Re: [Asterisk-Users] TDM400P crackel
I made the ISA/jumpers comment just the other day. God I miss the old days. On Mon, 2005-03-14 at 19:26, Ron Joffe wrote: On Monday 14 March 2005 16:18, Eric Wieling wrote: Ron Joffe wrote: Hey folks I have a new setup with a TDM400P for a pair of analog extensions and a few SIP phones. We seem to be experiencing a bunch of Crackeling when talking between the analog and SIP extensions. Any ideas? Yes. Check the suggestions given to the other guy that posted this question earlier TODAY. Sorry, I litterally Subscribed right after that post was made, and my search didn't reveal it. I've got to use more liberal search terms. The solution as was suggested was to determine interrupts. I disabled onboard USB,Serial,Parallel, and then moved the card around until it found it's own IRQ. hmm. Wish I was back on an ISA machine where i could have just set that :) Excellent support from the Digium help desk solved this problem Thanks, Ron ___ 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] FW: AntiSpam Alert from Rusten McKenzie
Title: AntiSpam Alert: Request For Authentication I wondered where that came from. I just deleted it and continued. On Tue, 2005-03-15 at 12:04, dean collins wrote: Is there anyway we can get this shit off the asterisk list apart from posting their email address [EMAIL PROTECTED] here for the spambots to pick up? Dean From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 15, 2005 1:02 PM To: [EMAIL PROTECTED] Subject: AntiSpam Alert from Rusten McKenzie Importance: High Rusten McKenzie needs you to authenticate yourself before your email will be accepted. To authorize yourself, here's all you have to do: 1. Press Reply 2. In the body of the reply, type in my AntiSpam Passcode: 3. Press Send. Bongosoft AntiSpam authenticates the SENDER of each incoming email. For more information about Bongosoft AntiSpam, please visit us at www.bongosoft.com X-Bongosoft-AntiSpamRFA ([EMAIL PROTECTED]) ___ 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] asterisk 1.0.6
I had this issue. I configured my sip phones to use rfc2293(?) instead of inband. Note:the rfc number is incorrect but I don't feel like looking up the correct one right now. Just look in sip.conf example and it will tell you the right number. On Mon, 2005-03-14 at 04:51, Bashir Ullah - www.Lamsre.Com wrote: Hi after upgrade from R2 to 1.0.6 , my dtmf not working and i cant dial . can any 1.0.6 user help me why i cant do that. Bashir ___ 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] Some Hardware Advice
I know the g4 370's(I think that's the model) have issues with zaptel cards and the new intel chipset. I've also seen people have issues with the 3ware SATA raid cards if that is what ships in the machine. The last thing I saw recently was a dell server having trouble with NMI, and disabling the USB seemed to fix it. Google the list for the above things to find more info. It seems the main issue with hardware is irq and pci latency. I've also seen people mention hyperthreading causing some issues. On Fri, 2005-03-11 at 07:47, Brett, Gary wrote: Hi there Just a quick post to ask you guys if you've had any bad (even good) experiences using current model HP or Dell servers ?? specifically the HP proliant ML110 and the Dell Poweredge 1800 SATA, (but I will welcome your recommendations on any current Models) . I will be rolling out some small to medium systems with a max 100 Sip extensions and 60 outbound (2 x e1) for the larger rollouts and as little as 5-10 users for the smaller systems ( zap channels on TDM400P's) . Any advice would be greatly appreciated Gary ___ 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] how do i get rid of this blasted echo !!!
