search for: incoming3

...mode=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 extensions. [sip] exten => 1000,1,Playback,welcome Thanks in advance. Bart

...ID}) exten => s,n,Goto(incoming1,s,1) [from-pstn3] exten => s,1,Set(FROM_DID="6780000000) exten => s,n,NoOp(id is ${FROM_DID}) 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...

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

...gt;> >>> [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 >> >> -- >> skamp <skamp@arkayinc.com> >> >&gt...

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

...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 TE405P card is connected to the telco and the port #3 is connected to the pbx. I've changed the line (chan_unicall.c): uc_callparm_calling_pa...