search for: pstns

Hello All, I have a small * server in my home office with several IP phones. The system is not fully in service yet as I'm still hunting for a cost effective FXO adapter that I can rely upon for my two primary PSTNs. That said, I'd like to move it into service for another application...which brings up a question. I'd really like to stop making international calls from my cell phone when I'm travelling. Can someone point me to an example of extension logic that accepts an incomming call on a known...

...(I understand this can be done with the Mediatrix -- http://lists.digium.com/pipermail/asterisk-users/2004-February/035926.html), and only answer two of the three lines (the third line will be answered by a directly connected FAX machine, but we can use that line for outgoing calls if the first two PSTNs are in use). Does the higher cost of something like the Mediatrix ($634) bring advantages to the table over the TDM04b ($337)? Does the TDM card provide functionality which the sip gateways do not (e.g. incoming / outgoing call progress)? I appreciate hearing from folks who have experience wi...

...me = 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 [trunk_3] allow = all context = DID_trunk_3 dialformat = ${EXTEN:1} hasexten = no hasiax = no hassi...

...me = 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 [trunk_3] allow = all context = DID_trunk_3 dialformat = ${EXTEN:1} hasexten = no hasiax = no hassi...

...l application and I (!) think, that it is connected to the dahdi parameter "overlapdial=yes". Sangoma support does not see any connection between this. But when enabling this option I face with some(!) dial-partners a audio one-way issue (the called party can not be heard). Only using PSTNS (germany e1 trunk) - no voip. Is there any other setup out there with this issue? Kernel: Linux 2.6.26-2-amd64 #1 SMP Sun Jun 20 20:16:30 UTC 2010 x86_64 GNU/Linux Asterisk: Asterisk 1.6.1.20 built by root @ xxx on a x86_64 running Linux on 2010-09-03 07:47:24 UTC LibPRI: 1.4.11.4 DAHDI: 2.3.0...

f: Force the callerid of the *calling* channel to be set as the extension associated with the channel using a dialplan 'hint'. For exam= ple, some PSTNs do not allow CallerID to be set to anything other than the number assigned to the caller.

...FILE* is not defined, then the default behaviour is to announce ("You have [XX minutes] YY seconds"). o *f*: forces callerid to be set as the extension of the line making/redirecting the outgoing call. For example, some PSTNs don't allow callerids from other extensions than the ones that are assigned to you. o *w*: Allow the /called/ user to start recording after pressing *1 or what defined in features.conf <http://www.voip-info.org/wiki/index.php?page=Asterisk+config...

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern:...

Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian

...lt;o:p></o:p></p> <p class=3DMsoNormal>dtmfmode =3D rfc2833<o:p></o:p></p> <p class=3DMsoNormal>vmsecret =3D 1234<o:p></o:p></p> <p class=3DMsoNormal><o:p>&nbsp;</o:p></p> <p class=3DMsoNormal>;some PSTNS<o:p></o:p></p> <p class=3DMsoNormal>[trunk_2]<o:p></o:p></p> <p class=3DMsoNormal>callerid =3D asreceived<o:p></o:p></p> <p class=3DMsoNormal>context =3D DID_trunk_2<o:p></o:p></p> <p class=3DMsoNormal&...

...lt;o:p></o:p></p> <p class=3DMsoNormal>dtmfmode =3D rfc2833<o:p></o:p></p> <p class=3DMsoNormal>vmsecret =3D 1234<o:p></o:p></p> <p class=3DMsoNormal><o:p>&nbsp;</o:p></p> <p class=3DMsoNormal>;some PSTNS<o:p></o:p></p> <p class=3DMsoNormal>[trunk_2]<o:p></o:p></p> <p class=3DMsoNormal>callerid =3D asreceived<o:p></o:p></p> <p class=3DMsoNormal>context =3D DID_trunk_2<o:p></o:p></p> <p class=3DMsoNormal&...

...lt;o:p></o:p></p> <p class=3DMsoNormal>dtmfmode =3D rfc2833<o:p></o:p></p> <p class=3DMsoNormal>vmsecret =3D 1234<o:p></o:p></p> <p class=3DMsoNormal><o:p>&nbsp;</o:p></p> <p class=3DMsoNormal>;some PSTNS<o:p></o:p></p> <p class=3DMsoNormal>[trunk_2]<o:p></o:p></p> <p class=3DMsoNormal>callerid =3D asreceived<o:p></o:p></p> <p class=3DMsoNormal>context =3D DID_trunk_2<o:p></o:p></p> <p class=3DMsoNormal&...

...lt;o:p></o:p></p> <p class=3DMsoNormal>dtmfmode =3D rfc2833<o:p></o:p></p> <p class=3DMsoNormal>vmsecret =3D 1234<o:p></o:p></p> <p class=3DMsoNormal><o:p>&nbsp;</o:p></p> <p class=3DMsoNormal>;some PSTNS<o:p></o:p></p> <p class=3DMsoNormal>[trunk_2]<o:p></o:p></p> <p class=3DMsoNormal>callerid =3D asreceived<o:p></o:p></p> <p class=3DMsoNormal>context =3D DID_trunk_2<o:p></o:p></p> <p class=3DMsoNormal&...