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Raj Roy Ghandhi

2015-Feb-23 11:51 UTC

[asterisk-users] Asterisk does not listed to port 5060

Hi Friends,
I encountered a strange issue.
I am running Asterisk 11.8.1 on Cent OS with no firewall running.
It has 3 NIC interfaces.

in my sip.conf I have

allowguest=yes
bindaddr=0.0.0.0
udpbindaddr = 0.0.0.0

But my Asterisk instance does not pick the call at all.

When I check the listening apps using lsof -i I get

asterisk   3046  asterisk    7u  IPv4 1191172      0t0  TCP *:5038 (LISTEN)
asterisk   3046  asterisk   10u  IPv4 1191186      0t0  UDP *:sip
asterisk   3046  asterisk   11u  IPv4 1191187      0t0  TCP *:sip (LISTEN)
asterisk   3046  asterisk   13u  IPv4 1191196      0t0  UDP *:iax
asterisk   3046  asterisk   15u  IPv4 1191199      0t0  UDP *:commplex-main
asterisk   3046  asterisk   16u  IPv4 1191201      0t0  UDP *:4520
asterisk   3046  asterisk   19u  IPv4 1191232      0t0  TCP
localhost:5038->localhost:43353 (ESTABLISHED)


But I van see the SIP Invite that comes into server and I can ngrep it as

U 10.85.0.24:5060 -> 10.25.172.10:5060
INVITE sip:+91712442211 at 10.25.172.10:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1.
Content-Type: application/sdp.
To: <sip:+91712442211 at unknown.invalid>.
Contact: <sip:+91711189078 at 10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078 at
10.25.84.3;user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078 at 10.25.84.3;user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+91712442211 at
unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1,
<sip:+91712442211 at unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E at GPAS_GWCS6_ipm_1_2_6.
CSeq: 1 INVITE.
Content-Length: 171.
.
v=0.
o=- 10000 10000 IN IP4 10.85.0.24.
s=-.
t=0 0.
m=audio 36740 RTP/AVP 8 101.
c=IN IP4 10.85.0.24.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=ptime:20.


U 10.85.0.24:5060 -> 10.25.172.10:5060
INVITE sip:+91712442211 at 10.25.172.10:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1.
Content-Type: application/sdp.
To: <sip:+91712442211 at unknown.invalid>.
Contact: <sip:+91711189078 at 10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078 at
10.25.84.3;user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078 at 10.25.84.3;user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+91712442211 at
unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1,
<sip:+91712442211 at unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E at GPAS_GWCS6_ipm_1_2_6.
CSeq: 1 INVITE.
Content-Length: 171.
.
v=0.
o=- 10000 10000 IN IP4 10.85.0.24.
s=-.
t=0 0.
m=audio 36740 RTP/AVP 8 101.
c=IN IP4 10.85.0.24.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=ptime:20.


U 10.85.0.24:5060 -> 10.25.172.10:5060
INVITE sip:+91712442211 at 10.25.172.10:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1.
Content-Type: application/sdp.
To: <sip:+94722442200 at unknown.invalid>.
Contact: <sip:+91711189078 at 10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078 at
10.25.84.3;user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078 at 10.25.84.3;user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+94722442200 at
unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1,
<sip:+91712442211 at unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E at GPAS_GWCS6_ipm_1_2_6.
CSeq: 1 INVITE.
Content-Length: 171.
.
v=0.
o=- 10000 10000 IN IP4 10.85.0.24.
s=-.
t=0 0.
m=audio 36740 RTP/AVP 8 101.
c=IN IP4 10.85.0.24.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=ptime:20.


U 10.85.0.24:5060 -> 10.25.172.10:5060
INVITE sip:+91712442211 at 10.25.172.10:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1.
Content-Type: application/sdp.
To: <sip:+94722442200 at unknown.invalid>.
Contact: <sip:+91711189078 at 10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078 at
10.25.84.3;user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078 at 10.25.84.3;user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+94722442200 at
unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1,
<sip:+91712442211 at unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E at GPAS_GWCS6_ipm_1_2_6.
CSeq: 1 INVITE.
Content-Length: 171.
.
v=0.
o=- 10000 10000 IN IP4 10.85.0.24.
s=-.
t=0 0.
m=audio 36740 RTP/AVP 8 101.
c=IN IP4 10.85.0.24.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=ptime:20.


U 10.85.0.24:5060 -> 10.25.172.10:5060
INVITE sip:+91712442211 at 10.25.172.10:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1.
Content-Type: application/sdp.
To: <sip:+94722442200 at unknown.invalid>.
Contact: <sip:+91711189078 at 10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078 at
10.25.84.3;user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078 at 10.25.84.3;user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+94722442200 at
unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1,
<sip:+91712442211 at unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E at GPAS_GWCS6_ipm_1_2_6.
CSeq: 1 INVITE.
Content-Length: 171.
.
v=0.
o=- 10000 10000 IN IP4 10.85.0.24.
s=-.
t=0 0.
m=audio 36740 RTP/AVP 8 101.
c=IN IP4 10.85.0.24.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=ptime:20.


