Incoming Calls Not Coming Properly in SIP Asterisk

Asterisk Asterisk General
1

Hi Team,

Hope you all are doing well…

We need Your Support for incoming call issue in Asterisk Server.

I have installed Asterisk 11.8.1 in Centos 6.5 with Core I3 Processor which has 4GB RAM and 500GB Hard-Drive.

I have taken PRI-Lines from Tata Teletel India Telecom Company. I have mapped our Asterisk server with SIP Trunk. When our front-end customer wants to dialed on our PRI(Pilot) number i.e ‘011-12345678’ its get connected successfully with my asterisk machine and proper IVR Plays and entire application working fine in which out-bound called happened. But “some time” or “some fraction of time slot” some callers again wants to connect with our PRI server they can’t able to make call on my PRI, if they called it from mobile number call-disconnected within 1 second if they called it from Landline or pstn it’s say Dialed number doesn’t exist ,Please check the number’ and I am not able to see any SIP logs on my asterisk server after 30 to 40 second its works fine. All incoming calls get connected successfully and no issue But this issue come 20 to 30 times in a Day and our customer would irritates just because of call not connected:
Please suggest me what should I do to resolve this issue

Below is my SIP Logs
Scheduling destruction of SIP dialog ['038d200307191120@XX.X.XX.XX](mailto:'038d200307191120@XX.X.XX.XX)' in 32000 ms (Method: OPTIONS) Really destroying SIP dialog ['038d200307191120@XX.X.XX.XX](mailto:'038d200307191120@XX.X.XX.XX)' Method: OPTIONS [2015-02-06 16:27:09] NOTICE[24508]: chan_sip.c:15059 sip_reregister: -- Re-registration for [XXXXXXX@XX.X.XX.XX](mailto:XXXXXXX@XX.X.XX.XX) REGISTER 11 headers, 0 lines Reliably Transmitting (no NAT) to XX.X.XX.XX:5060: REGISTER sip:XX.X.XX.XX SIP/2.0 Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK192c93fb Max-Forwards: 70 From: <sip:XXXXXXX@XX.X.XX.XX>;tag=as645301b7 To: <sip:XXXXXXX@XX.X.XX.XX> Call-ID: 18e1b9d23f9158fb259b22566ae35ca3@[::1]
CSeq: 778 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XX”, nonce=“67a22382fd5f88b5a1a239e8d69884ad”, response="7be6c3df2c8f02824fbeedd1c0b1d900"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0
<— SIP read from UDP:XX.X.XX.XX:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK192c93fb
To: sip:XXXXXXX@XX.X.XX.XX
From: sip:XXXXXXX@XX.X.XX.XX;tag=as645301b7
Call-ID: 18e1b9d23f9158fb259b22566ae35ca3@[::1]
CSeq: 778 REGISTER
User-Agent: ZTE-SoftSwitch
Content-Length: 0
WWW-Authenticate: Digest realm=“SIP-XXXXXXX”,nonce=“225e999d00e77800bc32c8c51079444c”,ZTE-ID=a59211495dab314eb5ded58cc9e4e6e1
<------------->
— (9 headers 0 lines) —
Responding to challenge, registration to domain/host name XX.X.XX.XX
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to XX.X.XX.XX:5060:
REGISTER sip:XX.X.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK102ef31e
Max-Forwards: 70
From: sip:XXXXXXX@XX.X.XX.XX;tag=as645301b7
To: sip:XXXXXXX@XX.X.XX.XX
Call-ID: 18e1b9d23f9158fb259b22566ae35ca3@[::1]
CSeq: 779 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XX”, nonce=“225e999d00e77800bc32c8c51079444c”, response="23237f4c532606a0bd61462ab6cdbb10"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0
<— SIP read from UDP:XX.X.XX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK102ef31e
To: sip:XXXXXXX@XX.X.XX.XX
From: sip:XXXXXXX@XX.X.XX.XX;tag=as645301b7
Call-ID: 18e1b9d23f9158fb259b22566ae35ca3@[::1]
CSeq: 779 REGISTER
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060;expires=300
User-Agent: ZTE-SoftSwitch
Date: Fri, 06 Feb 2015 16:28:57 GMT
Content-Length: 0
<------------->
— (10 headers 0 lines) —
[2015-02-06 16:27:09] NOTICE[24508]: chan_sip.c:23519 handle_response_register: Outbound Registration: Expiry for XX.X.XX.XX is 300 sec (Scheduling reregistration in 285 s)
Really destroying SIP dialog ‘18e1b9d23f9158fb259b22566ae35ca3@[::1]’ Method: REGISTER
<— SIP read from UDP:XX.X.XX.XX:5060 —>
OPTIONS sip:XXXXXXX@XX.X.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK1d2e39ec06ce4bdctaN0
Max-Forwards: 70
To: sip:XXXXXXX@XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
Call-ID: 216d200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Accept: application/sdp
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Sending to XX.X.XX.XX:5060 (no NAT)
Looking for XXXXXXX in from-general (domain XX.X.XX.XXX)
<— Transmitting (no NAT) to XX.X.XX.XX:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK1d2e39ec06ce4bdctaN0;received=XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
To: sip:XXXXXXX@XX.X.XX.XX;tag=as5fe200cf
Call-ID: 216d200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '216d200307191120@XX.X.XX.XX’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '216d200307191120@XX.X.XX.XX’ Method: OPTIONS
asterisk_serverCLI>
asterisk_serverCLI>
[2015-02-06 16:31:54] NOTICE[24508]: chan_sip.c:15059 sip_reregister: – Re-registration for XXXXXXX@XX.X.XX.XX
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to XX.X.XX.XX:5060:
REGISTER sip:XX.X.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK4277869c
Max-Forwards: 70
From: sip:XXXXXXX@XX.X.XX.XX;tag=as645301b7
To: sip:XXXXXXX@XX.X.XX.XX
Call-ID: 18e1b9d23f9158fb259b22566ae35ca3@[::1]
CSeq: 780 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XX”, nonce=“225e999d00e77800bc32c8c51079444c”, response="23237f4c532606a0bd61462ab6cdbb10"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0
<— SIP read from UDP:XX.X.XX.XX:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK4277869c
To: sip:XXXXXXX@XX.X.XX.XX
From: sip:XXXXXXX@XX.X.XX.XX;tag=as645301b7
Call-ID: 18e1b9d23f9158fb259b22566ae35ca3@[::1]
CSeq: 780 REGISTER
User-Agent: ZTE-SoftSwitch
Content-Length: 0
WWW-Authenticate: Digest realm=“SIP-XXXXXXX”,nonce=“227a17cfad5fe1450b8d4fdaead6a124”,ZTE-ID=123545f9287fd760cf11b2a18f16981f

