similar to: recursibve listing of file owner, possible?

Is there a way of temporarily suspending participants in a conference? Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the confbridge session that D and E can't hear, is there a way to suspend D and E for a while (whilst they are played music or whatever) and later join them back in? Failing that, I was considering kicking them and using an AGI script to rejoin

On a zaptel TE410p, when a call is bridged PRI - PRI how much involvement does the processor have? We're now seeing chunks of missing audio and I can't tell whether this is due to a kernel upgrade or to a zaptel/libpri/asterisk upgrade. I'm not seeing missed interrupts (from a cat of the proc/zaptel files), any other ideas on how I could go about tracking this down? I'm

As I understand it, you'd enter the extension at which you wish to be called back at, your 9665 has nothing to do with it. Instead of dialling 28 you could dial 9665 and that would add that SIP phone as an agent to the cytelcs queue. Steve -----Original Message----- From: Harold Workman [mailto:hworkman@cytelcom.com] Sent: 22 June 2004 18:54 To: asterisk-users@lists.digium.com Subject:

Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if that was all I could get. Steve -----Original Message----- From: Storer, Darren [mailto:starusers@comgate.tv] Sent: 01 July 2004 09:35 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve,

Hi, We're using our legacy PBX as a channel bank with asterisk sitting between the pbx and our telco provider spliced by a TE410P. If it were a straight analog FXS card then we'd use a hook flash to break into asterisk for transfers etc, does anybody know what the equivalent is for the PRI zaptel support? Regards Steve Steve Hanselman Brendata (UK) Ltd Tel:

Hi, ? I'm trying to figure out what the issue is splicing Asterisk between our Telewest PRI and a GDK-186 with a PRI card. ? We're using the Digium TE405P ? Our telco provider is Telewest, and Telco directly into switch is fine. ? When I splice Asterisk in, I can make and receive calls from Asterisk extensions, I can make outbound calls from the GDK, but inbound calls do not seem to pass

That sound's like the right thing to do, you'd probably have contexts related to what phones could do "unrestricted","localdialling","extensionsonly" and then within those contexts include the relevant contexts. If you look at the sample configs you can see how this is done for the international and local and you can extend that concept for your extension

I'm going to modify the queue announcements to allow for rounded seconds (e.g. we want to know to the tens of seconds. E.g. Average wait 1 minute 20 seconds). I'm going to add the optional announce of seconds to the queue config and a rounding factor (e.g. 10 in our case). The following parameters will be added Queue-announce-seconds (default is off) Queue-seconds (default

It'll work, either as a SIP phone with the SIP image, or as skinny using wither chan_sccp or chan_skinny (check the wiki). Steve -----Original Message----- From: Matt Davies | MattDavies.Net [mailto:matt@mattdavies.net] Sent: 02 July 2004 15:46 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960G and * I have been doing so much reading on phones lately that I have

It's so you can have agents that are less likely to take calls (e.g. imagine a sales queue, you'd have the sales people with no penalty, you might have the receptionists with a penalty of 1 and us propeller heads in technical support with a penalty of 2). The technical support people would only be offered a call from the sales queue if all the sales people and the receptionists were busy.

Are there any other GDK users out there with Asterisk? I've got all the integration working, except voicemail. Does anybody know a way of disabling the forward to voicemail on a per extension or per DDI basis (I can disable the voicemail hunt group but then I can't light the MWI indicators as it seems that only ports marked in the voicemail group can issue the MWI on/off commands).

Can anybody see a way of detecting the current number of retries remaining to a call file in the extension context that it is calling? E.g. If I want to schedule a fax and I want to feed an email back to the sender stating that the number is busy 2/5 retries remaining? Steve The information contained in this email is intended for the personal and confidential use of the

Throwing errors relating to utils.h: /usr/include/asterisk/strings.h:264: parse error before `__extension__' /usr/include/asterisk/strings.h:264: parse error before `;' /usr/include/asterisk/strings.h:264: warning: type defaults to `int' in declaration of `__retval' /usr/include/asterisk/strings.h:264: `__len' undeclared here (not in a function)

[vested interest: I reported that bug!] I agree with this, a system that has x bugs still has x bugs even if you close them, I think they should be closed by houskeeping if they can't be reproduced and the person that reported them can't provide a way to replicate, but they shouldn't be closed because nobody has fixed them for a while. This seems to be the case on a number of

I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) "three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz)" I cannot get the card working in any of the slots.

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial

We're using Cisco phones running skinny protocol. When I call other extensions I don't get a ringtone, although the remote end does ring and when answered we get clear two way audio. When I call a queue from a skinny phone then I don't hear the announcements. Likewise we don't hear music on hold on these phones, although we can see mpg123 in the process list and ls -l the fd

Hi, Is anybody in the UK using Telewest as a PRI Telco provider? Are you sending them caller ID? I've been told by Telewest that:- 1. Oftel doesn't allow them to accept caller ID (this is rubbish, and I replied pointing out where in the link to Oftel that they sent me it was stated. We need Type 2 caller ID) 2. Telewest can't do this. (this is rubbish, I'm certain

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On the /proc/zaptel it was lost interrupts, you haven't got any so that's good! On opencall.org there's a known issues link and that mentions some fax machines that have issues, might be worth a quick check there. I can receive from our fax machine this end, if you'd like I'll send you a test fax if you send me your fax number, if you can receive from us and tpc then it's