[SOLVED]Asterisk can't answer the phone now

#1

31/3/2019 21:34 HKT
i asked for carrier and repaired.

yesterday it can answer .

IP Debugging enabled

<--- SIP read from UDP:carrier-ims-ipv4.17:5060 --->
INVITE sip:s@Asterisk-IPv4:5060 SIP/2.0
Via: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
Call-ID: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
From: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
To: <tel:+8xmycallerid>
CSeq: 1000 INVITE
Max-Forwards: 64
Supported: 100rel,histinfo,timer
Accept: application/sdp,application/isup,multipart/mixed,application/dtmf,application/dtmf-relay
Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY
Privacy: id
User-Agent: ZTE-MGCF
Min-SE: 90
Contact: <sip:Anonymous@carrier-ims-ipv4.17:5060;zte-did=9-7-20481-3639-12-330-65535>
P-Called-Party-ID: <sip:+8xmycallerid@ims.domain>
Session-Expires: 1800;refresher=uac
X-ZTE-Cookie: 7zs4rm4;id=_WwEf5FvZvEWrvInJFMkQbDQ@mgcf22.ims.domain
Content-Length: 157
Conten▒--- (21 headers 8 lines) ---
Sending to carrier-ims-ipv4.17:5060 (no NAT)
Sending to carrier-ims-ipv4.17:5060 (no NAT)
Using INVITE request as basis request - asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
Found peer 'trunk_ims2' for 'Anonymous' from carrier-ims-ipv4.17:5060
[Mar 31 07:14:06] ERROR[1222][C-00000001]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("OpenWrt", "(null)", ...): Name does not resolve
[Mar 31 07:14:06] WARNING[1222][C-00000001]: acl.c:833 resolve_first: Unable to lookup 'OpenWrt'
[Mar 31 07:14:06] ERROR[1222][C-00000001]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo("OpenWrt", "(null)", ...): Name does not resolve
[Mar 31 07:14:06] WARNING[1222][C-00000001]: acl.c:833 resolve_first: Unable to lookup 'OpenWrt'
Found RTP audio format 8
Found audio description format PCMA for ID 8
Capabilities: us - (ulaw|alaw|gsm|h263|ilbc|h263p|h264), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port carrier-ims-ipv4.48:42684
Peer doesn't provide video
Looking for s in from-trunk2 (domain Asterisk-IPv4)
sip_route_dump: route/path hop: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>

<--- Transmitting (no NAT) to carrier-ims-ipv4.17:5060 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
l: 0


<------------>
Audio is at 15532
Adding codec alaw to SDP

<--- Reliably Transmitting (no NAT) to carrier-ims-ipv4.17:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
c: application/sdp
Require: timer
l: 2777

v=0
o=root 2016854071 2016854071 IN IP4 Asterisk-IPv4
s=Asterisk PBX 15.3.0
c=IN IP4 Asterisk-IPv4
t=0 0
m=audio 15532 RTP/AVP 8
a=maxptime:150
a=ice-ufrag:5ae5e49c2ff2f99f453e316c4c5e329c
a=ice-pwd:73f435282cb3f98d2b64e5bf709b8937
a=candidate:H2aab888d 1 UDP 213070Retransmitting #1 (no NAT) to carrier-ims-ipv4.17:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
c: application/sdp
Require: timer
l: 2777

v=0
o=root 2016854071 2016854071 IN IP4 Asterisk-IPv4
s=Asterisk PBX 15.3.0
c=IN IP4 Asterisk-IPv4
t=0 0
m=audio 15532 RTP/AVP 8
a=maxptime:150
a=ice-ufrag:5ae5e49c2ff2f99f453e316c4c5e329c
a=ice-pwd:73f435282cb3f98d2b64e5bf709b8937
a=candidate:H2aab888d 1 UDP 2130706431 fe80::7aa3Retransmitting #2 (no NAT) to carrier-ims-ipv4.17:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
c: application/sdp
Require: timer
l: 2777