<------------->
— (9 headers 0 lines) —
Responding to challenge, registration to domain/host name XX.X.XX.XX
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to XX.X.XX.XX:5060:
REGISTER sip:XX.X.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK6b89cc96
Max-Forwards: 70
From: sip:XXXXXXX@XX.X.XX.XX;tag=as645301b7
To: sip:XXXXXXX@XX.X.XX.XX
Call-ID: 18e1b9d23f9158fb259b22566ae35ca3@[::1]
CSeq: 781 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XX”, nonce=“227a17cfad5fe1450b8d4fdaead6a124”, response="8639103a65278ce6176708745da06ccc"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

<— SIP read from UDP:XX.X.XX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK6b89cc96
To: sip:XXXXXXX@XX.X.XX.XX
From: sip:XXXXXXX@XX.X.XX.XX;tag=as645301b7
Call-ID: 18e1b9d23f9158fb259b22566ae35ca3@[::1]
CSeq: 781 REGISTER
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060;expires=300
User-Agent: ZTE-SoftSwitch
Date: Fri, 06 Feb 2015 16:33:42 GMT
Content-Length: 0

<------------->
— (10 headers 0 lines) —
[2015-02-06 16:31:54] NOTICE[24508]: chan_sip.c:23519 handle_response_register: Outbound Registration: Expiry for XX.X.XX.XX is 300 sec (Scheduling reregistration in 285 s)
Really destroying SIP dialog ‘18e1b9d23f915enter code here8fb259b22566ae35ca3@[::1]’ Method: REGISTER `

2

Hi Team ,
Please suggest me the solution on priority basis.

3

This is the wrong forum for support requests.

For priority requests, please use the Biz and Jobs forum and indicate how much you are prepared to pay. For non-priority, peer support, requests, use the Asterisk Support forum.

There are no errors in the SIP trace and SIP is not PRI. There aren’t even any calls in the trace.

