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 Post subject: Calls from customer PBX dropping after ~6-8 seconds
PostPosted: Sun Aug 04, 2013 11:35 pm 
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Joined: Tue Aug 14, 2012 10:58 pm
Posts: 6
Hi All,

I'm having an issue with a customer setup where outbound calls from the customer PBX are dropping after approx 6-8 seconds. Calls inbound to the PBX work fine.

I can see from the trace on the A2billing server that once the call is established there is a continuous exchange between the A2biling server & the PBX in which a@billing sends a 200 OK message and the PBX replies with an ACK, but then the A2billing re-transmits the OK ( See debug in next post)

For simplicity & security I have replaced the real IP & hostname of the A2billing server in the logs to: 9.9.9.9 (my.a2billing.server) and the PBX IP to 5.5.5.5

I know this is a problem, but cannot see what i can do to resolve. Any assistance would be greatly appreciated.


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 Post subject: Re: Calls from customer PBX dropping after ~6-8 seconds
PostPosted: Sun Aug 04, 2013 11:35 pm 
Offline

Joined: Tue Aug 14, 2012 10:58 pm
Posts: 6
E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
INVITE sip:[email protected];transport=tcp SIP/2.0
From: <sip:[email protected]>;tag=A2443246313536410009F6A0
To: <sip:[email protected]:5060>
Contact: sip:[email protected]:5060;transport=tcp
Content-Type: application/sdp
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=no;privacy=full
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: [email protected]
CSeq: 1 INVITE
Route: <sip:my.a2billing.server;lr>
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK5F8006C6C7104D15
Content-Length: 304

v=0
o=- 0 0 IN IP4 5.5.5.5
s=T035
c=IN IP4 5.5.5.5
t=0 0
m=audio 10020 RTP/AVP 8 2 18 9 110
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:9 G722/8000
a=ptime:30
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
<------------->

E2-SIP-01*CLI>
--- (16 headers 16 lines) ---
Sending to 5.5.5.5:5060 (NAT)
Using INVITE request as basis request - [email protected]

E2-SIP-01*CLI>
Found peer '0744461921' for 'anonymous' from 5.5.5.5:5060

<--- Reliably Transmitting (NAT) to 5.5.5.5:5060 --->
SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK5F8006C6C7104D15;received=5.5.5.5;rport=5060From: <sip:[email protected]>;tag=A2443246313536410009F6A0To: <sip:[email protected]:5060>;tag=as4bf6427aCall-ID: [email protected]: 1 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL,

OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4928fcee"Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]' in 8576 ms (Method: INVITE)

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
ACK sip:[email protected];transport=tcp SIP/2.0
Call-ID: [email protected]
CSeq: 1 ACK
From: <sip:[email protected]>;tag=A2443246313536410009F6A0
To: <sip:[email protected]:5060>;tag=as4bf6427a
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK5F8006C6C7104D15
Max-Forwards: 70
Route: <sip:my.a2billing.server;lr>
User-Agent: NEC-i SV8100-GE 08.00
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

E2-SIP-01*CLI>
 -- SIP/SOUL_OUTBOUND-0000dbdb is ringing

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
INVITE sip:[email protected];transport=tcp SIP/2.0
From: <sip:[email protected]>;tag=A2653246313536410009F6A3
To: <sip:[email protected]:5060>
Contact: sip:[email protected]:5060;transport=tcp
Content-Type: application/sdp
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=no;privacy=full
CSeq: 2 INVITE
Authorization: Digest username="0744461921",realm="asterisk",algorithm=MD5,nonce="4928fcee",opaque="",uri="sip:[email protected]",response="0027de822c0886593adc7b6415bbb60a"
Allow: INVITE,ACK,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,timer
Expires: 180
Call-ID: [email protected]
Route: <sip:my.a2billing.server;lr>
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818
Content-Length: 304

v=0
o=- 0 0 IN IP4 5.5.5.5
s=T035
c=IN IP4 5.5.5.5
t=0 0
m=audio 10020 RTP/AVP 8 2 18 9 110
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:2 G726-32/8000
a=ptime:30
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:9 G722/8000
a=ptime:30
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
<------------->

E2-SIP-01*CLI>
--- (17 headers 16 lines) ---
Sending to 5.5.5.5:5060 (NAT)
Using INVITE request as basis request - [email protected]

E2-SIP-01*CLI>
Found peer '0744461921' for 'anonymous' from 5.5.5.5:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 9

E2-SIP-01*CLI>
Found RTP audio format 110
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 110
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x1908 (alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

E2-SIP-01*CLI>
Peer audio RTP is at port 5.5.5.5:10020
Looking for 0386969318 in empower-private (domain my.a2billing.server)
list_route: hop: <sip:[email protected]:5060;transport=tcp>

E2-SIP-01*CLI>

<--- Transmitting (NAT) to 5.5.5.5:5060 --->
SIP/2.0 100 TryingVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: <sip:[email protected]>;tag=A2653246313536410009F6A3To: <sip:[email protected]:5060>Call-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,

REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: <sip:[email protected]:5060>Content-Length: 0
> Limit Data for this call:

E2-SIP-01*CLI>
 > timelimit = 7200000 ms (7200.000 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =

E2-SIP-01*CLI>
 == Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

E2-SIP-01*CLI>
 -- Called SIP/SOUL_OUTBOUND/0386969318

<--- Transmitting (NAT) to 5.5.5.5:5060 --->
SIP/2.0 180 RingingVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: <sip:[email protected]>;tag=A2653246313536410009F6A3To: <sip:[email protected]:5060>;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL,

OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: <sip:[email protected]:5060>Content-Length: 0
<------------>

E2-SIP-01*CLI>
 -- SIP/SOUL_OUTBOUND-0000dbdb is ringing

E2-SIP-01*CLI>
 -- SIP/SOUL_OUTBOUND-0000dbdb answered SIP/0884936015-0000dbda

E2-SIP-01*CLI>
 -- SIP/SOUL_OUTBOUND-0000dbdd is ringing

<--- Transmitting (NAT) to 5.5.5.5:5060 --->
SIP/2.0 180 RingingVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: <sip:[email protected]>;tag=A2653246313536410009F6A3To: <sip:[email protected]:5060>;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL,

OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: <sip:[email protected]:5060>Content-Length: 0
<------------>

E2-SIP-01*CLI>
 -- SIP/SOUL_OUTBOUND-0000dbdd is making progress passing it to SIP/0744461921-0000dbdc

E2-SIP-01*CLI>
 -- SIP/SOUL_OUTBOUND-0000dbdd answered SIP/0744461921-0000dbdc

E2-SIP-01*CLI>
Audio is at 19900
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 5.5.5.5:5060 --->
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: <sip:[email protected]>;tag=A2653246313536410009F6A3To: <sip:[email protected]:5060>;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL,

OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: <sip:[email protected]:5060>Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110

telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv
<------------>

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: <sip:[email protected]:5060>;tag=as6df6ebba
From: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

E2-SIP-01*CLI>
Retransmitting #1 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: <sip:[email protected]>;tag=A2653246313536410009F6A3To: <sip:[email protected]:5060>;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL,

OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: <sip:[email protected]:5060>Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110

telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: <sip:[email protected]:5060>;tag=as6df6ebba
From: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

E2-SIP-01*CLI>
Retransmitting #2 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: <sip:[email protected]>;tag=A2653246313536410009F6A3To: <sip:[email protected]:5060>;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL,

OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: <sip:[email protected]:5060>Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110

telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: <sip:[email protected]:5060>;tag=as6df6ebba
From: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

E2-SIP-01*CLI>
Retransmitting #3 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: <sip:[email protected]>;tag=A2653246313536410009F6A3To: <sip:[email protected]:5060>;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL,

OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: <sip:[email protected]:5060>Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110

telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: <sip:[email protected]:5060>;tag=as6df6ebba
From: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

E2-SIP-01*CLI>
Retransmitting #4 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: <sip:[email protected]>;tag=A2653246313536410009F6A3To: <sip:[email protected]:5060>;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL,

OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: <sip:[email protected]:5060>Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110

telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: <sip:[email protected]:5060>;tag=as6df6ebba
From: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

E2-SIP-01*CLI>
 == Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> Limit Data for this call:
> timelimit = 7200000 ms (7200.000 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =

E2-SIP-01*CLI>
 == Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

E2-SIP-01*CLI>
 -- Called SIP/SOUL_OUTBOUND/0396207531

E2-SIP-01*CLI>
 -- SIP/SOUL_OUTBOUND-0000dbdf is ringing

E2-SIP-01*CLI>
 -- SIP/SOUL_OUTBOUND-0000dbdf is making progress passing it to SIP/2657549926-0000dbde

E2-SIP-01*CLI>
Retransmitting #5 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: <sip:[email protected]>;tag=A2653246313536410009F6A3To: <sip:[email protected]:5060>;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL,

OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: <sip:[email protected]:5060>Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110

telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: <sip:[email protected]:5060>;tag=as6df6ebba
From: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

E2-SIP-01*CLI>
Retransmitting #6 (NAT) to 5.5.5.5:5060:
SIP/2.0 200 OKVia: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK27D573A1AB3FE818;received=5.5.5.5;rport=5060From: <sip:[email protected]>;tag=A2653246313536410009F6A3To: <sip:[email protected]:5060>;tag=as6df6ebbaCall-ID: [email protected]: 2 INVITEServer: FPBX-2.8.1 (1.8.15.0)Allow: INVITE, ACK, CANCEL,

OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISHSupported: replaces, timerContact: <sip:[email protected]:5060>Content-Type: application/sdpContent-Length: 284v=0o=root 990094168 990094168 IN IP4 9.9.9.9s=Asterisk PBX 1.8.15.0c=IN IP4 9.9.9.9t=0 0m=audio 19900 RTP/AVP 8 18 110a=rtpmap:8 PCMA/8000a=rtpmap:18 G729/8000a=fmtp:18 annexb=noa=rtpmap:110

telephone-event/8000a=fmtp:110 0-16a=ptime:20a=sendrecv
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
CSeq: 2 ACK
To: <sip:[email protected]:5060>;tag=as6df6ebba
From: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
Max-Forwards: 70
User-Agent: NEC-i SV8100-GE 08.00
Via: SIP/2.0/UDP 5.5.5.5:5060;branch=z9hG4bK987C97F161461804
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---


E2-SIP-01*CLI>
Reliably Transmitting (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: <sip:[email protected]:5060>;tag=as6df6ebbaTo: <sip:[email protected]>;tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-

Authorization: Digest username="0744461921", realm="asterisk", algorithm=MD5, uri="sip:my.a2billing.server", nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
SIP/2.0 200 OK
From: <sip:[email protected]:5060>;tag=as6df6ebba
To: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

E2-SIP-01*CLI>
Retransmitting #1 (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: <sip:[email protected]:5060>;tag=as6df6ebbaTo: <sip:[email protected]>;tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-

Authorization: Digest username="0744461921", realm="asterisk", algorithm=MD5, uri="sip:my.a2billing.server", nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
SIP/2.0 200 OK
From: <sip:[email protected]:5060>;tag=as6df6ebba
To: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

E2-SIP-01*CLI>
Retransmitting #2 (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: <sip:[email protected]:5060>;tag=as6df6ebbaTo: <sip:[email protected]>;tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-

Authorization: Digest username="0744461921", realm="asterisk", algorithm=MD5, uri="sip:my.a2billing.server", nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
SIP/2.0 200 OK
From: <sip:[email protected]:5060>;tag=as6df6ebba
To: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

E2-SIP-01*CLI>
Retransmitting #3 (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: <sip:[email protected]:5060>;tag=as6df6ebbaTo: <sip:[email protected]>;tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-

Authorization: Digest username="0744461921", realm="asterisk", algorithm=MD5, uri="sip:my.a2billing.server", nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
SIP/2.0 200 OK
From: <sip:[email protected]:5060>;tag=as6df6ebba
To: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

E2-SIP-01*CLI>
Retransmitting #4 (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: <sip:[email protected]:5060>;tag=as6df6ebbaTo: <sip:[email protected]>;tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-

Authorization: Digest username="0744461921", realm="asterisk", algorithm=MD5, uri="sip:my.a2billing.server", nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
SIP/2.0 200 OK
From: <sip:[email protected]:5060>;tag=as6df6ebba
To: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

E2-SIP-01*CLI>
Retransmitting #5 (NAT) to 5.5.5.5:5060:
BYE sip:[email protected]:5060;transport=tcp SIP/2.0Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rportMax-Forwards: 70From: <sip:[email protected]:5060>;tag=as6df6ebbaTo: <sip:[email protected]>;tag=A2653246313536410009F6A3Call-ID: [email protected]: 102 BYEUser-Agent: FPBX-2.8.1 (1.8.15.0)Proxy-

Authorization: Digest username="0744461921", realm="asterisk", algorithm=MD5, uri="sip:my.a2billing.server", nonce="", response="963caa5a84822bd1da888a3022080ae3"X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0
---

E2-SIP-01*CLI>

<--- SIP read from UDP:5.5.5.5:5060 --->
SIP/2.0 200 OK
From: <sip:[email protected]:5060>;tag=as6df6ebba
To: <sip:[email protected]>;tag=A2653246313536410009F6A3
Call-ID: [email protected]
CSeq: 102 BYE
Server: NEC-i SV8100-GE 08.00/2.1
Via: SIP/2.0/UDP 9.9.9.9:5060;branch=z9hG4bK7128067b;rport
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

E2-SIP-01*CLI> quit]0;baismanage@E2-SIP-01: ~baismanage@E2-SIP-01:~$


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 Post subject: Re: Calls from customer PBX dropping after ~6-8 seconds
PostPosted: Wed Aug 07, 2013 9:00 am 
Offline

Joined: Tue Aug 14, 2012 10:58 pm
Posts: 6
Hi,

I think the issue is realted to the reinvites, as i can see the asterisk is re-transmitting the same invite during the call several times.

I want to trun off invites for this account, but setting canreinvite to no appear to have no impact. can some one advise how i can set this up..

Regards


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 Post subject: Re: Calls from customer PBX dropping after ~6-8 seconds
PostPosted: Wed Aug 07, 2013 12:35 pm 
Offline

Joined: Mon Mar 02, 2009 8:56 pm
Posts: 271
You're right that canreinvite should do it. Do you add it to both the a2billing customer SIP account and also the outbound trunk from a2billing?


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