Help : Call failure in SIP

Asterisk Asterisk Support
1

Hello all,
I was configuring asterisk to make VoIP call.I brought a sip account from a service provider an changed the relevant sip.conf and extension.conf.
The account got properly registered as i can see in the asterisk CLI console.

*CLI> sip show registry
Host Username Refresh State Reg.Time
sip.net4india.com:5060 052280680 105 Registered Sat, 24 May 2008 02:11:03

This is what i get in the console.

My problem is that i cant make any calls from it.As a testing purpose i had make from the asterisk console using the dial command.

The ‘dial’ command is deprecated and will be removed in a future release. Please use ‘console dial’ instead.
*CLI> – Executing [s@localterm:1] Dial(“OSS/dsp”, “SIP/971501737250@sip.net4india.com”) in new stack
– Called 971501737250@sip.net4india.com
[May 24 02:16:21] NOTICE[6106]: chan_sip.c:12186 handle_response_invite: Failed to authenticate on INVITE to ‘“asterisk” sip:asterisk@121.245.3.140;tag=as413460c8’
– SIP/sip.net4india.com-081e5290 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘OSS/dsp’ status is ‘CONGESTION’

This is the error message im getting in the console.

when i enabled the sip debugging.I noticed this message

<— SIP read from 202.71.134.13:5060 —>
SIP/2.0 407 Proxy Authentication required
Via: SIP/2.0/UDP 121.245.3.140:5060;branch=z9hG4bK0632c562;rport
From: “asterisk” sip:asterisk@121.245.3.140;tag=as7ff7c3b9
To: sip:971501737250@sip.net4india.com
Call-ID: 3b5694210bdf8c631fdab266729baee4@121.245.3.140
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“202.71.134.13”,nonce=“e2e79444072a0f447f007900cbc1dfe2”,opaque="",stale=FALSE,algorithm=MD5
Content-Length: 0

What is this proxy authentcation.is it done from our side or the service provider side.I dont have any firewall set in my system.

This is my sip.conf

[general]
context=default
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all
;allow=gsm
;externip= 172.23.129.14
;localip=121.245.52.165
;srvlookup=yes
register => 052280680:XXXXXXX@sip.net4india.com
username=052280680
canreinvite=no
secret=XXXXXX

And this is my extension.conf

[localterm]
exten => s,1,Dial(SIP/XXXXXXX@sip.net4india.com)
;exten => s,2,Congestion
;exten => s,3,Busy()

dialing form the console by using

*CLI> dial s@localterm

command…

What is wrong in the thing i have done???Help please…

2

I don’t see your registration string. That is what you are missing!

3

sorry i didnt get what you told.Will you please explain in detail.
I have given my sip.conf in the above message.About the registration string
i have given it in this way.And showing registered in console.

register => 052280680:XXXXX@sip.net4india.com

4

In most cases you do not have to register in order to place outgoing calls. Registration is mandatory for receiving calls.
401 (or 407) response from you provider is not the error but normal part of the dialog.

At the same time I see some mess in your sip.conf - you should not put all the settings related to your provider into general section, you should have a separate block of lines, see the example provided with Asterisk.

Fix your config and if the problem persists post the complete sip debug, i.e. messages following this 407 message.

5

Hi

You need to set up a sip peer for example

[net4india]
username=052280680
secret=xxxxxx
host=sip.net4india.com
disallow=all
allow=alaw
allow=ulaw

then the dial string

exten => s,1,Dial(SIP/XXXXXXX@net4india)

6

hello all,
Thanks for the replays, i have changed my sip.conf and extension.conf
now it looks like this

sip.conf
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=all
canreinvite=no

[net4india]
username=052280680
type=friend
secret=XXXXXXX
host=sip.net4india.com
insecure=very
;qualify=yes
disallow=all
allow=all
port = 5060 ; Port to bind to (SIP is 5060)
canreinvite=no
nat=no
outboundproxy=sip.net4india.com
;register => 052280680:XXXXX@sip.net4india.com

extension.conf
[localterm]
exten => s,1,Dial(SIP/971508473704@net4india.com)

Now it seems to be no Proxy Authentication error.But some the called phone is not seems to be ringing. Showing some congesting or busy always. I have pasted the the sip debug log below.Is it due to our mistake.

