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darryl
05-04-2009, 05:53 AM
HI
Does anyone have the FXO disconnect settings for South Africa as i believe FXO's not releasing is a common problem which is resolved by entering the custom fxo settings.

Thanks

davrays
05-06-2009, 12:58 PM
Darril, if such disconnect settings existed, they would have been applied by default by Epygi, as soon as you select SA as your locale...

Unfortunately, in your beautiful country there are too many differents standards used (better to say - no standard is used) for disconnect tones.
When doing trainings in SA I noticed several tones used by different telcos, and the FXO disconnection issues really look to be one of the bigger problems resellers face in SA.

There are methods to resolve those problems however. The easiest is the following:
1. Route your FXO line to some extension, which has voicemail service.
2. In "Voice Mail Common Settings" set the "Recording Codec" to G.711u.
3. Make an incoming call through that FXO and wait until the voicemail of the target extension activates. As soon as you listen the beep indicating that the voicemail recording is started, put down the handset.
4. Wait for some time. During that time voicemail system will record the disconnection signal from your telco. Typically you have to wait 5 minutes, which equals to the Max mail duration on the extension by default (if you want it to be faster, decrease the "Maximum mail message duration" from the "Voice Mail Settings" of that extension before making the test call).
5. Open the "Voice Mailbox" of that extension in the web browser, and download the voicemail.

After that you have two choices:
a) either send that to Epygi to analyse and give you the values to enter to the "fxocfg.cg";
b) or analyse the recording yourself using a audio editing tool, such as Cooledit/Adobe Audition.
If you open that recording in such program, you will see a periodic signal - tone-silence-tone-silence-.... You need to measure three parameters:
1. the duration of the tone period
2. the duration of the silence period
3. the frequency of the tone.
If you get some experience using those programs, this should be very easy. For me that takes about half a minute, for example.
Enter those parameters to the "fxocfg.cgi" and save.

Best regards,
David

IT Guyz
07-07-2009, 01:57 PM
Darril, if such disconnect settings existed, they would have been applied by default by Epygi, as soon as you select SA as your locale...

Unfortunately, in your beautiful country there are too many differents standards used (better to say - no standard is used) for disconnect tones.
When doing trainings in SA I noticed several tones used by different telcos, and the FXO disconnection issues really look to be one of the bigger problems resellers face in SA.

There are methods to resolve those problems however. The easiest is the following:
1. Route your FXO line to some extension, which has voicemail service.
2. In "Voice Mail Common Settings" set the "Recording Codec" to G.711u.
3. Make an incoming call through that FXO and wait until the voicemail of the target extension activates. As soon as you listen the beep indicating that the voicemail recording is started, put down the handset.
4. Wait for some time. During that time voicemail system will record the disconnection signal from your telco. Typically you have to wait 5 minutes, which equals to the Max mail duration on the extension by default (if you want it to be faster, decrease the "Maximum mail message duration" from the "Voice Mail Settings" of that extension before making the test call).
5. Open the "Voice Mailbox" of that extension in the web browser, and download the voicemail.

After that you have two choices:
a) either send that to Epygi to analyse and give you the values to enter to the "fxocfg.cg";
b) or analyse the recording yourself using a audio editing tool, such as Cooledit/Adobe Audition.
If you open that recording in such program, you will see a periodic signal - tone-silence-tone-silence-.... You need to measure three parameters:
1. the duration of the tone period
2. the duration of the silence period
3. the frequency of the tone.
If you get some experience using those programs, this should be very easy. For me that takes about half a minute, for example.
Enter those parameters to the "fxocfg.cgi" and save.

Best regards,
David

Hi David

Could you maybe elaborate on measuring the disconnect settings? I have Audition (same as Cool Edit) and have the recordings as you suggested.
would you mind describing how to measure the following settings? (I've been Googling for a long time without success)


Frequency1:
Frequency2:
Duration:
Duration Disp:
Silence:
Silence Disp:
Pattern Count:
Disconnect Detection
Disconnect Duration:
Silence Detection
Min Duration:
Threshold:

Many, many thanks

davrays
07-14-2009, 11:34 AM
Hi Anton

actually there are some links on the web, which can help you with this, though they are not easy to find. Here is one of them: http://forum.voxilla.com/linksys-sipura-voip-support-forum/translating-your-disconnect-tone-12874.html. The procedur described there, is similiar to what can be done for Quadro using Adobe Audition or CoolEdit...

