similar to: volume "fading in and out"

I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p EuroISDN. We have 2 ISDN lines, one we had for testing, and one for general (40+ users) use. During the testing phase, we had 10 phones linked to the second ISDN line, and there were no problems with echo at all. Lucky me. However, since we have started rolling out, we've had quite loud complaints that there is

I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls "popping in and out". Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? TIA Julian.

Has anyone else experienced a problem with the LED for span 1 on a TE410P or TE405P? I had a TE410P on which the span 1 LED would not light red, but once the span was connected, it did correctly light green. I RMAed the board to our UK distrbutor and received a replacement. However, the replacement board displayed the same problem! Wondering if it was related to the computer I was putting it

Thanks for the feedback. I do agree with having multiple smaller servers. When I was first approached with this task I mentioned as much. However, the current desire is to work with already existing hardware. That is out of my hands at the moment unless it just can't be done. I will explore Freeswitch a bit soon to compare it as well. I am struggling to find what the bottle neck is in

Is there anything to choose, in performance, between a TE405P and a TE410P? I understand the difference between the PCI bus voltages, and certainly don't intend to try Andrew's hacksaw operation :-). But if I choose the card first, and a compatible mobo second, does it make any difference which? Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play:

Thanks for the suggestion Tony, I installed each codec for MoH, core sounds, and extra sound packages. Unfortunately the tests produce the same results. [Sep 1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0 ( continuously for a while followed by a [Sep 1 20:36:46] WARNING[7761][C-0000770d]:

I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks. The application relies on a DTMF digit string sent by the phone after the call has connected. This DTMF is detected by Asterisk under the control of WAIT FOR DIGIT commands send from an AGI processor over a FastAGI connection. Usually the DTMF is detected without error, but on a significant minority of calls, Asterisk is missing

Hi We are having a significant (> 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes. All the other legs are PSTN (TE410P). The example configuration Slave box 1 meetme <--- IAX2 ---> Master box meetme <--- IAX2 ---> Slave box 2 meetme The delay is between Slave box 1 and Slave box 2 The primary suspect is our iax configuration

We have a Te410P with 3 Telco T1's (D4 SF ) with DID's (non-PRI). ANI & DNIS is received in-band DTMF in a format such as *7145551212*8002* What happens when there are 30 or more calls, asterisk might see is DNIS = 802 or ANI = 4551212 for examples, where parts of the numbers are dropped. All the traffic arrives into a simple IVR script where a message is played. We are

In article <E02FA554-9D6D-4E7D-8A78-5FBDE1DE939D at kicp.uchicago.edu>, Valeri Galtsev <galtsev at kicp.uchicago.edu> wrote: > > > > On Sep 26, 2020, at 8:05 AM, Jerry Geis <jerry.geis at gmail.com> wrote: > > > > I have a disk that is flagging errors, attempting to rescue the data. > > > > I tried dd first - if gets about 117G of 320G disk

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

has anyone done caller announce in MeetMe's before? Dave P >>> brian@bkw.org 5/18/2004 5:50:49 PM >>> With multiple parking lots you can give each person their own lot thus exten 800 for everyone will connect them with just their call passing the lot name which you know for X customer. bkw ----- Original Message ----- From: "Andrew Kohlsmith"

I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on this console is made up of weird characters instead of normal. Please see http://www.softins.co.uk/dsc00018.jpg for an example. If I do "asterisk -rvvvvv" on a normal login, either via the

Hello List, I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.

I have been writing a lot of AGI programs in C with good success. I would like somehow to have an AGI program continue in the background while the pbx execution returns to the dialplan and continues. Is this possible? I was thinking that perhaps I could fork or create another thread within the AGI prog. The reason I want to do so is in order to monitor external information (e.g. credit limit and

I am having problems getting the latest CVS right now. A cvs checkout asterisk -t gets to this part and sits forever: S-> server_register(fpm-world-mix.mp3, 1.1, , , , , ) S-> Register(fpm-world-mix.mp3, 1.1, , , ) Anyone know how I can just skip the file? Travis Conway EFS, Inc. Information Technology Desk:?? (334) 215-6551 Mobile: (334) 391-4450 mailto:travis@homeoffice.quikpawn.com

I've noticed that with FastAGI (and maybe AGI) that when you sequentially send a sequence of dial commands, if the call is picked up, that after the call ends, the Fast AGI script keeps executing the commands! Is there anyway to stop execution once a call is picked up? I think looking at the result codes after the Dial to determine if the call was picked up or not is not a good idea... if it

I have a puzzling situation, and would be grateful for any insight. I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g. [from-siptrunk] exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) Now, if I use a different SIP trunk for the outbound call, than the inbound call came on, the call is set up

Does anyone know how a program, script or shell user can best determine whether the machine is running on bare metal or is a VMware guest? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org