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Inbound Call without sound and drop after 6 sec

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Inbound Call without sound and drop after 6 sec

Postby iboam » Mon Apr 11, 2016 11:19 am

I HAVE A CLUSTER SERVER Running ViciBox_v.7.x86_64-7.0.2
1 - DB / VERSION: 2.12-549a BUILD: 160404-0940
1 - ASTERISK / VERSION: 2.12-549a BUILD: 160404-0940
1 - WEB / VERSION: 2.12-549a BUILD: 160404-0940


I'm able to make outbound calls without any issue, but inbound calls have no sound and call drop after 5 - 6 sec

I'm forwarding my did to my personal cellphone
DID = 91INBOUND
Personal Cell Phone = 91FORWARD-DID

Code: Select all
[Apr 11 12:07:18]     -- Executing [91INBOUND-DID@default:1] AGI("SIP/2000-00000068", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 11 12:07:18]     -- <SIP/2000-00000068>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 11 12:07:18]     -- Executing [91INBOUND-DID@default:2] Dial("SIP/2000-00000068", "SIP/SIP-Provider/+1INBOUND-DID,,To") in new stack
[Apr 11 12:07:18]   == Using SIP RTP CoS mark 5
[Apr 11 12:07:18]     -- Called SIP/SIP-Provider/+1INBOUND-DID
[Apr 11 12:07:19]   == Using SIP RTP CoS mark 5
[Apr 11 12:07:19]     -- Executing [+1INBOUND-DID@trunkinbound:1] NoOp("SIP/SIP-Provider-0000006a", "Stripping + from start of number, for annoying carriers who insist") in new stack
[Apr 11 12:07:19]     -- Executing [+1INBOUND-DID@trunkinbound:2] Goto("SIP/SIP-Provider-0000006a", "trunkinbound,1INBOUND-DID,1") in new stack
[Apr 11 12:07:19]     -- Goto (trunkinbound,1INBOUND-DID,1)
[Apr 11 12:07:19]     -- Executing [1INBOUND-DID@trunkinbound:1] NoOp("SIP/SIP-Provider-0000006a", "X,1INBOUND-DID,1INBOUND-DID") in new stack
[Apr 11 12:07:19]     -- Executing [1INBOUND-DID@trunkinbound:2] AGI("SIP/SIP-Provider-0000006a", "agi-DID_route.agi") in new stack
[Apr 11 12:07:19]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Apr 11 12:07:19]     -- <SIP/SIP-Provider-0000006a>AGI Script agi-DID_route.agi completed, returning 0
[Apr 11 12:07:19]     -- Executing [91FORWARD-DID@default:1] AGI("SIP/SIP-Provider-0000006a", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 11 12:07:19]     -- <SIP/SIP-Provider-0000006a>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 11 12:07:19]     -- Executing [91FORWARD-DID@default:2] Dial("SIP/SIP-Provider-0000006a", "SIP/SIP-Provider/+1FORWARD-DID,,To") in new stack
[Apr 11 12:07:19]   == Using SIP RTP CoS mark 5
[Apr 11 12:07:19]     -- Called SIP/SIP-Provider/+1FORWARD-DID
[Apr 11 12:07:22]     -- SIP/SIP-Provider-0000006b is making progress passing it to SIP/SIP-Provider-0000006a
[Apr 11 12:07:22]     -- SIP/SIP-Provider-00000069 is making progress passing it to SIP/2000-00000068
[Apr 11 12:07:22]        > 0x7fd3e005e220 -- Probation passed - setting RTP source address to 10.1.100.54:58458
[Apr 11 12:07:25]     -- SIP/SIP-Provider-0000006b answered SIP/SIP-Provider-0000006a
[Apr 11 12:07:25]     -- SIP/SIP-Provider-00000069 answered SIP/2000-00000068
[Apr 11 12:07:32] WARNING[1653]: chan_sip.c:4031 retrans_pkt: Retransmission timeout reached on transmission 08d894c200f91dce5b7e3414d9d0e948@0.0.0.0 for seqno 103 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Apr 11 12:07:32] WARNING[1653]: chan_sip.c:4060 retrans_pkt: Hanging up call 08d894c200f91dce5b7e3414d9d0e948@0.0.0.0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Apr 11 12:07:32]     -- Executing [h@default:1] AGI("SIP/SIP-Provider-0000006a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----ANSWER-----13-----7") in new stack
[Apr 11 12:07:32]     -- <SIP/SIP-Provider-0000006a>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----ANSWER-----13-----7 completed, returning 0
[Apr 11 12:07:32]   == Spawn extension (default, 91FORWARD-DID, 2) exited non-zero on 'SIP/SIP-Provider-0000006a'