4

Hi Team,

Below is the SIP Trace logs for more clarity:
[2015-02-09 10:57:04] NOTICE[2218]: chan_sip.c:15059 sip_reregister: – Re-registration for XXXXXXX@XX.X.XX.XXX
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to XX.X.XX.XXX:5060:
REGISTER sip:XX.X.XX.XXX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK7d85152e
Max-Forwards: 70
From: sip:XXXXXXX@XX.X.XX.XXX;tag=as570c1dcb
To: sip:XXXXXXX@XX.X.XX.XXX
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3468 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XXX”, nonce=“8023d7ac165d1c6df66ba53f525b1a15”, response="e1f9bbcf3512f0bbaf3cc7b9719665cf"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

<— SIP read from UDP:XX.X.XX.XXX:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK7d85152e
To: sip:XXXXXXX@XX.X.XX.XXX
From: sip:XXXXXXX@XX.X.XX.XXX;tag=as570c1dcb
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3468 REGISTER
User-Agent: ZTE-SoftSwitch
Content-Length: 0
WWW-Authenticate: Digest realm=“SIP-XXXXXXX”,nonce=“bc721f1ac13395545d2fc282824e73ed”,ZTE-ID=cad530ad1690262d494e00ce1577da0d

<------------->
— (9 headers 0 lines) —
Responding to challenge, registration to domain/host name XX.X.XX.XXX
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to XX.X.XX.XXX:5060:
REGISTER sip:XX.X.XX.XXX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK102e23d7
Max-Forwards: 70
From: sip:XXXXXXX@XX.X.XX.XXX;tag=as570c1dcb
To: sip:XXXXXXX@XX.X.XX.XXX
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3469 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XXX”, nonce=“bc721f1ac13395545d2fc282824e73ed”, response="6e35df47959458488a2a4825f32b553f"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

<— SIP read from UDP:XX.X.XX.XXX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK102e23d7
To: sip:XXXXXXX@XX.X.XX.XXX
From: sip:XXXXXXX@XX.X.XX.XXX;tag=as570c1dcb
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3469 REGISTER
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060;expires=300
User-Agent: ZTE-SoftSwitch
Date: Mon, 09 Feb 2015 10:58:47 GMT
Content-Length: 0

<------------->
— (10 headers 0 lines) —
[2015-02-09 10:57:04] NOTICE[2218]: chan_sip.c:23519 handle_response_register: Outbound Registration: Expiry for XX.X.XX.XXX is 300 sec (Scheduling reregistration in 285 s)
Really destroying SIP dialog ‘13c2f04307b1b3fd3b9742ac090a437f@[::1]’ Method: REGISTER

<— SIP read from UDP:XX.X.XX.XXX:5060 —>
INVITE sip:XXXXXXX@XX.X.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK53416181987902e300f0taN0
To: "124XXXXXXX"sip:124XXXXXXX@XX.X.XX.XXX
From: "0XXXXXXXXXX"sip:XXXXXXXXXX@XX.X.XX.XXX;tag=aa2c906-cXvn103886ebd2
Call-ID: 75ac103886ec-0210-0200@10.162.201.6
CSeq: 27041 INVITE
Max-Forwards: 7
Contact: sip:XXXXXXXXXX@XX.X.XX.XXX
Supported: 100rel
User-Agent: ZTE Softswitch/1.0.0
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Content-Type: application/sdp
Content-Length: 177

v=0
o=ZTE 320 16624 IN IP4 XX.X.XX.XXX
s=phone-call
c=IN IP4 XX.X.XX.XXX
t=0 0
m=audio 38118 RTP/AVP 8 0 18 97
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=ptime:20
<------------->
— (13 headers 9 lines) —
Sending to XX.X.XX.XXX:5060 (no NAT)
Sending to XX.X.XX.XXX:5060 (no NAT)
Using INVITE request as basis request - 75ac103886ec-0210-0200@10.162.201.6
Found peer ‘tatasip’ for ‘XXXXXXXXXX’ from XX.X.XX.XXX:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 97
Found audio description format telephone-event for ID 97
Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port XX.X.XX.XXX:38118
Looking for XXXXXXX in tata (domain XX.X.XX.XXX)
list_route: hop: sip:XXXXXXXXXX@XX.X.XX.XXX

<— Transmitting (no NAT) to XX.X.XX.XXX:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK53416181987902e300f0taN0;received=XX.X.XX.XXX
From: "0XXXXXXXXXX"sip:XXXXXXXXXX@XX.X.XX.XXX;tag=aa2c906-cXvn103886ebd2
To: "124XXXXXXX"sip:124XXXXXXX@XX.X.XX.XXX
Call-ID: 75ac103886ec-0210-0200@10.162.201.6
CSeq: 27041 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

<------------>

5

One More SIP Trace logs for your referenc :