I've added MMX and it didn't help. I also did the CFLAGS+=-march=pentium4 with no help there either. The more I search, the more I found and I'm down to disabling any hardware not used in the box such as USB and recompiling the kernel with a fresh copy from kernel.org. It seems there were a lot of problems solved when 2.6.9 came out. If I ever get mine fixed, I will try to post everything I did. On Fri, 2005-03-11 at 07:50, Brett, Gary wrote: So is it accepted as standard that compiling with MMX will help improve echo type issues ? -Original Message- From: Herman Cremer [mailto:[EMAIL PROTECTED] Sent: 11 March 2005 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] how do i get rid of this blasted echo !!! Thanks Error. I have switched to IAX looong agomuch better ! Just battle when doing double NAT :) I dont have the phones here with me, but lets say its different...is there away to adjust the channel to fix the err ? -herman On Fri, 2005-03-11 at 13:24, [EMAIL PROTECTED] wrote: Hi Herman, Look at the bottom of your phones and compare the REN values of both. Do they both value of REN 1.0? I think the one with the problem might have an REN value other than one. You tell me! Errol Samuels Don't let SIP Drive you crazy, use IAX2 On the echo... I have 2 extensions, with different analog phones. The one works fine, the other echos and scratches like mad !! I have switched the ports, cables etc but its ALWAYS the same phone... Maybe this could be it ? Is it ok from a SIP phone ? Herman cremer ___ 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 ___ 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] TDM04B lock up
I had an issue with the same setup except only channel 1 on each card would work for incoming. All would work for outgoing, but asterisk never saw the other channels ringing. Restarting asterisk didn't help either. I panicked and just rebooted and the problem went away. I wish I had taken the time to have tried unloading and reloading the drivers. The system had been up for 22 days when this happened so I now just restart every sunday morning to be safe. On Fri, 2005-03-11 at 09:36, Goutam Shaw wrote: Hi I have a strange situation. Once in a while (non-deterministic) the 2 TDM04B cards lock up at the same time and stop processing incoming and outgoing calls even though * shows that it is trying to communicate to ZAP channels (at least on the outgoing). The only cure is to reboot the system when it happens. It makes me very apprehensive of the system Has anyone seen this problem. Could this be something to do with the IRQ sharing. Here is the output of lspci -v. I see that one of the cards shares IRQ # with VGA controller and the other one with ICH4 IDE. Any help would be appreciated. 00:00.0 Host bridge: Intel Corp. 82845G/GL [Brookdale-G] Chipset Host Bridge (rev 01) Subsystem: Dell Computer Corporation: Unknown device 0160 Flags: bus master, fast devsel, latency 0 Memory at f000 (32-bit, prefetchable) [size=128M] Capabilities: [e4] #09 [1105] 00:02.0 VGA compatible controller: Intel Corp. 82845G/GL [Brookdale-G] Chipset Integrated Graphics Device (rev 01) (prog-if 00 [VGA]) Subsystem: Dell Computer Corporation: Unknown device 0160 Flags: fast devsel, IRQ 11 Memory at e800 (32-bit, prefetchable) [size=128M] Memory at feb8 (32-bit, non-prefetchable) [size=512K] Capabilities: [d0] Power Management version 1 00:1e.0 PCI bridge: Intel Corp. 82801BA/CA/DB PCI Bridge (rev 81) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=32 I/O behind bridge: d000-dfff Memory behind bridge: fe90-feaf 00:1f.0 ISA bridge: Intel Corp. 82801DB ISA Bridge (LPC) (rev 01) Flags: bus master, medium devsel, latency 0 00:1f.1 IDE interface: Intel Corp. 82801DB ICH4 IDE (rev 01) (prog-if 8a [Master SecP PriP]) Subsystem: Dell Computer Corporation: Unknown device 0160 Flags: bus master, medium devsel, latency 0, IRQ 9 I/O ports at ignored I/O ports at ignored I/O ports at ignored I/O ports at ignored I/O ports at ffa0 [size=16] Memory at feb7fc00 (32-bit, non-prefetchable) [size=1K] 00:1f.3 SMBus: Intel Corp. 82801DB SMBus (rev 01) Subsystem: Dell Computer Corporation: Unknown device 0160 Flags: medium devsel, IRQ 3 I/O ports at efe0 [size=32] 01:04.0 Communication controller: Tiger Jet Network Inc. Intel 537 Subsystem: Unknown device b100:0003 Flags: bus master, medium devsel, latency 64, IRQ 11 I/O ports at dc00 [size=256] Memory at fe9fc000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 01:05.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) Subsystem: Realtek Semiconductor Co., Ltd. RT8139 Flags: bus master, medium devsel, latency 64, IRQ 3 I/O ports at dd00 [size=256] Memory at fe9fbf00 (32-bit, non-prefetchable) [size=256] Capabilities: [50] Power Management version 2 01:06.0 Communication controller: Tiger Jet Network Inc. Intel 537 Subsystem: Unknown device b100:0003 Flags: bus master, medium devsel, latency 64, IRQ 9 I/O ports at de00 [size=256] Memory at fe9fd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Regards Goutam Shaw ___ 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] Phone suggestions
