Re: [asterisk-users] Minimum hardware requirements for 10 concurrent calls?

You could try using a Intel Little Falls motherboard for that if you are not going to be recording calls. It comes with the processor on board. Garth van Sittert BSC (Physics Comp Sci) Technical Director Tel: 08600 24826 supp...@bitco.co.za vese...@campbell

Re: [asterisk-users] MusicOnHold works Externally, but not internally

Do you not have to answer the channel before the MOH can happen? Joseph wrote: No, the same happens when I use SIP phone, no music on internal call. -- Joseph On 11/03/09 13:10, Danny Nicholas wrote: I suspect that IAX is the culprit... -Original Message- From:

Re: [asterisk-users] SNOM 870

We have the 870 working great in our test environment so far. Garth van Sittert BSC (Physics Comp Sci) Technical Director BitCo 08600 24826 www.bitco.co.za --[ UxBoD ]-- wrote: Anybody tried one with Asterisk yet ? Views ? Best Regards

Re: [asterisk-users] Asterisk capacity

Depends on what you want to do and what your server platform is like. Garth van Sittert Technical Director BitCo 08600 24826 www.bitco.co.za abdelkader wrote: Hello, What is the maximum number of simultaneous calls supported by asterisk. thks

Re: [asterisk-users] QoS VPN

I would think that VoIP over VPN is a bad idea as UDP packets need to be in realtime not corrected by the TCP of the VPN. Garth van Sittert Technical Director BitCo 08600 24826 www.bitco.co.za Aurimas Skirgaila wrote: Despite the VPN overhead, running VOIP through VPN is good idea because

Re: [asterisk-users] How to send 404 Not found SIP reply?

As a quick workaround you could use a goto to send to an invalid extension. Goto(nowhere,1) Garth van Sittert Technical Director BitCo 08600 24826 www.bitco.co.za Chris Maciejewski wrote: Hi, I am trying to send 404 Not found reply, without any luck with the following: exten = 555,1

[asterisk-users] mISDN ports and dstchannel CDR logging

asterisk 1.4.20 and misdn 1.1.8. This never used to happen on asterisk 1.2. I have also tried the latest chan_misdn on 1.4 with the exact same results. I have found no other useful documentation on this. Kind Regards Garth -- Garth van Sittert Technical Director BitCo 08600 24826

Re: [asterisk-users] asterisk queue agent problem

Hi Satish You would want to investigate Local channels on Asterisk for this. Garth Garth van Sittert BSc (Physics Computer Science) - Main: 08600 BITCO Phone: +27 (0)11 875 6900 Fax:+27 (0)11 875 6901 Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] MSN:[EMAIL

Re: [asterisk-users] extensions.conf vs. AEL

Where would you suggest all the logic goes Brian? Garth Garth van Sittert BSc (Physics Computer Science) - Main: 08600 BITCO Phone: +27 (0)11 875 6900 Fax:+27 (0)11 875 6901 Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] MSN:[EMAIL PROTECTED] Web

[asterisk-users] Asterisk-1.2 and Centos 5

it to Centos 5 I experienced a complete OS crash when calling over HFC misdn channels. Didn't really have time to investigate and dumps as it was a live machine. I have tried updating all relevant asterisk software, but to no avail. Anyone have any ideas? Kind Regards Garth

Re: [asterisk-users] Kirk IP600 V3 DECT Wireless server

Hi Remco I have used the IP600 v3 with SIP support on Asterisk... apparently I was the 1st person globally to run it at a site. The 1st firmware was a bit buggy at times, but seems to be much better on the later versions. Kind Regards Garth Garth van Sittert BSc (Physics Computer Science

[asterisk-users] Asterisk 1.2 and mixmonitor stopping short

a b410p card on the box. When using HFC based cards I have no problem with the recording. Does anyone have any ideas? Is it possible to reopen this bug on the old 1.2 code? Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

[asterisk-users] Spandsp-0.0.3 and asterisk 1.2

: Fax receive not successful - result (11) Unexpected message received. The files are only 8 bytes long??? Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] Digium b410p and 2.6.17 kernel bug?