<— SIP read from UDP:XX.X.XX.XX:5060 —>
OPTIONS sip:XXXXXXX@XX.X.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK100a543d0edb6454taN0
Max-Forwards: 70
To: sip:XXXXXXX@XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
Call-ID: 494f200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Accept: application/sdp
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Sending to XX.X.XX.XX:5060 (no NAT)
Looking for XXXXXXX in from-general (domain XX.X.XX.XXX)

<— Transmitting (no NAT) to XX.X.XX.XX:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK100a543d0edb6454taN0;received=XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
To: sip:XXXXXXX@XX.X.XX.XX;tag=as7592e6a2
Call-ID: 494f200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '494f200307191120@XX.X.XX.XX’ in 32000 ms (Method: OPTIONS)
localhostCLI>
localhostCLI>

<— SIP read from UDP:XX.X.XX.XX:5060 —>
INVITE sip:XXXXXXX@XX.X.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bKc8b2bf967ac2ea7bccabtaN0
To: "124XXXXXXX"sip:124XXXXXXX@XX.X.XX.XX
From: "0XXXXXXXXXX"sip:XXXXXXXXXX@XX.X.XX.XX;tag=aa2c906-EWOI10389f67d2
Call-ID: 2dfc10389f68-0210-0200@10.162.201.6
CSeq: 15055 INVITE
Max-Forwards: 7
Contact: sip:XXXXXXXXXX@XX.X.XX.XX
Supported: 100rel
User-Agent: ZTE Softswitch/1.0.0
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Content-Type: application/sdp
Content-Length: 178
v=0
o=ZTE 3241 27042 IN IP4 XX.X.XX.XX
s=phone-call
c=IN IP4 XX.X.XX.XX
t=0 0
m=audio 34822 RTP/AVP 8 0 18 97
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=ptime:20
<------------->
— (13 headers 9 lines) —
Sending to XX.X.XX.XX:5060 (no NAT)
Sending to XX.X.XX.XX:5060 (no NAT)
Using INVITE request as basis request - 2dfc10389f68-0210-0200@10.162.201.6
Found peer ‘tatasip’ for ‘XXXXXXXXXX’ from XX.X.XX.XX:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 97
Found audio description format telephone-event for ID 97
Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port XX.X.XX.XX:34822
Looking for XXXXXXX in tata (domain XX.X.XX.XXX)
list_route: hop: sip:XXXXXXXXXX@XX.X.XX.XX

<— Transmitting (no NAT) to XX.X.XX.XX:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bKc8b2bf967ac2ea7bccabtaN0;received=XX.X.XX.XX
From: "0XXXXXXXXXX"sip:XXXXXXXXXX@XX.X.XX.XX;tag=aa2c906-EWOI10389f67d2
To: "124XXXXXXX"sip:124XXXXXXX@XX.X.XX.XX
Call-ID: 2dfc10389f68-0210-0200@10.162.201.6
CSeq: 15055 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

6

One ore logs :

<— SIP read from UDP:XX.X.XX.XX:5060 —>
OPTIONS sip:XXXXXXX@XX.X.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK100a543d0edb6454taN0
Max-Forwards: 70
To: sip:XXXXXXX@XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
Call-ID: 494f200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Accept: application/sdp
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Sending to XX.X.XX.XX:5060 (no NAT)
Looking for XXXXXXX in from-general (domain XX.X.XX.XXX)

<— Transmitting (no NAT) to XX.X.XX.XX:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK100a543d0edb6454taN0;received=XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
To: sip:XXXXXXX@XX.X.XX.XX;tag=as7592e6a2
Call-ID: 494f200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '494f200307191120@XX.X.XX.XX’ in 32000 ms (Method: OPTIONS)
localhostCLI>
localhostCLI>

