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...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 [trunk_3] allow = all context = DID_trunk_3 dialformat = ${EXTEN:1} hasexten = no hasiax = no hassip = yes host = gw02.mytel.net.au port = 5060 registeriax = no registersip = yes secret = XXXXXXXX trunkname = Cus...

...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 [trunk_3] allow = all context = DID_trunk_3 dialformat = ${EXTEN:1} hasexten = no hasiax = no hassip = yes host = gw02.mytel.net.au port = 5060 registeriax = no registersip = yes secret = XXXXXXXX trunkname = Cus...

...on 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/attachment.htm

...erid = 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:1}) comment = _0X!,1,All Numbers,standard The t...

...=transport-tls context=from-twilio disallow=all allow=ulaw dtmf_mode=inband media_encryption=sdes rtp_symmetric=yes rewrite_contact=yes force_rport=yes canreinvite=no tlsdontverifyserver=yes [auth-out](!) type=auth auth_type=userpass [twilio] aors=twilio-aors [twilio-aors] type=aor contact=sips:trunkname.pstn.twilio.com:5061 ;tried with sip: also [twilio] type=identify endpoint=twilio match=54.172.60.0 match=54.172.60.1 match=54.172.60.2 match=54.172.60.3 [endpoint-basic](!) type=endpoint transport=transport-tls context=from-phones disallow=all allow=ulaw [auth-userpass](!) type=auth auth_type=...

...) B8ZS/ESF ClockSource group=0,11 context=DID_span_1 switchtype = national signalling = pri_cpe channel => 1-23 context = default group = 63 --- /etc/asterisk/users.conf (asterisk 1.4.22 w/ SVN-branch-2.0-r4657 GUI) --- [span_1] group = 1 hasexten = no switchtype = national signalling = pri_cpe trunkname = Span 1 trunkstyle = digital ; GUI metadata hassip = no hasiax = no context = DID_span_1 zapchan = 1-23 --- /etc/asterisk/users.conf (asterisk 1.4.24[0-1] w/ SVN-branch-2.0-r4661 GUI) --- group = 1 hasexten = no switchtype = national signalling = pri_cpe trunkname = Span 1 trunkstyle = digital...

...wing doc [1] stating you can pass username/password in a dial string. My goal is to dial from asterisk through one SIP trunk or another without touching my sip.conf file. In other words, I'm planning to use: Dial(technology/user:password at host/extension,timeout,options) instead of Dial(SIP/trunkname/extension) with a sip.conf that includes something like: [trunkname] username=username secret=password host=host My first question: is such thing possible, for a trunk which requires a password for each outgoing call ? Regards [1] http://the-asterisk-book.com/1.6/applikationen-dial.html

...be to see if that offsite location is still on a call forwarded to it by Asterisk. This way a receptionist could choose to transfer calls to a mobile phone only if it's finished with the last call the receptionist forwarded to it. If I configure a custom extension with the destination SIP/TrunkName/NXXNXXXXXX, the calls transfer fine but don't show as busy using the Flash Operator Panel (as an example). Any thoughts? Thanks in advance, Alex -- Alexander Burke, A+, CCNA Kingston, Ontario, Canada -------------- next part -------------- An HTML attachment was scrubbed... URL: http://l...

...unk 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/f00522ad/attachment.htm

...ment. 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. Changed to inband because we were having DTMF issues. [bandwidth.com_i...

...with but when we got pushed to another server the DTMF would 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...

...group =3D 2<o:p></o:p></p> <p class=3DMsoNormal>hasexten =3D no<o:p></o:p></p> <p class=3DMsoNormal>hasiax =3D no<o:p></o:p></p> <p class=3DMsoNormal>hassip =3D no<o:p></o:p></p> <p class=3DMsoNormal>trunkname =3D Ports 1,2,3,4<o:p></o:p></p> <p class=3DMsoNormal>trunkstyle =3D analog<o:p></o:p></p> <p class=3DMsoNormal>zapchan =3D 1,2,3,4<o:p></o:p></p> <p class=3DMsoNormal><o:p>&nbsp;</o:p></p> <p clas...

...ded 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 = proton threewaycalling = yes vmsecret = 1234 re...

...timert1: NULL timerb: NULL qualifyfreq: NULL constantssrc: NULL contactpermit: NULL contactdeny: NULL usereqphone: NULL textsupport: NULL faxdetect: NULL buggymwi: NULL auth: NULL fullname: NULL trunkname: NULL cid_number: NULL callingpres: NULL mohinterpret: NULL mohsuggest: NULL parkinglot: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL autoframing: NULL rtpkeepalive: NULL call-limit: NULL g726nonstandar...

...group =3D 2<o:p></o:p></p> <p class=3DMsoNormal>hasexten =3D no<o:p></o:p></p> <p class=3DMsoNormal>hasiax =3D no<o:p></o:p></p> <p class=3DMsoNormal>hassip =3D no<o:p></o:p></p> <p class=3DMsoNormal>trunkname =3D Ports 1,2,3,4<o:p></o:p></p> <p class=3DMsoNormal>trunkstyle =3D analog<o:p></o:p></p> <p class=3DMsoNormal>zapchan =3D 1,2,3,4<o:p></o:p></p> <p class=3DMsoNormal><o:p>&nbsp;</o:p></p> <p clas...

...with but when we got pushed to another server the DTMF would 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...

....org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure But it seems that it is not as per the Asterisk 1.8 SIP options. i.e. it contains 'call-limit' which is deprecated in 1.8 and not the 'callcounter' as one of the fields. Pardon my ignorance, but are 'cid_number','trunkname','fullname' from given link sip parameters to be set? sip.conf has no such entries. I also looked at .../contrib/realtime/mysql, and didn't find 'callcounter' in sipfriends.sql. I also couldn't find tables for queue and queue members in it. Yes, I can update or add la...

...no dtmfmode = rfc2833 disallow = all allow = all [dtrr] ;bi-directional trunk to 2nd asterisk system. allow = ulaw,alaw context = from-outside dialformat = ${EXTEN} hasexten = no hasiax = no hassip = yes host = 10.10.0.10 port = 5060 username = dtrr secret = dtrr registeriax = no registersip = no trunkname = Custom - Corporate trunkstyle = customvoip disallow = gsm,ilbc,speex,g726,adpcm,lpc10,g729 --extensions.conf-- [globals] trunk_2 => SIP/dtrr [macro-trunkdial] exten => s,1,Dial(${ARG1}) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-BUSY,1,Hangup() exten => s-NOANSWER,1,Hangup...

...Primary Proxy port: 5060 Line1 Number: 1002 2. Security Config Line1 Account: 1002 Line1 Password: ****** 3. Line Configuration Line1: Type=FXO, Hunting Group=2, Hot Line = 88621002 Asterisk settings: users.conf: [1002] context = DID_1002 host = *.133 username = 1002 secret = ****** trunkname = WellGate-1002 ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no host = dynamic disallow = all allow = ulaw,alaw,gsm,g726,g729 extensions.conf 1002 = SIP/1002 ... [DID_1002] exten => _88621002,1,NoOp(${CALLERID(num)}) exten => _886...

...ia;talk, have tried auto, rfc2833, and inband without success with any of them, and yes we had via:talk change their end too. Here is the users.conf entry or the connection to via:talk. [trunk_1] context = DID_trunk_1 host = galvatron.vtnoc.net username = <phone number> secret = blablabla trunkname = via:talk ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = <phone number> authuser = <phone number> insecure = port,invite dtmf = inband dtmfmode = inband relaxdtmf = yes ;rfc2833compensate = yes port = 5060 canreinv...