plase help
ViciBox: 11 | VERSION: 2.14-897a BUILD: 230927-0857 | Clusters: 1 DB-WEB-ASTX | SSL | WebRTC | Wallboard | DNC Nightly Scrubber
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Re: Inbound Call without sound and drop after 6 sec

Postby williamconley » Mon Apr 11, 2016 2:18 pm

Temporarily turn off your firewall. Test inbound sound. Turn back on your firewall.

If you had sound ... you must contact your carrier and ask ALL their IPs and add all of them to the firewall.

Good job posting your specs. 8-)
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Re: Inbound Call without sound and drop after 6 sec

Postby iboam » Mon Apr 11, 2016 2:30 pm

Yes i think is something related firewall / ports, I have this cluster and another single server up and running, inbound and outbound, both server are with different public ip same carrier, both public ip are registered into the carrier, i have this error only in the cluster the single server is OK
ViciBox: 11 | VERSION: 2.14-897a BUILD: 230927-0857 | Clusters: 1 DB-WEB-ASTX | SSL | WebRTC | Wallboard | DNC Nightly Scrubber
iboam
 
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Re: Inbound Call without sound and drop after 6 sec

Postby williamconley » Mon Apr 11, 2016 3:14 pm

"I Think" this is firewall related. Please turn off the firewall for the purpose of testing one call. Then you can remove the "I Think" from your statement and get the problem resolved.
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Re: Inbound Call without sound and drop after 6 sec

Postby iboam » Tue Apr 12, 2016 10:33 am

I moved my single server (express) to where the asterisk (cluster) is connected, and i was able to make outbound and get inbound calls, now i just switch and still getting same error

