Re: [Asterisk-Users] SPA3000 to Asterisk Server - Asterisk server not answering calls

I resolved this - I'm in the UK, and the problem was due to the cable (you need a two wire RJ cable) - I replaced it and it worked fine. Thanks On 7/13/05, Luki [EMAIL PROTECTED] wrote: John,all this ringing makes me think that your PSTN Ring Timeout is too low. Increase it by a second or two and

[Asterisk-Users] SPA3000 to Asterisk Server - Asterisk server not answering calls

My porblem is incoing PSTN calls are being forwarde to the * box, the phone rings, but when the phone is picked up, the call is not taken - it continues to ring. I am forwarding the call to (S0:) in my dial plan Can anyone assist? This is driving my crazy! Extract from the * console

[Asterisk-Users] OT: Open source CRM systems with * integration

Has anyone any experience of the above. Key feature for me is tracking incoming and outgoing emails and linking them to the contact record. Thanks, sorry for the OT ;-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] TAPI integration with * using Identapop software

Hi, I've got Outlook to call the number on * using the TAPI interface documented on the Wiki. Its working OK. I have downloaded the Indentapop application, and it appears to connect to * Ok using the Debug modes, but It isnt detecting incoming calls. Has anyone git identapop working? Care to

Re: [Asterisk-Users] Encrypted VOIP?

http://www.telephonyworld.com/cgi-bin/news/viewnews.cgi?category=allid=1107474518 On Fri, 04 Feb 2005 17:26:22 -0800, Steven P. Donegan [EMAIL PROTECTED] wrote: Is there any support in Asterisk for encryption of IAX and/or any other VOIP protocols? I haven't seen anything on this in the wiki or

[Asterisk-Users] Intertex IX66 incoming IAX

Hi, Has anyone got incoming IAX to work on the above router. I can call out, but incoming calls are not reaching the * box. Has anyone got this working? Could they give me some configuration hints. Thanks John ___ Asterisk-Users mailing list

[Asterisk-Users] Vocera Badges

Anyone got any experiences of these with *, and also costings? Thanks John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] IAX Softphone

www.signate.co.uk There is an e-book version. I bought mine from the states, arrived very quickly to the UK - around 5 days, and no postage cost. I ordered the CD of Asterisk with it, but didnt use it, and dont see it as having much value. Book is quite good for getting * running from basics

[Asterisk-Users] cepstral integration with * using AGI?

Hi, I've looked at the Wiki for this, have seen the Swift.agi details, but has anyone got a current script for Cepstral and an example of integraton in * please? I'm a * and linux newbie, so please be gentle ;-) Thanks John ___ Asterisk-Users mailing

[Asterisk-Users] Mediatrix III FXO 4 Port

Has anyone used one of these with *, any observations/comments please? Thanks John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] VoIP-to-TDM processing on-card?

Steve, thanks for that post, very useful and constructive. Thanks ;-) snip Check out the following link - proclaims to work with Asterisk http://www.sangoma.com/products/p_voice-data.htm -- They that give up essential liberty to obtain temporary safety, deserve neither liberty nor

[Asterisk-Users] R2 - Stable Asterisk

Now that I have your attention ;-) Anyone know if a new release is planned, and if so when? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] What is the best and easiest flavor to be used with Asterisk.

I use www.piebox.com (a Redhat AS Clone) which provides a good compromise between stability/release testing and cost - however I'm just using it on a test machine. John On Wed, 12 Jan 2005 16:58:29 +1300, Imran Sadiq [EMAIL PROTECTED] wrote: Could anyone please advise me on the best flavor

Re: [Asterisk-Users] So many Asterisk Patches - Which do I choose and use?

When you say CVS HEAD is the the same as stable? where do you get it from and what params do you use? On Wed, 12 Jan 2005 17:03:34 +, Niksa Baldun [EMAIL PROTECTED] wrote: There is no easy answer to your question. If you ask me, I prefer not to use any patches, except that I am forced to

Re: [Asterisk-Users] not sharing IRQ's

Isn't this used as a timer source by zaptel? On Wed, 12 Jan 2005 00:14:30 +0200, Shoval Tomer [EMAIL PROTECTED] wrote: You can disable the USB in the BIOS of the machine if you don't plan on using it. -Original Message- From: Michael Welter [mailto:[EMAIL PROTECTED] Sent:

[Asterisk-Users] Connecting a Home based worker with An Iaxy

Hi, If I need to connect a home based user to an Asterisk server, how does the above work? Is it (after being configured/provisioned) plug and play? Anyone done this got any comments Thanks John ___ Asterisk-Users mailing list

