Don Fanning
2006-Jan-02 07:44 UTC
[Asterisk-Users] Q: How to dial out / transfer calls with manager
Greetings, Here's my issue. My local free VSP isn't transfering proper DTMF (inband or converting to RFC2833) so I'm stuck with making a php interface so my roommates whom are not using softphone/ata devices to call out via * (and thusly get the better deals in Long Distance). I've tried using the Manager interface to creating the connection however when I create a Channel: it needs to be something virtually attached to the system. I'm trying to see if there is a way around it. IE: Currently I drop fputs($socket, "Secret: ibanez\r\n\r\n"); fputs($socket, "Action: Originate\r\n"); fputs($socket, "Channel: $mytelephone\r\n"); fputs($socket, "Exten: 1$callnumber\r\n"); fputs($socket, "Priority: 1\r\n\r\n");>From a php script with $mytelephone being the home phone via sip likeSIP/1235551212@sipprovider and $callnumber is the destination number which would default to my $TRUNK. However since the channel isn't registered on the system it will fail. Is there a way of cheating this via callpark or meetme? How about a dummy iaxclient to originate then dumps to a meetme with the $callnumber doing the same? I find this very limiting as I can't route calls the way I want to. (the DTMF issue is worse... Don't get me started. ;) Ideas? Thanks.
Moises Silva
2006-Jan-02 08:25 UTC
[Asterisk-Users] Q: How to dial out / transfer calls with manager
Your problem is related to not reading the documentation in voip-info.org :) You can originate a call to anyplace doing: - First a LoginAction. - Then an Originate action with the proper arguments. In the example you put, you are doing neither of them. You can test manually how the protocol works doing a login from a telnet client: telnet localhost 5038 (in case your in the asterisk box) Action: Login\r\n Username: someuser\r\n Secret: somesecretpassword\r\n\r\n Action: Originate Channel: SIP/13 <-- this should be the first phone you want to ring (your own phone usually) Context: somecontextwithoutbountpatterns Exten: XXXXXXXXXXXX <--- extension that will make your call Priority: 1 (usually one is fine) Best Regards On 1/2/06, Don Fanning <don@00100100.net> wrote:> Greetings, > > Here's my issue. My local free VSP isn't transfering proper DTMF > (inband or converting to RFC2833) so I'm stuck with making a php > interface so my roommates whom are not using softphone/ata devices to > call out via * (and thusly get the better deals in Long Distance). > > I've tried using the Manager interface to creating the connection > however when I create a Channel: it needs to be something virtually > attached to the system. I'm trying to see if there is a way around it. > > IE: Currently I drop > > fputs($socket, "Secret: ibanez\r\n\r\n"); > fputs($socket, "Action: Originate\r\n"); > fputs($socket, "Channel: $mytelephone\r\n"); > fputs($socket, "Exten: 1$callnumber\r\n"); > fputs($socket, "Priority: 1\r\n\r\n"); > > >From a php script with $mytelephone being the home phone via sip like > SIP/1235551212@sipprovider and $callnumber is the destination number > which would default to my $TRUNK. However since the channel isn't > registered on the system it will fail. > > Is there a way of cheating this via callpark or meetme? How about a > dummy iaxclient to originate then dumps to a meetme with the $callnumber > doing the same? I find this very limiting as I can't route calls the > way I want to. (the DTMF issue is worse... Don't get me started. ;) > > Ideas? Thanks. > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"
Don Fanning
2006-Jan-02 10:35 UTC
[Asterisk-Users] Q: How to dial out / transfer calls with manager
>From: asterisk-users-bounces@lists.digium.com[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Moises Silva>Sent: Monday, January 02, 2006 7:26 AM>Your problem is related to not reading the documentation invoip-info.org :) Umm.. Yeah I have. Otherwise I wouldn't be a pain in the ass right now. I'd just be clueless. :)>You can originate a call to anyplace doing: > >- First a LoginAction. >- Then an Originate action with the proper arguments. > >In the example you put, you are doing neither of them. You can testmanually how the protocol works doing a login from a >telnet client:> >telnet localhost 5038 (in case your in the asterisk box) > >Action: Login\r\n >Username: someuser\r\n >Secret: somesecretpassword\r\n\r\n >I can see myself login to the manager port just fine (even after I changed my password from my post slippage ;)>Action: Originate >Channel: SIP/13 <-- this should be the first phone you want to ring(your own phone usually) I don't want it to ring a REGISTERED device (SIP/IAX/ZAP) that is on the system. I want it to make a outbound call externally through my VSP and when it's answered, then make another outbound call on another channel.>Context: somecontextwithoutbountpatternsNot essential>Exten: XXXXXXXXXXXX <--- extension that will make your call >Priority: 1 (usually one is fine)Again, Ideas? Thanks, Don