<— SIP read from UDP:XX.X.XX.XX:5060 —>
INVITE sip:XXXXXXX@XX.X.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bKc8b2bf967ac2ea7bccabtaN0
To: "124XXXXXXX"sip:124XXXXXXX@XX.X.XX.XX
From: "0XXXXXXXXXX"sip:XXXXXXXXXX@XX.X.XX.XX;tag=aa2c906-EWOI10389f67d2
Call-ID: 2dfc10389f68-0210-0200@10.162.201.6
CSeq: 15055 INVITE
Max-Forwards: 7
Contact: sip:XXXXXXXXXX@XX.X.XX.XX
Supported: 100rel
User-Agent: ZTE Softswitch/1.0.0
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Content-Type: application/sdp
Content-Length: 178
v=0
o=ZTE 3241 27042 IN IP4 XX.X.XX.XX
s=phone-call
c=IN IP4 XX.X.XX.XX
t=0 0
m=audio 34822 RTP/AVP 8 0 18 97
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=ptime:20
<------------->
— (13 headers 9 lines) —
Sending to XX.X.XX.XX:5060 (no NAT)
Sending to XX.X.XX.XX:5060 (no NAT)
Using INVITE request as basis request - 2dfc10389f68-0210-0200@10.162.201.6
Found peer ‘tatasip’ for ‘XXXXXXXXXX’ from XX.X.XX.XX:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 97
Found audio description format telephone-event for ID 97
Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port XX.X.XX.XX:34822
Looking for XXXXXXX in tata (domain XX.X.XX.XXX)
list_route: hop: sip:XXXXXXXXXX@XX.X.XX.XX

<— Transmitting (no NAT) to XX.X.XX.XX:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bKc8b2bf967ac2ea7bccabtaN0;received=XX.X.XX.XX
From: "0XXXXXXXXXX"sip:XXXXXXXXXX@XX.X.XX.XX;tag=aa2c906-EWOI10389f67d2
To: "124XXXXXXX"sip:124XXXXXXX@XX.X.XX.XX
Call-ID: 2dfc10389f68-0210-0200@10.162.201.6
CSeq: 15055 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

7

There is no team; this is a peer support forum.

Your traces are incomplete. You need to go through to the end of the 2xx, 3xx, 4xx, 5xx or 6xx response, except that if you get 401, you need to go through to the end of the result of the repeat of the INVITE with authentication.

8

Hi David thanks for your support.

Now I am sharing complete SIP Trace logs with you

Scheduling destruction of SIP dialog '464db7d7773c847645f31d3a7c5d47a5@XX.X.XX.XXX:5060’ in 32000 ms (Method: ACK)
set_destination: Parsing sip:XXXXXXXXXX@XX.X.XX.XX for address/port to send to
set_destination: set destination to XX.X.XX.XX:5060
Reliably Transmitting (no NAT) to XX.X.XX.XX:5060:
BYE sip:XXXXXXXXXX@XX.X.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK766f7ff0
Max-Forwards: 70
From: "0XXXXXXXXXX"sip:XXXXXXX@XX.X.XX.XXX;tag=as3093bba2
To: sip:XXXXXXXXXX@XX.X.XX.XX;tag=aa2c906-S4fY1038f103d2
Call-ID: 464db7d7773c847645f31d3a7c5d47a5@XX.X.XX.XXX:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.8.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

== Spawn extension (macro-To-ADG-IBD-Call-Flow, transfer_number, 9) exited non-zero on ‘SIP/testsip-00000026’ in macro ‘To-ADG-IBD-Call-Flow’
== Spawn extension (tata, XXXXXXX, 1) exited non-zero on ‘SIP/testsip-00000026’

<— SIP read from UDP:XX.X.XX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK766f7ff0
To: sip:XXXXXXXXXX@XX.X.XX.XX;tag=aa2c906-S4fY1038f103d2
From: "0XXXXXXXXXX"sip:XXXXXXX@XX.X.XX.XXX;tag=as3093bba2
Call-ID: 464db7d7773c847645f31d3a7c5d47a5@XX.X.XX.XXX:5060
CSeq: 103 BYE
User-Agent: ZTE Softswitch/1.0.0
Content-Length: 0

<------------->
— (8 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '464db7d7773c847645f31d3a7c5d47a5@XX.X.XX.XXX:5060’ Method: ACK
Really destroying SIP dialog '1bda1038f0ec-0210-0200@XX.XXX.XXX.X’ Method: BYE
localhostCLI>
localhostCLI>
localhostCLI>
localhostCLI>
localhost*CLI>
[2015-02-09 15:18:22] NOTICE[2218]: chan_sip.c:15059 sip_reregister: – Re-registration for XXXXXXX@XX.X.XX.XX
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to XX.X.XX.XX:5060:
REGISTER sip:XX.X.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK5861d856
Max-Forwards: 70
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
To: sip:XXXXXXX@XX.X.XX.XX
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3578 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XX”, nonce=“f1576610e8abf77a141ef5509121f6b5”, response="dd0aa163e315367be19ddab43cabcaea"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

