Outbound calls being dropped after update to Version 1.6.17 -34-0fc0946 64bit
Hi, Getting a strange issue, all incoming calls work great, but outgoing calls are getting dropped. Initially though that it was a NAT issue but verified the NAT and it is working properly. The proper external IP is being used and the gateway is registering with the service provider without issue. Checked the call flow with sngrep and wireshark and found some strange issues. 1. From my end, making the outbound call I see the call being connected and I receive a 200 OK that the call is connected and I have a connection. It will last about 1.5 minutes while at the same time I see 5-6 INVITE being sent from my Freeswitch to my service provider and then a BYE that it is hanging up. 2. From my service provider, they see the call established and then: SIP/SDP 82 Request: INVITE sip: telephonnumber@IPxxxxxx:5060; transport=udp, in-dialog The question is what is the "in-dialog" it seems to be causing the call to not be recognize that it is active and INVITE being generated and not being answered which causes the Bye Command to occur. Not certain how to fix. Any suggestions would be appreciated.
I have no idea if this is actually the case or not (logs should probably confirm this) but it's possible that the subsequent re-invites being sent by FS are session timer refresh requests. If your gateway advertises support for session timers, but doesn't respond to re-invites, that would be a problem with your gateway. Of course you could disable session timers on FS if that is the case. Best, Colin On Mon, May 1, 2017 at 3:13 PM, Charles Bujold <cjbujold@accra.ca> wrote:
Hi,
Getting a strange issue, all incoming calls work great, but outgoing calls are getting dropped.
Initially though that it was a NAT issue but verified the NAT and it is working properly. The proper external IP is being used and the gateway is registering with the service provider without issue.
Checked the call flow with sngrep and wireshark and found some strange issues.
1. From my end, making the outbound call I see the call being connected and I receive a 200 OK that the call is connected and I have a connection. It will last about 1.5 minutes while at the same time I see 5-6 INVITE being sent from my Freeswitch to my service provider and then a BYE that it is hanging up.
1. From my service provider, they see the call established and then: SIP/SDP 82 Request: INVITE sip: telephonnumber@IPxxxxxx:5060; transport=udp, in-dialog
The question is what is the “in-dialog” it seems to be causing the call to not be recognize that it is active and INVITE being generated and not being answered which causes the Bye Command to occur.
Not certain how to fix. Any suggestions would be appreciated.
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[image: cid:image003.png@01D2C294.73F37890]
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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you can’t tell just from the data provided, but these issues are usually nat related. Inspect the IP addrs and ports in the sip packet and on the wire and see if all looks right.
On May 1, 2017, at 3:13 PM, Charles Bujold <cjbujold@accra.ca> wrote:
Hi,
Getting a strange issue, all incoming calls work great, but outgoing calls are getting dropped.
Initially though that it was a NAT issue but verified the NAT and it is working properly. The proper external IP is being used and the gateway is registering with the service provider without issue.
Checked the call flow with sngrep and wireshark and found some strange issues.
From my end, making the outbound call I see the call being connected and I receive a 200 OK that the call is connected and I have a connection. It will last about 1.5 minutes while at the same time I see 5-6 INVITE being sent from my Freeswitch to my service provider and then a BYE that it is hanging up.
From my service provider, they see the call established and then: SIP/SDP 82 Request: INVITE sip: telephonnumber@IPxxxxxx:5060; transport=udp, in-dialog
The question is what is the “in-dialog” it seems to be causing the call to not be recognize that it is active and INVITE being generated and not being answered which causes the Bye Command to occur.
Not certain how to fix. Any suggestions would be appreciated.
I can confirm that issue related to fragmented UDP packets from carrier equipment. Try swith to TCP transport. пн, 1 мая 2017, 23:12 Michael Jerris <mike@jerris.com>:
you can’t tell just from the data provided, but these issues are usually nat related. Inspect the IP addrs and ports in the sip packet and on the wire and see if all looks right.
On May 1, 2017, at 3:13 PM, Charles Bujold <cjbujold@accra.ca> wrote:
Hi,
Getting a strange issue, all incoming calls work great, but outgoing calls are getting dropped.
Initially though that it was a NAT issue but verified the NAT and it is working properly. The proper external IP is being used and the gateway is registering with the service provider without issue.
Checked the call flow with sngrep and wireshark and found some strange issues.
1. From my end, making the outbound call I see the call being connected and I receive a 200 OK that the call is connected and I have a connection. It will last about 1.5 minutes while at the same time I see 5-6 INVITE being sent from my Freeswitch to my service provider and then a BYE that it is hanging up.
1. From my service provider, they see the call established and then: SIP/SDP 82 Request: INVITE sip: telephonnumber@IPxxxxxx:5060; transport=udp, in-dialog
The question is what is the “in-dialog” it seems to be causing the call to not be recognize that it is active and INVITE being generated and not being answered which causes the Bye Command to occur.
Not certain how to fix. Any suggestions would be appreciated.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You can not possibly confirm that from the data provided. What you stated is worth a try, but there was not enough information to even indicate that is the issue.
On May 1, 2017, at 9:19 PM, Sergey Safarov <s.safarov@gmail.com> wrote:
I can confirm that issue related to fragmented UDP packets from carrier equipment. Try swith to TCP transport.
пн, 1 мая 2017, 23:12 Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>>: you can’t tell just from the data provided, but these issues are usually nat related. Inspect the IP addrs and ports in the sip packet and on the wire and see if all looks right.
On May 1, 2017, at 3:13 PM, Charles Bujold <cjbujold@accra.ca <mailto:cjbujold@accra.ca>> wrote:
Hi,
Getting a strange issue, all incoming calls work great, but outgoing calls are getting dropped.
Initially though that it was a NAT issue but verified the NAT and it is working properly. The proper external IP is being used and the gateway is registering with the service provider without issue.
Checked the call flow with sngrep and wireshark and found some strange issues.
From my end, making the outbound call I see the call being connected and I receive a 200 OK that the call is connected and I have a connection. It will last about 1.5 minutes while at the same time I see 5-6 INVITE being sent from my Freeswitch to my service provider and then a BYE that it is hanging up.
From my service provider, they see the call established and then: SIP/SDP 82 Request: INVITE sip: telephonnumber@IPxxxxxx:5060; transport=udp, in-dialog
The question is what is the “in-dialog” it seems to be causing the call to not be recognize that it is active and INVITE being generated and not being answered which causes the Bye Command to occur.
Not certain how to fix. Any suggestions would be appreciated.
participants (4)
-
Charles Bujold
-
Colin Morelli
-
Michael Jerris
-
Sergey Safarov