Actually you need to enable the Busy tone detection, and set only the "Frequency1", "Frequency2", "Duration", "Silence", "Pattern Count". The rest of the parameters can be left intact. Also enable the disconnect detection with default settings, as it is helpful to have that enabled in any case.

Now about the busy disconnection signal:

typical busy disconnection signal consists of several periods of tone (single of dual frequescy) and silence.

1. What do Quadro parameters mean:
The "Frequency1" is the main frequency used in the "tone" part of disconnection pattern. The "Frequncy2" is the second frequency (only actual if a dual-frequency signal is used by telco) used in the "tone" part of disconnection pattern. If the signal has just single frequency, then the "Frequency2" shoul be set to 0.
The "Duration" parameter is the duration of the "tone" part in milliseconds. The "Silence" parameter is the duration of the "silence" part in milliseconds.
The "Pattren Count" is the number of repetitions, after which the signal is recognised as the disconnect signal. It typically should be set to 3 or 2.

2. How to determine the needed parameters:
"Duration" can be determined by selecting the "tone" part of the signal and looking at the right-bottom corner of the Audition window ("Lenght" field). Typically should be something between 100 and 1500 msec.
"Silence" can be determined by selecting the "silence" part of the signal and looking at the right-bottom corner of the Audition window ("Lenght" field). Typically should be something between 100 and 1500 msec.
Frequencies can be found using Audition's "Analyse"->"Show Frequency Analys" menu (select the "tone" part only, then click on that menu item). As sooin as you call that, you will see a graph. Press on "Scan" to make the graph better. On the graph you will see one or two peaks. Point the mouse on the peak, and you will see the frequecny in the left-bottom corner of "Frequency Analysis" wondow (it is measured in Hz). If there is just one peak - it means you have single frequency tone, Frequency1=<what you see on the graph>, Frequency2=0. If you see two peaks: Frequency1=<highest peak on the graph>, Frequency2=<second peak on the graph>. Typically should be something between 200-1000 Hz (more commonly - between 300 and 700 Hz).

Of course this will look better with screenshots, but I am not sure I can attach them here.. I'll see.

UPDATE:

Ok, here are attachments. They are in very poor quality, and to be honest this is not a true disconnect signal (though it is very similiar), but they can serve as an basic illustration.
Fig 1: this is how disconnect signal looks
Fig 2: shows how to measure the "tone" part duration
Fig 3. shows how to measure the "silence" part duration
Fig 4. shows how to measure frequences (here is a dual frequency tone, with two peaks very close to each other)

IT Guyz
10-13-2009, 06:58 AM
Thanks, that did the trick!

Kristoph
09-23-2011, 09:11 AM
Davrays, would it be possible for me to send you the file I got on my voice mail to check the frequencies for Poland?

I think my configuration of the voicemail is right but the pattern I get from the .wav file looks nothing like the one from your attachment.

The strange thing is that the voice mail stops recording very quickly as if it was aware of the fact the call ended, but the extension keeps ringing long after caller disconnected and I get voicemail after each such scenario.

I think I got three parameters:

duration: 100
silence: 130
pattern: 1

davrays
09-24-2011, 04:34 PM
If there is no pattern as shown above on the wav file, I suppose there is nothing to look for in that file... If voicemail really stops recrding quickly, it means the disconnect signal is ok. Another test you could do is to call from PSTN, pickup the call on any local extension, talk a bit to yourself :), then terminate the call from PSTN side, and listen what happens on local extension phone. What you hear there? Also look/refresh at the "Active Calls" page to see how long you see that call there, after you have terminated it from PSTN.

If the call disappears from the Active Calls table in 2-5 seconds, then apparently you have no problem with disconnect settings, and should look for the problem in different place. If the call stays in the table for long (20 seconds or more), then lets think further about how to catch the diosconnect signal.

Kristoph
09-26-2011, 10:37 AM
After picking up a phone and disconnecting quickly the call gests terminated in the Active Calls" quickly (2sec) but if the call gets dropped before anyone answers it it stays in the active calls until it gets to the voice mail.

If I pick up on Panasonic phone the incoming call and then disconnect I get a one and everlasting biiiiiiiii...

If I pick up on Philips (other extension) phone the incoming call and then disconnect I get a series of quick bi.bi.bi.bi....