<— SIP read from UDP:XX.X.XX.XX:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK5861d856
To: sip:XXXXXXX@XX.X.XX.XX
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3578 REGISTER
User-Agent: ZTE-SoftSwitch
Content-Length: 0
WWW-Authenticate: Digest realm=“SIP-XXXXXXX”,nonce=“f80a810d333bc59bd755cff17fe4a826”,ZTE-ID=210a73ab63e68f9dcd0b7af3a7b42fba

<------------->
— (9 headers 0 lines) —
Responding to challenge, registration to domain/host name XX.X.XX.XX
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to XX.X.XX.XX:5060:
REGISTER sip:XX.X.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK68bf5d5c
Max-Forwards: 70
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
To: sip:XXXXXXX@XX.X.XX.XX
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3579 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XX”, nonce=“f80a810d333bc59bd755cff17fe4a826”, response="a2f2075702bea83b5485e6fc3a1ab6cc"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

<— SIP read from UDP:XX.X.XX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK68bf5d5c
To: sip:XXXXXXX@XX.X.XX.XX
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3579 REGISTER
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060;expires=300
User-Agent: ZTE-SoftSwitch
Date: Mon, 09 Feb 2015 15:20:05 GMT
Content-Length: 0

<------------->
— (10 headers 0 lines) —
[2015-02-09 15:18:22] NOTICE[2218]: chan_sip.c:23519 handle_response_register: Outbound Registration: Expiry for XX.X.XX.XX is 300 sec (Scheduling reregistration in 285 s)
Really destroying SIP dialog ‘13c2f04307b1b3fd3b9742ac090a437f@[::1]’ Method: REGISTER

9

Successful registration, but not calls even attempted.

10

Yes ,Please suggest me on my configuration ,if it is ok or not

11

I think you need to hire a consultant as you are just not providing the information requested.

12

SIP Trace logs when Incoming Calls not Connected

Really destroying SIP dialog ‘394110393963-0210-0200@10.162.201.6’ Method: BYE
Really destroying SIP dialog '7dea832e06bd39653eb7d57a15ea56a1@XX.X.XX.XXX:5060’ Method: BYE
<— SIP read from UDP:XX.X.XX.XX:5060 —>
OPTIONS sip:XXXXXXX@XX.X.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK140d562e6a426d06taN0
Max-Forwards: 70
To: sip:XXXXXXX@XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
Call-ID: 2243200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Accept: application/sdp
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Sending to XX.X.XX.XX:5060 (no NAT)
Looking for XXXXXXX in from-general (domain XX.X.XX.XXX)

<— Transmitting (no NAT) to XX.X.XX.XX:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK140d562e6a426d06taN0;received=XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
To: sip:XXXXXXX@XX.X.XX.XX;tag=as128e555b
Call-ID: 2243200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '2243200307191120@XX.X.XX.XX’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '2243200307191120@XX.X.XX.XX’ Method: OPTIONS

<— SIP read from UDP:XX.X.XX.XX:5060 —>
OPTIONS sip:XXXXXXX@XX.X.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK606669350da068bctaN0
Max-Forwards: 70
To: sip:XXXXXXX@XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
Call-ID: 51ee200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Accept: application/sdp
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to XX.X.XX.XX:5060 (no NAT)
Looking for XXXXXXX in from-general (domain XX.X.XX.XXX)

<— Transmitting (no NAT) to XX.X.XX.XX:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK606669350da068bctaN0;received=XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
To: sip:XXXXXXX@XX.X.XX.XX;tag=as390f980b
Call-ID: 51ee200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '51ee200307191120@XX.X.XX.XX’ in 32000 ms (Method: OPTIONS)

Really destroying SIP dialog '51ee200307191120@XX.X.XX.XX’ Method: OPTIONS

<— SIP read from UDP:XX.X.XX.XX:5060 —>
OPTIONS sip:XXXXXXX@XX.X.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK106250a57d2a2e9ataN0
Max-Forwards: 70
To: sip:XXXXXXX@XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
Call-ID: 7a40200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Accept: application/sdp
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to XX.X.XX.XX:5060 (no NAT)
Looking for XXXXXXX in from-general (domain XX.X.XX.XXX)

<— Transmitting (no NAT) to XX.X.XX.XX:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK106250a57d2a2e9ataN0;received=XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
To: sip:XXXXXXX@XX.X.XX.XX;tag=as024b1cd3
Call-ID: 7a40200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '7a40200307191120@XX.X.XX.XX’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '7a40200307191120@XX.X.XX.XX’ Method: OPTIONS

