Frequently Incoming Call Issue on SIP-Trunk

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Frequently Incoming Call Issue on SIP-Trunk

Postby mbmannubisht » Thu Feb 12, 2015 12:28 am

Hi Team,

Hope you all are doing well,I am facing some incoming calls issue with SIP-Trunk frequently. Normally Incoming calls is working fine but some fraction of time when customer calls on my SIP he/she listen "Dialed number doesn't exist ,Please check the number" etc etc. Please help me out to solve this problem. I checked all the configuration of sip.conf or extension.conf from my end unable to get any conclusion, so need your support.
mbmannubisht
Oldsterisk
 
Posts: 64
Joined: Fri Jan 10, 2014 6:14 am
Location: India

Re: Frequently Incoming Call Issue on SIP-Trunk

Postby mbmannubisht » Thu Feb 12, 2015 12:31 am

Below I am sharing SIP Logs and sip.conf file for the same:

<------------>
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.conf

[general]

defaultexpiry=300
progressinband=yes
localnet=XX.X.XX.XXX/XXX.XXX.XXX.XXX
nat=no
;qualify=yes

;### Appended fileds just because of incoming issues

bindaddr=0.0.0.0
bindport=5060
context=from-general
udpbindaddr=0.0.0.0
tcpenable=yes
srvlookup=yes

register => XXXXXXX:XXXX:XXXXXXX@XX.X.XX.XX/XXXXXXX

[tatasip]

type=friend
disallow=all
allow=alaw
allow=ulaw
allow=g729
host=XX.X.XX.XX
dtmfmode=rfc2833
nat=no
canreinvite=no
context=tata

[1000]

username=1000
secret=1000
type=friend
context=1000
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=g723
allow=g729
dtmfmode=rfc2833
nat=no
mbmannubisht
Oldsterisk
 
Posts: 64
Joined: Fri Jan 10, 2014 6:14 am
Location: India

Re: Frequently Incoming Call Issue on SIP-Trunk

Postby ambiorixg12 » Thu Feb 12, 2015 9:16 pm

Your SIP trace is not completed.. There is no initiation of call on the SIP trace, but based on the

404 Not Found

The server has definitive information that the user does not exist at
the domain specified in the Request-URI. This status is also
returned if the domain in the Request-URI does not match any of the
domains handled by the recipient of the request.

As you already said
" "Dialed number doesn't exist ,Please check the number"
I think the message is played by your provider I, So will be good if you also address this issue with them, Check on the Invite the number dialed and if is in the correct format
ambiorixg12
Astmaster
 
Posts: 967
Joined: Sun Mar 04, 2007 9:32 pm
Location: Dominican Republic

Re: Frequently Incoming Call Issue on SIP-Trunk

Postby mbmannubisht » Thu Feb 12, 2015 11:04 pm

Dear ambiorixg12/Team,

Thanks for your time to see my query and give your valuable feedback.
I am so happy you all are here for us,also congrats to Asterisk Forum member.

Good news for every body Issue has been resolved till now.

Thanks to all ...... :D :D :D :D :D :D
mbmannubisht
Oldsterisk
 
Posts: 64
Joined: Fri Jan 10, 2014 6:14 am
Location: India


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