Polycom SIP300 works good with all the features except the echo cancellation. It says in the manual it has an echo can, but other sources say otherwise. Not to advertise, but voipsupply.com lists their sip phones by price and might make your search a little easier. I am not affiliated with the above site, but just used it for reference. Sorry if I am breaking the rules. On Fri, 2005-03-11 at 09:58, James Murray wrote: Can anyone offer any suggestions for quality hardware sip phones under $150. Preferable one with a 2 line caller id screen and the ability to disable call waiting. It would also be very useful if it had a good voice echo cancellation built into the phone. ___ 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] Question with email notification
127.0.0.1 localhost 192.168.1.9 mypc mypc.mydomain replace mypc with hostname and mydomain with hostname. On Fri, 2005-03-11 at 13:05, J P Edmund wrote: ummm... can someone give me a default settings for the hosts.conf file for the basic, or the correct format. Dummy here seems to have screwed it up and overwritten my backup. Now my pager notification stopped. urrrg. J.P. Edmund If you think it's not a game, you've already lost On Mar 10, 2005, at 1:20 PM, J P Edmund wrote: Thanks to all who answered my question. Most of the suggestions to sources I had tried with no success and retried again. Which leads me to believe that I messed thing else up. I also neglected to say that I am using the asterisk at home version. Right now the email thing isn't a big issue for me. Notification does go out to the cell phone email addresses fine as they seem to accept the from [EMAIL PROTECTED] Once I remember where I put the password for the email server here, I am just going to forward the email thru it and leave asterisk alone with that for now. J.P. Edmund If you think it's not a game, you've already lost J.P. Edmund If you think it's not a game, you've already lost ___ 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] Phone suggestions
I wonder if the GS is better than the 100 series. The 100's are too cheap for business use. On Fri, 2005-03-11 at 12:52, Wiley Siler wrote: I am a Polycom guy so IP300 or IP500 comes to mind. Also this Grandstream looks interesting... http://www.voipsupply.com/product_info.php?products_id=331osCsid=adc1f6 01a12939e2a97ca342d73173b7 Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, March 11, 2005 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phone suggestions James Murray wrote: Can anyone offer any suggestions for quality hardware sip phones under $150. Preferable one with a 2 line caller id screen and the ability to disable call waiting. It would also be very useful if it had a good voice echo cancellation built into the phone. SIPura SPA-841 -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] TDM04B lock up
Should ACPI be turned off in the kernel? In the bios I can only set cards to the 1-15 interrupt range, but linux and acpi it seems moves these to the 20's. I looked last night on the lists and found no true answer to this question. I have 4 TDM's so interrupts below 15 are few and far between, but I do have enough. On Fri, 2005-03-11 at 13:50, Richard Scobie wrote: Goutam Shaw wrote: Hi I have a strange situation. Once in a while (non-deterministic) the 2 TDM04B cards lock up at the same time and stop processing incoming and outgoing calls even though * shows that it is trying to communicate to ZAP channels (at least on the outgoing). The only cure is to reboot the system when it happens. It makes me very apprehensive of the system Has anyone seen this problem. Could this be something to do with the IRQ sharing. Here is the output of lspci -v. Get the cards on their own interrupts - use the BIOS, turn off unneeded onboard devices etc. If you still have problems and the FXO modules are marked Rev C on the non pin side, talk to Digium support. I had ongoing problems with FXO modules stopping responding and requiring reboots to restore. Regdumps of the offending module show ff loaded in almost all registers. After contacting Digium, I was told this was a hardware issue and after having them replaced with modules marked X100B RevB, I have so far had no problems. Regards, Richard ___ 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] tdm400p and dell 2600 poweredge
