V. 1.8.4 fake ring


#1

Hello,
I have an asterisk box runing under the latest v. 1.8.4 and billing the CDR with a2billing 1.9.4. Where everything it runing properly.

But, I’m providing Voip Wholesale service, si some of my partners communicate that they are having FASE answer calls, meanwhile the calls still in progress.

I Have as dialcommand programm the following:

Inside a2billing i have the answer_calls=no

Then the asterisk dialplan, when i have this one:

[quote][a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,AGI(a2billing.php,2)
exten => _X.,n,Hangup[/quote]

Everything it’s working properly. But the FAKE is there…

When i edit the dialplan by removing the answer line, nothing work, and no way to progress calls…

and when I change the dialplan to this one:

[quote][a2billing-sip]
exten => _X.,1,AGI(a2billing.php,2)
exten => _X.,n,Hangup
[/quote]
But when i do that I see in cli:

Any body can help to figure out this issue with v.1.8.4??

Regards,


#2

the message is pretty clear… your context are not identical (a2billign and a2billing-sip) and the extension is not found… check the exten => lines


#3

Thanks Jean for your fast prompt.
In the fist case, when all it’s working, I’m refering to a2billing context, then in the second I changed the contest in the peer to a2billing-sip as a second possibility, but as you see asterisk still turning, looking for that extension, and not finding it…

let’s say easer…

When I have this context:

[quote][a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,AGI(a2billing.php,2)
exten => _X.,n,Hangup[/quote]

Everything it’s working, progressing, and correct, but there’s FAKE issue…

So, when I edit the context as below:

[quote][a2billing]
exten => _X.,1,AGI(a2billing.php,2)
exten => _X.,n,Hangup[/quote]

Asterisk is showing this:

Thanks again,


#4

check the syntax around your context definition… it has to do with this… post your exact extensions.conf and the CLI exact message


#5

Thanks again Jean,

in extensions_a2billing.conf I have the following:
In fact A.
[a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,AGI(a2billing.php,2)
exten => _X.,n,Hangup

User is appointing to context=a2billing

As I said, it’s working, but giving FAKE, meanwhile the call still in progress… In the server Cli I have the following:

[quote]Connected to Asterisk 1.8.4.2 currently running on myserver (pid = 247)
Verbosity is at least 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [XXXXXXXXX@a2billing:1] Answer(“SIP/XXXX-00000064”, “”) in new stack
– Executing [XXXXXXXX@a2billing:2] Wait(“SIP/XXXXX-00000064”, “1”) in new stack
[Jun 28 16:18:04] NOTICE[10409]: channel.c:4071 __ast_read: Dropping incompatible voice frame on SIP/XXXXXXX-00000064 of format gsm since our native format has changed to 0x4 (ulaw)
– Executing [xxxxxxx@a2billing:3] AGI(“SIP/06604-00000064”, “a2billing.php,2”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
– AGI Script Executing Application: (DIAL) Options: (SIP/TrunkC/xxxxxxxxx,60,HL(715380000:61000:30000))
> Limit Data for this call:
> timelimit = 715380000 ms (715380.000 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 =
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called TrunkC/XXXXXXXXX
– SIP/TrunkC-00000065 is making progress passing it to SIP/XXXXX-00000064
– SIP/TrunkC-00000065 is ringing
– SIP/TrunkC-00000065 is making progress passing it to SIP/XXXXX-00000064
– <SIP/06604-00000064>AGI Script a2billing.php completed, returning -1
[/quote]

The problem here, in this scenario is, just when asterisk server is ringing as [quote] – SIP/TrunkC-00000065 is ringing[/quote]
Is showing immediatly answer call in the client server…

So, the solution which I wanted to do is to change the dialplan as below:

[quote][a2billing]
exten = _X.,n,AGI(a2billing.php,2)
exten = _X.,n,Hangup[/quote]

So, when i make a call, asterisk is just showing this message:

