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 try again (it probably is onthe PSTN tab in the web config menu of the SPA). I may be wrong, but Iknow for sure that my setup shows only one ringing message... --Luki ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Executing Dial(SIP/3001-047c, SIP/2004) in new stack -- Called 2004 -- SIP/2004-2f34 is ringing -- SIP/2004-2f34 is ringing -- SIP/2004-2f34 is ringing -- SIP/2004-2f34 is ringing -- SIP/2004-2f34 answered SIP/3001-047c -- Attempting native bridge of SIP/3001-047c and SIP/2004-2f34 == Spawn extension (sipura, , 1) exited non-zero on 'SIP/3001-047c' -- Executing Dial(SIP/3001-3d80, SIP/2004) in new stack -- Called 2004 -- SIP/2004-33ec is ringing -- SIP/2004-33ec is ringing -- SIP/2004-33ec is ringing -- SIP/2004-33ec is ringing SIP.conf [3001] type=friend host=dynamic context=sipura username=3001 secret=xx dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very canreinvite=no cantransfer=yes and extension.conf [sipura] exten =,1,Dial(SIP/2004) exten =,2,Answer exten =,3,Hangup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 share configuration details? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 on the list. Just curious. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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: http://lists.digium.com/mailman/listinfo/asterisk-users
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 IMHO when used in conjuction with the Wiki. John On Wed, 26 Jan 2005 16:50:13 +0100, Germán Micale [EMAIL PROTECTED] wrote: Hi, Does someone know an ActiveX IAX softphone? I need a free softphone to connect with Asterisk from a web page. Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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: http://lists.digium.com/mailman/listinfo/asterisk-users
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 safety. (Ben Franklin) The course of history shows that as a government grows, liberty decreases. (Thomas Jefferson) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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: http://lists.digium.com/mailman/listinfo/asterisk-users
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 of Linux on which Asterisk is easiest to install. I am currently using RH8.0, everything over the IP works fine but when I want to call a physical line I can only have conversation for about 3 sec and everything freezes after that. I have to hard reset the machine to bring it back up. Any suggestions will be greatly appreciated. Thanks Imran Sadiq Systems Engineer Tel: +64 9 377 8282 World Class Support for any business Fax: +64 9 377 7900 with between 7 and 70 computers. Mob: 027 286 9269 LANcom Technology Limited: 25 Union St, Auckland, New Zealand ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 use bristuff because I have quadBRI ISDN cards. Bristuff patches Zaptel in order to enable using quadBRI and octoBRI cards, and also adds some features to *. More info on www.junghanns.net. Like you said, really valuable patches will make it to the CVS sooner or later, so I prefer to wait because it makes installation and maintenance easier. I use Gentoo with 2.6 kernel. I am not sure whether you will get any benefits from upgrading, but I didn't have any problems with it (except that I had to migrate from devfs to udev, but that issue exists with 2.4 kernel too). Paul Rodan wrote: Ok, I usually use the latest stable CVS, with no patches or modifications. If figured if there was a worthwhile patch, Mark would have already included it. However, there was that neat patch about being able to press a certain key and it'd begin recording in mid-stream, that was an awesome feature and I patched my latest features.c file with that patch. But I keep seeing mentions of other patches, specifically something about the MOH patch, the BRISTUFFED patch, and now I'm hearing about a Super Parking Lot patch? For now I've been using the mpg123 method, it tends to work for me, but if I can save CPU/RAM and other troubles by using another format, which one do I go with? What is BRISTUFFED? And if I'm right, the super parking lot patch allows for call parking based on context, a way to break it apart, instead of making it universal across the whole system (where can I find this patch)? So I'm going to ask the question, if I were to install the latest CVS Stable tonight, which patches should I install on it before compiling? Also, I'm using Gentoo Linux, with the 2.4.26-r9 gentoo kernel. I've seen issues with people making Asterisk work perfectly with the 2.6 kernel so I've stayed clear of it, but I still see people fighting to make it work and such, I saw one post a while back about the benefits using Asterisk w/ the 2.6 kernel, can somebody please refresh my memory? What are the benefits of using Asterisk with the 2.6 kernel? I'm trying to get the most out of my system. Any help in making tonights compile/upgrade go perfect would be greatly appreciated. Thanks, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: Wednesday, January 12, 2005 12:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] not sharing IRQ's You absolutely do need to worry about usb module. http://www.microsoft.com/whdc/system/sysperf/apic.mspx Warren Burstein wrote: I'm not having any trouble with interrupts, but here's my /proc/interrupts on Fedora Core 2 on a hyper-threading CPU and using the SMP kernel (2.6.5-1.138). I don't think I need to worry about uhci_hcd, nothing is plugged into USB, but libata is the disk driver. How do I get libata and wctdm to use different interrupts? $ cat /proc/interrupts CPU0 CPU1 0:39957053931405IO-APIC-edge timer 1:530489IO-APIC-edge i8042 2: 0 0 XT-PIC cascade 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 12: 56 0IO-APIC-edge i8042 15:489 0IO-APIC-edge ide1 169:51072365082420 IO-APIC-level libata, uhci_hcd, wctdm 177:2136633 0 IO-APIC-level eth0, Intel ICH5 185:10019076889735 IO-APIC-level uhci_hcd, wctdm 193: 0 0 IO-APIC-level uhci_hcd 201: 0 0 IO-APIC-level ehci_hcd 217:59781561900756 IO-APIC-level wctdm 225:19173325960110 IO-APIC-level wctdm NMI: 0 0 LOC:79268527926712 ERR: 0 MIS: 0 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 Products, but they are very expensive. Can anyone point us to a well priced Enterprise SIP aware Firewall? SIP Phones - Firewall - Asterisk Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 DOESN'T actually pick up the call, i.e. it continues ringing. I'm using KS signalling, and connected to the UK phone system. Anyone seen this before, or give me some pointers, or suggest what I should post for diagnostic purposes? 