Hi, all. I've used the perl/AGI interface, and... well, I found it kind of hokey. Granted, this was in 1.2 days -- perhaps things have changed. Regardless, I guess I have two questions: 1) Has the Perl/AGI "binding" improved since then? 2) Is there any chance of a "real" API for Perl? Thanks much! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.
Ken D'Ambrosio wrote:> Hi, all. I've used the perl/AGI interface, and... well, I found it kind > of hokey. Granted, this was in 1.2 days -- perhaps things have changed. > Regardless, I guess I have two questions: > 1) Has the Perl/AGI "binding" improved since then? > 2) Is there any chance of a "real" API for Perl?What is your criterion of "real"? That is to say, what do you need that it does not provide? I've used AGI and FastAGI in Perl extensively and it is yet to fail to serve my purposes. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
Greg Oliver
2008-Feb-02 21:15 UTC
[asterisk-users] Multiple SIP phones behind a Linksys firewall
On Feb 2, 2008, at 2:11 PM, John Von Essen <john at quonix.net> wrote:> I posted an email a few days regarding a problem with hearing the > voicemail greeting on my sip phones. > > It turns out to be a phone/stun/linksys issue - not an asterisk issue. > Which brings up a couple of questions.... > > I always assumed that you can have multiple SIP phones behind a > Linksys > firewall/router (WRT54G) all using the same STUN server/port. > > But apparently thats not the case. Is it a Linksys bug, a > Grandstream bug > in the BudgeTone-100 phone, or am I off base and just doing something > wrong? > > I cleary have problems as soon as I try to use a second phone behind > the > Linksys - registration issues, cant hear voicemail greeting, etc.,. > > My next test was to run multiple STUN servers on the same machine with > different ports. Then, for my multiple SIP phones behind the > Linksys, have > each phone use a different stun port. > > Any thoughts? > > JohnI have 3 phones connected to 2 servers behind a 54g running openwrt with no stun or any special configuration. I am running cisco phones which do nat well natively. -greg> > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
john at quonix.net
2008-Feb-02 21:43 UTC
[asterisk-users] Multiple SIP phones behind a Linksys firewall
Greg, Without STUN how are the phones able to register? I was unable to get the Grandstream phones to work at all without STUN. -John ----------------------------------------------------
Greg Oliver
2008-Feb-02 23:00 UTC
[asterisk-users] Multiple SIP phones behind a Linksys firewall
On Feb 2, 2008, at 3:43 PM, john at quonix.net wrote:> Greg, > > Without STUN how are the phones able to register? I was unable to > get the > Grandstream phones to work at all without STUN. > > -John >I have nat on in sip.conf and off on the phones. Works perfect for 7960/1 and 7971. When I get back home, I will login to the asterisk servers and tell you what IPs the registration requests have in them.> ---------------------------------------------------- > From : Greg Oliver <greg.oliver at cistera.com> > To : Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Subject : Re: [asterisk-users] Multiple SIP phones behind a Linksys > firewall > Date : Sat, 2 Feb 2008 15:15:34 -0600 >> >> >> On Feb 2, 2008, at 2:11 PM, John Von Essen <john at quonix.net> wrote: >> >>> I posted an email a few days regarding a problem with hearing the >>> voicemail greeting on my sip phones. >>> >>> It turns out to be a phone/stun/linksys issue - not an asterisk >>> issue. >>> Which brings up a couple of questions.... >>> >>> I always assumed that you can have multiple SIP phones behind a >>> Linksys >>> firewall/router (WRT54G) all using the same STUN server/port. >>> >>> But apparently thats not the case. Is it a Linksys bug, a >>> Grandstream bug >>> in the BudgeTone-100 phone, or am I off base and just doing >>> something >>> wrong? >>> >>> I cleary have problems as soon as I try to use a second phone behind >>> the >>> Linksys - registration issues, cant hear voicemail greeting, etc.,. >>> >>> My next test was to run multiple STUN servers on the same machine >>> with >>> different ports. Then, for my multiple SIP phones behind the >>> Linksys, have >>> each phone use a different stun port. >>> >>> Any thoughts? >>> >>> John >> >> I have 3 phones connected to 2 servers behind a 54g running openwrt >> with no stun or any special configuration. I am running cisco phones >> which do nat well natively. >> >> -greg >>> >>> _______________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com >>> -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users