I'm using tdm400's here and was curious about the irq misses. My zttool won't compile for some reason and I haven't researched it enough to be worried about it yet. Is zttool the way to diagnose the irq issue and does zttool work with the tdm400 boards. I don't have a shared irq issue but here's the output of /proc/interrupts. is there anything else to check? cat /proc/interrupts CPU0 CPU1 0: 172897707 0 IO-APIC-edge timer 1: 11 0 IO-APIC-edge i8042 2: 0 0 XT-PIC cascade 8: 2 0 IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 11: 0 0 IO-APIC-level ohci_hcd 12: 50 0 IO-APIC-edge i8042 14: 76 1 IO-APIC-edge ide0 18: 172811936 0 IO-APIC-level wctdm 20: 172802850 0 IO-APIC-level wctdm 22: 172807318 0 IO-APIC-level wctdm 24: 172791427 0 IO-APIC-level wctdm 26: 361826 0 IO-APIC-level cciss0 28: 11479585 0 IO-APIC-level eth0 31: 293841 0 IO-APIC-level aic7xxx, aic7xxx NMI: 5767 0 LOC: 172894484 172894411 ERR: 0 MIS: 0 On Thu, 2005-03-10 at 11:11, Jeb Campbell wrote: Grant McInnes wrote: Hi all: I've been developing and testing on a tdm400p card and it's been going well. As you probably know, the tdm400p needs an ide power supply, but the dell poweredge 2600 that this card is destined for eventually has all the power supplied on the backplane with no ide cables. I used a molex splitter from another drive. On the topic of the dell 2600, I had to change servers as I could not get the t100p and tdm400p on their own irq's (It also had the raid card which is not supported in the new megaraid driver, but that is another story). Anyway in the bios, I could only set the irq to share with the raid controller or the gigabit ethernet. I was getting tons of irq misses which caused frame slips (and bad echo) on the t1. Anyway, please let me know how it goes and watch for irq misses. ** This was a while back, and it could have been a 2650, so please verify this for yourself ** Jeb Campbell [EMAIL PROTECTED] ___ 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] Polycom phones do not talk to each other and cannot answer when we pickup
Never used pbxware, but the context the sip phones dial out using specified in sip.conf needs to include the dialplan context of the phones in extensions.conf. On Thu, 2005-03-10 at 15:08, Kanuri, Seshu (Company IT) wrote: We have bought PBXware GUI from Bicom systems and configured extensions with Polycom Phones as UAs. The Polycom Phones can dial out and make calls but I cannot make extension to extension calling. Googling did not help much. As you may be aware PBXware is a closed source software GUI from Bicom Systems for configuring extensions. It is a good tool to configure and manage users and phones but it does not allow to do any of the customization tasks that are possible by directly editing the .conf files, which may be required in for Polycom. However if this is an issue of configuration on the Phone itself, we want to be able to make changes and fix this problem. Any tips? Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] Telecom echo cancel disable
Yeah. Edit zconfig.h and there's an option to ignore 2100hz. I didn't know what caused the 2100 until you said something. On Wed, 2005-03-09 at 09:47, Matt Schulte wrote: Disabled echo canceller because of tone (tx) on channel 10 I understand that the PSTN companies use their own echo canceller's, send a tone across 2100hz, the problem we're having is people are complaining of echo on random calls. I'm assuming this may be the cause. Is their anyway to 'ignore' the disabling of EC? Or would be just be a manual code change.. Matt ___ 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] Assistance with Overhead Paging
If you use polycoms, there's a great script I use from the wiki that works perfect. A little tweaking and you can even do zone paging. The main trick is being able to autoanswer on speaker on the sip phone depending on either line or as polycom uses, a variable. On Wed, 2005-03-09 at 12:26, Nathan C. Smith wrote: A quick search of the list (google: paging site:digium.com) would show you this has been discussed in the past. There is at least 1 SIP paging product. Modified grandstream phones set to autoanswer have been suggested. Sound cards using the console/DSP have been suggested. There are references to these on the WIKI as well -Nate ___ 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] Re: Upgrading Asterisk