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: http://lists.digium.com/mailman/listinfo/asterisk-users
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 answer the call before that ringing will stop. Hope this helps, Phil. In * when a call comes in, it follows my dialplan and answers the call according to the log, but IT DOESN'T actually pick up the call, i.e. it continues ringing. I'm using KS signalling, and connected to the UK phone system. Anyone seen this before, or give me some pointers, or suggest what I should post for diagnostic purposes? 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: http://lists.digium.com/mailman/listinfo/asterisk-users -- Phil Quinney IT Consultant - Any-Ideas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 DOESN'T actually pick up the call, i.e. it continues ringing. I'm using KS signalling, and connected to the UK phone system. Anyone seen this before, or give me some pointers, or suggest what I should post for diagnostic purposes? For the FXO ports on the card, you need to use FXS signalling. FXS_KS in your case. Confused? you should be :-) HTH Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines
They are set to No HMMM I get message Ring/off-hook in strange state 6 on channel1 in the log Anyone seen this before On Thu, 6 Jan 2005 23:07:39 +, Phil Quinney [EMAIL PROTECTED] wrote: They should be set to no - sorry, I should have been clearer. Busydetect and callprogress cause 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 work in the UK. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM4000P with 4 FXO's not picking up ringing lines
Sorry, maybe I didn't explain myself - could you send me your file please. Thanks John On Thu, 6 Jan 2005 16:41:18 -0700, Ernie Ankele [EMAIL PROTECTED] wrote: Exactly. I know this is not preferred, but it is the only way I can get around the problem currently (strange state message). I am 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 Ankele [EMAIL PROTECTED] wrote: Yes, I am getting the exact same messages on my console if I do not answer the call as my first priority in the dialplan. On Jan 6, 2005, at 4:11 PM, John Middleton wrote: They are set to No HMMM I get message Ring/off-hook in strange state 6 on channel1 in the log Anyone seen this before On Thu, 6 Jan 2005 23:07:39 +, Phil Quinney [EMAIL PROTECTED] wrote: They should be set to no - sorry, I should have been clearer. Busydetect and callprogress cause 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 work in the UK. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 with * modules refusing to build I replaced the RHEL kernel with stock 2.6.10. Asterisk seems to be working but when I dial voicemail I hear nothing. When I hangup I see a message on the console that the calller did not specify a mailbox number so I guess voicemail app is working. The phone(Grandstream BT100) is connected directly to the * server so it's not any NAT or firewall trouble (no firewall installed). Any ideas? Which kernel options are required for Asterisk to function properly. Any recommendations on that? Several options do come to mind like, also to prevent timing problems like: - HPET Timer Support (CONFIG_HPET_TIMER) - Provide RTC interrupt (CONFIG_HPET_EMULATE_RTC) - Preemptible Kernel (CONFIG_PREEMPT) (even though the info in kernel describes this for desktop) - Message Signaled Interrupts (MSI and MSI-X) (CONFIG_PCI_MSI) - Enhanced Real Time Clock Support (CONFIG_RTC) - HPET - High Precision Event Timer (CONFIG_HPET) For general Astrisk with ISDN operation: Is telephony support and ISDN support in the kernel required? Thanks for any suggestions! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 incoming callers on a particular PSTN line? In the UK 1471 is a BT number which plays back the most recent callers number, it also gives you the option to call this number back (now charging you for this service too!). Is there anything similar in asterisk-land? thanks Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music from Freeplay music included in * ??
Hi, Hmmm they aren't there - I did a cvs checkout -r v1-0_stable asterisk from the digium web server - Whats the CVS command for a 'head' install ? Thanks On Wed, 5 Jan 2005 14:43:23 -0500, Steven Frazier [EMAIL PROTECTED] wrote: Hi John, Yes when you do the cvs head install, look in /var/lib/asterisk/moh -rw-r--r--1 root root 1939812 Jan 5 14:07 fpm-calm-river.mp3 -rw-r--r--1 root root 2582496 Jan 5 14:07 fpm-sunshine.mp3 -rw-r--r--1 root root 2217563 Jan 5 14:07 fpm-world-mix.mp3 [EMAIL PROTECTED] mohmp3]# Steve -Original 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 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 http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 (My first posting be gentle) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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: http://lists.digium.com/mailman/listinfo/asterisk-users
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: http://lists.digium.com/mailman/listinfo/asterisk-users