The first is working, but causing FAKE, and the second is not working at all…

Regards,


#6

here we go…

[a2billing] exten = _X.,n,AGI(a2billing.php,2) exten = _X.,n,Hangup

replace the ,n, in the first line with ,1, -> you always need a prio #1


#7

Thanks again Jean, this one work, but also it’s still causing FAKE, in the client server it’s showing answered call, meanwhile in the Asterisk Server it still progressing the call, so it’s causing billing problems, and this is what we need to figure out…

it’s really strange, here we’re putting asterisk dial plan in A2billing agi, and there I have:

[quote]
dialcommand_param= ,60,HL(%timeout%:61000:30000)
dialcommand_param_sipiax_friend= ,60,HL(3600000:61000:30000)
answer_call = NO [/quote]

I don’t understand from where asterisk still getting the answer command…

Regards,


#8

you may want to do a sip set debug and check the exact times at which the call is answered, by who


#9

I stay without word, to say Thanks again Jean,

After sip set debug I don’t see any answer message in the server, but the client stil rescieving the answer, I start to suspect if it’s coming from the trunk provider, but in that case, my server should read it also, I don’t see that… below is the trace:


#10

You haven’t got versbose output in the trace so you cannot see which part of the dialplan is answering the call.


#11

sorry, i post incompletely…

the answer is there:

<--- Reliably Transmitting (NAT) to XXXXXXXXx:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP XXXXXXXXx:5060;branch=z9hG4bK73371c39;received=XXXXXXXXx;rport=5060 From: "XXXXXXXXXXX" <sip:xxxx@xxxxxxxx>;tag=as1ebf164a To: <sip:XXXXXXXXXXX@XXXXXXXXx>;tag=as0f12d91f Call-ID: 78de8ecc6c16f45f06a4c3915048cf07@XXXXXXXXx:5060 CSeq: 103 INVITE

however, as we cant see the time, nor the ip, this doesnt say much.

I would recommend using ngrep -t -d eth0 -O myfile.pcap port 5060 to capture the exchange, and then use wireshark to open it, and go to Telephony / Voip Call / Flow … you will see the exchange

Having said that, you also want to add debug traces in A2B, via the menu system config / agi setting , config #2, verbose & debug options. you will see better what is happening

rgds,
J.


#12

You can get the time from the full log. However, my guess is that it is immediate, in which case you need the verbose trace of the dialplan to see where the answer is being generated.


#13

Hello again Guys, sorry i forget to turn the verbose to on, now verbose is 5 in asterisk and 4 in a2billing agi, the trace is bellow:

[quote]To: sip:xxxx%2334693XXXXXX@89.140.201.40;tag=008082590C3212CFAF6496DA9AED
Call-ID: 2515d7320f93bc6c14832f040c355af1@178.63.XXX.XXX8:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Content-Length: 183

v=0
o=- 3951158040 1 IN IP4 89.140.201.40
s=session
t=0 0
m=audio 5680 RTP/AVP 8 101
c=IN IP4 89.140.201.56
a=rtpmap:8 PCMA/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
<------------->
— (9 headers 9 lines) —
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 89.140.201.56:5680
– SIP/TrunkC-00000095 is ringing
– SIP/TrunkC-00000095 is making progress passing it to SIP/887867-00000094

<— SIP read from UDP:178.63.XXX.XXXX:5060 —>
BYE sip:0034xxxxxxxx@178.63.XXX.XXX8:5060 SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXXX:5060;branch=z9hG4bK44baf346;rport
Max-Forwards: 70
From: “0034xxxxxxxx” sip:887867@178.63.XXX.XXXX;tag=as299e0dd3
To: sip:0034xxxxxxxx@178.63.XXX.XXX8;tag=as5eea9cf4
Call-ID: 5f5f064a592c56ce1ab9928b37bfde8f@178.63.XXX.XXXX:5060
CSeq: 104 BYE
User-Agent: FPBX-2.9.0(1.8.4.2)
Authorization: Digest username=“887867”, realm=“asterisk”, algorithm=MD5, uri=“sip:0034xxxxxxxx@178.63.XXX.XXX8:5060”, nonce=“6dc835a2”, response="ff94a710d5b6b3071d08f6e843dad4a8"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 178.63.XXX.XXXX:5060 (NAT)
Scheduling destruction of SIP dialog '5f5f064a592c56ce1ab9928b37bfde8f@178.63.XXX.XXXX:5060’ in 6400 ms (Method: BYE)