I hope this helps to solve the mystery.

davrays
09-29-2011, 10:27 AM
Well, this means you don't have regular fxo disconnection problem. The problem on your setup seems to happen because the Quadro doesn't recognise the end of ring signals from telco. Truly saying, I have never listened about such problem before... As far as I understand, you have tweaked some settings on the hidden fxocfg.cgi. May it be that you have touched the "Fxo Ring Stop Timeout" value on the same page? Try to set it to the default 7000. If that doesn't help, you can try to play with that setting a bit - set it to 5000 (it should work with standard Poland ring signal). Try other settings between 4000 and 10000 (I would not recommend to set it lower than 4000, as polish ring signal has 4 sec timeout between rings).
If it makes no absolutely no effect, try setting it to 5000 and reboot the Quadro (I tell this because I am not 100% sure this setting applies without reboot).

Hope this helps... If not, you'll have to contact Epygi support, so they can connect to the device and make some test calls while capturing actual fxo data. In any case, keep us informed.

Kristoph
10-03-2011, 07:02 AM
Everything works fine for FXO connections now but what about the SIP connections?

When I call SIP number and disconnect before someone answers the phone the problem remains. Is there a way to correct this?

davrays
10-03-2011, 11:11 AM
Kristoph, first of all - what did you change to solve the FXO problem? Did you try anything I wrote above or the problem resolved itself somehow?
After writing the long text above, I am interested to know what was the reason of the issue, and whether something I suggested did help or not...

Second - if phones continue ring after disconnecting SIP call, then you may have network problem at your intallation. Look at the SIP logs.

Kristoph
10-18-2011, 07:28 AM
Sorry for not replying so long I was out of office. Your suggestion helped with PSTN calls, but had no influence on SIP calls.

I can send you some logs but I need to know which ones.

davrays
10-18-2011, 07:55 AM
Its extremely strange that you hav such problem with SIP calls. I don't see any reason for this other than delayd response from ITSP or some network problem (big delays in network because of huge traffic or connectivity problems, like loops, collisions or ip conflict).

You could go to "System->Diagnostics->Show System Logs", marks the logs (press "Mark the Logs" button), make a call to reproduce the problem (make a call, wait 1-2 seconds to listen a ringing, then hangup, make sure the ringing continues for 3-4 seconds after that), then mark the logs again. Copy the "SIP Agent Logs" in between two mark signs, and paste them here. We could see which of the SIP messages was delayed, and whether it has delayed at all...

Kristoph
10-18-2011, 10:55 AM
Here we go

<<======================LOGS MARKER======================>>
<<================================================== =====>>
<<User's Comment: SIP disconnect test start1>>
<<================================================== =====>>
14:27:02 Receive SIP message # (18/10/2011 12:27:02:833 GMT) # UDP # 954 bytes # buff size 0 # from: SIPserver:5060 # to: QuadroIP:5060

***************************** SIP message buffer start *****************************
INVITE sip:SIPnumber@QuadroIP:5060 SIP/2.0
Via: SIP/2.0/UDP SIPserver:5060;branch=z9hG4bK2f4be87a;rport
Max-Forwards: 70
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
To: <sip:SIPnumber@QuadroIP:5060>
Contact: <sip:caller1@SIPserver>
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
CSeq: 102 INVITE
User-Agent: DHGW
Date: Tue, 18 Oct 2011 12:27:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 393

v=0
o=root 1004707011 1004707011 IN IP4 SIPserver
s=Asterisk PBX 1.6.2.9-2+squeeze3
c=IN IP4 SIPserver
t=0 0
m=audio 12900 RTP/AVP 0 111 4 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
***************************** SIP message buffer end ******************************

14:27:02 Try to send SIP message # (18/10/2011 12:27:02:837 GMT) # UDP # 289 bytes # buff size 0 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SIPserver:5060;rport=5060;branch=z9hG4bK2f4be87a
To: <sip:SIPnumber@QuadroIP:5060>
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
CSeq: 102 INVITE
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:02 TLayer::MsgToTU # Msg type: 51 # TID: 2057128 # DID: 0
14:27:02 UACore::TLReqMsgProc # Got INVITE SIP request
14:27:02 SipSessDlg: # Call ID Info # SID: 5664807695850202111 # 14:27:02 SipID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
14:27:02 SipSessDlg::ChangeSessTimerFlag # user caller1 # SessionTimer = 0
14:27:02 Receive SIP message # (18/10/2011 12:27:02:855 GMT) # UDP # 956 bytes # buff size 1 # from: SIPserver:5060 # to: QuadroIP:5060

