Moises,<br><br>Thank you for your reply and the lesson of MFC/R2 !<br><br>My configs for the unicall.conf is:<br><pre>[channels]<br>language=br<br>context=from-pstn<br>usecallerid=yes<br>hidecallerid=no<br>immediate=no<br>
callwaitingcallerid=yes<br>threewaycalling=yes<br>transfer=yes<br>cancallforward=yes<br>callreturn=yes<br>echocancel=yes<br>echocancelwhenbridged=yes<br>rxgain=0.0<br>txgain=0.0<br>faxdetect=both<br>loglevel=0<br>protocolclass=mfcr2
<br>protocolvariant=br,20,4<br>protocolend=cpe<br>group=1<br>callerid=asreceived<br>channel=>1-15<br>channel=>17-31<br>channel=>32-46<br>channel=>48-62<br>channel=>63-77<br>channel=>79-93<br>protocolclass=mfcr2
</pre>
<br>The teleco who provides the links E1s is Brasil Telecom, I use the protocolvariant as shown in <a href="http://voip-info.org">voip-info.org</a>:<br><h4> Brasil Telecom</h4><h4>protocolvariant=br,20,4 <br></h4><div id="result_box" dir="ltr">
But I have a question in relation to variable: <br> protocolend=co<br><br>I was using "=co" and others configs I saw are using "=cpe". I have change it, but don't seams to have effect to me.<br></div>
<br>I read something on the internet which suggested changes in the file mfcr2.c to correct variables of timing. I believe that that should be the way to solution, but I do not feel safe to do this changes.<br><br><div id="result_box" dir="ltr">
Some
research later, I saw information that in future versions of libunicall
would not be necessary to rebuild lib to change parameters of timing, but I believe that's not implemented yet.<br><br>How I can set a time of increased response of "Seize ACK" ?<br><br>Thank you !<br></div>
<br><div><span class="gmail_quote">2007/12/11, Moises Silva <<a href="mailto:moises.silva@gmail.com">moises.silva@gmail.com</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Roger,<br><br>The "seize ack timeout" problem is because libmfcr2 is expecting a<br>response ( an ACK ) from the far end and it does not arrive in a R2<br>variant dependant amount of time. Which protocolvariant do you have
<br>configured in unicall.conf?<br><br>This is how the process to start a call goes:<br><br>1. When you Dial(Unicall/blah), Asterisk will ask chan_unicall.c to<br>initiate the call. chan_unicall will ask libunicall to start the call,
<br>and libunicall will ask libmfcr2 to start the call.<br><br>2. libmfcr2 will set the ABCD bits to 0x0 (000) ( normally the ABCD<br>bits are in Idle 1001 ). Setting the ABCD bits to 0x0 is our way to<br>tell the far end ( the telco ) that we want to start a call, this is
<br>known as the "Seize".<br><br>3. The far end should detect this bit pattern change and answer with a<br>"Seize ACK" ( ABCD bits in 0xC ), in this case, libmfcr2 does not<br>receive the expected ACK in 2000ms unless you are in Kuwait ( 4000ms )
<br>or Nigeria (10000ms ).<br><br>So, let us know your R2 variant, probably your country require more<br>time to wait for the Seize ACK.<br><br>Regards,<br><br>Moisés Silva<br><br><br>On Dec 11, 2007 7:03 AM, Roger C. Beraldi Martins
<br><<a href="mailto:rogerberaldi@gmail.com">rogerberaldi@gmail.com</a>> wrote:<br>> Dears,<br>><br>> After having set up the board Digium TE420 to receive 3 E1s, I can receive<br>> calls without difficulties. As you can see in the log below:
<br>><br>> -- Executing [5908@from-pstn:1] NoOp("UniCall/14-1", "Catch-All DID Match<br>> - Found 5908 - You probably want a DID for this.") in new stack<br>> -- Executing [5908@from-pstn
:2] Goto("UniCall/14-1", "ext-did|s|1") in<br>> new stack<br>> -- Goto (ext-did,s,1)<br>> -- Executing [s@ext-did:1] Set("UniCall/14-1", "__FROM_DID=s") in new<br>
