search for: greyman

Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into

Do anyone has an idea about an open source SIP API written in C# that can communicate with Asterisk, to call out? Regards, Sanjay.

Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '2e4a02807750b7024d5ff09c668fa0f7 at 10.0.0.2'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '0755ad8f40b9d09d491b635e70bb8905 at

...es.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong detstination. Regards, Greyman.

...dge changes or the bridge is hungup a CDR(s) would be generated. The implementation would undoubtedly be a lot more difficult but if the design could be agreed upon at least those of us in between a rock and a hard place on this could decide to sponsor development, offer a bounty etc. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail

...but then email in complaining about duplicate billing to try and get one of the CDR's refunded. On a separate note does anyone know how to block transfers on a SIP channel? I can block REFER requests from my SIP Proxy but I have to support some transfers so that's not an option. Regards, Greyman. ----- Original Message ---- From: Steve Murphy <murf at digium.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Sent: Monday, 15 October, 2007 6:22:45 PM Subject: Re: [asterisk-users] CDR Sorry! I've gotten some complaints on...

...es.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong detstination. Regards, Greyman.

each Asterisk channel should be handled after by it's own unique thread and save the need for any locking on the channel data structures in the first place. Regards, Greyman.

...rs at the drop of a hat to save costs and cope with peak and quite times, with dedicated servers you're stuff with 12 or 24 month contracts for the number of servers you'd need under maximum load. And then of course the major factor for both is what the call quality will be like. Regards, Greyman.

...ase there is a Record-Route header in the response so the ACK request should be being sent to that address. Perhaps your firewall is not correctly mangling that to allow the request to find its way back to your Asterisk server. Record-Route: <sip:216.82.224.202;lr;ftag=as3ed791f3> Regards, Greyman.

...be billed for incoming calls and that can be forwarded out to billable destinations then I want a CDR for both ends of the bridge. In your first blind transfer example what if the initial incoming call to A is billable? I can't see any easy way to get the duration of that call leg. Regards, Greyman.

...nformation is used when a SIP call is authenticated by IP address? I'd guess it's the socket the call was received on but was wondering (or more correctly hoping) it would be done on the address in the bottom Via header. That way IP authentication could work through a SIP proxy. Regards, Greyman. Sick of deleting your inbox? Yahoo!7 Mail has free unlimited storage. http://au.docs.yahoo.com/mail/unlimitedstorage.html

Hello, I'm having troubles while using the "Goto" function in a realtime extension. Here is the error message : -- Executing Goto("SIP/siemens1-081f56b0", "script_13_0|s|1") -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context

...to do with the e164.org ENUM site monitors this list could you check your signup page as the Captcha's (the test to see if you are human) fails for both the text and audio tests every time. I'd post a message on the e164.org forums but the signup page there has the test missing altogether. Greyman.

Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Thanks in advance, regards, Rob.

Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool> or can u just sniff? regards, ron

Hi, I am in a weird situation where a variable seemed ignored, but not always. That variable is __TRANSFER_CONTEXT. Basically, I have a phone registered with asterisk. It's context is "internal". Outgoing calls go through that context (all good). When I get an incoming call which I want transferred, I don't want it to go through the context "internal" but

Hello, We've had this problem happen twice with retail customers already and still have no solution. Basically there are times when customers can't get any ring at all. It happens that they call our switch and even though we are receiving ring from the carrier they hear no ring. We have even put a fake-ring(with Rr) back at their request and they are unable to get this ring either. The

Hi folks I'm not sure what I am missing but I cannot find a predefined channel variable to identify the SIP peer/user which has initiated a call and established the channel. The one option is to extract it from the CHANNEL variable, but that is fraught with difficulties. Is there another variable I don't know about or another way to do this? Thanks in advance! Richard -- Richard

Hi All, I am using asterisk 1.4.21 with iaxmodem and hylafax which is sending fax from my system with zap device. I am facing a problem that some times my asterisk CLI got freeze and i am not able to get any information from asterisk. I need to restart the asterisk compulsory to work it again. And because of this my iaxmodems are also getting time out from asterisk. Please provide some help