similar to: Missing Request URI in SIP message

Hi everyone, Sorry if I'm repeating the e-mail, but I'm having problems with the list. I'm currently trying to enable call transfer to different domains in asterisk box (Asterisk 1.2.13 running on Debian etch). I have a configuration that requires me to transfer call to separate domains like ext at 10.10.10.10:5050. My calls come from a R2 channels in a board installed in the machine.

Hullo :) I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using CAPI for ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from sipgate.co.uk to any other extension. My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to blind transfer, simply dial the number you want to transfer to, and press 'FWD'... This is what

Hi, I am running a Asterisk 1.6 box in our Data Centre, and have a number of users connecting to that box, as their PBX. Calls in and out work fine, as does voicemail. The PBX at the Data Centre has an External IP, Nat?d to it by the firewall, and the relevant ports are open. The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this

I am having issues with transfers (SIP/REFER) using Asterisk 1.6. You will find the SIP debug below. There are three phones in this setup. 5253 and 5258 are Aastra 53i telephones, 101 is a standard phone connected through an Audiocodes gateway. All phones are registered in context "phones" and are set to not allow reinvites. All phones can dial each other directly. The dialplan

Hey all, Alright. So we decided to not go with Avaya for our next PBX and we are now full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP gateway and call center and Lync is our internal UC and IP-PBX server. I've already got Asterisk tied with our Nortel/Merridian Option 11 with QSig and all is beautiful (except for the Opt11 not receiving names from * but

Hi, I am using Asterisk 1.6.2.11 to test 3rd party Voice Message server (VMS), currently when VMS send NOTIFY message (standalone NOTIFY, no previous SUBSCRIBE for the dialog for SIP), asterisk responded with 489 Bad Event, in the debug log it indicates as the following: [Feb 4 13:27:06] DEBUG[8353] chan_sip.c: Invalid SIP message - rejected , no callid, len 771 I have googled around

Hi We've been running asterisk 1.4.17 (deb package) in a production environment for some while now and are finally taken the plunge to update to 1.4.38 (Ubuntu servers). All of this is using the RealTime Architecture I have upgraded the asterisk version in one of our test environments and blind transferring seems to have suddenly stopped working. It was working fine under 1.4.17 So, call

I am having a problem with SJphone registration, having read the list and wathced it for a while for similar problems. I just can't seem to figure out the problem. I tryed to follow a tutorial from http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone, but in SJphone (SIP tab), I can't find the following setting. Use local outbound proxy - checked. Proxy IP Address:

hello, Although my previous posts in this forum have not received satisfying answers, here is another question from me. my question is can i use ChanIsAvail function to get the status of a user in the format SPI/user-id if i provide user in sip uri like this ChanIsAvail(SIP/user at 153.18.x.x:5062) calling user with this sip uri works fine. I once tried but status returned was "unknow

Hi, I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ as target user which is registered. Asterisk server IP: 70.118.x.x calling user IP: 117.58.x.x called user IP: 117.58.x.x:5062 First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I dialed the registered user via

I don't think it's the snom, (the break key is set to "off") the "#" key is not being interpereted by the PBX as an attempt to initiate a transfer. Is this an error in my extensions.conf? Brian > >Message: 4 >Date: Wed, 15 Dec 2004 19:39:39 -0500 >From: Info <info@idatasys.com> >Subject: Re: [Asterisk-Users] Help with transferring a second call

Hello List- I'm having a problem getting snom 190 phones to transfer a call to another local extension. Here is the scenario: A call (call1) comes in from the PSTN to (exten1). (via pri, if that matters) Another call (call2) comes in to (exten1). (call1) is put on hold while (call2) is answered. (call2) is then transferred to (exten2) using the "Xfer" button on the snom phone. This

Hi, I`ve just made a leap from * 1.2.7 to 1.4.19. It took a while to fix all the deprecated stuff, but everything seems to be working fine now, except for a little tiny thing. I lost all color in my CLI, which makes it harder to debug. Is there something that needs doing? I didn't explicitely disable colorization from the command line, and I did try using nocolor=no in the config files.

Hello, I''m getting an URI::InvalidURIError in my controller when I use german characters in the uri. Google accepts them but open-uri not. How to encode them? That''s my controller simplified: def index require ''open-uri'' address = "Bürgerstraße+68+01127+Dresden" api_key = "my api key" json =

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

Is there any reason for Mongrel to be stripping request.uri as described in this bug report: http://rubyforge.org/tracker/index.php?func=detail&aid=5103&group_id=1306&atid=5145 I was planning on using mongrel to deploy my app but my login code depends on the full request.uri to redirect back after login. Greetings, Pedro.

Is there any particular reason why mongrel rails hard codes ''/'' as the URI it registers as the URI node? If not would there be any objection to a patch that allows you to specify it as an option? -- Neil Wilson (neil at aldur.co.uk) -------------- next part -------------- An HTML attachment was scrubbed... URL:

URI.escape() is supposed to be able to take a second parameter listing unsafe characters in the URI. This may be a regexp or string. If a string, it''s supposed to represent a character set listing all unsafe characters. An example given in the core documentation at: http://www.ruby-doc.org/stdlib/libdoc/uri/rdoc/classes/URI/Escape.html#M008992 ...is: p URI.escape("@?@!",

Hey All, I''ve posted the first release of FakeWeb, a little library to help with all your http client testing needs. This helper makes it trivial to setup an idempodent environment for you to test any web service requests in your applications. Available on RubyForge, http://rubyforge.org/projects/fakeweb/ == Overview * Force Net::HTTP (and any dependent libraries, e.g. open-uri) to

Sorry if the answer has been posted but the search terms (like URI) are so generic I get tons of useless results. I have a controller (in this case "forum_controller") so if I want to read a forum topic the URI is "/forum/topic/1". As of right now for adding a new topic my URI is setup as "/forum/new_topic". I''d really like it to be something like