Question about the make linux26 command. I use a 2.6 kernel and always do a straight make. Does adding the linux26 do anything except help the makefile know that it's a 26 kernel if it has trouble detecting for some reason? I might go ahead and try 1.0.7 and need to know if make linux26 does anything special a simple make doesn't. On Wed, 2005-03-09 at 14:01, Martin Roy wrote: I did the upgrade everything went well. Now I hope it fixed my echo problem... When I installed version 1.0.3 I forgot to do make linux26 so maybe my echo problem was coming from that. I'll know soon enough... Martin Martin Roy wrote: I'm about to upgrade my currently running asterisk server from 1.0.3 to 1.0.6 is there anything I should do before doing the upgrade? I know since I'm using Fedora Core 3 that I must do for Zaptel : make clean then make linux26 and finally make install. Do I have to remove the version 1.0.3 first or only doing install will replace all existing modules without removing my config files? Once I have installed Zaptel 1.0.6 then do I have to do something special to upgrade asterisk from 1.0.3 to 1.0.6? As I have a lot of people using the asterisk server I can't put it down for a long time so I want to be sure I don't make any stupid mistake before doing the upgrade as I don't have time to reinstall everything from scratch if something goes bad. Thanks Martin ___ 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] Tweaking AGGRESSIVE_SUPPRESSOR
Well this morning I'm running without AGGRESSIVE_SUPPRESSOR but with the MMX optimization. If I have more trouble, I might try the pentium4 instructions flag. I've searched everywhere and found nothing on the wtcdm driver bug with it dropping frames. Never heard of this until Steve mentioned it. The Adit 600 with mgcp is looking better everyday, not sure if I want MGCP or T1 interface though. On Tue, 2005-03-08 at 06:56, Andrew Kohlsmith wrote: On March 8, 2005 04:32 am, Adam Goryachev wrote: Is this the same thing that I might be getting on a TE405p? Most of the time, everything works nicely, but sometimes, things go astray, and echo comes into the conversation. Also seems to affect faxes with some lines with errors during the fax. Finally, also sometimes get a red alarm on the line, shortly followed by it going green again, of course, it drops all calls in progress... It sounds like you are getting clock slips on the span. Are you sure you're syncing to the other side (or that the other side is syncing to you)? FWIW I have *zero* trouble sending or receiving faxes through a 1-hop IAX2 span with TE405Ps on either side. It is only with the TDM400P that I cannot reliably receive faxes. Note that I am *not* using the agressive canceller, just the default (MARK2 I think). It is an interesting datapoint that on our Bell Canada PRI I was unable to achieve decent echo cancellation until I enabled MMX optimzation in zconfig.h and turned on pentium4 instructions and ordering with CFLAGS+=-march=pentium4 in the Makefile. This is on a Xeon 2.8 system. -A. ___ 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] Queue and SetGroup
I use queues here and if you have a multiple presence (i think that's the term, you know where you have call waiting on an extension) phone, it will do this. My polycoms do this and I actually like it because my queues are just single user queues. Try turning multiple presence off and see if it helps. On Tue, 2005-03-08 at 06:06, James Murray wrote: I manage the PBX system for amedium sizedcall center. Where all calls are distributed via a few call Queues. However I am having an issue where reps are being distributed calls regardlessof wether they are on a call. I have looked into using SetGroup but I don't think this works with Call Queues. I have also looked into incomingcalllimit and that seems to no longer work. Any sugestions? Not sure if it matters but I am using AgentCallbackLogin as a login method. ___ 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] Retreiving the called number
I might be misunderstanding the question but wouldn't ${EXTEN} work?
On Tue, 2005-03-08 at 04:31, Guy Decarpentrie wrote:
Hi all,
I've note that the variable DIALEDPEERNUMBER is broken.
Now i want to know if exist another method to retreive the called number on *,
and, if it's possible, an example ;)
Regards.