v=0
o=root 2016854071 2016854071 IN IP4 Asterisk-IPv4
s=Asterisk PBX 15.3.0
c=IN IP4 Asterisk-IPv4
t=0 0
m=audio 15532 RTP/AVP 8
a=maxptime:150
a=ice-ufrag:5ae5e49c2ff2f99f453e316c4c5e329c
a=ice-pwd:73f435282cb3f98d2b64e5bf709b8937
a=candidate:H2aab888d 1 UDP 2130706431 fe80::7aa3Retransmitting #3 (no NAT) to carrier-ims-ipv4.17:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
c: application/sdp
Require: timer
l: 2777

v=0
o=root 2016854071 2016854071 IN IP4 Asterisk-IPv4
s=Asterisk PBX 15.3.0
c=IN IP4 Asterisk-IPv4
t=0 0
m=audio 15532 RTP/AVP 8
a=maxptime:150
a=ice-ufrag:5ae5e49c2ff2f99f453e316c4c5e329c
a=ice-pwd:73f435282cb3f98d2b64e5bf709b8937
a=candidate:H2aab888d 1 UDP 2130706431 fe80::7aa3Retransmitting #4 (no NAT) to carrier-ims-ipv4.17:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
c: application/sdp
Require: timer
l: 2777

v=0
o=root 2016854071 2016854071 IN IP4 Asterisk-IPv4
s=Asterisk PBX 15.3.0
c=IN IP4 Asterisk-IPv4
t=0 0
m=audio 15532 RTP/AVP 8
a=maxptime:150
a=ice-ufrag:5ae5e49c2ff2f99f453e316c4c5e329c
a=ice-pwd:73f435282cb3f98d2b64e5bf709b8937
a=candidate:H2aab888d 1 UDP 2130706431 fe80::7aa3Retransmitting #5 (no NAT) to carrier-ims-ipv4.17:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
c: application/sdp
Require: timer
l: 2777

v=0
o=root 2016854071 2016854071 IN IP4 Asterisk-IPv4
s=Asterisk PBX 15.3.0
c=IN IP4 Asterisk-IPv4
t=0 0
m=audio 15532 RTP/AVP 8
a=maxptime:150
a=ice-ufrag:5ae5e49c2ff2f99f453e316c4c5e329c
a=ice-pwd:73f435282cb3f98d2b64e5bf709b8937
a=candidate:H2aab888d 1 UDP 2130706431 fe80::7aa3
<--- SIP read from UDP:carrier-ims-ipv4.17:5060 --->
CANCEL sip:s@Asterisk-IPv4:5060 SIP/2.0
Via: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384
Call-ID: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
From: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
To: <tel:+8xmycallerid>
CSeq: 1000 CANCEL
Reason: Q.850;cause=18;text="No user responding";sss-internal
Max-Forwards: 70
X-ZTE-Cause: "PSS-9.7.0x8E000326.mmtel23.ims.domain"
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to carrier-ims-ipv4.17:5060 (no NAT)

<--- Transmitting (no NAT) to carrier-ims-ipv4.17:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 CANCEL
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
l: 0


<------------>
Retransmitting #6 (no NAT) to carrier-ims-ipv4.17:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
c: application/sdp
Require: timer
l: 2777

v=0
o=root 2016854071 2016854071 IN IP4 Asterisk-IPv4
s=Asterisk PBX 15.3.0
c=IN IP4 Asterisk-IPv4
t=0 0
m=audio 15532 RTP/AVP 8
a=maxptime:150
a=ice-ufrag:5ae5e49c2ff2f99f453e316c4c5e329c
a=ice-pwd:73f435282cb3f98d2b64e5bf709b8937
a=candidate:H2aab888d 1 UDP 2130706431 fe80::7aa3Retransmitting #7 (no NAT) to carrier-ims-ipv4.17:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
c: application/sdp
Require: timer
l: 2777