<— SIP read from UDP:XX.X.XX.XX:5060 —>
OPTIONS sip:XXXXXXX@XX.X.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK58421cbd46933340taN0
Max-Forwards: 70
To: sip:XXXXXXX@XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
Call-ID: 22b3200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Accept: application/sdp
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to XX.X.XX.XX:5060 (no NAT)
Looking for XXXXXXX in from-general (domain XX.X.XX.XXX)

<— Transmitting (no NAT) to XX.X.XX.XX:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK58421cbd46933340taN0;received=XX.X.XX.XX
From: BgwLinkTest5031 sip:bgw@XX.X.XX.XX;tag=200307191120
To: sip:XXXXXXX@XX.X.XX.XX;tag=as4d92a3f9
Call-ID: 22b3200307191120@XX.X.XX.XX
CSeq: 1000 OPTIONS
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '22b3200307191120@XX.X.XX.XX’ in 32000 ms (Method: OPTIONS)
[2015-02-09 17:36:08] NOTICE[2218]: chan_sip.c:15059 sip_reregister: – Re-registration for XXXXXXX@XX.X.XX.XX
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to XX.X.XX.XX:5060:
REGISTER sip:XX.X.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK41ba62b7
Max-Forwards: 70
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
To: sip:XXXXXXX@XX.X.XX.XX
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3636 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XX”, nonce=“ceed9bf8aed36001e20182bf1da37d30”, response="ad2b665c0b8dfae94d631c202632d453"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

<— SIP read from UDP:XX.X.XX.XX:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK41ba62b7
To: sip:XXXXXXX@XX.X.XX.XX
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3636 REGISTER
User-Agent: ZTE-SoftSwitch
Content-Length: 0
WWW-Authenticate: Digest realm=“SIP-XXXXXXX”,nonce=“1a886290cff88b63a9079bf6cbb85f29”,ZTE-ID=ac6008cc9bbebe61359711818baa3375

<------------->
— (9 headers 0 lines) —
Responding to challenge, registration to domain/host name XX.X.XX.XX
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to XX.X.XX.XX:5060:
REGISTER sip:XX.X.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK374e6cae
Max-Forwards: 70
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
To: sip:XXXXXXX@XX.X.XX.XX
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3637 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XX”, nonce=“1a886290cff88b63a9079bf6cbb85f29”, response="d2579ccb61bb762f6f28b042d2dd457e"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

<— SIP read from UDP:XX.X.XX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK374e6cae
To: sip:XXXXXXX@XX.X.XX.XX
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3637 REGISTER
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060;expires=300
User-Agent: ZTE-SoftSwitch
Date: Mon, 09 Feb 2015 17:37:52 GMT
Content-Length: 0

<------------->
— (10 headers 0 lines) —
[2015-02-09 17:36:09] NOTICE[2218]: chan_sip.c:23519 handle_response_register: Outbound Registration: Expiry for XX.X.XX.XX is 300 sec (Scheduling reregistration in 285 s)
Really destroying SIP dialog ‘13c2f04307b1b3fd3b9742ac090a437f@[::1]’ Method: REGISTER
Really destroying SIP dialog '22b3200307191120@XX.X.XX.XX’ Method: OPTIONS

SIP Trace Logs When Calls get Connected

[2015-02-09 17:40:54] NOTICE[2218]: chan_sip.c:15059 sip_reregister: – Re-registration for XXXXXXX@XX.X.XX.XX
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to XX.X.XX.XX:5060:
REGISTER sip:XX.X.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK657564a5
Max-Forwards: 70
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
To: sip:XXXXXXX@XX.X.XX.XX
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3638 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XX”, nonce=“1a886290cff88b63a9079bf6cbb85f29”, response="d2579ccb61bb762f6f28b042d2dd457e"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

<— SIP read from UDP:XX.X.XX.XX:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK657564a5
To: sip:XXXXXXX@XX.X.XX.XX
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3638 REGISTER
User-Agent: ZTE-SoftSwitch
Content-Length: 0
WWW-Authenticate: Digest realm=“SIP-XXXXXXX”,nonce=“a8305fcc03248186d264bb698f38af31”,ZTE-ID=e01587776ec1d005b58988b70c99d038