DSP_CANCEL_INIT called Disabling EC Disabling Hardware EC vpm_echocan_off called on timeslot 1 mode_hfcmulti: channel 0 protocol 0 slot -1 bank 0 (TX) slot -1 bank 0 (RX) dsp_from_down: change tx volume to 0 handle_bmsg: unknown PH_CONTROL info 0 Any ideas would be greatly appreciated. Kind Regards Garth

[asterisk-users] MySQL cmd % pattern matching

${connid} ${query}) But there seems to be a problem with the % sign and I don't know how to hash it out. It works without the % sign. Thanks Kind Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

Re: [asterisk-users] MySQL cmd % pattern matching

Hi Jon No luck - it works with the quotes and no % sign but as soon as I add the % it doesn't work. Garth Jon Farmer wrote: Try enclosing in single quotes. ie. SELECT name from contacts where tel like '%${number}' Jon Farmer Telford, Shropshire, UK - Original Message From

Re: [asterisk-users] MySQL cmd % pattern matching

I have it working as your example, Doug, but unfortunately I need the like phrase as the numbers all contain spaces or sometimes even brackets. Garth Doug Lytle wrote: Garth van Sittert wrote: exten = s,5,Set(query=SELECT name from contacts where tel like %${number}) exten = s,6,MySQL

Re: [asterisk-users] MySQL cmd % pattern matching

Garth Garth van Sittert wrote: Hi All Does anyone know how to use the MySQL cmd in Asterisk with LIKE and % in the query? I have: exten = s,5,Set(query=SELECT name from contacts where tel like %${number}) exten = s,6,MySQL(Connect connid hostname username password dbname) exten

Re: [asterisk-users] Echo Issues

Hi Matt Check that your volumes are not too high. Kind Regards Garth Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO MSN:[EMAIL PROTECTED] Web:www.bitco.co.za Matt wrote: Hello, I had had

[asterisk-users] Cubix / Firefly softphone and Asterisk

. Idefisk works 100% on the same setup. Kind Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CDR - mysql with asterisk 1.2.12 not working

/ suggestions? Have you tried turning on debug in logger.conf. You should be able to see what is wrong from there. Kind Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

[asterisk-users] Budgetones - multiple phones losing IP address during day

what could be causing this? Thanks Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk + Samsung OffServ 500

Hi Eugeniy You should set the Asterisk configuration as if you are connecting to your local Telco provider and set the Samsung to fit Asterisk. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

Re: [asterisk-users] How to test TE405P T1

part of the processing for the Digium cards (that's why they are so cheap) so you cannot test the PRI without getting asterisk up and running. The zaptel drivers will get Layer 1 and I think Layer 2 up, but Asterisk is needed for more than this. Kind Regards Garth

[asterisk-users] AgentCallBackLogin and cdrupdate

the agent still show as the static extension. Kind Regards Garth -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO MSN:[EMAIL PROTECTED] Web:www.bitco.co.za

[asterisk-users] Asterisk queues and dynamic members

know that the agent is busy on a call and try another free agent? I have worked around this by using call-limit=1 in the sip.conf file. Kind Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

[asterisk-users] Snom phones locking up

be the network switches etc, but Cisco? I fail to see how a switch could bring down a device. Kind Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] GSM analogue router

Garth -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO MSN:[EMAIL PROTECTED] Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com

[asterisk-users] Rxfax and squashed TIFF

the metric units in the TIFF header. Convert them to imperial units? Kind Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[asterisk-users] PRI channels filling up

channel = 1-15,17-31,32-46,48-62 I have since changed the switchtype to QSIG and the Samsung is now set up with QSIG. Kind Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?

-users Hi I had stability issues with queues on 1.2.9.1. 1.2.7.1 also has queue issues, but it is a LOT more stable. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

[asterisk-users] Echo on PRI

correctly. Echo problems in the past have been straight forward to remove with the correct echotraning and volumes set. I am quite certain it is due to calls going through the Samsung. Any ideas? Kind Regards Garth ___ --Bandwidth and Colocation provided

Re: [asterisk-users] Asterisk + fax

cancellation on your line is causing this. I had the same issue with faxing. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[Asterisk-Users] Rxfax with Sirrix quad BRI

Hi All Has anyone had experience with rxfax on asterisk 1.2.x with a sirrix quad BRI card? Does it work with the Sirrix cards? Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

Re: [Asterisk-Users] Any IP phones with pro-audio connections?