v=0
o=root 2016854071 2016854071 IN IP4 Asterisk-IPv4
s=Asterisk PBX 15.3.0
c=IN IP4 Asterisk-IPv4
t=0 0
m=audio 15532 RTP/AVP 8
a=maxptime:150
a=ice-ufrag:5ae5e49c2ff2f99f453e316c4c5e329c
a=ice-pwd:73f435282cb3f98d2b64e5bf709b8937
a=candidate:H2aab888d 1 UDP 2130706431 fe80::7aa3Retransmitting #8 (no NAT) to carrier-ims-ipv4.17:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
c: application/sdp
Require: timer
l: 2777

v=0
o=root 2016854071 2016854071 IN IP4 Asterisk-IPv4
s=Asterisk PBX 15.3.0
c=IN IP4 Asterisk-IPv4
t=0 0
m=audio 15532 RTP/AVP 8
a=maxptime:150
a=ice-ufrag:5ae5e49c2ff2f99f453e316c4c5e329c
a=ice-pwd:73f435282cb3f98d2b64e5bf709b8937
a=candidate:H2aab888d 1 UDP 2130706431 fe80::7aa3Retransmitting #9 (no NAT) to carrier-ims-ipv4.17:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
c: application/sdp
Require: timer
l: 2777

v=0
o=root 2016854071 2016854071 IN IP4 Asterisk-IPv4
s=Asterisk PBX 15.3.0
c=IN IP4 Asterisk-IPv4
t=0 0
m=audio 15532 RTP/AVP 8
a=maxptime:150
a=ice-ufrag:5ae5e49c2ff2f99f453e316c4c5e329c
a=ice-pwd:73f435282cb3f98d2b64e5bf709b8937
a=candidate:H2aab888d 1 UDP 2130706431 fe80::7aa3Retransmitting #10 (no NAT) to carrier-ims-ipv4.17:5060:
SIP/2.0 200 OK
v: SIP/2.0/UDP carrier-ims-ipv4.17:5060;branch=z9hG4bK*6-5-16648-1521-12-330-0*1NGil7Qm5bhehbhjccg.6T12384;received=carrier-ims-ipv4.17
Record-Route: <sip:carrier-ims-ipv4.17:5060;transport=udp;lr>
f: "Anonymous"<sip:Anonymous@anonymous.invalid>;tag=sbc1205ztesipoT5da5GG*9-7-20481*jfbi.9
t: <tel:+8xmycallerid>;tag=as5e4b75b8
i: asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims
CSeq: 1000 INVITE
Server: (null)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
x: 1800;refresher=uac
m: <sip:s@Asterisk-IPv4:5060>
c: application/sdp
Require: timer
l: 2777

v=0
o=root 2016854071 2016854071 IN IP4 Asterisk-IPv4
s=Asterisk PBX 15.3.0
c=IN IP4 Asterisk-IPv4
t=0 0
m=audio 15532 RTP/AVP 8
a=maxptime:150
a=ice-ufrag:5ae5e49c2ff2f99f453e316c4c5e329c
a=ice-pwd:73f435282cb3f98d2b64e5bf709b8937
a=candidate:H2aab888d 1 UDP 2130706431 fe80::7aa[Mar 31 07:14:38] WARNING[1222]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims for seqno 1000 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31998ms with no response
[Mar 31 07:14:38] WARNING[1222]: chan_sip.c:4096 retrans_pkt: Hanging up call asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Really destroying SIP dialog 'asbcHMGep2s_L6YpROVs1nPV8zG_QGSSTCVbuYjKF0kVqYTicJdi6fTDg7chda@zteims' Method: CANCEL


Reason: Q.850;cause=18;text="No user responding";sss-internal
Max-Forwards: 70
X-ZTE-Cause: "PSS-9.7.0x8E000326.mmtel23.ims"
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#2

Asterisk has answered. The carrier appears not t be receiving that answer.

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#3

oh , should i call them for repair?

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#4

yes. i call for carrier for reset my status

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