
[Apr 2 09:19:50] WARNING[25266][C-00000038]: app_dial.c:2510 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
-- Called PJSIP/102
== 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.
-- Called PJSIP/103
-- Called Local/mobilephones@default
-- Executing [mobilephones@default:1] Ringing("Local/mobilephones@default-0000001d;2", "") in new stack
== Using SIP RTP Video TOS bits 136
== Using SIP RTP Video TOS bits 136 in TCLASS field.
-- Executing [mobilephones@default:2] System("Local/mobilephones@default-0000001d;2", "/bin/sleep 6") in new stack
-- PJSIP/102-000000bb connected line has changed. Saving it until answer for PJSIP/161-000000ba
-- Local/mobilephones@default-0000001d;1 is ringing
<--- Transmitting SIP response (454 bytes) to UDP:192.168.1.161:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.161:5060;rport=5060;received=192.168.1.161;branch=z9hG4bK365732804
Call-ID: 1753293872
From: <sip:161@asterisk>;tag=69824255
To: <sip:601@asterisk>;tag=4b6d3cbd-6020-4f34-ae3b-db0812529718
CSeq: 21 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
Content-Length: 0
-- PJSIP/103-000000bc connected line has changed. Saving it until answer for PJSIP/161-000000ba
<--- Transmitting SIP request (1094 bytes) to UDP:192.168.1.243:54583 --->
INVITE sip:103@192.168.1.243:54583 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPjb1423509-4f17-4c1d-b151-d8a4d73285c2
From: "161" <sip:Main_door@192.168.1.18>;tag=a6474041-3b14-48e9-9c99-9d566c641b75
To: <sip:103@192.168.1.243>
Contact: <sip:asterisk@192.168.1.18:5060>
Call-ID: 91de4ca1-4a14-4208-b7b2-8c97bc44be7b
CSeq: 32140 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: "161" <sip:Main_door@192.168.1.18>
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Type: application/sdp
Content-Length: 375
v=0
o=- 790870005 790870005 IN IP4 192.168.1.18
s=Asterisk
c=IN IP4 192.168.1.18
t=0 0
m=audio 38192 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpma<--- Transmitting SIP request (1092 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=z9hG4bKPj700a280e-98ba-40a2-a69f-61f424371b7d
From: "161" <sip:Main_door@192.168.1.18>;tag=7aa17525-f030-431c-a7cf-2d3d6b9003fe
To: <sip:102@192.168.1.236>
Contact: <sip:asterisk@192.168.1.18:5060>
Call-ID: 519c691b-61fc-40fc-a5e9-8bf119b968f2
CSeq: 4556 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: "161" <sip:Main_door@192.168.1.18>
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Type: application/sdp
Content-Length: 375
v=0
o=- 753944810 753944810 IN IP4 192.168.1.18
s=Asterisk
c=IN IP4 192.168.1.18
t=0 0
m=audio 30542 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:1<--- Received SIP response (491 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=z9hG4bKPj700a280e-98ba-40a2-a69f-61f424371b7d
From: "161" <sip:Main_door@192.168.1.18>;tag=7aa17525-f030-431c-a7cf-2d3d6b9003fe
To: <sip:102@192.168.1.236>
Call-ID: 519c691b-61fc-40fc-a5e9-8bf119b968f2
CSeq: 4556 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 (591 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=z9hG4bKPjb083ae9a-700a-437d-8d52-b6e0c0d98d7d
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=9003d673-da48-41f5-b4f9-e3d92a8a045f
To: <sip:102@192.168.1.236>;tag=1054244180
Call-ID: 13decb8d-401d-4e1c-b6d1-6e436492a180
CSeq: 24199 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/102-000000b8 is ringing
<--- Received SIP response (578 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=z9hG4bKPj700a280e-98ba-40a2-a69f-61f424371b7d
From: "161" <sip:Main_door@192.168.1.18>;tag=7aa17525-f030-431c-a7cf-2d3d6b9003fe
To: <sip:102@192.168.1.236>;tag=184018702
Call-ID: 519c691b-61fc-40fc-a5e9-8bf119b968f2
CSeq: 4556 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/102-000000bb is ringing
<--- Received SIP response (499 bytes) from UDP:192.168.1.243:54583 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPjc26dff16-83a8-486f-9c90-a18b4dd78c78
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=d628cfe8-e045-4f5e-a262-8cbec0ec7489
To: <sip:103@192.168.1.243>
Call-ID: ed974515-9563-4a8c-9889-f5b520e02ba8
CSeq: 30497 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.26
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<--- Received SIP response (488 bytes) from UDP:192.168.1.243:54583 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPjb1423509-4f17-4c1d-b151-d8a4d73285c2
From: "161" <sip:Main_door@192.168.1.18>;tag=a6474041-3b14-48e9-9c99-9d566c641b75
To: <sip:103@192.168.1.243>
Call-ID: 91de4ca1-4a14-4208-b7b2-8c97bc44be7b
CSeq: 32140 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.26
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<--- Received SIP response (588 bytes) from UDP:192.168.1.243:54583 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPjc26dff16-83a8-486f-9c90-a18b4dd78c78
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=d628cfe8-e045-4f5e-a262-8cbec0ec7489
To: <sip:103@192.168.1.243>;tag=1148571086
Call-ID: ed974515-9563-4a8c-9889-f5b520e02ba8
CSeq: 30497 INVITE
Contact: <sip:103@192.168.1.243:54583>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.26
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
-- PJSIP/103-000000b9 is ringing
<--- Received SIP response (577 bytes) from UDP:192.168.1.243:54583 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPjb1423509-4f17-4c1d-b151-d8a4d73285c2
From: "161" <sip:Main_door@192.168.1.18>;tag=a6474041-3b14-48e9-9c99-9d566c641b75
To: <sip:103@192.168.1.243>;tag=1037328968
Call-ID: 91de4ca1-4a14-4208-b7b2-8c97bc44be7b
CSeq: 32140 INVITE
Contact: <sip:103@192.168.1.243:54583>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.26
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
-- PJSIP/103-000000bc is ringing
<--- 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=z9hG4bK981381306
From: <sip:161@192.168.1.18>;tag=452779857
To: <sip:192.168.1.18>
Call-ID: 1849217048
CSeq: 20 OPTIONS
Accept: application/sdp
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 0