Re: [Asterisk-Users] OT: SIP Aware Firewall with Asterisk

Not an enterprise level system, but anyone used the www.intertex.se IX66? On Mon, 10 Jan 2005 10:14:46 -, Craig Waddington [EMAIL PROTECTED] wrote: We are on the lookout for a Firewall which is SIP aware, to pass the voice stream to Asterisk. We have looked at the Ingate

[Asterisk-Users] TDM4000 FXS and UK Caller ID

Anyone know in the current zaptel drivers and stable asterisk what the parameters are to receive caller ID in the UK over BT lines? Thanks Looked at the Wiki and bugs.digium but more confused, perhaps someone can help me John ___ Asterisk-Users

[Asterisk-Users] OT help with rmdir pls

I am tying to clear down an asterisk source directory before CVS'ing a new version the --ignore... option is being used but its still not being deleted, can anyone give me some clues. Sorry I'm new to Linux, as if you havent guessed. Googling hasnt helped so far Thanks

[Asterisk-Users] off topic - SSH configuration for Digium Support

I've an issue with my TDM4000P card and I will be calling Digium later to ask for their help. Could anyone help me with a basic configuration so they can SSH to me? Thanks John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] How do I get version 1.x from theDigium CVS or elsewhere?

Anyone help me, I've looked at the Wiki and cant see anything ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines

Ive just installed a TDM4000P with 4 fxos. The zaptel config is fine, zttest comes back with configured. If i call a line when zttest it shows on the display,and then goes when the line drops. In * when a call comes in, it follows my dialplan and answers the call according to the log, but IT

Re: [Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines

Yeah, it executes the answer in the script, and goes on to play music etc, but the line isnt actually answred IE continues ringing On Thu, 6 Jan 2005 22:41:09 +, Phil Quinney [EMAIL PROTECTED] wrote: Hi John, Have you got a line like this: Exten = s,1,Answer You need to actually

Re: [Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines

Yeah, it is set to the right signalling.. On Thu, 6 Jan 2005 22:47:45 + (GMT), Chris Glover [EMAIL PROTECTED] wrote: On Thu, 6 Jan 2005, John Middleton wrote: Ive just installed a TDM4000P with 4 fxos. The zaptel config is fine, zttest comes back with configured. If i call a line when

Re: [Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines

my X100P's to not answer calls if I have them enabled. Phil. On 6 Jan 2005, at 22:56, John Middleton wrote: what should those two settings say? should i set them to yes, or take the lines out? OK: Do you have these in zapata.conf? busydetect=no callprogress=no They won't

Re: [Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines

sure the call progress and busy detect are both no in my conf. I am looking for a More Correct answer to this as well. On Jan 6, 2005, at 4:34 PM, John Middleton wrote: What do you mean, as your first priority, you mean exten = s,1,Answer? On Thu, 6 Jan 2005 16:21:35 -0700, Ernie

Re: [Asterisk-Users] New asterisk installation but no audible voicemail prompts?

use - on the command line for debugging information, there should be detailed tracking information provided that will help On Wed, 5 Jan 2005 12:41:23 +0100 (CET), Remco Barende [EMAIL PROTECTED] wrote: Hi List! I installed Asterisk 1.0.3 stable on a RHEL rebuild. Due to problems

[Asterisk-Users] Versions of * what do they do/where is the change history/docs?

Could you please explain or tell me where it is explained the version and contents of * that is retrieved with CVS. I am wondering whether there is a change list or something. If you tell me here I will update the Wiki ;-) Thanks John ___

Re: [Asterisk-Users] Last callers script?

See http://www.wheely-bin.co.uk/asterisk/ check this link - I've implemented it and it works, at least in the test environment. John On Wed, 5 Jan 2005 16:00:56 +, Mike Dent [EMAIL PROTECTED] wrote: Hi, Is there some script which can be called from a * extension to playback the recent

[Asterisk-Users] Music from Freeplay music included in * ??

Hi On the www.asterisk.org main page it says Music provided by Freeplay Music with a link - Where is the music in the *config? I cant find any supplied music - is there any? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] Music from Freeplay music included in * ??

Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Middleton Sent: Wednesday, January 05, 2005 2:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Music from Freeplay music included in * ?? Hi On the www.asterisk.org main page

[Asterisk-Users] Dell Poweredge 6300 4 analogue lines

I'm just about to start implementing this project. I have a test server working well with SIP phones and IAX for incoming and outgoing, but when I golive will need 4 analogue lines coming in. 1. Anyone got this config working with a 4 port FXO digium card 2. Any tips/hints/traps Thanks John

[Asterisk-Users] voiptalk.org IAX service - user experiences

Hi, Anyone used this service, any comments on reliability/support? Thanks John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] voiptalk.org IAX service - user experiences

Peter Thanks for your response - have u experimented with the codec selections, or has anyone? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: