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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150223/21a9546a/attachment.html>
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150226/15910557/attachment.html>
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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150227/5a73058d/attachment.html>