<— Transmitting (NAT) to 178.63.XXX.XXXX:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 178.63.XXX.XXXX:5060;branch=z9hG4bK44baf346;received=178.63.XXX.XXXX;rport=5060
From: “0034xxxxxxxx” sip:887867@178.63.XXX.XXXX;tag=as299e0dd3
To: sip:0034xxxxxxxx@178.63.XXX.XXX8;tag=as5eea9cf4
Call-ID: 5f5f064a592c56ce1ab9928b37bfde8f@178.63.XXX.XXXX:5060
CSeq: 104 BYE
Server: FPBX-2.9.0(1.8.4.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '2515d7320f93bc6c14832f040c355af1@178.63.XXX.XXX8:5060’ in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 89.140.201.40:5060:
CANCEL sip:xxxx#34693XXXXXX@89.140.201.40 SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK12d8e34f;rport
Max-Forwards: 70
From: “693775992” sip:693775992@178.63.XXX.XXX8;tag=as4b522c45
To: sip:xxxx#34693XXXXXX@89.140.201.40
Call-ID: 2515d7320f93bc6c14832f040c355af1@178.63.XXX.XXX8:5060
CSeq: 102 CANCEL
User-Agent: FPBX-2.9.0(1.8.4.2)
Content-Length: 0


Scheduling destruction of SIP dialog '2515d7320f93bc6c14832f040c355af1@178.63.XXX.XXX8:5060’ in 6400 ms (Method: INVITE)
a2billing.php,2: file:Class.RateEngine.php - line:1274 - uniqueid:1309364844.148 - DIAL SIP/TrunkC/xxxx#34693XXXXXX,60,HL(714930000:61000:30000)
a2billing.php,2: file:Class.RateEngine.php - line:1157 - uniqueid:1309364844.148 - [TRUNK STATUS UPDATE : UPDATE cc_trunk SET inuse=inuse-1 WHERE id_trunk=‘3’]
a2billing.php,2: file:Class.RateEngine.php - line:1426 - uniqueid:1309364844.148 - [USEDRATECARD=0]
a2billing.php,2: file:Class.RateEngine.php - line:959 - uniqueid:1309364844.148 - :[sessiontime:0 - id_cc_package_offer:-1 - package2apply:]
a2billing.php,2:
a2billing.php,2:
a2billing.php,2: file:Class.RateEngine.php - line:1042 - uniqueid:1309364844.148 - [CC_RATE_ENGINE_UPDATESYSTEM: usedratecard K=0 - (sessiontime=0 :: dialstatus=CANCEL :: buycost=0 :: cost= : signe_cc_call=-: signe=+)]
a2billing.php,2: file:Class.RateEngine.php - line:1107 - uniqueid:1309364844.148 - [CC_asterisk_stop : SQL: DONE : result=1]
a2billing.php,2: file:Class.RateEngine.php - line:1108 - uniqueid:1309364844.148 - [CC_asterisk_stop : SQL: INSERT INTO cc_call (uniqueid, sessionid, card_id, nasipaddress, starttime, sessiontime, real_sessiontime, calledstation, terminatecauseid, stoptime, sessionbill, id_tariffgroup, id_tariffplan, id_ratecard, id_trunk, src, sipiax, buycost, id_card_package_offer, dnid, destination) VALUES (‘1309364844.148’, ‘SIP/887867-00000094’, ‘1’, ‘’, SUBDATE(CURRENT_TIMESTAMP, INTERVAL 0 SECOND) , ‘0’, ‘0’, ‘34693XXXXXX’, 4, now() , ‘-0’, ‘1’, ‘6’, ‘90931’, ‘3’, ‘693775992’, ‘0’, ‘0’, NULL, ‘0034xxxxxxxx’, ‘346’)]
a2billing.php,2: file:a2billing.php - line:588 - uniqueid:1309364844.148 - [a2billing account stop]
a2billing.php,2: file:Class.A2Billing.php - line:750 - uniqueid:1309364844.148 - [CARD STATUS UPDATE]
a2billing.php,2: file:Class.A2Billing.php - line:759 - uniqueid:1309364844.148 - [QUERY USING CARD UPDATE::> UPDATE cc_card SET inuse=inuse-1, credit=credit+0.25 WHERE username=‘887867’]
– <SIP/887867-00000094>AGI Script a2billing.php completed, returning -1