U 10.85.0.24:5060 -> 10.25.172.10:5060
INVITE sip:+91712442211 at 10.25.172.10:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1.
Content-Type: application/sdp.
To: <sip:+94722442200 at unknown.invalid>.
Contact: <sip:+91711189078 at 10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078 at
10.25.84.3;user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078 at 10.25.84.3;user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+94722442200 at
unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1,
<sip:+91712442211 at unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E at GPAS_GWCS6_ipm_1_2_6.
CSeq: 1 INVITE.
Content-Length: 171.
.
v=0.
o=- 10000 10000 IN IP4 10.85.0.24.
s=-.
t=0 0.
m=audio 36740 RTP/AVP 8 101.
c=IN IP4 10.85.0.24.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=ptime:20.

Please let me know what I miss in this configuration.

Best Regards,
Roy.
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Rusty Newton

2015-Feb-27 01:03 UTC

head link

[asterisk-users] Asterisk does not listed to port 5060

On Mon, Feb 23, 2015 at 5:51 AM, Raj Roy Ghandhi <roy.gandhi at gmail.com>
wrote:
> Hi Friends,
> I encountered a strange issue.
> I am running Asterisk 11.8.1 on Cent OS with no firewall running.
> It has 3 NIC interfaces.
>
> in my sip.conf I have
>
> allowguest=yes
> bindaddr=0.0.0.0
> udpbindaddr = 0.0.0.0
>
> But my Asterisk instance does not pick the call at all.
>
> When I check the listening apps using lsof -i I get
>
> asterisk   3046  asterisk    7u  IPv4 1191172      0t0  TCP *:5038 (LISTEN)
> asterisk   3046  asterisk   10u  IPv4 1191186      0t0  UDP *:sip
> asterisk   3046  asterisk   11u  IPv4 1191187      0t0  TCP *:sip (LISTEN)
> asterisk   3046  asterisk   13u  IPv4 1191196      0t0  UDP *:iax
> asterisk   3046  asterisk   15u  IPv4 1191199      0t0  UDP *:commplex-main
> asterisk   3046  asterisk   16u  IPv4 1191201      0t0  UDP *:4520
> asterisk   3046  asterisk   19u  IPv4 1191232      0t0  TCP
> localhost:5038->localhost:43353 (ESTABLISHED)
>
>
> But I van see the SIP Invite that comes into server and I can ngrep it as
>
>I believe UDP ports don't provide the state in lsof.

Asterisk is listening here:
asterisk   3046  asterisk   10u  IPv4 1191186      0t0  UDP *:sip

My system shows similar output for lsof and it works fine.

Have you tried using the Asterisk CLI with "sip set debug on" to see
if
Asterisk shows any SIP packets?

You might consider collecting a debug log with "sip set debug on"
output :
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Once you have that, provide a pastebin link to the output and someone may
be able to help you out.

-- 

Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806 - USdirect: +1 256 428 6200
Check us out at: http://digium.com & http://asterisk.org
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Amit Patkar

2015-Feb-27 05:49 UTC

head link

[asterisk-users] Asterisk does not listed to port 5060

You can use following command to check

netstat -an

This will show host and ports in numeric format.*

Regards,*
Amit Patkar


On 2/27/2015 6:33 AM, Rusty Newton wrote:>
>
> On Mon, Feb 23, 2015 at 5:51 AM, Raj Roy Ghandhi <roy.gandhi at
gmail.com
> <mailto:roy.gandhi at gmail.com>> wrote:
>
>     Hi Friends,
>     I encountered a strange issue.
>     I am running Asterisk 11.8.1 on Cent OS with no firewall running.
>     It has 3 NIC interfaces.
>
>     in my sip.conf I have
>
>     allowguest=yes
>     bindaddr=0.0.0.0
>     udpbindaddr = 0.0.0.0
>
>     But my Asterisk instance does not pick the call at all.
>
>     When I check the listening apps using lsof -i I get
>     asterisk   3046  asterisk    7u  IPv4 1191172  0t0  TCP *:5038
>     (LISTEN)
>     asterisk   3046  asterisk   10u  IPv4 1191186  0t0  UDP *:sip
>     asterisk   3046  asterisk   11u  IPv4 1191187  0t0  TCP *:sip (LISTEN)
>     asterisk   3046  asterisk   13u  IPv4 1191196  0t0  UDP *:iax
>     asterisk   3046  asterisk   15u  IPv4 1191199  0t0  UDP
>     *:commplex-main
>     asterisk   3046  asterisk   16u  IPv4 1191201  0t0  UDP *:4520
>     asterisk   3046  asterisk   19u  IPv4 1191232  0t0  TCP
>     localhost:5038->localhost:43353 (ESTABLISHED)
>
>
>     But I van see the SIP Invite that comes into server and I can
>     ngrep it as
>
>
> I believe UDP ports don't provide the state in lsof.
>
> Asterisk is listening here:
> asterisk 3046  asterisk   10u  IPv4 1191186      0t0  UDP *:sip
>
> My system shows similar output for lsof and it works fine.
>
> Have you tried using the Asterisk CLI with "sip set debug on" to
see
> if Asterisk shows any SIP packets?
>
> You might consider collecting a debug log with "sip set debug on"
> output 
> :https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
>
> Once you have that, provide a pastebin link to the output and someone 
> may be able to help you out.
>
> -- 
> Rusty Newton
> Digium, Inc. |Community Support Manager
> 445 Jan Davis Drive NW  -Huntsville, AL 35806  -US
> direct:  +1 256 428 6200  
>
> Check us out at:http://digium.com  &http://asterisk.org
>
>

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