*CLI> – Executing [s@localterm:1] Dial(“OSS/dsp”, “SIP/971508473704@net4india.com”) in new stack
Audio is at 121.245.5.54 port 12654
Adding codec 0x40 (slin) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 202.71.130.4:5060:
INVITE sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Contact: sip:asterisk@121.245.5.54
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 25 May 2008 03:05:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 469

v=0
o=root 6131 6131 IN IP4 121.245.5.54
s=session
c=IN IP4 121.245.5.54
t=0 0
m=audio 12654 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called [971508473704@net4india.com](mailto:971508473704@net4india.com)

Retransmitting #1 (no NAT) to 202.71.130.4:5060:
INVITE sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Contact: sip:asterisk@121.245.5.54
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 25 May 2008 03:05:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 469

v=0
o=root 6131 6131 IN IP4 121.245.5.54
s=session
c=IN IP4 121.245.5.54
t=0 0
m=audio 12654 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #2 (no NAT) to 202.71.130.4:5060:
INVITE sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Contact: sip:asterisk@121.245.5.54
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 25 May 2008 03:05:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 469

v=0
o=root 6131 6131 IN IP4 121.245.5.54
s=session
c=IN IP4 121.245.5.54
t=0 0
m=audio 12654 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #3 (no NAT) to 202.71.130.4:5060:
INVITE sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Contact: sip:asterisk@121.245.5.54
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 25 May 2008 03:05:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 469

v=0
o=root 6131 6131 IN IP4 121.245.5.54
s=session
c=IN IP4 121.245.5.54
t=0 0
m=audio 12654 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #4 (no NAT) to 202.71.130.4:5060:
INVITE sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Contact: sip:asterisk@121.245.5.54
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 25 May 2008 03:05:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 469

v=0
o=root 6131 6131 IN IP4 121.245.5.54
s=session
c=IN IP4 121.245.5.54
t=0 0
m=audio 12654 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #5 (no NAT) to 202.71.130.4:5060:
INVITE sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Contact: sip:asterisk@121.245.5.54
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 25 May 2008 03:05:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 469

v=0
o=root 6131 6131 IN IP4 121.245.5.54
s=session
c=IN IP4 121.245.5.54
t=0 0
m=audio 12654 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #6 (no NAT) to 202.71.130.4:5060:
INVITE sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Contact: sip:asterisk@121.245.5.54
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 25 May 2008 03:05:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 469

v=0
o=root 6131 6131 IN IP4 121.245.5.54
s=session
c=IN IP4 121.245.5.54
t=0 0
m=audio 12654 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[May 25 08:36:22] NOTICE[6312]: chan_sip.c:2927 auto_congest: Auto-congesting SIP/net4india.com-081f2770
– SIP/net4india.com-081f2770 is circuit-busy
Reliably Transmitting (no NAT) to 202.71.130.4:5060:
CANCEL sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Scheduling destruction of SIP dialog ‘5988bf2720deab79358de2161320063d@121.245.5.54’ in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘OSS/dsp’ status is ‘CONGESTION’
[May 25 08:36:22] WARNING[6306]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: Device or resource busy
Retransmitting #1 (no NAT) to 202.71.130.4:5060:
CANCEL sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Retransmitting #2 (no NAT) to 202.71.130.4:5060:
CANCEL sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Retransmitting #3 (no NAT) to 202.71.130.4:5060:
CANCEL sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Retransmitting #4 (no NAT) to 202.71.130.4:5060:
CANCEL sip:971508473704@net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.5.54:5060;branch=z9hG4bK29111d7d;rport
From: “asterisk” sip:asterisk@121.245.5.54;tag=as0dab7afe
To: sip:971508473704@net4india.com
Call-ID: 5988bf2720deab79358de2161320063d@121.245.5.54
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<< Hangup on console >>

7

Again, Proxy Authentication is not an error!

Edit your sip.conf:

[net4india]
username=052280680
type=friend
secret=XXXXXXX
host=sip.net4india.com
insecure=invite
;qualify=yes
allow=all
canreinvite=no
nat=no

Check the general section.
Make sure you put the right values for externip (or externhost) and localnet.

