Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don't have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and only one T1 worth of phone lines (24) connected to it using an Adtran TA750 channel bank. Is Asterisk capable of handling multiple incoming VoIP calls arriving from the same source (IP) or do I need to get something else to take the incoming traffic and pass it on to Asterisk? (I've read about using SER as a SIP proxy, but it's not clear to me wheather I need it or not). Can I use the OpenH.323 module to take care of the incoming VoIP traffic? I am totally new to all this. Any help will be really appreciated. Please give me your input on which is the best implementation of a VoIP->PSTN gateway and the some sample configuration files for the plattforms involved (Asterisk, SER, OpenH323, etc.) Thanks in advance. Alejandro Sosa. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040719/22af2867/attachment.htm
you can do that, in my experiencie, using oh323 I could not handle more than 30 active calls, doing g729 passthru... I dont know how to do IP limitation.... for restrict ip access use iptables.... I did basic dialpeers like this: exten -> 1305.,1,dial(OH323/xxxx) exten -> 1305.,2,congestion I am right? _____ De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de Alejandro Sosa Enviado el: Lunes, 19 de Julio de 2004 03:25 p.m. Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] PSTN gateway implementation? Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don't have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and only one T1 worth of phone lines (24) connected to it using an Adtran TA750 channel bank. Is Asterisk capable of handling multiple incoming VoIP calls arriving from the same source (IP) or do I need to get something else to take the incoming traffic and pass it on to Asterisk? (I've read about using SER as a SIP proxy, but it's not clear to me wheather I need it or not). Can I use the OpenH.323 module to take care of the incoming VoIP traffic? I am totally new to all this. Any help will be really appreciated. Please give me your input on which is the best implementation of a VoIP->PSTN gateway and the some sample configuration files for the plattforms involved (Asterisk, SER, OpenH323, etc.) Thanks in advance. Alejandro Sosa. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040719/49661657/attachment.htm
Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they don't have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and only one T1 worth of phone lines (24) connected to it using an Adtran TA750 channel bank. Is Asterisk capable of handling multiple incoming VoIP calls arriving from the same source (IP) or do I need to get something else to take the incoming traffic and pass it on to Asterisk? (I've read about using SER as a SIP proxy, but it's not clear to me wheather I need it or not). Can I use the OpenH.323 module to take care of the incoming VoIP traffic? I am totally new to all this. Any help will be really appreciated. Please give me your input on which is the best implementation of a VoIP->PSTN gateway and the some sample configuration files for the plattforms involved (Asterisk, SER, OpenH323, etc.) Thanks in advance. Alejandro Sosa. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040719/3da1e957/attachment.htm
This is an upgrade from a previous system. The old one didn't handle PRI, so they had analog phone lines as trunks. Management won't invest the money right now to get a PRI circuit. Any suggestions?> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Andrew Kohlsmith > Sent: Monday, July 19, 2004 4:59 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] PSTN gateway implementation? > > > -I have a TE405P board and only one T1 worth of phone lines (24) > > connected to it using an Adtran TA750 channel bank. > > Any particular reason against using PRI from your telco? > > > Is Asterisk capable of handling multiple incoming VoIP callsarriving> > from the same source (IP) or do I need to get something else to takethe> > incoming traffic and pass it on to Asterisk? (I've read about usingSER> > as a SIP proxy, but it's not clear to me wheather I need it or not).Can> > I use the OpenH.323 module to take care of the incoming VoIPtraffic?> > Asterisk can handle multiple calls from the same IP without any worry. > Your > main worry is the lack of real billing since you're terminating toanalog> PSTN instead of using PRI -- you have no way of actually knowing ifthe> call > was answered or not, so he'll be billed on every call. I doubt youwant> to > try and work with callprogress=yes. > > -A. > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users