<------------->
— (9 headers 0 lines) —
Responding to challenge, registration to domain/host name XX.X.XX.XX
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to XX.X.XX.XX:5060:
REGISTER sip:XX.X.XX.XX SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK5473285e
Max-Forwards: 70
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
To: sip:XXXXXXX@XX.X.XX.XX
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3639 REGISTER
User-Agent: Asterisk PBX 11.8.1
Authorization: Digest username=“XXXXXXX”, realm=“SIP-XXXXXXX”, algorithm=MD5, uri=“sip:XX.X.XX.XX”, nonce=“a8305fcc03248186d264bb698f38af31”, response="3bcd2878e23063e88eee65772968c3e7"
Expires: 300
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

<— SIP read from UDP:XX.X.XX.XX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP XX.X.XX.XXX:5060;branch=z9hG4bK5473285e
To: sip:XXXXXXX@XX.X.XX.XX
From: sip:XXXXXXX@XX.X.XX.XX;tag=as570c1dcb
Call-ID: 13c2f04307b1b3fd3b9742ac090a437f@[::1]
CSeq: 3639 REGISTER
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060;expires=300
User-Agent: ZTE-SoftSwitch
Date: Mon, 09 Feb 2015 17:42:37 GMT
Content-Length: 0

<------------->
— (10 headers 0 lines) —
[2015-02-09 17:40:54] NOTICE[2218]: chan_sip.c:23519 handle_response_register: Outbound Registration: Expiry for XX.X.XX.XX is 300 sec (Scheduling reregistration in 285 s)
Really destroying SIP dialog ‘13c2f04307b1b3fd3b9742ac090a437f@[::1]’ Method: REGISTER

<— SIP read from UDP:XX.X.XX.XX:5060 —>
INVITE sip:XXXXXXX@XX.X.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK6a89e12ad513e8454e66taN0
To: "124XXXXXXX"sip:124XXXXXXX@XX.X.XX.XX
From: "0XXXXXXXXXX"sip:XXXXXXXXXX@XX.X.XX.XX;tag=aa2c906-yMhm10394001d2
Call-ID: 030710394002-0210-0200@10.162.201.6
CSeq: 9965 INVITE
Max-Forwards: 7
Contact: sip:XXXXXXXXXX@XX.X.XX.XX
Supported: 100rel
User-Agent: ZTE Softswitch/1.0.0
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,REFER,NOTIFY,SUBSCRIBE,PRACK,UPDATE
Content-Type: application/sdp
Content-Length: 177

v=0
o=ZTE 4678 2782 IN IP4 XX.X.XX.XX
s=phone-call
c=IN IP4 XX.X.XX.XX
t=0 0
m=audio 35060 RTP/AVP 8 0 18 97
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=ptime:20
<------------->
— (13 headers 9 lines) —
Sending to XX.X.XX.XX:5060 (no NAT)
Sending to XX.X.XX.XX:5060 (no NAT)
Using INVITE request as basis request - 030710394002-0210-0200@10.162.201.6
Found peer ‘tatasip’ for ‘XXXXXXXXXX’ from XX.X.XX.XX:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 97
Found audio description format telephone-event for ID 97
Capabilities: us - (ulaw|alaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port XX.X.XX.XX:35060
Looking for XXXXXXX in tata (domain XX.X.XX.XXX)
list_route: hop: sip:XXXXXXXXXX@XX.X.XX.XX

<— Transmitting (no NAT) to XX.X.XX.XX:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP XX.X.XX.XX:5060;branch=z9hG4bK6a89e12ad513e8454e66taN0;received=XX.X.XX.XX
From: "0XXXXXXXXXX"sip:XXXXXXXXXX@XX.X.XX.XX;tag=aa2c906-yMhm10394001d2
To: "124XXXXXXX"sip:124XXXXXXX@XX.X.XX.XX
Call-ID: 030710394002-0210-0200@10.162.201.6
CSeq: 9965 INVITE
Server: Asterisk PBX 11.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:XXXXXXX@XX.X.XX.XXX:5060
Content-Length: 0

13

As david55 already told You - You’ll need to hire a consultant for solving Your problems:

Your trace in [quote]SIP Trace logs when Incoming Calls not Connected[/quote] doesn’t show any INVITE but an OPTIONS-request (probably in conjunction with “qualify=yes” in sip.conf) which is failing as the extension is not found in “from-general”. This has nothing to do with a failing call.

The other traces shows a correct call coming in from an IP associated with the sip-peer “tatasip”.

As long as You think, that the first trace shows an incoming call, You’ll need professional help to dig in further …

And: BTW: Pls use codetags when posting logs. It’s really awful tracking down logs as normal text … :unamused:

14

Ok Please suggest me one Consultant

15

You need to ask on Biz and Jobs. It would be a breach of etiquette to answer the question on this forum.