Post the log if error occurs.

8

hello as per your suggestions have made changes in my sip.conf
now my sip.conf seems to me like this

[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=all
canreinvite=no
context=localterm
srvlookup=yes
nat=yes
externip= 122.166.XX.XX
localnet=192.168.1.1/255.255.255.0
[net4india]
username=052280680
authuser=052280680
secret=XXXXX
type=friend
host=sip.net4india.com
insecure=invite
allow=all
canreinvite=no
nat=yes
fromdomain=sip.net4india.com

This is wat i am getting in the console when i dial the number

*CLI> – Executing [s@localterm:1] Dial(“OSS/dsp”, “SIP/966556749501@net4india.com”) in new stack
Audio is at 122.166.40.72 port 11666
Adding codec 0x40 (slin) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 202.71.130.4:5060:
INVITE sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Contact: sip:asterisk@122.166.40.72
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 May 2008 10:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 8223 8223 IN IP4 122.166.40.72
s=session
c=IN IP4 122.166.40.72
t=0 0
m=audio 11666 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called [966556749501@net4india.com](mailto:966556749501@net4india.com)

Retransmitting #1 (NAT) to 202.71.130.4:5060:
INVITE sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Contact: sip:asterisk@122.166.40.72
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 May 2008 10:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 8223 8223 IN IP4 122.166.40.72
s=session
c=IN IP4 122.166.40.72
t=0 0
m=audio 11666 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #2 (NAT) to 202.71.130.4:5060:
INVITE sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Contact: sip:asterisk@122.166.40.72
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 May 2008 10:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 8223 8223 IN IP4 122.166.40.72
s=session
c=IN IP4 122.166.40.72
t=0 0
m=audio 11666 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #3 (NAT) to 202.71.130.4:5060:
INVITE sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Contact: sip:asterisk@122.166.40.72
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 May 2008 10:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 8223 8223 IN IP4 122.166.40.72
s=session
c=IN IP4 122.166.40.72
t=0 0
m=audio 11666 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #4 (NAT) to 202.71.130.4:5060:
INVITE sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Contact: sip:asterisk@122.166.40.72
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 May 2008 10:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 8223 8223 IN IP4 122.166.40.72
s=session
c=IN IP4 122.166.40.72
t=0 0
m=audio 11666 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #5 (NAT) to 202.71.130.4:5060:
INVITE sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Contact: sip:asterisk@122.166.40.72
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 May 2008 10:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 8223 8223 IN IP4 122.166.40.72
s=session
c=IN IP4 122.166.40.72
t=0 0
m=audio 11666 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #6 (NAT) to 202.71.130.4:5060:
INVITE sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Contact: sip:asterisk@122.166.40.72
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 May 2008 10:46:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 471

v=0
o=root 8223 8223 IN IP4 122.166.40.72
s=session
c=IN IP4 122.166.40.72
t=0 0
m=audio 11666 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[May 26 16:17:10] NOTICE[8236]: chan_sip.c:2927 auto_congest: Auto-congesting SIP/net4india.com-081f1300
– SIP/net4india.com-081f1300 is circuit-busy
Reliably Transmitting (NAT) to 202.71.130.4:5060:
CANCEL sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Scheduling destruction of SIP dialog ‘26554ed338ed81c9675c1bb04cc81f14@122.166.40.72’ in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘OSS/dsp’ status is ‘CONGESTION’
[May 26 16:17:10] WARNING[8230]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: Device or resource busy
Retransmitting #1 (NAT) to 202.71.130.4:5060:
CANCEL sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Retransmitting #2 (NAT) to 202.71.130.4:5060:
CANCEL sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Retransmitting #3 (NAT) to 202.71.130.4:5060:
CANCEL sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