> stack<br>> -- Executing [s@ext-did:2] GotoIf("UniCall/14-1", "0 ?cidok") in new<br>> stack<br>> -- Executing [s@ext-did:3] Set("UniCall/14-1",<br>> "CALLERID(name)=4133602900") in new stack
<br>> -- Executing [s@ext-did:4] NoOp("UniCall/14-1", "CallerID is<br>> "4133602900" <4133602900>") in new stack<br>> -- Executing [s@ext-did:5] Goto("UniCall/14-1", "ivr-3|s|1") in new
<br>> stack<br>> -- Goto (ivr-3,s,1)<br>> *snip*<br>> -- Executing [s@ivr-3:10] BackGround("UniCall/14-1", "custom/celia") in<br>> new stack<br>> -- <UniCall/14-1> Playing 'custom/celia' (language 'br')
<br>> -- Executing [h@ivr-3:1] Hangup("UniCall/14-1", "") in new stack<br>> -- Hungup 'UniCall/14-1'<br>> -- Unicall/14 released<br>><br>><br>><br>> Now I am having problems to make calls using the libunicall. The problem is
<br>> clear in this line of the full log:<br>> [Dec 11 10:03:54] ERROR[12935] chan_unicall.c: Unicall/1 protocol error.<br>> Cause 32776<br>><br>> Searching for the error I discovered it is "Seize ack timed out", but I do
<br>> not know exactly of what it means or how to fix it. Here is de version of<br>> softwares/libs I have use (<br>> <a href="http://www.voip-info.org/wiki/view/Asterisk+MFC+R2">http://www.voip-info.org/wiki/view/Asterisk+MFC+R2
</a>).<br>><br>> asterisk-1.4.9<br>> spandsp-0.0.4<br>> unicall-0.0.5pre1<br>> zaptel-1.4.4<br>><br>> I already try using asterisk 1.4.10 but the comportment is the same. I<br>> don't believe the problem is in asterisk. I think my configs are correctly
<br>> but not sure. Attached in text file follow the tests I have done using<br>> testunicall, config files from zaptel.conf and unicall.conf I using on this<br>> solution. More logs is in the same file.<br>>
<br>> This can be caused by a problem with signaling between my settings and the<br>> standard of telephony service ?<br>><br>> I'm using FreePBX with a Custon Trunk (Custon String Dial: UniCall/g1), my<br>
> extensions_aditional has<br>> the "OUT_3 = AMP:UniCall/g1" and "OUTMAXCHANS_3 = 10".<br>><br>> Someone has already gone through a problem like this ? I would be grateful<br>> if received suggestions to correct it.
<br>><br>> Log Full:<br>><br>> [Dec 11 10:03:51] VERBOSE[12935] logger.c: -- Executing<br>> [s@macro-dialout-trunk :32] Dial("SIP/2290-09b18a68", "UniCall/g1|300|") in<br>> new stack
<br>> [Dec 11 10:03:51] DEBUG[12935] chan_unicall.c: unicall_call called - 'g1'<br>> [Dec 11 10:03:51] DEBUG[12935] chan_unicall.c: unicall_call caller id -<br>> '2290'<br>> [Dec 11 10:03:51] VERBOSE[12935]
logger.c: -- Called g1<br>> [Dec 11 10:03:51] NOTICE[12935] chan_unicall.c: Unicall/1 event Dialing<br>> [Dec 11 10:03:54] NOTICE[12935] chan_unicall.c: Unicall/1 event Protocol<br>> failure<br>> [Dec 11 10:03:54] ERROR[12935] chan_unicall.c: Unicall/1 protocol error.
<br>> Cause 32776<br>> [Dec 11 10:03:54] DEBUG[12935] chan_unicall.c: disabled echo cancellation on<br>> channel 1<br>> [Dec 11 10:03:54] WARNING[12935] app_dial.c: Unable to forward voice or dtmf<br>> [Dec 11 10:03:54] DEBUG[12935] chan_unicall.c: Hangup: channel: 1 index = 0,
<br>> normal = 10, callwait = -1, thirdcall = -1<br>> [Dec 11 10:03:54] DEBUG[12935] chan_unicall.c: Updated conferencing on 1,<br>> with 0 conference users<br>> [Dec 11 10:03:54] VERBOSE[12935] logger.c: -- Hungup 'UniCall/1-1'
<br>> [Dec 11 10:03:54] VERBOSE[12935] logger.c: == Everyone is busy/congested<br>> at this time (1:0/0/1)<br>><br>><br>><br>><br>> --<br>> Atenciosamente,<br>><br>> Roger C. Beraldi Martins
<br>> Fone: 41-8828-7068<br>> _______________________________________________<br>> --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--</a><br>><br>> asterisk-users mailing list
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-- <br>Atenciosamente,<br><br>Roger C. Beraldi Martins<br>Fone: 41-8828-7068