search for: rfc2543

Hi, Does anyone know the difference between RFC2543 and RFC3261? They are both SIP, but apparently incompatible. We are testing some hardware devices that support RFC3261 and it appears Asterisk is supporting RFC2543 and not completely compatible with RFC3261 (see below). Does anyone know how to configure Asterisk so it can do what is missing?...

...y the call will go through and it will successfully re-register itself without needing a restart. What can this be? Surely Polycom is re-registering every 3600 before Asterisk times it out. But Asterisk is just refusing it. By the way, anyone know whether Asterisk is geared towards RFC3261 or RFC2543? I know Asterisk is not a fully SIP Proxy but lets say if a SIP PSTNGW or a SIP phone is designed under the spec 2543 as suppose to 3261, will it work better or the same with Asterisk? David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pi...

[Resent due to non-descriptive subject line.] Hi folks When connecting two SIP users, is there any way to stop Asterisk from sending SIP 183 Session Progress messages, either globally or per-peer? Scenario as follows: Call from UA1 to Asterisk (UA2) to UA3. UA3 sends RTP before SIP OK to Asterisk (UA2). Asterisk (UA2) detects early audio from UA3 and sends 183 Session Progress with SDP to

I was previously having an issue with a Polycom phone and Polycom support said that Asterisk didn't play well with Polycom firmware versions 1.6.7 and newer due to SIP compatibility issues. I believe I heard a lot of things were fixed\adjusted in 1.4 and was wondering if anyone has had success with Asterisk 1.4 and the latest Polycom firmware releases. -------------- next part

Hi, My setup is : PSTN --------- ISTP Network ----------- Router ------------- Asterisk ---------- SIP Phones Phones are located in the same location. I'm thinking about installing new phones in other locations (small agency, home workers), registering those phones to the same Asterisk server. As every location has DSL access, I think I should have those phones directly exchanging RTP data

For someone that's network-aware, but hasn't sat down and plowed through umpteen SIP-related RFC's and memorized the standards, is there a good primer on troubleshooting SIP issues? I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk and my Sipura 942's, for instance... Not sure what these are... perhaps the qualify keepalives? In which case, I guess

Andrew, I don't think it's a Polycom issue. We took Asterisk out of the picture and had our Polycom phones communicate directly with an Audiocodes PSTN gateway. Unlike Asterisk, the audiocodes do not send 180 Ringing before sending 183 Session Progress, and the polycom's play the correct tones in this case. We WANT Asterisk to send progress tones in band. In our case it IS needed.

Hello all, One of our clients reported that they are experiencing echo in SIP calls (even on internal ones). What do you think could be causing echo in internal SIP calls? We're using Polycom telephones, do you think they could be causing it? Thanks, Alex

Hi..... I am new to this software, and I want to implement a client (SIP or IAX) with PHP or at least to pass the main functions (connection,call, transfer, hangup, call id etc) to a CRM. Does anyone know if I could achive a project like that with AGI ? Any example using AGI with PHP ? Do I have all the functionality with AGI ? What about call id ? What is depend on ? (As I know * does not