<— SIP read from UDP:89.140.201.40:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK12d8e34f;rport=5060;received=178.63.XXX.XXX8
From: “693775992” sip:693775992@178.63.XXX.XXX8;tag=as4b522c45
To: sip:xxxx%2334693XXXXXX@89.140.201.40
Call-ID: 2515d7320f93bc6c14832f040c355af1@178.63.XXX.XXX8:5060
CSeq: 102 CANCEL
Contact: sip:89.140.201.40:5060
Content-Length: 0

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

<— SIP read from UDP:89.140.201.40:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK12d8e34f;rport=5060;received=178.63.XXX.XXX8
From: “693775992” sip:693775992@178.63.XXX.XXX8;tag=as4b522c45
To: sip:xxxx%2334693XXXXXX@89.140.201.40;tag=008082590C3212CFAF6496DA9AED
Call-ID: 2515d7320f93bc6c14832f040c355af1@178.63.XXX.XXX8:5060
CSeq: 102 INVITE
Reason: Q.850;cause=16;text="Normal call clearing"
Content-Length: 0

<------------->
— (8 headers 0 lines) —
ransmitting (NAT) to 89.140.201.40:5060:
ACK sip:xxxx#34693XXXXXX@89.140.201.40 SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK12d8e34f;rport
Max-Forwards: 70
From: “693775992” sip:693775992@178.63.XXX.XXX8;tag=as4b522c45
To: sip:xxxx#34693XXXXXX@89.140.201.40;tag=008082590C3212CFAF6496DA9AED
Contact: sip:693775992@178.63.XXX.XXX8:5060
Call-ID: 2515d7320f93bc6c14832f040c355af1@178.63.XXX.XXX8:5060
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.4.2)
Content-Length: 0


Really destroying SIP dialog '2515d7320f93bc6c14832f040c355af1@178.63.XXX.XXX8:5060’ Method: INVITE
Reliably Transmitting (NAT) to 178.63.XXX.XXX9:5060:
OPTIONS sip:178.63.XXX.XXX9 SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK218b58d6;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as40c57e7c
To: sip:178.63.XXX.XXX9
Contact: sip:Unknown@178.63.XXX.XXX8:5060
Call-ID: 04ace1fa4b301a0372eb148b260cc67d@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.4.2)
Date: Wed, 29 Jun 2011 16:27:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:178.63.XXX.XXX9:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK218b58d6;received=178.63.XXX.XXX8;rport=5060
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as40c57e7c
To: sip:178.63.XXX.XXX9;tag=as6e5839bd
Call-ID: 04ace1fa4b301a0372eb148b260cc67d@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
Server: FPBX-2.9.0(1.8.4.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog '04ace1fa4b301a0372eb148b260cc67d@178.63.XXX.XXX8:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 77.72.169.134:5060:
OPTIONS sip:sip.nonoh.net SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK48a9cc25;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as2568b2d9
To: sip:sip.nonoh.net
Contact: sip:Unknown@178.63.XXX.XXX8:5060
Call-ID: 5405b6fa70f2f9fd29d35b4f766b5d51@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.4.2)
Date: Wed, 29 Jun 2011 16:27:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:77.72.169.134:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK48a9cc25;rport
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as2568b2d9
To: sip:sip.nonoh.net
Contact: sip:77.72.169.134:5060
Call-ID: 5405b6fa70f2f9fd29d35b4f766b5d51@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
Supported: foo
User-Agent: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Accept: application/sdp

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog '5405b6fa70f2f9fd29d35b4f766b5d51@178.63.XXX.XXX8:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 77.72.174.132:5060:
OPTIONS sip:77.72.174.132 SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK4ddf7c67;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as1e21e21f
To: sip:77.72.174.132
Contact: sip:Unknown@178.63.XXX.XXX8:5060
Call-ID: 5f1e7fe4591eb0997cc3197b36f3c83f@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.4.2)
Date: Wed, 29 Jun 2011 16:27:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:77.72.174.132:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK4ddf7c67;rport
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as1e21e21f
To: sip:77.72.174.132
Contact: sip:77.72.174.132:5060
Call-ID: 5f1e7fe4591eb0997cc3197b36f3c83f@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
Supported: foo
User-Agent: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Accept: application/sdp

