similar to: send # with transfer enabled

Evening All, I was wondering how I would go about enabling the usual # + ext transfer on a call requiring user input followed by a #. For example, when I say ring a bank, they ask me for my account number, I key it in and then press the # key. Of course this doesn't work if I have the "T" option in the dial command, as asterisk tries to transfer the call. How to overcome this

Can anyone tell me how I can implement the features added in the following link for call transfer? The authors seem to feel they are finished but it doesn't appear to have been integrated into what everyone can download. It is referred to as a patch but I don't understand how it could be applied. Here is the link: http://bugs.digium.com/bug_view_page.php?bug_id=0002010 I guess I just

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

Has anyone been able to change the way that asterisk performs transfers? Instead of using the # key, I would like to due something else, such as flash. # is just causing too many problems with transfers and menus when calling out.

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

Hello everyone. I am having problems getting speex support. It seems * is not loading speex. When i did a make in the codecs sub dir, the following error pops up when making speex: codec_speex.c:34:19: speex.h: No such file or directory is this file missing in the cvs as i just removed the whole * dir and did a new checkout and still seem to get this error, or do i need to get/install

Hello all, I recently noticed the realTime effort and must say it is a nice idea! I would appreciate any help to get it running .. I downloaded the code & patches and succefully patched my asterisk (CVS-HEAD-11/29/04-12). - created a DB called asterisk, and a table sip using the schema supplied at http://bugs.digium.com/bug_view_page.php?bug_id=0002613. - entered an entry: insert into

I just ordered the Asterisk Developers Lite kit. My environment will be the RH9 Linux server and a Windows workstation with Samba. I also of course have analog lines and DSL. I am interested in SIP development. I already downloaded the Asterisk software. What else should I download. Is there a doc that basically tells you the steps to install Asterisk and get it up and running? I would like a

Hi there, I have serious problems with MGCP and Swissvoice ip10s, and it appears that recent CVS also introduced trouble for other MGCP users. Please check and add comments in the bugtracker so that we can get a clearer picture - thanks! Also comment if things are working fine for you. http://bugs.digium.com/bug_view_page.php?bug_id=0001542

Hello list, I was looking for a way to implement non-blind call transfers with *, i.e. the usual behaviour of most PBXs when pressing the flash button: - A and B are talking - A pushes flash - A is free to compose a new number - B hears music on hold - A's call is answered by C - A hangs up - B and C are in conversation As much as I can understand, * only supports blind transfers, where if

Can someone take a look, tell me if this is a bug, a possible resources issue, or my own damn fault? http://bugs.digium.com/bug_view_page.php?bug_id=0001381 Thanks, Derek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040407/f8f4d79b/attachment.htm

Hi Is this patch (http://bugs.digium.com/bug_view_page.php?bug_id=0001719) the best/only way to get callerid working in the UK with a tdm400p? I thought I'd seen a patch that'd gone into cvs, but maybe I was just imagining things ;) Should this patch work against current cvs? Of the 3 files 2 are .patch and one is .diff - what's the difference between them, and how should I

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

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem,

I'm using a modified "default config" file for extensions.conf, the one that uses macro-stdexten to handle the stations. We use a TDM30 card for our stations. When a call that has been rung in using that macro transfers the call things work just fine as far as the "other" instrument ringing. But once the ring timeout has expired, the call then drops into the *original

I updated from CVS yesterday and today and still have the problem. IaxComm cannot transfer the call when it's an outgoing call. ('outgoing' is from the dial plan point of view). details : First I call the IaxComm phone and accept the call. Then I'm not able to transfer it from the IaxComm phone. If the call is an incoming call it works fine. details : First I call a phone

Any body used patch, http://bugs.digium.com/bug_view_page.php?bug_id=0001719 to get the callerid for BT Line. I applied the patch successfully but could not get it to work. Any help. Here are the logs: -- Starting simple switch on 'Zap/1-1' Jun 17 18:22:31 NOTICE[426000]: chan_zap.c:4811 ss_thread: Got event 2 (Ring/Answered)... Jun 17 18:22:34 NOTICE[426000]: chan_zap.c:4811

Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=0000254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using

I am trying to find the app ChanSpy for Asterisk v1.0.7. I have tried looking on VOIP-info.org's ChanSpy page (http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and also referred to the link regarding bug 3836 (http://bugs.digium.com/bug_view_page.php?bug_id=0003836). I downloaded the attachments and tried to use the patch and compile the source. However, it seems that

Hi Everyone! I live in Sweden and can not get CallerID to work on analog incoming lines. I m trying to find out if DTMF style CallerID works on a FXO card (X100). I`v seen one solution with a modem attached in parallel with the X100 just to provide the ID on its serial port. It must be much better if this can be implemented in to the X100 driver. Any info about this would be highly appreciated.