I recently installed one of these plugged into an ATA with a dialplan that calls a predetermined number when the line is picked up. The sound guys have only to push one button and we get audio into the pbx. Works flawlessly. We did have to boost the input level more that I think we should have

[Asterisk-Users] Billing when forwarding incomming calls from SIP phone

and dst forwarded number in the CDR. No SIP details at all. How do we track this? Billing for standard calls from the SIP user to the PSTN is fine. Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] Voicemail bomb

for a release. Can anyone give me some more info? Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: Voicemail bomb

Thanks for the input guys. Nice to understand what's going on. Steven, how do you know that particular diff is the fix for that particular bug (not that I doubt you, just curious how to reference the code to the bug). Thanks again, Garth On 5/8/06, Steven [EMAIL PROTECTED] wrote: Or, if you

[Asterisk-Users] Voicemail indication for analog phones

(probably because the above examples are time based)? Thanks for any input. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

[Asterisk-Users] Zaptel compile errors

.ELsmp. I have both the kernel and kernel sources installed: kernel-2.4.21-37.EL kernel-pcmcia-cs-3.1.31-13 kernel-utils-2.4-8.37.12 kernel-source-2.4.21-37.EL kernel-doc-2.4.21-37.EL kernel-smp-2.4.21-37.EL Can anyone help with this? Garth

Re: [Asterisk-Users] Zaptel compile errors

Upgraded kernel and sources. Seemed to sort it out. Garth Garth van Sittert wrote: Hi When trying to compile zaptel-1.2.5 I am getting the following errors: /usr/include/linux/modversions.h:1:2: #error Modules should never use kernel-headers system headers, /usr/include/linux/modversions.h

[Asterisk-Users] Passing Digits between ISDN PBX and Asterisk

long or too short. Anyone know what could be causing this? I would like to find some more info on the ISDN layers and protocols, but I haven't found a good source on this. Thanks Garth ___ --Bandwidth and Colocation provided by Easynews.com

Re: [Asterisk-Users] Asterisk hints

Hi Jean-Marc I tried removing the call-limit setting. It still doesn't work. I am using a SNOM 360 to monitor the line status'. Do I still need to activate the busy lamp on the IP10S' or is this only if you want the IP10S' to monitor the hints? Garth Jean-Marc Salsa wrote: I am using

Re: [Asterisk-Users] Asterisk hints

How does the hints work? Do you know anything about the flow? Thanks Garth Mike Pollitt wrote: Hi Garth -- Other users have also reported problems with the status being set by the SwissVoice phones - oh wait a minute... that was you! Have you tried setting call-limit=1 in sip.conf

[Asterisk-Users] Asterisk hints

Hi All Does anyone know how the hints in asterisk works? How does a SIP phone interact with the hints? I am having a problem with certain phone models that do not set the hints correctly when I list the hints with a 'show hints'. Thanks Garth

Re: [Asterisk-Users] need help

Have you checked the permissions on the file? Is it executable? Garth Dirgan Putra wrote: hi All need help, iam installing areskiCC and have a problem after that create extension for calling card and after dial exten = 17000,3,DeadAgi,a2billing.php i see messages : a2billing.php

Re: [Asterisk-Users] Asterisk hints

I am using Swissvoice IP10S phones. Garth Mike Pollitt wrote: Garth -- What kind of phones are you using? Mike. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Garth van Sittert Sent: Wednesday, 22 February 2006 7:29 PM To: Asterisk Users Mailing

[Asterisk-Users] Sirrix BRI errors

Hi I have a test setup of a sirrix card installed in NT mode connected to a PBX. I keep getting the following error: D-Channel receive message aborted, discarding frame (RSTAD=0x1c) What does this mean? What could be causing it? Garth

Re: [Asterisk-Users] MixMonitor and command

Yes, you need to remove the 'System' part. You should only have: exten = s,n,MixMonitor(${CALLDIR}${CALLFILENAME}.wav||touch/tmp/test${UNIQUEID}) Garth Alex Barnes wrote: Has anyone had any success using the MixMonitor() plus command as nothing I have tried works. I am using 1.2.1 I

Re: [Asterisk-Users] Handset phone to replace Flash Operator Pane l

I have that set up, but I cannot get some of the phones to change the hint State. The SNOM phone show State:InUse, but Swissvoice phones show State:Idle even when on a call. I use 'show hints' to see this. Kind Regards Garth Colin Anderson wrote: Breeze to set up, too. To monitor

Re: [Asterisk-Users] Asterisk and Snom 360

Could we possibly see your settings to get this right? I am trying to get it working at the moment. I can see the phone buttons have subscribed to asterisk, but they just don't light up. We are using 4.1 firmware and are upgrading to 5.3 to see if it helps. Regards Garth Darrell Long

[Asterisk-Users] Hint priority

dialog-info+xml Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] asterisk silence suppression?

