search for: registeriax

...w_an_extns=no hasagent=no hasdirectory=no [250] callwaiting=yes cid_number= context=numberplan-custom-2 email= fullname=Winanjaya group= hasagent=yes hasdirectory=no hasiax=yes hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=250 secret=1234 threewaycalling=yes vmsecret=1234 zapchan= registeriax=yes registersip=yes canreinvite=no nat=no dtmfmode=rfc2833 disallow=all allow=all type=peer [998] callwaiting=yes cid_number= context=numberplan-custom-2 email= fullname=MyLine1 group= hasagent=yes hasdirectory=no hasiax=yes hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=999 secret...

...ike this... [6001] callwaiting = yes context = numberplan-custom-1 email = jsmall at visinet.com.au fullname = Joshua Small hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6001 secret = XXXXX threewaycalling = yes registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 vmsecret = 1234 ;some PSTNS [trunk_2] callerid = asreceived context = DID_trunk_2 group = 2 hasexten = no hasiax = no hassip = no trunkname = Ports 1,2,3,4 trunkstyle = analog zapchan = 1,2,3,4 ;my IP trunk...

...ike this... [6001] callwaiting = yes context = numberplan-custom-1 email = jsmall at visinet.com.au fullname = Joshua Small hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6001 secret = XXXXX threewaycalling = yes registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 vmsecret = 1234 ;some PSTNS [trunk_2] callerid = asreceived context = DID_trunk_2 group = 2 hasexten = no hasiax = no hassip = no trunkname = Ports 1,2,3,4 trunkstyle = analog zapchan = 1,2,3,4 ;my IP trunk...

...] callwaiting = yes cid_number = 951XXXXXX context = numberplan-custom-1 email = fullname = New User group = hasagent = no hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 951XXXXXX secret = 000000 threewaycalling = yes vmsecret = 1234 zapchan = registeriax = no registersip = yes

....conf entry for Asterisk registration to the Sip Provider. (I know I don't have T38 as allowed codecs, not sure what to add for T38) [trunk_66] ;register allow = ulaw dialformat = ${EXTEN:1} canreinvite = no hasexten = no hasiax = no hassip = yes host = provider.com insecure = very port = 5060 registeriax = no registersip = yes trunkname = abc username = abc disallow = gsm,g726,alaw contact = abc secret = abc Any ideas appreciated. Thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091005/66fc15b2/attac...

...fo Trunk Info [trunk_1] disallow = allow = all callerid = 028012xxxx contact = context = DID_trunk_1 dialformat = ${EXTEN:1} fromdomain = iinetphone.iinet.net.au fromuser = 028012xxxx group = hasexten = no hasiax = no hassip = yes host = sip.nsw.iinet.net.au insecure = very port = 5060 provider = registeriax = no registersip = yes secret = xxxxxxxx trunkname = Custom - iinet trunkstyle = customvoip username = 028012xxxx The dialplan, Just dial 0, then number, then strip the first 0 and dial [numberplan-custom-2] include = default plancomment = home exten = _0X!,1,Macro(trunkdial,${trunk_1}/? ${EXTEN...

...asedrules] *******part of extensions.conf that was added by asterisk-gui (svn)******* *******part of users.conf that was added by asterisk-gui (svn)******* [trunk_1] allow = all context = DID_trunk_1 dialformat = ${EXTEN:1} hasexten = no hasiax = yes hassip = no host = iax2.fwdnet.net port = 4569 registeriax = yes registersip = no secret = rycort4e trunkname = Custom - fwd trunkstyle = customvoip username = 788694 [6000] callwaiting = yes cid_number = 6000 fullname = proton hasagent = yes hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host = dynamic mailbox = 6000 secret...

...gt; #72 ; Park call (one step parking) --users.conf-- [6003] callwaiting = no context = from-internal fullname = IT Support hasagent = no hasdirectory = no hasiax = yes hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6003 threewaycalling = no vmsecret = 1234 registeriax = yes registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 disallow = all allow = all [dtrr] ;bi-directional trunk to 2nd asterisk system. allow = ulaw,alaw context = from-outside dialformat = ${EXTEN} hasexten = no hasiax = no hassip = yes host = 10.10.0.10 port = 5060 username = dtrr s...

...Asterisk does match the correct trunk and seems to send out correct Auth credentials...but not the one below.. [trunk_1] ;register to SP allow = ulaw ;context = test dialformat = ${EXTEN:1} canreinvite = no hasexten = no hasiax = no hassip = yes host = provider.sip.com insecure = very port = 5060 registeriax = no registersip = yes trunkname = test trunkstyle = customvoip username = 3035551122 disallow = gsm,g726,alaw contact = 3035551122 secret = xxxxx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101004/f...

