similar to: Default From and Contact header domain

Hi, I have noticed that when I am calling from my Snom-phone to another Snom-phone through Asterisk, the SIP-message's Contact -header could be sometimes empty and for example other Snom get no BYE-message. Here is example of that kind of message: 10 headers, 0 lines Sending to 192.168.0.32 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

Hi, I'm observing wrong From/Contact header values. When I try to set CallerID(num) it has no effect in the From and Contact Headers, and these values are the same as the dialed number. SIP Peers are defined using asterisk realtime. If I define the SIP Peers using sip.conf then From/Contact header value are correct. extentions.conf [test] exten=> 1000, 1,NoOp() same=>

Hi list! I'm trying to get a DID routed to me and the provider seems to have an unusual setup. Or maybe not? From looking at their SIP header they are using "BroadWorks". The problem: they're sending their SIP invite from port 36252. My Asterisk 10.7.1 is answering to that port 36252 but their BroadWorks thingie is not listening on that port, but instead on port 5060. So

In an attempt to connect our Asterisk 1.6 phone system with another phone system called "Broadsmart", they gave me credentials to register to. Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365) watermelon*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time {broadsmart_ip}:5060 N

Hi all, I'm not sure wether it is a bug or not, so I'm asking for your opinion before submitting it to the bugtracker. The problem: I use asterisk with in sip.conf a non standard bind port of 5070 set. Now when asterisk sends out an Invite message to my sip proxy, the contact header in de request is something like: Contact: <sip:12329123 at 123.123.123.123> The call succeeds and

Asterisk: 1.4.17 with sip realtime Openser 1.3.x Hi, I had this setup working fine until I try putting OpenSER in the picture as a proxy. Unauthenticated calls go to a PRI based app via a ZAP channel, calls to sip users get send to them etc. Now with a proxy in the picture asterisk asks the incoming calls for authentication "407 Proxy Authentication Required". It seems that the

I'm migrating an application running on a fairly old 1.4 (or 1.2?) version of Asterisk to some boxes running 1.6.0.27 The application takes an inbound INVITE like: mumble-fratz-sip%3Afoo%40bar.com at asteriskbox.abc.com:5062 The older version of asterisk replies with a 200 OK and a Contact: header that looks like: Contact: <sip:mumble-fratz-sip%3Afoo%40bar.com at

I'm a bit confused about how * constructs the RURI when it generates a BYE. For the situation where * send the initial INVITE it constructs the RURI for the BYE from the contact header of the 200 OK response which is well and good. However when * receives the initial INVITE it does not use the contact header contained within to construct the BYE's RURI but constructs it from scratch. This

Hi All, I'd just like to verify what the correct operation of the timeout parameter is for the dial application. I'm not sure if I've encountered a bug or a configuration issue. When a sip phone is not responding to invites on an outbound call, the dial application still waits the duration of timeout before continuing with dialplan execution. I was under the impression that app_dial

Hello! I encounter the following problem (asterisk 11 and 13) with Teconisy as trunk provider with enabled qualify and default t1min (100ms): Teconisy most often doesn't answer the first invite before asterisk default t1min ended. Therefore asterisk sends one more invite. This second invite is answered by Teconisy with status 486 - Request terminated - Channel limit exceeded. (The second

Antonio G?mez Soto wrote: > > So basically, the 'contact' in the AOR is just an ip address (or > 'dynamic', in which case it accepts > incoming registrations). A contact is a SIP term, it's a way of getting to something. (IP address+port) > So what happens if one endpoint has multiple AOR's which are registered > from different ip addresses. > And

Hi, I'm seeing a very strange error when dealing with Diversions. If a call setup to a number comes to an Asterisk server, that server sends a request to a third proxy, that proxy sends the call back with a Diversion flag, Asterisk complains about the host not existing (and the host is the number). Here's the output from the Asterisk CLI with SIP debugging enabled: <--- SIP read from

Well, not so solved unfortunately... Now I am back to where I have the situation the Asterisk sends out 183 Media Progress from one interface, containing a Contact Header with the local IP of the other interface breaking audio. Is there any way to completely bind all IP Addresses within headers sent out one interface to the IP of that interface? Mit freundlichen Grüssen -Benoît Panizzon- -- I

Thanks for responding, On Sun, Jan 4, 2015 at 5:45 PM, George Joseph <george.joseph at fairview5.com> wrote: > On Sun, Jan 4, 2015 at 3:29 PM, Antonio G?mez Soto < > antonio.gomez.soto at gmail.com> wrote: > >> Hello, >> >> I am slightly confused by the difference between chan_sip and pjsip. >> Especially the new (to me) objects aor and contact.

Hi I am currently using 1.2.x and 1.4.x behind OpenSER. One of the things I do on INVITES is to re-authenticate the user from OpenSER. Then when the INVITE gets passed to Asterisk I capture the AUTH to a variable in the dialplan and pass to an AGI script. I am now trying to set the same thing up in 1.6 However because the argument delimter in 1.6 has changed from pipe to comma this breaks as the

Hello, I noticed a strange thing while testing my Asterisk-Kamailio Realtime setup. In an INVITE the From and To headers contain the same number when calling through a Realtime integration setup. This happens when the INVITE leaves Asterisk. Can you guys tell me what might be causing this? I have 660 at testers.com as a websocket client and 700 at testers.com (caller) using a Zoiper client (db

Hello everyone, Once again I'm trying to set up our Samba server to operate as a PDC, now that I've verified that I have a clean install of the Samba 3 software. However there are some difficulties with this. On my client machine when I try to set the domain name in the system control panel, I get an error message saying that the domain could not be contacted. I've set up my

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden

That sounds like asterisk was working 100% correctly. If you receive an INVITE from an unknown IP address, then it should fail. Unless you want to allow anonymous, which is genearlly a very bad idea. If you are registering to IP X, but the provider may be transmitting invites from any number of other IP addresses, then you need a list of IP addresses, and have a trunk configuration set up for