Here it got stuck for a while and then the rest occurs

[May 26 16:17:10] NOTICE[8236]: chan_sip.c:2927 auto_congest: Auto-congesting SIP/net4india.com-081f1300
– SIP/net4india.com-081f1300 is circuit-busy
Reliably Transmitting (NAT) to 202.71.130.4:5060:
CANCEL sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Scheduling destruction of SIP dialog ‘26554ed338ed81c9675c1bb04cc81f14@122.166.40.72’ in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘OSS/dsp’ status is ‘CONGESTION’
[May 26 16:17:10] WARNING[8230]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: Device or resource busy
Retransmitting #1 (NAT) to 202.71.130.4:5060:
CANCEL sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Retransmitting #2 (NAT) to 202.71.130.4:5060:
CANCEL sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Retransmitting #3 (NAT) to 202.71.130.4:5060:
CANCEL sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Retransmitting #4 (NAT) to 202.71.130.4:5060:
CANCEL sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<< Hangup on console >>
Retransmitting #5 (NAT) to 202.71.130.4:5060:
CANCEL sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Retransmitting #6 (NAT) to 202.71.130.4:5060:
CANCEL sip:966556749501@net4india.com SIP/2.0
Via: SIP/2.0/UDP 122.166.40.72:5060;branch=z9hG4bK092569f4;rport
From: “asterisk” sip:asterisk@122.166.40.72;tag=as6c5d3994
To: sip:966556749501@net4india.com
Call-ID: 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

[May 26 16:17:30] WARNING[8236]: chan_sip.c:1949 retrans_pkt: Maximum retries exceeded on transmission 26554ed338ed81c9675c1bb04cc81f14@122.166.40.72 for seqno 102 (Non-critical Request)
Really destroying SIP dialog ‘26554ed338ed81c9675c1bb04cc81f14@122.166.40.72’ Method: INVITE

Is it any problem with my extension.conf file syntax in the number format im trying to call the dubai number also tried us and saudi numbers the result is still the same???

9

there are 2 problems :frowning:

You’re not following the recommendations. Just compare what I told you and what you have in your sip.conf peer definition.
You’re sending REGISTER to the wrong address. You should send to sip.net4india.com but in fact you’re sending to www.net4india.com
Make sure your DNS works and you did sip reload or restarted asterisk.

10

hello andrewz,
First of all thanking for your patience.The thing is that i had tried wat you have given but was not got working.That time i got the error as shown below.And about the DNS.I have tried both with the domain name and direct IP.But still got the same result.

[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=all
canreinvite=no
context=localterm
srvlookup=yes
nat=no
externip= 121.245.48.116

[sip.conf]
[net4india]
username=052280680
secret=XXXXXX
type=friend
host=sip.net4india.com
;host=202.71.134.13
insecure=invite
allow=all
canreinvite=no
nat=no

*CLI> – Executing [s@localterm:1] Dial(“OSS/dsp”, “SIP/971507690802@sip.net4india.com”) in new stack
Audio is at 121.245.48.116 port 14942
Adding codec 0x40 (slin) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 202.71.134.13:5060:
INVITE sip:971507690802@sip.net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.48.116:5060;branch=z9hG4bK0292a2b6;rport
From: “asterisk” sip:asterisk@121.245.48.116;tag=as6d917402
To: sip:971507690802@sip.net4india.com
Contact: sip:asterisk@121.245.48.116
Call-ID: 2dd7df402145810222db999000a8c9bf@121.245.48.116
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 May 2008 15:59:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 473

v=0
o=root 6154 6154 IN IP4 121.245.48.116
s=session
c=IN IP4 121.245.48.116
t=0 0
m=audio 14942 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called [971507690802@sip.net4india.com](mailto:971507690802@sip.net4india.com)

Retransmitting #1 (no NAT) to 202.71.134.13:5060:
INVITE sip:971507690802@sip.net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.48.116:5060;branch=z9hG4bK0292a2b6;rport
From: “asterisk” sip:asterisk@121.245.48.116;tag=as6d917402
To: sip:971507690802@sip.net4india.com
Contact: sip:asterisk@121.245.48.116
Call-ID: 2dd7df402145810222db999000a8c9bf@121.245.48.116
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 May 2008 15:59:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 473