...callwaiting = yes cid_number = 6058 context = default email = user@mydomain.com fullname = Bob smith group = hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = yes host=dynamic mailbox = 6058 secret = myPassword threewaycalling = yes vmsecret = 1337 zapchan = registeriax = no registersip = yes Any idea why this doesn't work but registering normal SIP configured accounts does? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070419/c30c94e0/attachment.htm

...se it seems to add each phone to the group, which breaks my GotoIf() statement. Here's some relevant information: Users.conf (added by Asterisk-GUI) [trunk_2] provider = Bandwidth (SIP) ; GUI metadata context = DID_trunk_2 hasexten = no hasiax = no hassip = yes host = 216.82.224.202 registeriax = no registersip = no usecallerid = yes nat = no ;Testing trunkname = Bandwidth.com (Sip) ; GUI metadata username = secret = disallow = all allow = ulaw,alaw,g726 sip.conf [general] context = frombandwidth ;other variables, etc. ;Added according to Bandwidth.com's wiki entry....

...d fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = rfc2833 relaxdtmf = yes rfc2833compensate = yes port = 5060 canreinvite = no fromdomain = galvatron.vtnoc.net disallow = all allow = ulaw,gs...

...Users.conf [6001] callwaiting = yes context = numberplan-custom-1 email = jsmall at visinet.com.au fullname = Joshua Small hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6001 secret = SECRET threewaycalling = yes registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 Queues.conf [6003] fullname = All of us strategy = ringall timeout = wrapuptime = autofill = yes autopause = no maxlen = joinempty = no leavewhenempty = no reportholdtime = no musicclass = member = Agent/600...

...[trunk] type = friend callwaiting = yes caller id = contact = context = default fullname = group = hasagent = no hasdirectory = yes hasiax = no hasmanager = no hassip = yes host = 172.21.235.1 secret = [password] threewaycalling = yes registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 registeriax = no disallow = all allow = gsm register=>trunk:[password]@172.21.235.1 Here is the applicable portion of extensions.conf: [default] exten = _5XX,1,Dial(SIP/trunk/${EXTEN},,Tt) Here is the SIP Debug output: INVITE sip:510 at 172.21.235.1 SIP/2.0 Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9...

...;o:p></o:p></p> <p class=3DMsoNormal>mailbox =3D 6001<o:p></o:p></p> <p class=3DMsoNormal>secret =3D XXXXX<o:p></o:p></p> <p class=3DMsoNormal>threewaycalling =3D yes<o:p></o:p></p> <p class=3DMsoNormal>registeriax =3D no<o:p></o:p></p> <p class=3DMsoNormal>registersip =3D yes<o:p></o:p></p> <p class=3DMsoNormal>canreinvite =3D no<o:p></o:p></p> <p class=3DMsoNormal>nat =3D no<o:p></o:p></p> <p class=3DMsoNormal&g...

Hello everyone, I have previously asked this question on the asterisk-video list, but I got directed here. I have a setup consisting of asterisknow beta4 (not sure if that is crucial) and a few clients all running X-Lite 3.0 (not eyebeam) on the local network. My computer has a USB-Camera installed, and now I would like to do some video calling with it, at least, so that the other user can

...;o:p></o:p></p> <p class=3DMsoNormal>mailbox =3D 6001<o:p></o:p></p> <p class=3DMsoNormal>secret =3D XXXXX<o:p></o:p></p> <p class=3DMsoNormal>threewaycalling =3D yes<o:p></o:p></p> <p class=3DMsoNormal>registeriax =3D no<o:p></o:p></p> <p class=3DMsoNormal>registersip =3D yes<o:p></o:p></p> <p class=3DMsoNormal>canreinvite =3D no<o:p></o:p></p> <p class=3DMsoNormal>nat =3D no<o:p></o:p></p> <p class=3DMsoNormal&g...

...] callwaiting = yes cid_number = 60011 context = numberplan-custom-2 fullname = Roberto Bolivar Casa hasagent = yes hasdirectory = no hasiax = no hasmanager = no hassip = yes hasvoicemail = yes deletevoicemail = no host = dynamic mailbox = 60011 secret = 123654 threewaycalling = yes vmsecret = 1234 registeriax = no registersip = yes canreinvite = no nat = yes dtmfmode = rfc2833 disallow = all allow = all the client Xlite call to 66666 and the enter to musicold and not pass to the agent ... how to the problem ??? and the asterisk-gui not can edit the "Call Queues" rules and show message: &q...

...d fail till I did a sip reload or restarted Astersk. Now we get no DTMF ever. System set up. Asterisk 1.4.22 Asterisk GUI 2.0 users.conf [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = rfc2833 relaxdtmf = yes rfc2833compensate = yes port = 5060 canreinvite = no fromdomain = galvatron.vtnoc.net disallow = all allow = ulaw,gs...

I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk server at work from home. I've setup zoiper for iax, set the ip address to work's fixed ip address, user: home, password: password but the zoiper log shows: 11:02:35 Rejected registration for 'home@<my-office-ip-address>' with cause 'facility rejected' 11:03:35 Rejected registration for