Outgoing call dropped

Asterisk Asterisk Support
1

Hi alls,
I got sometimes outgoins calls suddenly dropped.
My asterisk version is 1.8.16.0 and I use aasra 6731i.
Call is dropped after 6seconds from receiving
[Oct 30 17:10:27] VERBOSE[13160] chan_sip.c: Scheduling destruction of SIP dialog ‘097f9e4a03984f19’ in 6400 ms (Method: SUBSCRIBE)

Peer has qualify = yes
Any idea ?

This is the logs with SIP debug
<------------->
[Oct 30 17:10:27] VERBOSE[13160] chan_sip.c: — (17 headers 0 lines) —
[Oct 30 17:10:27] VERBOSE[13160] chan_sip.c: Creating new subscription
[Oct 30 17:10:27] VERBOSE[13160] chan_sip.c: Sending to 10.11.5.34:5060 (NAT)
[Oct 30 17:10:27] VERBOSE[13160] chan_sip.c: list_route: hop: sip:7134@10.11.5.34:5060;transport=udp
[Oct 30 17:10:27] VERBOSE[13160] chan_sip.c: Found peer ‘7134’ for ‘7134’ from 10.11.5.34:5060
[Oct 30 17:10:27] VERBOSE[13160] chan_sip.c:
<— Transmitting (NAT) to 10.11.5.34:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.11.5.34;branch=z9hG4bK80d6d495db30e2da9;received=10.11.5.34;rport=5060
From: “7134” sip:7134@10.11.5.10:5060;tag=5c29b42119
To: sip:10.11.5.10:5060;tag=as1cb80f45
Call-ID: 097f9e4a03984f19
CSeq: 15379 SUBSCRIBE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0838693e"
Content-Length: 0

<------------>
[Oct 30 17:10:27] VERBOSE[13160] chan_sip.c: Scheduling destruction of SIP dialog ‘097f9e4a03984f19’ in 6400 ms (Method: SUBSCRIBE)
[Oct 30 17:10:31] VERBOSE[12935] asterisk.c: – Remote UNIX connection
[Oct 30 17:10:31] VERBOSE[1354] asterisk.c: – Remote UNIX connection disconnected
[Oct 30 17:10:33] VERBOSE[13160] chan_sip.c: Really destroying SIP dialog ‘097f9e4a03984f19’ Method: SUBSCRIBE
[Oct 30 17:10:36] VERBOSE[12935] asterisk.c: – Remote UNIX connection
[Oct 30 17:10:36] VERBOSE[1417] asterisk.c: – Remote UNIX connection disconnected
[Oct 30 17:10:38] VERBOSE[13158] sig_pri.c: – Accepting call from ‘00006212’ to ‘7002’ on channel 0/1, span 2
[Oct 30 17:10:38] VERBOSE[1449] pbx.c: – Executing [7002@DID_span_2:1] Goto(“DAHDI/i2/00006212-154a”, “app-adp-op,7002,1”) in new stack
[Oct 30 17:10:38] VERBOSE[1449] pbx.c: – Goto (app-adp-op,7002,1)
[Oct 30 17:10:38] VERBOSE[1449] pbx.c: – Executing [7002@app-adp-op:1] NoOp(“DAHDI/i2/00006212-154a”, “adp op700X”) in new stack
[Oct 30 17:10:38] VERBOSE[1449] pbx.c: – Executing [7002@app-adp-op:2] Dial(“DAHDI/i2/00006212-154a”, “SIP/7002,180,kKtHhTXxmi”) in new stack
[Oct 30 17:10:38] VERBOSE[1449] netsock2.c: == Using SIP RTP CoS mark 5
[Oct 30 17:10:38] VERBOSE[1449] app_dial.c: – Called SIP/7002
[Oct 30 17:10:38] VERBOSE[1449] res_musiconhold.c: – Started music on hold, class ‘default’, on DAHDI/i2/00006212-154a
[Oct 30 17:10:38] VERBOSE[1449] app_dial.c: – SIP/7002-0000607c is ringing
[Oct 30 17:10:40] VERBOSE[29280] pbx.c: – Executing [h@macro-trunkdial-failover-0.3:1] Macro(“SIP/7134-00006075”, “hangupcall,”) in new stack
[Oct 30 17:10:40] VERBOSE[29280] pbx.c: – Executing [s@macro-hangupcall:1] NoOp(“SIP/7134-00006075”, “macro hangupcall called”) in new stack
[Oct 30 17:10:40] VERBOSE[29280] pbx.c: – Executing [s@macro-hangupcall:2] Macro(“SIP/7134-00006075”, “complet-cdr,”) in new stack
[Oct 30 17:10:40] VERBOSE[29280] pbx.c: – Executing [s@macro-complet-cdr:1] Set(“SIP/7134-00006075”, “CDR(causecode)=16”) in new stack
[Oct 30 17:10:40] VERBOSE[29280] pbx.c: – Executing [s@macro-complet-cdr:2] Set(“SIP/7134-00006075”, “CDR(dialstatus)=ANSWER”) in new stack
[Oct 30 17:10:40] VERBOSE[29280] pbx.c: – Executing [s@macro-complet-cdr:3] Set(“SIP/7134-00006075”, “CDR(inoutstatus)=”) in new stack
[Oct 30 17:10:40] VERBOSE[29280] pbx.c: – Executing [s@macro-complet-cdr:4] Set(“SIP/7134-00006075”, “CDR(dnid)=00778663982”) in new stack
[Oct 30 17:10:40] VERBOSE[29280] pbx.c: – Executing [s@macro-complet-cdr:5] ResetCDR(“SIP/7134-00006075”, “w”) in new stack
[Oct 30 17:10:40] VERBOSE[29280] pbx.c: – Executing [s@macro-complet-cdr:6] NoCDR(“SIP/7134-00006075”, “”) in new stack
[Oct 30 17:10:40] VERBOSE[29280] pbx.c: – Executing [s@macro-complet-cdr:7] MacroExit(“SIP/7134-00006075”, “”) in new stack
[Oct 30 17:10:40] VERBOSE[29280] pbx.c: – Executing [s@macro-hangupcall:3] Hangup(“SIP/7134-00006075”, “”) in new stack
[Oct 30 17:10:40] VERBOSE[29280] app_macro.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘SIP/7134-00006075’ in macro ‘hangupcall’
[Oct 30 17:10:40] VERBOSE[29280] features.c: == Spawn extension (macro-trunkdial-failover-0.3, h, 1) exited non-zero on ‘SIP/7134-00006075’
[Oct 30 17:10:40] VERBOSE[29280] chan_dahdi.c: – Hungup ‘DAHDI/i1/0778663982-327c’

2

You provided too few details. But from the things that you provided it looks like the dialplan was automatically generated by a WebGUI. And those dialplans are a pain the a** to debug. If you are using FreepBX, it’s best to ask them :wink:

3

Hi
Thank for your response
You are right, I use gui but it is Asterisk-GUI :smile: and I don’t think that the problem is in dial macro cause it works but jsut sometimes calls dropped.
What kind of information can be useful ?

4

Having extensions.conf dialplans and a full debug of “sip set debug on” for the call would be a good start.

It’s hard to pin-point the problem where you have DAHDI, VoIP phone and a over-blown dialplan. That is the problem with the dialplans generated by WebGUI’s. Very hard to debug and pin point the problem.

5

Also note that AsteriskGUI is a dead product.

6

I know for asterisk-gui and used just for little people help :smile: .
I have some asterisks installed with the same configuration and it works well just this one broke me legs :smile:
I will change my dialplan for outgoing calls without gui macro and back later.