***************************** SIP message buffer start *****************************
INVITE sip:SIPnumber-1@QuadroIP:5060 SIP/2.0
Via: SIP/2.0/UDP SIPserver:5060;branch=z9hG4bK7e3154cc;rport
Max-Forwards: 70
From: "caller1" <sip:caller1@SIPserver>;tag=as35065dc1
To: <sip:SIPnumber-1@QuadroIP:5060>
Contact: <sip:caller1@SIPserver>
Call-ID: 478453cb2d45990a13a75e2d108fd95a@SIPserver
CSeq: 102 INVITE
User-Agent: DHGW
Date: Tue, 18 Oct 2011 12:27:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 391

v=0
o=root 684655146 684655146 IN IP4 SIPserver
s=Asterisk PBX 1.6.2.9-2+squeeze3
c=IN IP4 SIPserver
t=0 0
m=audio 18640 RTP/AVP 0 111 4 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
***************************** SIP message buffer end ******************************

14:27:02 Try to send SIP message # (18/10/2011 12:27:02:858 GMT) # UDP # 291 bytes # buff size 1 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SIPserver:5060;rport=5060;branch=z9hG4bK7e3154cc
To: <sip:SIPnumber-1@QuadroIP:5060>
From: "caller1" <sip:caller1@SIPserver>;tag=as35065dc1
CSeq: 102 INVITE
Call-ID: 478453cb2d45990a13a75e2d108fd95a@SIPserver
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:02 TLayer::MsgToTU # Msg type: 51 # TID: 2057130 # DID: 0
14:27:02 SipSessDlg::IncInviteProc # Got INVITE message # OID: 2057129 # SID: 5664807695850202111
14:27:02 TargetQualifier::QualifyTargets # Try to qualify targets for dest SIPserver # OID: 2057129
14:27:02 TargetQualifier::NatHandling # Don't need nat settings - nat disconnected # OID: 2057129
14:27:02 SipSessDlg::IncInviteProc # mLocalParams.mHostAddr = QuadroIP localHostName = quadro.epygi-config.com OID: 2057129 # SID: 5664807695850202111
14:27:02 UA --> CM # MakeCall # from: caller1@SIPserver:, to: SIPnumber, child: (empty), media exist, GUID: (empty) # ContactInfo: SIPserver # SID: 5664807695850202111
14:27:03 CM --> UA # OnUpdateUserInfo # OID: 2057129 # SID: 5664807695850202111
14:27:03 UACore::TLReqMsgProc # Got INVITE SIP request
14:27:03 SipSessDlg: # Call ID Info # SID: 5664807700144356199 # 14:27:03 SipID: 478453cb2d45990a13a75e2d108fd95a@SIPserver
14:27:03 SipSessDlg::ChangeSessTimerFlag # user caller1 # SessionTimer = 0
14:27:03 SipSessDlg::IncInviteProc # Got INVITE message # OID: 2057131 # SID: 5664807700144356199
14:27:03 TargetQualifier::QualifyTargets # Try to qualify targets for dest SIPserver # OID: 2057131
14:27:03 TargetQualifier::NatHandling # Don't need nat settings - nat disconnected # OID: 2057131
14:27:03 SipSessDlg::IncInviteProc # mLocalParams.mHostAddr = QuadroIP localHostName = quadro.epygi-config.com OID: 2057131 # SID: 5664807700144356199
14:27:03 UA --> CM # MakeCall # from: caller1@SIPserver:, to: SIPnumber-1, child: (empty), media exist, GUID: (empty) # ContactInfo: SIPserver # SID: 5664807700144356199
14:27:03 CM --> UA # OnRinging # Media not exist # OID: 2057129 # SID: 5664807695850202111
14:27:03 Try to send SIP message # (18/10/2011 12:27:03:164 GMT) # UDP # 448 bytes # buff size 0 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP SIPserver:5060;rport=5060;branch=z9hG4bK2f4be87a
To: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
CSeq: 102 INVITE
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
Contact: <sip:SIPnumber@QuadroIP:5060>
Server: Epygi Quadro SIP User Agent/v5.1.39 (QUADRO-4X/16X)
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:03 CM --> UA # OnUpdateUserInfo # OID: 2057131 # SID: 5664807700144356199
14:27:03 CM --> UA # OnRinging # Media not exist # OID: 2057131 # SID: 5664807700144356199
14:27:03 Try to send SIP message # (18/10/2011 12:27:03:184 GMT) # UDP # 452 bytes # buff size 0 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP SIPserver:5060;rport=5060;branch=z9hG4bK7e3154cc
To: <sip:SIPnumber-1@QuadroIP:5060>;tag=1317644155c017bfec-b0dc-4ff4-b02f-31a8b0b5b94b
From: "caller1" <sip:caller1@SIPserver>;tag=as35065dc1
CSeq: 102 INVITE
Call-ID: 478453cb2d45990a13a75e2d108fd95a@SIPserver
Contact: <sip:SIPnumber-1@QuadroIP:5060>
Server: Epygi Quadro SIP User Agent/v5.1.39 (QUADRO-4X/16X)
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:03 CM --> UA # OnAccept # Media exist #OID: 2057129 # SID: 5664807695850202111
14:27:03 SipSessDlg::SelfMediaHandling # State: complete, Sesssion Key: empty # OID: 2057129 # SID: 5664807695850202111
14:27:04 Try to send SIP message # (18/10/2011 12:27:03:1000 GMT) # UDP # 811 bytes # buff size 0 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPserver:5060;rport=5060;branch=z9hG4bK2f4be87a
To: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
CSeq: 102 INVITE
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, UPDATE
Contact: <sip:SIPnumber@QuadroIP:5060>
Content-Type: application/sdp
Supported: replaces, norefersub
Server: Epygi Quadro SIP User Agent/v5.1.39 (QUADRO-4X/16X)
Content-Length: 211

