Re: [asterisk-users] OT: OpenSIPS vs Kamailio -- which do you use and why?
Hi, I have been working on a project with asterisk and kamailio. I would prefer using kamailio because i have personally met with the developers and it has more active users and rapid developments. The developers are also very friendly and helpful. And well open ser is not gone, the name is changed to kamailio I guess. It had a fork, but now they have merged together. Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net On 3/4/2011 11:49 AM, Steve Edwards wrote: I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :) Which do you use and why? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to asterisk t.38
Hi Bryant, The 1.4 box has two interfaces one with 202 ip and the other with 172 ip , the audio code has 172 ip and the ast 1.6 has only 172 ip. Any ideas ? Both the trunks have t.38 enabled on it. And the way we use fax is fax machine connected to ata which supports t.38 in both the ends. Thank You Amit Nepal Systems Administrator Phoenix Internet Phone: 602-385-0731 602-234-0917#112 http://www.phoenixinternet.net On 1/20/2011 4:11 PM, Bryant Zimmerman wrote: Amit Make sure that the trunk you have between the two servers has the t.38 enabled on it. Do you have any NAT between the two servers or are they on the same lan. We do the t.38 faxing between 1.4 and 1.6 asterisk boxes all of the time. Our audio codes gateway dumps into a 1.4 box and all faxes calls are then sent to either 1.6.x or 1.8.x boxes and then on to the final ata. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 *From*: Amit Nepal ami...@phoenixinternet.net *Sent*: Thursday, January 20, 2011 4:27 PM *To*: asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Asterisk to asterisk t.38 Hi, I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is sending/receiving between ast 1.4 and ast 1.6. ATA (T.38 capable) AST 1.6 AUDIO CODEAST 1.4ATA (t.38 Capable) Thank You Amit Nepal On 1/20/2011 1:56 PM, David Backeberg wrote: On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepalami...@phoenixinternet.net wrote: I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? You don't say what's between the boxes as the medium over which the faxes are going. Try a fax between them without t.38 and see if it goes through. It might be a connection that is not reliable for any kind of faxing. That would not be an asterisk problem, it would be a faxing over a bad connection problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk to asterisk t.38
I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? -- Thank You Amit Nepal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to asterisk t.38
Hi, I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is sending/receiving between ast 1.4 and ast 1.6. ATA (T.38 capable) AST 1.6 AUDIO CODEAST 1.4ATA (t.38 Capable) Thank You Amit Nepal On 1/20/2011 1:56 PM, David Backeberg wrote: On Thu, Jan 20, 2011 at 3:14 PM, Amit Nepalami...@phoenixinternet.net wrote: I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? You don't say what's between the boxes as the medium over which the faxes are going. Try a fax between them without t.38 and see if it goes through. It might be a connection that is not reliable for any kind of faxing. That would not be an asterisk problem, it would be a faxing over a bad connection problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to asterisk t.38
Yes Tom, I am sending via the PSTN gateway which is audio code in my case. Thank You Amit Nepal On 1/20/2011 3:07 PM, Tom Rymes wrote: On 01/20/2011 4:26 PM, Amit Nepal wrote: I have an Audio code gateway between two asterisk servers. The audio code has PRI connected for PSTN. I can send faxes and receive faxes in ast 1.4 . Also I can send faxes for ast 1.6 to outside (PSTN) and receive faxes. The only problem I am having is sending/receiving between ast 1.4 and ast 1.6. ATA (T.38 capable) AST 1.6 AUDIO CODEAST 1.4ATA (t.38 Capable) It sounds like you are trying to send a fax directly from AST 1.6 to AST 1.4 via t.38 that never hits the PSTN. Have you tried sending a fax from AST 1.6 out via the PSTN, and then back in via the PSTN to AST 1.4? Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
