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...fromuser=NUM2 authuser=NUM2 secret=SECRET host=sip.broadvoice.com context=sip fromdomain=sip.broadvoice.com canreinvite=no nat=never dtmfmode=inband [broadvoice-incoming] type=peer host=147.135.8.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never [broadvoice-incoming2] type=peer host=147.135.0.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never [broadvoice-incoming3] type=peer host=147.135.4.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never In addition, I have this config in extension...

What can I do because my X100P don´t answer sometimes. I put these parameters in the zapata.conf, but sometimes my X100P don´t answer. usedistinctiveringdetection=yes dring1=363,137,0 dring1context=incoming2 Any sugestion? Thank´s --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040707/1b672f99/attachment...

...s rxgain = 0.0 txgain = 0.0 useincomingcalleridonzaptransfer = yes callreturn = yes callwaiting = yes threewaycalling = yes cancallforward = yes busydetect = yes busycount = 6 musiconhold = default callgroup=1 pickupgroup=1 signalling=fxs_ks context=incoming1 group = 1 channel => 1,3-6 context=incoming2 group = 2 channel => 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060515/dd59bc48/attachment.htm

...ID}) exten => s,n,Goto(incoming1,s,1) [from-pstn4] exten => s,1,Set(FROM_DID="6780000000) exten => s,n,NoOp(id is ${FROM_DID}) exten => s,n,Goto(incoming3,s,1) [from-pstn5] exten => s,1,Set(FROM_DID="6780000000) exten => s,n,NoOp(id is ${FROM_DID}) exten => s,n,Goto(incoming2,s,1) [from-pstn6] exten => s,1,Set(FROM_DID="6780000000) exten => s,n,NoOp(id is ${FROM_DID}) exten => s,n,Goto(incoming1,s,1) [from-pstn7] exten => s,1,Set(FROM_DID="6780000000) exten => s,n,NoOp(id is ${FROM_DID}) exten => s,n,Goto(incoming1,s,1) [from-pstn8] exten...

Today, We have added INVITE Authentication. This seems to bring a large amount of problems to people in the way since they can't make outbound calls. Here's what needs to be done. You need to add three variables to your peers or friends, username, authuser, and secret. username=<phonenumber> authuser=<phonenumber> secret=<registration password> Dan

...>> >>> [broadvoice-incoming] >>> type=peer >>> host=147.135.8.128 >>> context=from-broadvoice >>> qualify=yes >>> canreinvite=no >>> disallow=all >>> allow=ulaw >>> nat=never >>> >>> [broadvoice-incoming2] >>> type=peer >>> host=147.135.0.128 >>> context=from-broadvoice >>> qualify=yes >>> canreinvite=no >>> disallow=all >>> allow=ulaw >>> nat=never >>> >>> [broadvoice-incoming3] >>> type=peer >>...

...NAT canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw [broadvoice-incoming] context=from-broadvoice dtmf=inband dtmfmode=inband fromdomain=sip.broadvoice.com host=sip.broadvoice.com insecure=very nat=yes secret=MyPassword1 type=user user=305xxxxx22 username=305xxxxx22 [broadvoice-incoming2] context=from-broadvoice dtmf=inband dtmfmode=inband fromdomain=sip.broadvoice.com host=sip.broadvoice.com insecure=very nat=yes secret=MyPassword1 type=user user=305xxxxx66 username=305xxxxx66 extensions.conf [from-broadvoice] exten => s,1,Answer exten => s,2,Wait(2) ; Waits 2 Seco...

....com md5secret=<password> qualify=yes canreinvite=no disallow=all allow=ulaw [broadvoice-incoming] type=peer dtmfmode=inband host=147.135.8.128 context=incoming qualify=yes nat=yes canreinvite=no fromdomain=sip.broadvoice.com username=2129999999 fromuser=2129999999 insecure=very [broadvoice-incoming2] type=peer dtmfmode=inband host=147.135.0.128 context=incoming qualify=yes nat=yes canreinvite=no fromdomain=sip.broadvoice.com username=2129999999 fromuser=2129999999 insecure=very --------------------------------------------------------- extensions.conf ------------------------------------------...

Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers

...llforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=yes loglevel=255 protocolclass=mfcr2 protocolvariant=mx,10,4,16 group=1 protocolend=cpe context=incoming1 channel => 1-15 channel => 17-31 group=2 protocolend=cpe context=incoming2 channel => 32-46 channel => 48-62 protocolvariant=mx,10,8 group=3 immediate=yes usecallerid=yes protocolend=co context=incoming3 channel => 63-77 channel => 79-93 group=4 protocolend=co context=incoming4 channel => 94-103 channel => 110-124 The port #1 of a T...