search for: rfc3261

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? 3. The probl...

...uot; in the SIP > INVITE. We don't see that field poplated in this INVITE. What is the > originating gateway? What device is sending the call to the 827? We > should be seeing "remote-party-id" in the INVITE. The string "remote-party-id" does not even appear in RFC3261. A little bit of googling reveals it seems to be something "up in the air" and there is no RFC which seems to reference it. Further, in its absense (as I have learned), RFC3261 states of the "From" header: The From header field indicates the logical identity of the initiat...

Hi all, In the INVITE from my SIP provider to Asterisk i can see that the Allow Header includes an INFO Method, but Asterisk replies a 200 OK with an Allow Header without INFO Method. But in the RFC3261 (20.5) you can read: "All methods, including ACK and CANCEL, understood by the UA MUST be included in the list of methods in the Allow header field, when present. " My SIP provider seems to refuse to send SIP INFO DTMF and releases the call, because in 200 OK from * there is no INFO Met...

...sk seems to only accept these when there is also a Session Description (SDP) included in the message. I also verified this by looking at the code. However for a project we are working with a trunk to a third party system (Alcatel) and they are insisting that this behavior is non-compliant with RFC3261 (SIP). So can someone please tell me the reason, why Asterisk does not support 183 messages without SDP as this would really help me finding arguments in this situation. So far Alcatel just tells us that this is not SIP-compliant and that we have to change things on the Asterisk side, but I'm...

I've made a subtle but important SIP change as part of bug #155. According to Mediatrix, the URI in "Contact" should be used as the URI in the top part of the SIP request when sending follow up messages, and they're allowed to use the same IP for from and to. I made the change to support that but I want to be sure we didn't break anything else, so if it did, let me know, on

...sk/phones to try. It works like this: ./Options.py eth0 192.168.1.35(ip of the UA) Do you know why asterisk send me a 404 message and how can I ask him to answer correctly? Thanks a lot for your help, Thomas Ps: for more information about the OPTION method: http://www.ietf.org/rfc/rfc3261.txt ( page 66 and 67 ) Response of a Thomson: /----------------------------------------------------------------------- ----------------------- | SIP/2.0 200 OK | Via: SIP/2.0/UDP 212.147.65.204:5060 | From: "252"<sip:252@192.168.1.35:5060> |...

...t X.YYY.32.10 :5060",response="cb6306569b3047ac35064dcb5aee6db4" X-Enswitch-RURI: sip:8009499014 at X.YYY.32.10:5060 X-Enswitch-Source: X.YYY.33.178:5060 The only problem I see with this INVITE is the VIAs are not right after the INVITE line... although in https://www.ietf.org/rfc/rfc3261.txt, it explicitly states the the order of the headers is not a requirement, it seems Asterisk does make it one... "The relative order of header fields with different field names is not significant. However, it is RECOMMENDED that header fields which are needed for proxy processing (Vi...

...thing else (see [1]) [1] https://www.voip-info.org/p-asserted-identity-and-remote-party-id-header/ > Recommended? who knows? Implementations are all over the place. I've > always thought of the From header as identifying the user agent making the > request which kinda agrees with RFC3261. The PAI header should contain > the identity of the original caller. > > >> >> 2. When Bob forwards to Cory a call coming from Alice, would expect >> Diversion/History-Info header to include Alice's number ? >> > > No. The diversion header shows who th...

Hi, After a long discussion with a friend, I would like to ask here: 1.According SIP RFCs, is possible/recommended to have different values in >From and P-Asserted-Id fields ? For instance, From field showing 123456789 and P-Asserted-Id showing 987654321 (beside privacy considerations) ? 2. When Bob forwards to Cory a call coming from Alice, would expect Diversion/History-Info header to

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the

Hello, is there anywhere an overview of SIP error codes and under which condition they are reported by Asterisk? There are general definitions for SIP error codes, but they are quite general and it's Asterisk that actually checks what's wrong and then reports an error. Now, currently I could check the source code to get more informations what could have caused the error, but that's

Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even

...en allow_subscribe = yes auth = Biv_Exit outbound_auth = Biv_Sortie aors = Biv_Sortie Question how can I solve this character problem "@"? 2) resolution of the orange-obs.fr DNS.  I am attaching an extract from the documentation that Orange issued in 2015 SIP/Internet is described in RFC3261 and following. THE SIP/IMS is described by 3GPP standards. It's not the same SIP. In the Internet world, VoIP machines route SIP messages to the IP addresses of the FQDNs of the SIP URIs (VoIP domain). In the 3GPP world, SIP messages are routed to an I/P-CSCF (depending on whether we are in int...

FYI: Asterisk puts URIs in messages which violates the SIP spec and can't be accepted by URI parsers: username includes a whitespace. See for example the From header field. Attached is example of an incorrect message and related parts of RFC3261 specification. (Who doesn't want to dig into parser details may want to realize that whitespaces are used as uri delimitors in first request line and can't thus be a uri part.) I would recommend that the stack generally validates URIs for such glitches and uses other word for "no...

Can anyone give me a pointer to descriptions of the status codes my Grandstream phone displays? I've looked on Google but can't find a definitive listing of SIP codes. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * sjsobol@JustThe.net

...and usually 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.di...

Hi all, i need to study the SIP protocol. can anybody tell me about any ebook which could halp me understand the sip protocol, architecture, and how to read and understand the sip signalling when i use "sip debug" in asterisk? -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL:

[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

Hi, I have some problem with my provider. While the sip registration is successful, i intermittently encounter problem in dialing out. I receive 603 Declined error in my Sjphone client. The asterisk log shows line is busy/congestion. Appreciate if help or direction can be provided. Thanks. CK -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am going to change my Digest realm to match my DNS SVR record. I dug through the code in chan_sip.c and on line 2748 I found it hard coded <frown> : snprintf(tmp, sizeof(tmp), "Digest realm=\"asterisk\", nonce=\"%s\"", r\anddata); I'm going to change this to : snprintf(tmp, sizeof(tmp), "Digest realm=\"isdn.net\",