v=0
o=- 877 388 IN IP4 QuadroIP
s=-
c=IN IP4 QuadroIP
t=0 0
m=audio 6024 RTP/AVP 0 8 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
***************************** SIP message buffer end ******************************

14:27:04 Receive SIP message # (18/10/2011 12:27:04:013 GMT) # UDP # 434 bytes # buff size 1 # from: SIPserver:5060 # to: QuadroIP:5060

***************************** SIP message buffer start *****************************
ACK sip:SIPnumber@QuadroIP:5060 SIP/2.0
Via: SIP/2.0/UDP SIPserver:5060;branch=z9hG4bK4e88b46d;rport
Max-Forwards: 70
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
To: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
Contact: <sip:caller1@SIPserver>
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
CSeq: 102 ACK
User-Agent: DHGW
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:04 TLayer::MsgToTU # Msg type: 55 # TID: 2057128 # DID: 2057129
14:27:04 UACore::TLReqMsgProc # Got ACK SIP request
14:27:04 SipSessDlg::IncAckProc # Got ACK message # OID: 2057129 # SID: 5664807695850202111
14:27:04 UA --> CM # Done # media not exist # SID: 5664807695850202111
14:27:04 Receive SIP message # (18/10/2011 12:27:04:045 GMT) # UDP # 409 bytes # buff size 2 # from: SIPserver:5060 # to: QuadroIP:5060

***************************** SIP message buffer start *****************************
CANCEL sip:SIPnumber-1@QuadroIP:5060 SIP/2.0
Via: SIP/2.0/UDP SIPserver:5060;branch=z9hG4bK7e3154cc;rport
Max-Forwards: 70
From: "caller1" <sip:caller1@SIPserver>;tag=as35065dc1
To: <sip:SIPnumber-1@QuadroIP:5060>
Call-ID: 478453cb2d45990a13a75e2d108fd95a@SIPserver
CSeq: 102 CANCEL
User-Agent: DHGW
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:04 TLayer::MsgToTU # Msg type: 53 # TID: 2057132 # DID: 0
14:27:04 UACore::TLReqMsgProc # Got CANCEL SIP request
14:27:04 SipSessDlg::IncCancelProc # Got CANCEL message # OID: 2057131 # SID: 5664807700144356199
14:27:04 Receive SIP message # (18/10/2011 12:27:04:053 GMT) # UDP # 848 bytes # buff size 1 # from: SIPserver:5060 # to: QuadroIP:5060

***************************** SIP message buffer start *****************************
INVITE sip:SIPnumber@QuadroIP:5060 SIP/2.0
Via: SIP/2.0/UDP SIPserver:5060;branch=z9hG4bK2f483c58;rport
Max-Forwards: 70
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
To: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
Contact: <sip:caller1@SIPserver>
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
CSeq: 103 INVITE
User-Agent: DHGW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 273

v=0
o=root 1004707011 1004707012 IN IP4 91.202.124.51
s=Asterisk PBX 1.6.2.9-2+squeeze3
c=IN IP4 91.202.124.51
t=0 0
m=audio 12640 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
***************************** SIP message buffer end ******************************