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog '5f1e7fe4591eb0997cc3197b36f3c83f@178.63.XXX.XXX8:5060’ Method: OPTIONS
Really destroying SIP dialog '5f5f064a592c56ce1ab9928b37bfde8f@178.63.XXX.XXXX:5060’ Method: BYE
Reliably Transmitting (NAT) to 89.140.201.40:5060:
OPTIONS sip:89.140.201.40 SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK687efcb4;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as3ba46430
To: sip:89.140.201.40
Contact: sip:Unknown@178.63.XXX.XXX8:5060
Call-ID: 0b6330d0730c68eb1f18525911e12045@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.4.2)
Date: Wed, 29 Jun 2011 16:27:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:89.140.201.40:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK687efcb4;rport
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as3ba46430
To: sip:89.140.201.40
Call-ID: 0b6330d0730c68eb1f18525911e12045@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
Contact: sip:89.140.201.40:5060
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/x-session-info, application/tls-aoc99, application/dtmf-relay, message/sipfrag
Accept-Encoding:
Accept-Language: *
Supported: 100rel, timer, replaces
Content-Length: 701

v=0
o=- 1431161886 0 IN IP4 89.140.201.40
s=session
t=0 0
m=audio 0 RTP/AVP 18 4 4 96 97 98 99 100 3 8 0 101 102 103 104
c=IN IP4 127.0.0.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-32/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:99 G726-16/8000
a=rtpmap:100 GSM-EFR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 CLEARMODE/8000
a=rtpmap:102 G.nX64/8000
a=rtpmap:103 X-CCD/8000
a=rtpmap:104 X-NSE/8000
a=sendrecv
m=image 0 udptl t38
c=IN IP4 127.0.0.1
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxRateManagement:transferredTCF
<------------->
— (14 headers 29 lines) —
Really destroying SIP dialog '0b6330d0730c68eb1f18525911e12045@178.63.XXX.XXX8:5060’ Method: OPTIONS

<— SIP read from UDP:109.172.66.193:5064 —>

<------------->
Reliably Transmitting (NAT) to 213.236.11.72:5060:
OPTIONS sip:sip.voicetrunk.com SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK4d5e5f8b;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as7de066c2
To: sip:sip.voicetrunk.com
Contact: sip:Unknown@178.63.XXX.XXX8:5060
Call-ID: 7b646f286257b875691dc71a330a6f6b@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.4.2)
Date: Wed, 29 Jun 2011 16:27:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:213.236.11.72:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK4d5e5f8b;received=178.63.XXX.XXX8;rport=5060
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as7de066c2
To: sip:sip.voicetrunk.com;tag=as45867f80
Call-ID: 7b646f286257b875691dc71a330a6f6b@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
User-Agent: Telcom BS
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog '7b646f286257b875691dc71a330a6f6b@178.63.XXX.XXX8:5060’ Method: OPTIONS
Really destroying SIP dialog ‘612ba0e97c18745d’ Method: REGISTER

<— SIP read from UDP:178.63.XXX.XXX9:5060 —>
OPTIONS sip:178.63.XXX.XXX8 SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXX9:5060;branch=z9hG4bK656d5a91;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@178.63.XXX.XXX9;tag=as2f0d784b
To: sip:178.63.XXX.XXX8
Contact: sip:Unknown@178.63.XXX.XXX9:5060
Call-ID: 373a3a19182d5642669f665414d1f855@178.63.XXX.XXX9:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.4.2)
Date: Wed, 29 Jun 2011 16:27:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Looking for in from-sip-external (domain 178.63.XXX.XXX8)

