UTRINI@embratel.com.br
2004-Sep-29 14:15 UTC
[Asterisk-Users] HELP: Asterisk - SIP to H.323 translation
Hi all, I am new to this list... Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I want to implement PC-to-Phone calls in the following topology (for signalling): SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> PSTN The RTP audio packets would go direct through Softphone to gateway. Does someone have a configuration file example of Asterisk? Thanks for your help, Helaine
Asterisk .
2004-Sep-30 04:45 UTC
[Asterisk-Users] HELP: Asterisk - SIP to H.323 translation
Hello, --- UTRINI@embratel.com.br wrote: <snip>> Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? > I want to implement PC-to-Phone calls in the following topology (for > signalling): > SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> > PSTN > The RTP audio packets would go direct through Softphone to gateway.You can use Asterisk as a SIP-H323 translator. It is not a SIP proxy, but a PBX having a SIP channel. It also is a SIP UAS/Registrar. I dont think when it is used as a translator, RTP packets will go directly from softphone to gateway, since there are 2 different protocols involved. Asterisk will force the RTP packets to go through it.> > Helaine >Regards, Girish _______________________________ Do you Yahoo!? Express yourself with Y! Messenger! Free. Download now. http://messenger.yahoo.com
Hi: I have a question. What is a sip proxy and what is the benefit of having one with Asterisk? I am well aware that we have a sip channel in Asterisk and that we have SIP registration. I am not sure why you would need a SIP server. Second question, with Asterisk are you able to do video on VOIP video phones? Thanks Steve steve@17q.com -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Asterisk . Sent: Thursday, September 30, 2004 4:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HELP: Asterisk - SIP to H.323 translation Hello, --- UTRINI@embratel.com.br wrote: <snip>> Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? > I want to implement PC-to-Phone calls in the following topology (for > signalling): > SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> > PSTN > The RTP audio packets would go direct through Softphone to gateway.You can use Asterisk as a SIP-H323 translator. It is not a SIP proxy, but a PBX having a SIP channel. It also is a SIP UAS/Registrar. I dont think when it is used as a translator, RTP packets will go directly from softphone to gateway, since there are 2 different protocols involved. Asterisk will force the RTP packets to go through it.> > Helaine >Regards, Girish _______________________________ Do you Yahoo!? Express yourself with Y! Messenger! Free. Download now. http://messenger.yahoo.com _______________________________________________ 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
Alex Barnes
2004-Oct-01 03:40 UTC
[Asterisk-Users] HELP: Asterisk - SIP to H.323 translation
Hi, First Question: This is rather a difficult question to answer. Many people say that it allows * to scale better for SIP. For example assuming your SIP proxy is stateful the proxy will handle all retransmissions / redirections / register lookups / call logging / ????, all of which would be hidden from the *. Multiple registrations: * doesn't support multiple registrations. A registrar proxy will be able to do this, allowing such things as forking (sequential / parrellel / combinations of both). Grey Areas (Functionality that crosses over between * and a proxy) ------------------------------------------------------------------- CPL: Any proxy worth its salt will have a CPL engine built in allowing some pretty powerful scripting of incoming / outgoing calls on a individual user basis. Application Servers: A proxy may also be an application server. Which could be used to make any number of technologies available. For example Ubiquity make a HA SIP App Server that can do pretty much anything including RMI client/server apps, SOAP, link with Web Application Servers (Websphere / Weblogic), Java Eventlets that let you write your own interfaces using our SIP SDK's for call handling. HOWEVER!!!!!!!!!!! ---------------------- I don't think that Asterisk is quite ready to support all live deployment scenarios that include a 3rd party SIP proxy. One problem I ran into was Asterisk does not handle looped back calls. For example a call comes in over PSTN to Asterisk, Asterisk forwards to your SIP registrar proxy, Registrar does a lookup on the SIP address and finds that the user is register'd to an analogue phone. If the SIP registrar redirected using a 3xx response the * will play along happily, but if the proxy wishes to stay in the loop (maybe you have a billing application running on it) it would add a Record-Route header to the SIP request , to say it wishes to receive all subsequent messages for this call, and then proxy back to the *. The * will ignore this INVITE totally. If the user had been registered to a proper SIP end point then the loop back wouldn't have happened and this works a treat. Second question: Yes but looks like support isn't great / the community hasn't really investigated this much (http://www.voip-info.org/wiki-Asterisk+video) I plan to evaluate video over SIP next week / this weekend so if you get anywhere with this please let me know. I will of course do likewise. I hope this helps you. Alex -----Original Message----- From: steve [mailto:steve@17q.com] Sent: 01 October 2004 11:14 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] HELP: Asterisk - SIP to H.323 translation Hi: I have a question. What is a sip proxy and what is the benefit of having one with Asterisk? I am well aware that we have a sip channel in Asterisk and that we have SIP registration. I am not sure why you would need a SIP server. Second question, with Asterisk are you able to do video on VOIP video phones? Thanks Steve steve@17q.com -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Asterisk . Sent: Thursday, September 30, 2004 4:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HELP: Asterisk - SIP to H.323 translation Hello, --- UTRINI@embratel.com.br wrote: <snip>> Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I > want to implement PC-to-Phone calls in the following topology (for > signalling): > SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 > ---> PSTN The RTP audio packets would go direct through Softphone to > gateway.You can use Asterisk as a SIP-H323 translator. It is not a SIP proxy, but a PBX having a SIP channel. It also is a SIP UAS/Registrar. I dont think when it is used as a translator, RTP packets will go directly from softphone to gateway, since there are 2 different protocols involved. Asterisk will force the RTP packets to go through it.> > Helaine >Regards, Girish _______________________________ Do you Yahoo!? Express yourself with Y! Messenger! Free. Download now. http://messenger.yahoo.com _______________________________________________ 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 _______________________________________________ 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 Dear Friends of Ubiquity Software: As you may have noticed, Ubiquity Software began using the web domain ubiquity.com earlier this year in addition to the previously established ubiquity.net for our website and email communications to you. However, since that time, a dispute has emerged with respect to actual ownership of the ubiquity.com domain. As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net <http://www.ubiquity.net/> and via email at @ubiquity.net. However, we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com <http://www.ubiquitysoftware.com/> for web and @ubiquitysoftware.com for email communications. Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. Thank you. Regards, Ubiquity Software www.ubiquitysoftware.com <http://www.ubiquitysoftware.com/> info@ubiquitysoftware.com