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

Raj Roy Ghandhi roy.gandhi at gmail.com
Mon Feb 23 05:51:15 CST 2015 

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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150223/21a9546a/attachment.html>

More information about the asterisk-users mailing list