14:27:04 Try to send SIP message # (18/10/2011 12:27:04:160 GMT) # UDP # 340 bytes # buff size 1 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SIPserver:5060;rport=5060;branch=z9hG4bK2f483c58
To: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
CSeq: 103 INVITE
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:04 TLayer::MsgToTU # Msg type: 51 # TID: 2057133 # DID: 0
14:27:04 Try to send SIP message # (18/10/2011 12:27:04:195 GMT) # UDP # 399 bytes # buff size 0 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPserver:5060;rport=5060;branch=z9hG4bK7e3154cc
To: <sip:SIPnumber-1@QuadroIP:5060>;tag=1317644155c017bfec-b0dc-4ff4-b02f-31a8b0b5b94b
From: "caller1" <sip:caller1@SIPserver>;tag=as35065dc1
CSeq: 102 CANCEL
Call-ID: 478453cb2d45990a13a75e2d108fd95a@SIPserver
Server: Epygi Quadro SIP User Agent/v5.1.39 (QUADRO-4X/16X)
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:04 Try to send SIP message # (18/10/2011 12:27:04:201 GMT) # UDP # 415 bytes # buff size 0 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP SIPserver:5060;rport=5060;branch=z9hG4bK7e3154cc
To: <sip:SIPnumber-1@QuadroIP:5060>;tag=1317644155c017bfec-b0dc-4ff4-b02f-31a8b0b5b94b
From: "caller1" <sip:caller1@SIPserver>;tag=as35065dc1
CSeq: 102 INVITE
Call-ID: 478453cb2d45990a13a75e2d108fd95a@SIPserver
Server: Epygi Quadro SIP User Agent/v5.1.39 (QUADRO-4X/16X)
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:04 UA --> CM # CloseCall # reason: Got CANCEL message # SID: 5664807700144356199
14:27:04 Receive SIP message # (18/10/2011 12:27:04:224 GMT) # UDP # 438 bytes # buff size 1 # from: SIPserver:5060 # to: QuadroIP:5060

***************************** SIP message buffer start *****************************
ACK sip:SIPnumber-1@QuadroIP:5060 SIP/2.0
Via: SIP/2.0/UDP SIPserver:5060;branch=z9hG4bK7e3154cc;rport
Max-Forwards: 70
From: "caller1" <sip:caller1@SIPserver>;tag=as35065dc1
To: <sip:SIPnumber-1@QuadroIP:5060>;tag=1317644155c017bfec-b0dc-4ff4-b02f-31a8b0b5b94b
Contact: <sip:caller1@SIPserver>
Call-ID: 478453cb2d45990a13a75e2d108fd95a@SIPserver
CSeq: 102 ACK
User-Agent: DHGW
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:04 SipSessDlg::TerminateTransactions # OID: 2057131 # SIPID: 478453cb2d45990a13a75e2d108fd95a@SIPserver # SID: 5664807700144356199
14:27:04 CallsAgent::MsgFromCM # Got 4 message from CM for unknown call
14:27:04 UA --> CM # ReportError # Error: NoSuchCall # Sip error: 0 # SID: 5664807700144356199
14:27:04 UACore::TLReqMsgProc # Got INVITE SIP request
14:27:04 SipSessDlg::IncInviteProc # Got INVITE message # OID: 2057129 # SID: 5664807695850202111
14:27:04 UA --> CM # ChangeMedia # media exist # SID: 5664807695850202111
14:27:04 CM --> UA # OnAccept # Media exist #OID: 2057129 # SID: 5664807695850202111
14:27:04 SipSessDlg::SelfMediaHandling # State: complete, Sesssion Key: empty # OID: 2057129 # SID: 5664807695850202111
14:27:04 Try to send SIP message # (18/10/2011 12:27:04:270 GMT) # UDP # 720 bytes # buff size 0 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPserver:5060;rport=5060;branch=z9hG4bK2f483c58
To: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
CSeq: 103 INVITE
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
Contact: <sip:SIPnumber@QuadroIP:5060>
Content-Type: application/sdp
Supported: replaces, norefersub
Server: Epygi Quadro SIP User Agent/v5.1.39 (QUADRO-4X/16X)
Content-Length: 211

v=0
o=- 877 389 IN IP4 QuadroIP
s=-
c=IN IP4 QuadroIP
t=0 0
m=audio 6024 RTP/AVP 0 8 101
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
***************************** SIP message buffer end ******************************

14:27:04 Receive SIP message # (18/10/2011 12:27:04:313 GMT) # UDP # 434 bytes # buff size 1 # from: SIPserver:5060 # to: QuadroIP:5060

