similar to: Providing Telewest in the UK with per extension outbound callerID

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, ? 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

A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. On the adept line I got a digium card in an opteron supermicro server. ztcfg gives me over 99.99 pretty much all the time.

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

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Folks, I've seen a few reports that people have had problems with hang up detection on UK cable phone lines. I have a TDM400P with two FXO ports, one connected to my BT line and the other connected to my Telewest/Virgin Media cable line. If I ring the BT line and then clear down, Asterisk detects this and acts accordingly. If I ring the Telewest line, the clear down is not detected, hence

Hi, I'm writing a utility that needs to smbmount various shares from servers in numerous domains (no problem, all working) and then list the contents of the directories (no problem again) and obtain the windows file owner in a textual form..... Any ideas how I can achieve the last part efficiently? I see that smbcacls can do it 1 file at a time, I really need a way of doing it

Hi there, We've got a problem connecting Asterisk with a TE205P to a TeleWest E1 ISDN line in the UK. We get a lot of "HDLC Bad FCS (8) on Primary D-channel" errors, and every so often the Primary D-channel goes down and all the calls got dropped. We've fully tested the card and made sure it's got its own IRQ. I was just wondering if anyone out there has

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

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:

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:

Hi I posted earlier about having issues connecting to Telewest's ISDN, only to find out later Telewest had forgotten to turn it on - hopefully I'm not having a similar silly problem. My PRI span is now up and operational, but when I try to send a call out over it, I just get congestion tones. Occasionally, I get about one second of ring tones, only for it to cut out and play congestion.

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

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

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

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)