[asterisk-users] Detecting Called party Ring indication (and act on it)

2009-08-15 Thread Ketema Harris
is there a way to have asterisk short circuit the dial timeout parameter based on called party sending ring progress ? History: I have multiple routes that a call can take. Some routes are not so good and take a long time. Currently I use the Dial time out parameter, but it times out

Re: [asterisk-users] SIP app for iPhone that works well with Asterisk?

2009-08-11 Thread Ketema Harris
For the price WeePhone is decent. Sent from my iPhone On Aug 11, 2009, at 4:57 PM, Philip A. Prindeville philipp_s...@redfish-solutions.com wrote: Anyone have a chance to test any of the various iPhone SIP apps? I see there are a few out there, but most of the iTunes reviews aren't

[asterisk-users] SIP AND NAT

2009-08-03 Thread Ketema Harris
I recently did a set up where I replaced a simple D-link home router that was having trouble processing a T1's worth of bandwidth with a linux machine running iptables. the kernel was 2.6.29-r5 and I chose the SIP connection tracking modules from the menuconfig. Router worked fine for

Re: [asterisk-users] CallerPres SIP headers Analog Phone

2009-07-23 Thread Ketema Harris
Yes. I have sendrpid = yes in sip.conf. CallerPres works fine with sip handsets. On Jul 23, 2009, at 4:29 AM, Ishfaq Malik wrote: Ketema Harris wrote: hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file

[asterisk-users] CallerPres SIP headers Analog Phone

2009-07-22 Thread Ketema Harris
hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the

Re: [asterisk-users] CallerPres SIP headers Analog Phone

2009-07-22 Thread Ketema Harris
the CallerPres() Thanks On Jul 22, 2009, at 11:36 AM, Philipp Kempgen wrote: Ketema Harris schrieb: hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling

[asterisk-users] Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card

2009-02-24 Thread Ketema Harris
Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the Green FXS ports are not green, they just are not

Re: [asterisk-users] Configuring chan_dahdi.conf for Sangoma A200/Remora FXO/FXS Analog AFT card

2009-02-24 Thread Ketema Harris
is unsuccessful! On Feb 24, 2009, at 3:41 PM, Ketema Harris wrote: Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two

[asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code

2008-10-09 Thread Ketema Harris
Hi I have searched the mailing lists and come across similar threads, but no actual solution. I am trying to use a Cisco AS5300 as a gateway for PSTNr. I have been able to configure it to take outbound calls and send them to the PSTN just fine. Inbound calls however are rejected by

Re: [asterisk-users] Inbound Calls From AS5300 Rejected with 488 Code

2008-10-09 Thread Ketema Harris
is set. On Thu, October 9, 2008 9:25 am, Ketema Harris wrote: Hi I have searched the mailing lists and come across similar threads, but no actual solution. I am trying to use a Cisco AS5300 as a gateway for PSTNr. I have been able to configure it to take outbound calls and send them