***************************** SIP message buffer start *****************************
ACK sip:SIPnumber@QuadroIP:5060 SIP/2.0
Via: SIP/2.0/UDP SIPserver:5060;branch=z9hG4bK13eed06b;rport
Max-Forwards: 70
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
To: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
Contact: <sip:caller1@SIPserver>
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
CSeq: 103 ACK
User-Agent: DHGW
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:04 TLayer::MsgToTU # Msg type: 55 # TID: 2057133 # DID: 2057129
14:27:04 UACore::TLReqMsgProc # Got ACK SIP request
14:27:04 SipSessDlg::IncAckProc # Got ACK message # OID: 2057129 # SID: 5664807695850202111
14:27:04 UA --> CM # Done # media not exist # SID: 5664807695850202111
14:27:15 Receive SIP message # (18/10/2011 12:27:15:147 GMT) # UDP # 846 bytes # buff size 1 # from: SIPserver:5060 # to: QuadroIP:5060

***************************** SIP message buffer start *****************************
INVITE sip:SIPnumber@QuadroIP:5060 SIP/2.0
Via: SIP/2.0/UDP SIPserver:5060;branch=z9hG4bK2b4af37b;rport
Max-Forwards: 70
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
To: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
Contact: <sip:caller1@SIPserver>
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
CSeq: 104 INVITE
User-Agent: DHGW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 1004707011 1004707013 IN IP4 SIPserver
s=Asterisk PBX 1.6.2.9-2+squeeze3
c=IN IP4 SIPserver
t=0 0
m=audio 12900 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
***************************** SIP message buffer end ******************************

14:27:15 Try to send SIP message # (18/10/2011 12:27:15:151 GMT) # UDP # 340 bytes # buff size 1 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SIPserver:5060;rport=5060;branch=z9hG4bK2b4af37b
To: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
CSeq: 104 INVITE
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:15 TLayer::MsgToTU # Msg type: 51 # TID: 2057134 # DID: 0
14:27:15 UACore::TLReqMsgProc # Got INVITE SIP request
14:27:15 SipSessDlg::IncInviteProc # Got INVITE message # OID: 2057129 # SID: 5664807695850202111
14:27:15 UA --> CM # ChangeMedia # media exist # SID: 5664807695850202111
14:27:15 CM --> UA # OnReportError # Error: RequestRejected # SIP error: 0 # OID: 2057129 # SID: 5664807695850202111
14:27:15 Try to send SIP message # (18/10/2011 12:27:15:172 GMT) # UDP # 414 bytes # buff size 0 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP SIPserver:5060;rport=5060;branch=z9hG4bK2b4af37b
To: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
CSeq: 104 INVITE
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
Server: Epygi Quadro SIP User Agent/v5.1.39 (QUADRO-4X/16X)
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:15 Receive SIP message # (18/10/2011 12:27:15:184 GMT) # UDP # 434 bytes # buff size 1 # from: SIPserver:5060 # to: QuadroIP:5060

***************************** SIP message buffer start *****************************
ACK sip:SIPnumber@QuadroIP:5060 SIP/2.0
Via: SIP/2.0/UDP SIPserver:5060;branch=z9hG4bK2b4af37b;rport
Max-Forwards: 70
From: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
To: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
Contact: <sip:caller1@SIPserver>
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
CSeq: 104 ACK
User-Agent: DHGW
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:43 CM --> UA # OnCloseCall # OID: 2057129 # SID: 5664807695850202111
14:27:43 SipSessDlg::TerminateTransactions # OID: 2057129 # SIPID: 46fdc7537bc2f95135e871ab03f14545@SIPserver # SID: 0
14:27:43 Try to send SIP message # (18/10/2011 12:27:43:810 GMT) # UDP # 470 bytes # buff size 0 # from: QuadroIP:5060 # to: SIPserver:5060

***************************** SIP message buffer start *****************************
BYE sip:caller1@SIPserver SIP/2.0
Via: SIP/2.0/UDP QuadroIP:5060;rport;branch=z9hG4bKEPSVBUS05aae88c-4171-45e1-b48c-d26323ef3c5a
To: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
From: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
CSeq: 931 BYE
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
User-Agent: Epygi Quadro SIP User Agent/v5.1.39 (QUADRO-4X/16X)
Max-Forwards: 70
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:43 Receive SIP message # (18/10/2011 12:27:43:821 GMT) # UDP # 539 bytes # buff size 1 # from: SIPserver:5060 # to: QuadroIP:5060

