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roy.gandhi at gmail.com
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 Posted: Mon Feb 23, 2015 6:51 am Post subject: [asterisk-users] Asterisk does not listed to port 5060 |
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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@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@unknown.invalid>.
Contact: <sip:+91711189078@10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+91712442211@unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1, <sip:+91712442211@unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E@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@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@unknown.invalid>.
Contact: <sip:+91711189078@10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+91712442211@unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1, <sip:+91712442211@unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E@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@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@unknown.invalid>.
Contact: <sip:+91711189078@10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+94722442200@unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1, <sip:+91712442211@unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E@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@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@unknown.invalid>.
Contact: <sip:+91711189078@10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+94722442200@unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1, <sip:+91712442211@unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E@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@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@unknown.invalid>.
Contact: <sip:+91711189078@10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+94722442200@unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1, <sip:+91712442211@unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E@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@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@unknown.invalid>.
Contact: <sip:+91711189078@10.85.0.24:5060;transport=udp>.
From: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>;tag=5BD23246313536415F1CF602.
P-Asserted-Identity: <sip:+91711189078@10.25.84.3 ([email]sip%3A%2B91711189078@10.25.84.3[/email]);user=phone>.
Privacy: none.
Supported: histinfo,100rel.
Request-Disposition: no-fork.
P-Early-Media: supported.
History-Info: <sip:+94722442200@unknown.invalid;user=phone?Privacy=history&Reason=SIP%3Bcause%3D302%3Btext%3D%22Moved%20Temporarily%22>;index=1, <sip:+91712442211@unknown.invalid;user=phone>;index=1.1.
Max-Forwards: 69.
Accept: application/sdp.
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK.
Call-ID: 1207028FDD814000006CCD7E@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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rnewton at digium.com
Guest
|
| Posted: Thu Feb 26, 2015 8:03 pm Post subject: [asterisk-users] Asterisk does not listed to port 5060 |
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On Mon, Feb 23, 2015 at 5:51 AM, Raj Roy Ghandhi <roy.gandhi@gmail.com (roy.gandhi@gmail.com)> wrote:
| Quote: | 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.
--
| Quote: | 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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amit at avhan.com
Guest
|
| Posted: Fri Feb 27, 2015 12:50 am Post subject: [asterisk-users] Asterisk does not listed to port 5060 |
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|
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:
| Quote: |
On Mon, Feb 23, 2015 at 5:51 AM, Raj Roy Ghandhi <roy.gandhi@gmail.com (roy.gandhi@gmail.com)> wrote:
| Quote: | 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.
--
| Quote: | 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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