The silence suppression is a client setting. Asterisk does not have silence suppression as far as I know. Garth Dan Elder wrote: Hi all, I'm getting some noise gate like effects on our sip lines I think I need to disable silence supression, I'm searching docs not finding where this can

[Asterisk-Users] PRI Bridging and Recording

this, that would be great. Thanks Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] ATA's and faxing

purely for VoIP traffic. Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ATA's and faxing

I am using alaw and I have already enabled the pass through. Does alaw and ulaw work? I can fax out, but not receive faxes. Garth Johann Steinwendtner wrote: Enable pass thru fax mode on the HT486, or enable ulaw in your SIP config. Hans Garth van Sittert schrieb: Hi All Is there any

[Asterisk-Users] Handset phone to replace Flash Operator Panel

Hi All Has anyone come across a handset that can somehow replace FOP? Some users don't like FOP unless it is on a dedicated PC. Thanks Garth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

Re: [Asterisk-Users] Codec Selection

Hi Abdul You will need to download and install the Intel API which is then used to compile the patched G723 codec. Hope this helps. Kind Regards Garth Abdul Lateef wrote: Hi All, I have one Carrier which is supporting only G.723.1, how i can put in my extentions.conf to send calls

Re: [Asterisk-Users] Directed Call Pickup

] port = 5060 bindaddr = 0.0.0.0 canreinvite=no disallow=all allow=alaw context=internal [200] callerid=Reception 200 type=friend host=dynamic dtmfmode=rfc2833 username=200 secret=pbx Kind Regards Garth Garth van Sittert wrote: Show Features produces: Builtin

Re: [Asterisk-Users] Directed Call Pickup

Hi Alex I tried your exact example below and still the same thing. I am getting 403 Denied after I see the Pickup cmd in the CLI. If you do a show channel SIP/XXX when the phone is ringing, do you get a value for Extension:?? Kind Regards Garth Alex Barnes wrote: -Original Message

Re: [Asterisk-Users] fax possibilities

Hi James I would consider Hylaxfax if you are going to do purely faxing. Garth James Harper wrote: I am trying to set up a linux based faxing solution for a client, and have found that the modem they have (ancient dataplex external unit) just isn't up to the job. It talks to some remote fax

[Asterisk-Users] Outbound Caller ID number on E1

Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth -- Garth van

Re: [Asterisk-Users] Outbound Caller ID number on E1

? Do I need to send the complete number, 3 digit area code + 4 digit extension to the Telko? Does the zapata.conf add the prefix? How can I check what callerid number is being passed to the Telko? Garth Steve Underwood wrote: Garth van Sittert wrote: Hi All I am having a problem

[Asterisk-Users] Directed Call Pickup

Kind Regards Garth -- Garth van Sittert BSc (Physics Computer Science) - Mobile: +27 (0)83 791 6662 Email: [EMAIL PROTECTED] Phone: 08600 BITCO Web:www.bitco.co.za ___ --Bandwidth and Colocation provided by Easynews.com

Re: [Asterisk-Users] Rewind MusicOnHold?

Hi Dan Have a look at setting up queues. Kind Regards Garth Dan Journo wrote: I thought someone was going to say that. Does anyone know a way to do the following:- 1) Answer incoming call 2) Begin dialing an extension 3) While extension is ringing play a welcome message to the caller 4

Re: [Asterisk-Users] Directed Call Pickup

}) When I dial 812, in the CLI I can see: Executing Pickup(SIP/29-707f, 12) in new stack Any thoughts? Kind Regards Garth Bob Goddard wrote: On Thursday 02 Feb 2006 16:46, Garth van Sittert wrote: Hi All I am having problems with Directed Call Pickup in Asterisk 1.2.1 If extension

Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth connection

Hi Cosmin You should be able to get about 12 simultaneous calls on a 128k line and about 28 on a 256k line according to asteriskguru's bandwidth calculator http://www.asteriskguru.com/tools/bandwidth_calculator.php. Kind Regards Garth BitCo Data Communications http://www.bitco.co.za Cosmin

Re: [Asterisk-Users] (newby) Is PING a good indicator of latency?

