search for: 100,102

...04&SIP/105,,20) exten => 9043442342,1,Voicemail(u102) The issue I have is that I need each user of these extensions to have multiple line appearances ("roll over" lines). In a traditional PBX, usually this is accomplished by setting up a roll over lines... i.e my extension is 100, my roll over extension is 200, and next roll over extension is 300. So if i am on my first line, the next call will roll over to 200. I have this setup and it works great for calling a single phone by setting incomingcalllimit=1 and I can do something like: exten 100,1,DIAL(SIP/100,,20) exten...

...co ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack -- Called 62.213.36.100 -- OH323/L4366 answered SIP/519-3781 1:36.475 LogChanTx:8130bc0 PWLib Assertion fail: Invalid parameter, file rtp.cxx, line 385, Error=22 <A>bort, <C>ore dump, <I>gnore? ?and connection becomes one-way styl...

...; Allow codecs in order of preference ;allow=ilbc allow=all [fwd.pulver.com] type=friend secret=mypassword username=myfwd# host=fwd.pulver.com [myfwd#] type=friend host=dynamic dtfmode=inband ; Choices are inband,rcf2833, or info context=sip username= myfwd# secret=mypassword mailbox=100 ; Mailbox for message waiting indicator callid="Red " <myfwd#> register =>myfwd#:mypassword @fwd.pulver.com/100 [my2ndfwd#] type=friend host=dynamic username=my2ndfwd# secret=mypassword dtmfmode=inband mailbox=101 context=sip callid="Red2 " <my2ndfwd#> register...

Can anyone recommend some phone sets that are adsi compliant and work well with asterisk?

I am having problems transferring calls from one sip extension to another - the extensions use various phones hardware/software. From what I can tell I should just be able to press # and then dial an extension to blind xfer a call right? How do I do attended xfer? Either the phones (for this test I have tried xlite and budgetone102) are not sending DTMF correctly or something else is amiss...

I searched the 22490 messages I have in my own personal asterisk-users archive and have not found the answer, and it also does not appear on the wiki. I have a SIP phone and a regular phone on a TDM400P FXS interface. Extensions are 100 and 101, respectively. On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension I want to transfer to. No problem. I can do the same thing on the FXS port. My question is does anyone have a dialplan that will bring the call I transfered back to me if the transf...

...asterisk-1.4.12. I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk server and client phone are on different computers but are on the same LAN, i.e. no NAT. I have an "echo test" extension which works fine. But when I try to call into my voicemail account using 8100, I do not hear the first "Playing 'vm_password'" message (although do I hear the subsequent messages). And any numbers I enter on the ekiga keypad do not seem to be recognised by asterisk (I enabled debug rdp and rfc2833 packets appear to be transmitted and received). The soft...

...imeout,15 exten => s,5,SetMusicOnHold,Random exten => s,6,Background,tku4calling exten => s,7,Background,menu exten => 2,1,Goto(sales,s,1) exten => 3,1,Goto(finance,s,1) exten => 4,1,Goto(change,s,1) exten => 5,1,Goto(support,s,1) exten => 0,1,Dial,ZAP/2|10|mt exten => 100,1,Dial,Zap/2|20|m exten => 100,2,Voicemail,u100 exten => 100,102,Voicemail,b100 exten => 101,1,Dial,Zap/2|20|m exten => 101,2,WaitMusicOnHold,15 exten => 101,3,Voicemail,u101 exten => 101,102,Voicemail,b101 exten => 102,1,Dial,Zap/2|20|m exten => 102,2,Voicemail,u102 exten...