v=0
o=root 6154 6154 IN IP4 121.245.48.116
s=session
c=IN IP4 121.245.48.116
t=0 0
m=audio 14942 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from 202.71.134.13:5060 —>
SIP/2.0 407 Proxy Authentication required
Via: SIP/2.0/UDP 121.245.48.116:5060;branch=z9hG4bK0292a2b6;rport
From: “asterisk” sip:asterisk@121.245.48.116;tag=as6d917402
To: sip:971507690802@sip.net4india.com
Call-ID: 2dd7df402145810222db999000a8c9bf@121.245.48.116
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“202.71.134.13”,nonce=“efdcba694d405981cc5bd3d1add82db9”,opaque="",stale=FALSE,algorithm=MD5
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (no NAT) to 202.71.134.13:5060:
ACK sip:971507690802@sip.net4india.com SIP/2.0
Via: SIP/2.0/UDP 121.245.48.116:5060;branch=z9hG4bK0292a2b6;rport
From: “asterisk” sip:asterisk@121.245.48.116;tag=as6d917402
To: sip:971507690802@sip.net4india.com
Contact: sip:asterisk@121.245.48.116
Call-ID: 2dd7df402145810222db999000a8c9bf@121.245.48.116
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

[May 26 21:29:16] NOTICE[6168]: chan_sip.c:12186 handle_response_invite: Failed to authenticate on INVITE to ‘“asterisk” sip:asterisk@121.245.48.116;tag=as6d917402’
– SIP/sip.net4india.com-081f10b0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘OSS/dsp’ status is ‘CONGESTION’
[May 26 21:29:16] WARNING[6162]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: Device or resource busy
Really destroying SIP dialog ‘2dd7df402145810222db999000a8c9bf@121.245.48.116’ Method: INVITE

<— SIP read from 202.71.134.13:5060 —>
SIP/2.0 407 Proxy Authentication required
Via: SIP/2.0/UDP 121.245.48.116:5060;branch=z9hG4bK0292a2b6;rport
From: “asterisk” sip:asterisk@121.245.48.116;tag=as6d917402
To: sip:971507690802@sip.net4india.com
Call-ID: 2dd7df402145810222db999000a8c9bf@121.245.48.116
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“202.71.134.13”,nonce=“755afe9d6e17b89fe705e6611dbf7c6d”,opaque="",stale=FALSE,algorithm=MD5
Content-Length: 0

<------------->
— (8 headers 0 lines) —
<< Hangup on console >>

This is what i got then… :frowning:

11

Just tested the following setup. It works as it should, having in mind that I do not know your password.

[net4india]
username=052280680
secret=idontknowit
host=sip.net4india.com
insecure=invite
disallow=all
allow = alaw
allow = ulaw
canreinvite=no
nat=no
fromuser=052280680
fromdomain=sip.net4india.com
registersip=yes

The last line replaces the string you used before

register => 052280680:XXXXX@sip.net4india.com

Please note the right settings for [global]:
localnet=192.168.1.0/255.255.255.0

According to your log your public IP is not static. If that is right then you should not use externip

12

Sory Andrewz

Im getting the same result with the same configuration u had given tried some alternatives too .I also tried from a soft phone which is connected to the lan,But still the problem persists.I am sure that my account is working as i can make calls when i give the configuration directly in the X-lite.Will be thankful if you figure out a solution

This is my sip.conf now

[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=all
canreinvite=no
context=net4india
srvlookup=yes
nat=yes
registersip=yes
localnet=192.168.1.1/255.255.255.0
outboundproxy=sip.net4india.com
register => 052280680:XXXXXX@sip.net4india.com
[net4india]
username=052280680
secret=XXXXX
host=sip.net4india.com
insecure=invite
disallow=all
allow = alaw
allow = ulaw
canreinvite=no
nat=yes
fromuser=052280680
fromdomain=sip.net4india.com
outboundproxy=sip.net4india.com
registersip=yes
[200]
username=200
authuser=200
secret=200
context=net4india
callerid=“Bipin” <200>
type=friend
host=dynamic
insecure=invite
allow=all
canreinvite=no
nat=yes
registersip=yes
outboundproxy=sip.net4india.com