There is a good utility called iaxping to test IAX latency. Kind Regards Garth BitCo Data Communications http://www.bitco.co.za Cosmin Prund wrote: As the subject line says: Is PING a good indicator of network latency? If not, how can I measure latency? Thanks, Cosmin Prund

Re: [Asterisk-Users] Voipjet - No one is available to answer at this time

Don't think there is anything wrong with your setup. We get the same thing... Maybe they're down, but I would like a third opinion... G Michaël Gaudette wrote: Hi, I`ve just tried the Voipjet 0.25$ test account, following everything the web site told me to do (see below). When I dial a

Re: [Asterisk-Users] Testing AreskiCC

Not an answer to your questions, but just in case you don't know there is a lot of info on the wiki: http://www.voip-info.org/wiki/view/AreskiCC+CallingCard+Application We use Areskicc here, and it works great. However we do not use sip/iax friends, perhaps both of your problems lie there?

Re: [Asterisk-Users] Areski Calling Card GUI

If you haven't seen it already, this will be a lot of help to you. http://www.voip-info.org/tiki-index.php?page=AreskiCC+CallingCard+Application+The+idiots+guideV2 You should now be on step 12. :) G Omar McKenzie wrote: Hi I have gone thru the steps of installing AreskiCC, I

Re: [Asterisk-Users] wifi phones - desk

I've also only heard of the Clipcomm Along the same lines... Why doesn't anyone make a wireless ATA? Am I the only one with a need for such a thing? By the time I plug in a wireless bridge, an ata and a cordless phone, I need a five outlet powerstrip and shoebox to hide all the

Re: [Asterisk-Users] call to a particular 800 number never shows answered on Zap channel

This one drove me crazy for a while too. I found out that some companies don't exactly play fair and don't pass answer supervision on a call until you are actually speaking with a live person. The person I spoke to about this wasn't sure if that was even legal, but he said it happens quite a

[Asterisk-Users] Multiple PCI cards

Hi All Does anyone know if multiple Digium cards on a single machine will be a problem. Machine specs: Dual Zeon 3.0GHz on Intel server board. Cards: TE411P, TDM400P, TDM400P I will turn off all unnecessary PCI devices; USB, parallel, serial, etc... Thanks

Re: [Asterisk-Users] looking for failover ideas

I have had an idea of using two identical servers: Server A with IP x.x.x.a and server B with IP x.x.x.b. Server A is live while server B sits in the background monitoring server A. Server B rsync's asterisk config files daily with server A. In the event of server A going down, server B

Re: [Asterisk-Users] pins for users

We use the Areskicc calling card system as an authentication system. It does everything you are asking and can generate great reports and graphs. I like it very much. That being said, areskicc is tough to get going, but there is plenty of info here:

[Asterisk-Users] Voipjet experiment

Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using

Re: [Asterisk-Users] Voipjet experiment

not their problem, but at the same time I would think they want to be as functional as possible. Thanks for the help everyone, G Garth Summey wrote: Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation

[Asterisk-Users] DND Indication

Hi, Has anyone come up with a clever way of indicating DND is activated? I've thought of stutter dial tone and using the mwi, but have no idea how to implement these. I'm using Budgetones. My concern is that users will activate the DND, then forget about it not realizing that they are not

[Asterisk-Users] app_intercept

Hi, Can anyone give me any information at all to get app_intercept working? I've found these pages, but there is just not enough for me to get it going. http://www.pbxfreeware.org/archives/2005/06/new_download_--.html and http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002692

Re: [Asterisk-Users] [EMAIL PROTECTED] newbie extensions always busy

Let's start basic, we know that both PCs that are running the soft phones can see the aah server, but can both PCs see each other? Can they ping each other? (ie, they are not across a NAT router or something like that?) G Mark Anthony C. Delfin wrote: hi list, I'm running a newly

[Asterisk-Users] DND Indication

Has anyone come up with a clever way of indicating DND is activated? I've thought of stutter dial tone and using the mwi, but have no idea how to implement these. I'm using Budgetones. My concern is that users will activate the DND, then forget about it not realizing that they are not

[Asterisk-Users] Ring but now audio on answer

Ok, it now seems to be a firewall issue. When I turn the FW completely off, the calls work just fine. I only had one opening in the FW (5060:udp). Is there another port/protocol that should also be open? ___ Asterisk-Users mailing list

[Asterisk-Users] Ring but now audio on answer

I have my Asterisk server all setup. But have an odd problem and hope someone here can help. I have a Polycom IP 300, a Grandstream GXP-2000, and an installation of X-Lite. They can each call each other just fine (extension-to-extension). I can also dial-in from the outside (via