<— Transmitting (NAT) to 178.63.XXX.XXX9:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 178.63.XXX.XXX9:5060;branch=z9hG4bK656d5a91;received=178.63.XXX.XXX9;rport=5060
From: “Unknown” sip:Unknown@178.63.XXX.XXX9;tag=as2f0d784b
To: sip:178.63.XXX.XXX8;tag=as314fcc1d
Call-ID: 373a3a19182d5642669f665414d1f855@178.63.XXX.XXX9:5060
CSeq: 102 OPTIONS
Server: FPBX-2.9.0(1.8.4.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '373a3a19182d5642669f665414d1f855@178.63.XXX.XXX9:5060’ in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '031c734178b3fa9a1906519d499e1cd4@178.63.XXX.XXXX’ Method: REGISTER
Reliably Transmitting (NAT) to 174.133.195.194:5060:
OPTIONS sip:174.133.195.194 SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK74f027a1;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as6c3fcbef
To: sip:174.133.195.194
Contact: sip:Unknown@178.63.XXX.XXX8:5060
Call-ID: 6037b26c25f3679419521cf648d8cf05@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.4.2)
Date: Wed, 29 Jun 2011 16:27:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:174.133.195.194:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK74f027a1;received=178.63.XXX.XXX8;rport=5060
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as6c3fcbef
To: sip:174.133.195.194;tag=as13bb95dc
Call-ID: 6037b26c25f3679419521cf648d8cf05@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
User-Agent: DiDXsuPErTecSIP7
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:174.133.195.194
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog '6037b26c25f3679419521cf648d8cf05@178.63.XXX.XXX8:5060’ Method: OPTIONS
Really destroying SIP dialog '67cc99bd3e04d9d656f377121578acd3@178.63.XXX.XXXX:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 66.241.7.200:5060:
OPTIONS sip:66.241.7.200 SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK1f389c27;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as308b39a9
To: sip:66.241.7.200
Contact: sip:Unknown@178.63.XXX.XXX8:5060
Call-ID: 3c48bd102e405a2458b20ab1330a2ea3@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.4.2)
Date: Wed, 29 Jun 2011 16:27:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:66.241.7.200:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK1f389c27;rport
To: sip:66.241.7.200;tag=3518353668-948839
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as308b39a9
Call-ID: 3c48bd102e405a2458b20ab1330a2ea3@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: sip:66.241.7.200:5060
Content-Length: 0

<------------->
— (9 headers 0 lines) —
eally destroying SIP dialog '3c48bd102e405a2458b20ab1330a2ea3@178.63.XXX.XXX8:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 64.71.145.237:5060:
OPTIONS sip:64.71.145.237 SIP/2.0
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK41a5a7cb;rport
Max-Forwards: 70
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as4e8f98c8
To: sip:64.71.145.237
Contact: sip:Unknown@178.63.XXX.XXX8:5060
Call-ID: 799ff6ac189144642fb7ceb94008fc53@178.63.XXX.XXX8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.4.2)
Date: Wed, 29 Jun 2011 16:27:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:64.71.145.237:61137 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 178.63.XXX.XXX8:5060;branch=z9hG4bK41a5a7cb;rport
From: “Unknown” sip:Unknown@178.63.XXX.XXX8;tag=as4e8f98c8
To: sip:64.71.145.237;tag=22697758-1285
Date: Wed, 29 Jun 2011 16:31:25 GMT
Call-ID: 799ff6ac189144642fb7ceb94008fc53@178.63.XXX.XXX8:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 454

v=0
o=CiscoSystemsSIP-GW-UserAgent 6775 4968 IN IP4 64.71.145.237
s=SIP Call
c=IN IP4 64.71.145.237
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 64.71.145.237
m=image 0 udptl t38
c=IN IP4 64.71.145.237
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
— (14 headers 18 lines) —
Really destroying SIP dialog '799ff6ac189144642fb7ceb94008fc53@178.63.XXX.XXX8:5060’ Method: OPTIONS

<— SIP read from UDP:109.172.66.193:5064 —>[/quote]


#14

No incoming call is answered in that trace (no 200 OK to an INVITE).


#15

Hello,
Just a fedback to share, that I forgot to come back here when I solved this issue…

I had to modify the wholesale dialplan as below:
[voip-wholesale]
exten => _X.,1,NoOp(A2Billing Start)
exten => _X.,n,Agi(a2billing.php,2)
exten => _X.,n,Hangup

Then also the dialcommand in a2billing as: ,60,LIW(389220000:60000:30000)

and it’s work…

Regards,