*CLI> – Executing [971507690802@net4india:1] Dial(“OSS/dsp”, “SIP/971507690802@sip.net4india.com”) in new stack
Audio is at 192.168.1.4 port 10224
Adding codec 0x40 (slin) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 202.71.134.13:5060:
INVITE sip:971507690802@sip.net4india.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7496f445;rport
From: “asterisk” sip:asterisk@192.168.1.4;tag=as13674397
To: sip:971507690802@sip.net4india.com
Contact: sip:asterisk@192.168.1.4
Call-ID: 2fb94e0161ac8cbd191e15a14a0a42e4@192.168.1.4
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 27 May 2008 05:35:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 467

v=0
o=root 6663 6663 IN IP4 192.168.1.4
s=session
c=IN IP4 192.168.1.4
t=0 0
m=audio 10224 RTP/AVP 10 3 0 8 112 5 7 97 111 101
a=rtpmap:10 L16/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called [971507690802@sip.net4india.com](mailto:971507690802@sip.net4india.com)

<— SIP read from 202.71.134.13:5060 —>
SIP/2.0 407 Proxy Authentication required
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7496f445;rport
From: “asterisk” sip:asterisk@192.168.1.4;tag=as13674397
To: sip:971507690802@sip.net4india.com
Call-ID: 2fb94e0161ac8cbd191e15a14a0a42e4@192.168.1.4
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“202.71.134.13”,nonce=“79fe0af9aa9312b713215509e104965f”,opaque="",stale=FALSE,algorithm=MD5
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to 202.71.134.13:5060:
ACK sip:971507690802@sip.net4india.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK7496f445;rport
From: “asterisk” sip:asterisk@192.168.1.4;tag=as13674397
To: sip:971507690802@sip.net4india.com
Contact: sip:asterisk@192.168.1.4
Call-ID: 2fb94e0161ac8cbd191e15a14a0a42e4@192.168.1.4
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

[May 27 11:05:34] NOTICE[6676]: chan_sip.c:12186 handle_response_invite: Failed to authenticate on INVITE to '“asterisk” sip:asterisk@192.168.1.4;tag=as13674397’
– SIP/sip.net4india.com-081f1a10 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘OSS/dsp’ status is 'CONGESTION’
Really destroying SIP dialog ‘2fb94e0161ac8cbd191e15a14a0a42e4@192.168.1.4’ Method: INVITE
<< Hangup on console >>
[May 27 11:06:11] NOTICE[6676]: chan_sip.c:7403 sip_reregister: – Re-registration for 052280680@sip.net4india.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 202.71.134.13:5060:
REGISTER sip:sip.net4india.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK206cec7b;rport
From: sip:052280680@sip.net4india.com;tag=as285c5a8a
To: sip:052280680@sip.net4india.com
Call-ID: 2b01065a43a4f1565c05601d1f4ecb9f@127.0.1.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“052280680”, realm=“202.71.134.13”, algorithm=MD5, uri=“sip:sip.net4india.com”, nonce=“163b46e2f2178a2995faf4ed56753acd”, response=“37ef90e118e3388056c1a1cfff04d455”, opaque=""
Expires: 120
Contact: sip:s@192.168.1.4
Event: registration
Content-Length: 0


My extension.conf

[net4india]
exten=_X.,1,Dial(SIP/${EXTEN}@sip.net4india.com)

13

you’re not willing to read
sorry, cannot help
there is no technical issue here

14

sory Andrewz.
The thing is that when am in office i use intenet with network there and when i reach home, used to try with my data card in my laptopThats y the configuration difference.Any way thanks for your help.Now will try myself and will let u know if i get it.

15

sory Andrewz.
The thing is that when am in office i use internet with network there and when i reach home, used to try with my data card in my laptop Thats why the configuration difference.Any way thanks for your help.Now will try myself and will let u know if i get it.

16

sory Andrewz.
The thing is that when am in office i use internet with network there and when i reach home, used to try with my data card in my laptop thats why the configuration difference.Any way thanks for your help.Now will try myself and will let u know if i get it.

17

hello andrewz at last i made it working …

asteriskdocs.org/html/ch04s08.html

followed this link…
i think the problem was with my extension.conf

[outgoing_calls]
exten => _X.,1,NoOp()
exten => _X.,n,Dial(SIP/my_service_provider/${EXTEN})

sip.conf

[my_provider]
type=friend
username=052280680
fromuser=052280680
context=callout
fromdomain=sip.net4india.com
canreinvite=no
secret=XXXXXXXX
insecure=invite
host= 202.71.134.13
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
nat=no

Bipin
bipinb.com/

when i tried in this way it got worked…

Thanks for your helps.