***************************** SIP message buffer start *****************************
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP QuadroIP:5060;branch=z9hG4bKEPSVBUS05aae88c-4171-45e1-b48c-d26323ef3c5a;received=QuadroIP;rport=5060
From: <sip:SIPnumber@QuadroIP:5060>;tag=131764415503572366-4d6e-4254-8a17-96f6626826c4
To: "caller1" <sip:caller1@SIPserver>;tag=as6682c795
Call-ID: 46fdc7537bc2f95135e871ab03f14545@SIPserver
CSeq: 931 BYE
Server: DHGW
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

***************************** SIP message buffer end ******************************

14:27:43 TLayer::MsgToTU # Msg type: 481 # TID: 2057233 # DID: 2057129
14:27:43 SipSessDlg::Inc4xxProc # Got 481 message # OID: 2057129 # SID: 0
14:27:43 SipSessDlg::Inc4xxProc # Call incorrect termination # Got 481 error message for BYE request # OID: 2057129 # SID: 0
14:27:43 SipSessDlg::TerminateTransactions # OID: 2057129 # SIPID: 46fdc7537bc2f95135e871ab03f14545@SIPserver # SID: 0
<<================================================== =====>>
<<======================LOGS MARKER======================>>
<<================================================== =====>>
<<User's Comment: SIP disconnect test stop1>>
<<================================================== =====>>

changed for security reasons:
caller1 = caller's number (it was GSM phone)
quadroIP = my quadro IP address
SIPnumber = my sip numer
SIPnumber-1 = secondary sip number used for outgoing calls
SIPserver = Sip server IP

davrays
10-18-2011, 11:45 AM
Let me understand first:

1. In this logs the messages are sent from "QuadroIP" to "QuadroIP", so there is just one IP address, and the IP of Asterisk server (is that the ITSP?) is not visible. How that happened? Did you replace two different IP addresses in logs with the same "QuadroIP" word?

2. There is no ringing at all in those logs. The quadro accepted the call almost immediately, and there is no ringing process. What was the destination of the call - is that an Autoattendant, or voicemail, or FXS phone or something different?
To be short, could you please describe the call by words (what was the calling device; was the call from ITSP or some other device; what was the destination; how long it rang; how long you waited until hanging up the calling phone; when did you hangup - during the ringing, or when the call was already established)?

Kristoph
10-19-2011, 03:06 AM
You are right, sorry for this, similar IP addresses, my bad.

davrays
10-19-2011, 03:42 PM
Thanks for correcting the logs.... What about my second question?

Kristoph
10-24-2011, 09:06 AM
Testing routine:

All incoming calls go to "00" when the caller gets the welcome message he chooses the extension he wants to ring. In this test it was "14" so the caller had to choose "014" to bypass the regular menu in the autoattendant.

If you wouldlike I can send you what was recorded in th voice mailbox when the caller disconnected before the call was answered.

davrays
10-24-2011, 01:11 PM
Well, if this is the test scenario, then you can clearly see in the logs above that the problem is in the SIP server (it looks to be a kind of Asterisk machine). As you tell, you have hang up the calling phone (at the Asterisk side), but we don't see any BYE message coming from the Asterisk machine. Thats why Quadro is not informed that you have hangup the calling phone, and so the call stays active on the Quadro (so you have a voicemail recorded). After timeout the call was closed by Quadro itself, and you can see at this point the Astreisk server told it doesn't know anything about that call ("481 Call leg/transaction does not exist" message).

So my assumptions are the following:
It looks like the Asterisk machine have closed this call locally after getting "488 Not Acceptable Here" response ftrom Quadro at 14:27:15. But Asterisk did that silently, without sending any "close" message to Quadro. This is incorrect behaviour from that server machine side (as it SHOULD send "BYE" message). I suppose this is a bug in that server software.

Now about what you can do to fix this problem:
1. you either have to find somebody to fix the problem in Asterisk (so it doesn't silently close the call after getting "488 Not Acceptable Here"),
or
2. find out why the Asterisk server sends the mid-call re-INVITE message at 14:27:15, which causes this problem (as it causes Quadro to send the 488 response), and configure the server not to send those re-INVITES. I suppose this could be some kind of session timer initiated from Astyerisk machine. Probably you have to disable that.

In any case - you have to look for the problem in the Asterisk server software, but not in the Quadro.

Kristoph
10-26-2011, 04:02 AM
I will contact the SIP provider and show them the conclusion you came to, hopefully they can fix the problem.

Thank you for your time!