Inbound calls are forwarded to extension 2000 but drop after few seconds

Code: Select all
 == Using SIP RTP CoS mark 5
[Apr 12 11:27:22]     -- Executing [+1INBOUND-DID@trunkinbound:1] NoOp("SIP/SIP-PROVIDER-0000010a", "Stripping + from start of number, for annoying carriers who insist") in new stack
[Apr 12 11:27:22]     -- Executing [+1INBOUND-DID@trunkinbound:2] Goto("SIP/SIP-PROVIDER-0000010a", "trunkinbound,1INBOUND-DID,1") in new stack
[Apr 12 11:27:22]     -- Goto (trunkinbound,1INBOUND-DID,1)
[Apr 12 11:27:22]     -- Executing [1INBOUND-DID@trunkinbound:1] NoOp("SIP/SIP-PROVIDER-0000010a", "X,1INBOUND-DID,1INBOUND-DID") in new stack
[Apr 12 11:27:22]     -- Executing [1INBOUND-DID@trunkinbound:2] AGI("SIP/SIP-PROVIDER-0000010a", "agi-DID_route.agi") in new stack
[Apr 12 11:27:22]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Apr 12 11:27:22]     -- <SIP/SIP-PROVIDER-0000010a>AGI Script agi-DID_route.agi completed, returning 0
[Apr 12 11:27:22]     -- Executing [010*001*020*035*2000@default:1] Goto("SIP/SIP-PROVIDER-0000010a", "default,2000,1") in new stack
[Apr 12 11:27:22]     -- Goto (default,2000,1)
[Apr 12 11:27:22]     -- Executing [2000@default:1] Dial("SIP/SIP-PROVIDER-0000010a", "SIP/2000,60,") in new stack
[Apr 12 11:27:22]   == Using SIP RTP CoS mark 5
[Apr 12 11:27:22]     -- Called SIP/2000
[Apr 12 11:27:22]     -- SIP/2000-0000010b is ringing
[Apr 12 11:27:22]     -- SIP/SIP-PROVIDER-00000109 is ringing
[Apr 12 11:27:25]     -- SIP/2000-0000010b answered SIP/SIP-PROVIDER-0000010a
[Apr 12 11:27:25]     -- Locally bridging SIP/SIP-PROVIDER-0000010a and SIP/2000-0000010b
[Apr 12 11:27:25]        > 0x7fd38c014d30 -- Probation passed - setting RTP source address to 10.1.100.54:34076
[Apr 12 11:27:25]     -- SIP/SIP-PROVIDER-00000109 answered SIP/2000-00000108
[Apr 12 11:27:25]     -- Started music on hold, class 'default', on SIP/SIP-PROVIDER-00000109
[Apr 12 11:27:25]        > 0x7fd3e0377930 -- Probation passed - setting RTP source address to 10.1.100.54:17112
[Apr 12 11:27:25]        > 0x7fd3e05d7a90 -- Probation passed - setting RTP source address to 54.172.60.129:10140
[Apr 12 11:27:25]        > 0x7fd3a800c990 -- Probation passed - setting RTP source address to 54.172.60.135:18550
[Apr 12 11:27:31] WARNING[1653]: chan_sip.c:4031 retrans_pkt: Retransmission timeout reached on transmission dae288430843b8e8412447cd42357ffa@0.0.0.0 for seqno 103 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[Apr 12 11:27:31] WARNING[1653]: chan_sip.c:4060 retrans_pkt: Hanging up call dae288430843b8e8412447cd42357ffa@0.0.0.0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[Apr 12 11:27:31]     -- Executing [h@default:1] AGI("SIP/SIP-PROVIDER-0000010a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----ANSWER-----9-----6") in new stack
[Apr 12 11:27:31]     -- <SIP/SIP-PROVIDER-0000010a>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----ANSWER-----9-----6 completed, returning 0
[Apr 12 11:27:31]   == Spawn extension (default, 2000, 1) exited non-zero on 'SIP/SIP-PROVIDER-0000010a'
[Apr 12 11:27:34]     -- Stopped music on hold on SIP/SIP-PROVIDER-00000109
[Apr 12 11:27:34]        > 0x7fd3e0377930 -- Probation passed - setting RTP source address to 10.1.100.54:17112
[Apr 12 11:27:35]     -- Executing [h@default:1] AGI("SIP/2000-00000108", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----30-----10") in new stack
[Apr 12 11:27:35]     -- <SIP/2000-00000108>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----30-----10 completed, returning 0
[Apr 12 11:27:35]   == Spawn extension (default, 91INBOUND-DID, 2) exited non-zero on 'SIP/2000-00000108'
ViciBox: 11 | VERSION: 2.14-897a BUILD: 230927-0857 | Clusters: 1 DB-WEB-ASTX | SSL | WebRTC | Wallboard | DNC Nightly Scrubber
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Re: Inbound Call without sound and drop after 6 sec

Postby williamconley » Tue Apr 12, 2016 1:19 pm

iboam wrote:I moved my single server (express) to where the asterisk (cluster) is connected, and i was able to make outbound and get inbound calls, now i just switch and still getting same error

Inbound calls are forwarded to extension 2000 but drop after few seconds


"now i just switch ..."

what does that mean?
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
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Re: Inbound Call without sound and drop after 6 sec

Postby ambiorixg12 » Tue Apr 12, 2016 9:37 pm

This problem is related to NAT issue, open the link that is on your Asterisk CLI, and you will get some recomendations. Also on the Asterisk side there are some setting that need to be adjusted if you server is behind a NATTED network, also specify an stun server on the cliente side would help a lot

[Apr 11 12:07:32] WARNING[1653]: chan_sip.c:4031 retrans_pkt: Retransmission timeout reached on transmission 08d894c200f91dce5b7e3414d9d0e948@0.0.0.0 for seqno 103 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 6400ms with no response
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