similar to: doublehash patch doesn't work in asterisk 0.7.1

Hello, I recently tried an upgrade of CVS on my test server today and found that the res/res_parking.c file is completely gone. This is where I had to go into the code every time I do an upgrade and change the code to allow for doublehash transfers instead of single hash transfers: That means that you need to hit the pound key twice to initiate a transfer instead of once. Because of our inbound

Here's a patch that changes the behaviour of # transfers in asterisk. A single # is transferred to the remote phone/system. Two # in quick succession will trigger a transfer. This is very useful for users who have basic analogue phones connected to an ATA 186. For example, when calling a remote conference or IVR system you often want a single # to be sent to the remote system - not to

Hello, We've released another update to our Asterisk GUI Client suite: http://astguiclient.sf.net/ Screen shots: http://astguiclient.sourceforge.net/screenshots.html The client suite runs on both Windows and UNIX and includes a dialer (the suite is not an asterisk configuration tool) In addition to the usual bug fixes, this is mostly an update for the VICIDIAL dialer application.

Hello, Good news on the Polycom front for those that are interested. It looks like we may get a dedicated Engineer for Polycom/Asterisk!!! Happy Day! Here's the message I got tonight: Matt: I heard back from our VP of Engineering- she is prepared to have an individual dedicated to working on the Digium- Asterisk project. Can we discuss again Friday or mid next week? Scott Willard

Hello, Just confirmed this on my end, because of the massive changes that have been made to callerID handling in asterisk 1.0.5 many of the features of the astGUIclient suite will not work on this new version. The latest stable version recommended is Asterisk 1.0.3. We will work on trying to find ways around the new callerID rules that the asterisk developers have put in place and hope to have

Has anyone on the list been able to locate and try out the 1.1.0 firmware? It was released in November, but I have yet to get my hands on it. The Polycom site has way more docs online, but the link to the firmware only brings up the release notes. -sb

Is anyone else experiencing problems with Transfer via # and the 'T' or 't' flags passed to Dial()? I've tried both the latest CVS and 0.7.1 tarball. If I dial in from a pstn line and then choose an extension that dials a SIP phone with "Ttm" flags, when I press # on the SIP phone, the pstn caller hears the "Transfer" and the SIP phone gets the music on

Hi Ilia Thanks again for the reply. On Wed, 28 Oct 2020 at 14:59, Ilia Mirkin <imirkin at alum.mit.edu> wrote: > > On Wed, Oct 28, 2020 at 10:20 AM Dave Stevenson > <dave.stevenson at raspberrypi.com> wrote: > > > > Hi Ilia > > > > Thanks for taking the time to reply. > > > > On Wed, 28 Oct 2020 at 14:10, Ilia Mirkin <imirkin at

Hello. Was just curious to know if anyone is working on a graphical interface to Asterisk using X windows, or something else similar. Thanks! David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/2718fd7e/attachment.htm

Greetings, Hoping someone can shed some light on this. I've been searching for over a week and cannot find information on how Samba generates SID's from Unix UID's and GID's. I keep running into situations where after adding a new user to my CentOS server all other users are suddenly prevented from accessing shares that have a group ACL assigned. I finally figured out that

I have a couple of live T1s sitting around and they are not ISDN(like most of the people that are using Asterisk seem to be using), they are regular old 24 channel, robbed-bit, E&M wink start, D4AMI T1 circuits. Can I get these T1s to work with a T100P Digium card and asterisk? Searching through the lists and the documentation I haven't seen any examples of how to configure this kind

I have a Cisco 7960 & IpDialogs that I am not able to use the transfer button on it. What happens is that it puts the call on hold and then it gives you a dial tone. You can dial but it will not transfer the call. What we are trying to do is transfer to extension 700 for parking so another person can pick up the line. We can not use the # key to do this due to we have several IVR's

Hello, I have received yet another new phone today, the ClipComm 101 (http://www.clipcomm.co.kr/eng/e_product/e_product_voip_ip_phone.html) I bought it for $165 directly from the Korean Manufacturer(No US distributer yet). Here are the features: - Built-in NAT functionality, you can switch from Hub to Nat, great for home DSL/Cable users - This includes some limited port forwarding

Hi Ilia Thanks for taking the time to reply. On Wed, 28 Oct 2020 at 14:10, Ilia Mirkin <imirkin at alum.mit.edu> wrote: > > The most common issue on arm is that the pci memory window is too narrow to allocate all the BARs. Can you see if there are messages in the kernel to that effect? All the BAR allocations seem to succeed except for the IO one. AIUI I/O is deprecated, but is it

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer

Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, I am having trouble listening to GSM files created by Asterisk using a browser. I am noticing that some of my users succeed in listening to them and some others don't. I guess it is something of a codec problem that does not seem to be installed on all machines (though they are all WinXP). Anybody knows what one should do to listen to GSM files? I send files through the

Hello, In the last week I've been getting a lockup about every 2 days. during the lockup the people that are on the phone can keep talking, but noone can initiate any kind of call internal or external. I went into the manager interface and tried a Action: Hangup and Manager gave me a Success message back only to see that the Zap channel was still active in the "show channels"

Hello all, Has anyone used the SIP version of this phone with Asterisk? I see Polycom has a H.323 and MGCP version also, does anyone know if you flash the phone to swith protocols? Thanks in advance for the info. Ed -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031029/a9377305/attachment.htm