We are having a problem with outbound calls. We make the call, it works but after for about 30 seconds the calls suddenly closes(hangs off) and disappears. Sometimes it works and we talk without problem but sometimes that weird problems happens, especially when we try to make the 2nd call at the same time(when a user is in an active Outgoing call and the other one tries to make another outgoing call then this 2nd call it will definitely closes after max 30 seconds). We didn’t change any configuration recently and it worked before.
Any clue where should I start to investigate this issue ?
Thank you for answering. I use something like : Gateways --> Trunks . I am not sure if we are talking for same thing? I configured it and it worked, I still can do outbound calls landline but when I try to do the 2nd call from another user at same time it stops after some seconds. SOMETIMES even the only outbound landline calls stops after some seconds. This issue is not happening with internal outbound calls.
I have some new updates about this issue. The problem happens every time that you try to do the 2nd call (while another is active) BUT the problem has also more than 50% chance to happen even if you only make a single call.
For example: Sometimes when I call a number, it works and I can talk without any problem or without any limitation. Later that day, or just few minutes later, I try to call the same number, from same network, same user, exactly the same circumstances, the outgoing call drops after exactly 10 seconds. I tried that several times and it happens over and over again.
I guess the problem happens somewhere here (I compared the CLI output of working call and not working one):
== Begin MixMonitor Recording SIP/qdgolgn5MRBai1U-000001f1
-- Executing [5000-dial@mdc_ivr-1:3] Gosub("SIP/3126047321-000001f3", "sub_main-5000,5000,1") in new stack
-- Executing [5000@sub_main-5000:1] BackGround("SIP/3126047321-000001f3", "VIPTV-ivr") in new stack
-- <SIP/3126047321-000001f3> Playing 'VIPTV-ivr.slin' (language 'en')
[Oct 8 15:33:03] WARNING: chan_sip.c:4050 retrans_pkt: Retransmission timeout reached on transmission 39d445d6f62f44d3811c7ec02e0d02f6 for seqno 40 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 8960ms with no response
[Oct 8 15:33:03] WARNING: chan_sip.c:4079 retrans_pkt: Hanging up call 39d445d6f62f44d3811c7ec02e0d02f6 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
-- Executing [h@sub_trunk-outgoing-19:1] Gosub("SIP/qdgolgn5MRBai1U-000001f1", "def_hangup,s,1(,ANSWER,,CALL)") in new stack
-- Executing [s@def_hangup:1] NoOp("SIP/qdgolgn5MRBai1U-000001f1", ">>>def_hangup:: EXTEN: DIALSTATUS: ANSWER QUEUESTATUS: REASON: CALL") in new stack
== Spawn extension (sub_trunk-outgoing-19, 13129625248, 11) exited non-zero on 'SIP/qdgolgn5MRBai1U-000001f1'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/qdgolgn5MRBai1U-000001f1
I attached only this part of OUTPUT, certainly there is more at the beginning and at the end.
Can you help us to find those features(SIP-ALG and Increase the NAT Table TTL) please ? We used Nat=Yes but that didn’t fix the issue. Even when we try to do a Sip Reload, it appears a warning message which says that Nat=Yes is depricated and suggest to use Nat=force_rport,comedia , we used that also but no fix.