<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Times New Roman; font-size: 12pt; color: #000000'>Hey Moy, long time we don't talk :)<div><br></div><div>I was just wondering if that was curently possible or not. Thanks for the info.<br><br><hr id="zwchr"><blockquote style="border-left:2px solid rgb(16, 16, 255);margin-left:5px;padding-left:5px;color:#000;font-weight:normal;font-style:normal;text-decoration:none;font-family:Helvetica,Arial,sans-serif;font-size:12pt;">That is because Asterisk will wait for the first leg of the call to be ready before going to the second leg. Considerable code changes would have to be done in chan_dahdi to support what you need.<div><br clear="all">Moises Silva<br>
Senior Software Engineer, Software Development Manager<br>Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada<br>t. 1 905 474 1990 x128 | e. <a href="mailto:moy@sangoma.com" target="_blank">moy@sangoma.com</a><br>
<br><br><div class="gmail_quote">2011/4/26 Vinícius Fontes <span dir="ltr"><<a href="mailto:vinicius@canall.com.br" target="_blank">vinicius@canall.com.br</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Hello.<br>
<br>
Considering the following setup:<br>
<br>
Legacy PBX --(ISDN)--> Asterisk --(MFC/R2)--> PSTN<br>
<br>
<br>
When a user dials out, Asterisk receive overlap digits, matches them to an extension and dial the PSTN, completing the call. So far so good.<br>
<br>
The issue I'm trying to solve (or at least improve) is the call takes much longer to complete than the users were used to before having Asterisk between the PBX and the PSTN. It happens because the digits are sent to the PSTN only after the extension is matched in the dialplan, and dialing on MFC/R2 takes a few seconds.<br>
<br>
Here's the console log. Notice how it takes 6 seconds from the instant the user starts dialing to the instant the dialed number starts to ring. First 3 seconds is the user manually dialing plus Asterisk absolute timeout. Next 3 seconds are the time Asterisk takes to dial the number to the PSTN and the call be accepted.<br>
<br>
<br>
[Apr 26 10:57:13] -- Accepting overlap call from '7416' to '<unspecified>' on channel 0/1, span 2<br>
[Apr 26 10:57:13] -- Starting simple switch on 'DAHDI/32-1'<br>
<br>
*** User finished dialing + Asterisk absolute timeout ***<br>
<br>
[Apr 26 10:57:16] -- Executing [0145333114657@pbx:1] Answer("DAHDI/33-1", "") in new stack<br>
[Apr 26 10:57:16] -- Executing [0145333114657@pbx:2] Dial("DAHDI/33-1", "DAHDI/g1/0145333114657") in new stack<br>
[Apr 26 10:57:16] -- Called g1/0145333114657<br>
<br>
*** Asterisk starts dialing to the PSTN ***<br>
<br>
[Apr 26 10:57:19] MFC/R2 call has been accepted on forward channel 1<br>
[Apr 26 10:57:19] -- DAHDI/1-1 is ringing<br>
<br>
*** Dialed number finally rings ***<br>
<br>
So my question is: is there a way to fully overlap the digits from the user's phone on the PBX (ISDN) to the PSTN (MFC/R2), eliminating the need to wait for an extension to be matched? I already have overlapdial=yes in both spans, but that didn't made it. Also googled for it, even searched this list archives but found nothing.<br>
<br>
<br>
chan_dahdi.conf:<br>
<br>
[channels]<br>
<br>
signalling=mfcr2<br>
mfcr2_variant=br<br>
mfcr2_get_ani_first=no<br>
mfcr2_max_ani=20<br>
mfcr2_max_dnis=4<br>
mfcr2_category=national_subscriber<br>
mfcr2_logdir=span1<br>
mfcr2_call_files=no<br>
mfcr2_logging=all<br>
mfcr2_mfback_timeout=-1<br>
mfcr2_metering_pulse_timeout=-1<br>
mfcr2_allow_collect_calls=yes<br>
mfcr2_double_answer=no<br>
mfcr2_immediate_accept=no<br>
mfcr2_forced_release=no<br>
mfcr2_charge_calls=yes<br>
language=pt_BR<br>
echocancel=yes<br>
echocancelwhenbridged=no<br>
callgroup=0<br>
pickupgroup=0<br>
group=1<br>
context=telco<br>
overlapdial=yes<br>
channel => 1-15,17-31<br>
<br>
switchtype=euroisdn<br>
pridialplan=unknown<br>
prilocaldialplan=unknown<br>
priindication=outofband<br>
signalling=pri_net<br>
busydetect=yes<br>
busycount=5<br>
language=pt_BR<br>
echocancel=yes<br>
echocancelwhenbridged=no<br>
overlapdial=yes<br>
group=2<br>
context=pbx<br>
channel => 32-46,48-62<br>
<br>
<br>
<br>
extensions.conf:<br>
<br>
[telco]<br>
exten => _X.,1,Dial(DAHDI/g2/${EXTEN})<br>
<br>
[pbx]<br>
exten => _X.,1,Dial(DAHDI/g1/{$EXTEN})<br>
<br>
<br>
Sample console:<br>
<font color="#888888"><br>
<br>
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<br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br><br>asterisk-r2 mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-r2</blockquote><br></div></div></body></html>