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Re: [Asterisk-Users] Tweaking AGGRESSIVE_SUPPRESSOR
Mine is a p4 xeon so I'm good there. MMX didn't help me though. I have had to redefine aggressive at the moment. The updated mec2.h and mec2_const.h files from bug#2820 didn't seem to help either. I found an article about in the asterisk sources chan_zap.c a READ_SIZE 160 causing echo but I wasn't sure if this was just with the T1 cards or also the FXO cards. The thread is started here http://lists.digium.com/pipermail/asterisk-dev/2004-November/007423.html On Tue, 2005-03-08 at 11:56, Trevor Peirce wrote: Andrew Kohlsmith wrote: Note that it's only a good idea if your system has MMX capabilities (and a P4), obviously. :-) While this is being mentioned -- Never do this on a Celeron... sound quality is bad and there are random crashes if I recall correctly from my experience. Turning MMX off immediately fixed both the issues. ___ 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] GotoIf problem
Can you post your dialplan for that extension. Also, NoOp works great for debugging these issues.
On Tue, 2005-03-08 at 12:29, kurt x wrote:
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is
this venture. When I dial and access voicemail with an ani of 3000
the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) . In additon when I dial with a 10 digit
ani the s,3,Gotoif does not work. It also goes through each line(
1,2,3,4,5,6,7)
Any help is greatly appreciated.
Thanks
Kurt
Asterisk CVS-HEAD-07/14/04-16:28:29 built by
[EMAIL PROTECTED] on a i686 running Linux
[globals]
${ext}=0
SetGlobalVar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM}}) = ${DIGITS}]?s|5)
exten = s,4,GotoIf($[${CALLERIDNUM} = 3000]?s|6)
exten = s,5,Voicemail(u${ext})
exten = s,6,Background(pbx-invalid)
exten = s,7,Hangup
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[Asterisk-Users] Adit 600 for asterisk
Ok, I've pretty much decided to try the Adit route. Somebody who has experience with these tell me if I'm missing something. I have 15 incoming PSTN lines. T1 is not an option at current location. I want to put in an Adit 600 with 2 8-port FXO boards. The adit will then connect to * via a digium t1 board. I configure zaptel.conf for the T1. What other parts would be needed? How do the PSTN lines connect to the Adit, standard rj11 jacks? It looks to be about a $2500 investment and I need to know if there is anything special I am missing. 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] Im a noob
A common issue you will have with FXO/PSTN lines with sip is echo. Test thoroughly before you go live and have 20 people yelling, I can hear myself talk. On Fri, 2005-03-04 at 15:39, Ty Purcell wrote: Yes it does support a basic analog line (or many many lines...). It also supports T1's, ISDN, etc. FXO would provide an analog connection to the phone company (your wall jack) FXS would allow you to plug analog phones into Asterisk. Phone ---(FXS)---Asterisk(FXO)Phone Company You could eliminate the FXS need if you run SIP or IAX IP handsets. Then they would just connect to your network. Ty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, March 04, 2005 3:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Im a noob Im completly new to the whole PBX thing. I have a toshiba unit now and we're moving our office in the next few months. I want to use asterisk but would like to test it out first. Does it support a basic analog phone line like the one in my house? Is that FXS? Are there any FAQs I should read to learn more? Thanks for the reply! ___ 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] Is there a way to find free zap channels on remote servers ??
how about using chanisavail via manager api On Thu, 2005-03-03 at 16:21, Paco Perez wrote: Hello: I would like to know if there's a way to request free chanels from remote asterisk servers ? My idea is to make an agi returning a dial to inter-asterisk connected servers when there's not enought chanels on local server, maybe like a ping to all of them or maybe requesting to a central server where all the *s send and request information about available chanels each 2 or 3 seconds, it has not about dial plans because I make LCR first and I have a flat rate for national calls, It is about using less analog lines with constant costs every month. Maybe Asterisk has internals for manage this situation (like virtual group of different asterisks chanels) But I would like to be sure 90% that a free chanel is going to be available when I dial to another asterisk and not to have calls rounding over Internet. Thanks for your comments. Paco ___ 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] Tweaking AGGRESSIVE_SUPPRESSOR
Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while they are talking. I haven't been able to find anywhere in the MEC2 source to limit when it mutes the remote party. It seems to do it with just the slightest bit of sound coming from the room. What are my options besides getting mad and ordering a PRI and a TE100? ___ 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] Tweaking AGGRESSIVE_SUPPRESSOR
This seems to be how AGGRESSIVE_SUPPRESSOR works. To make sure you don't get echo, it does what a speakerphone does, mute the other party if it hears audio from your end. There is a setting in mec2_const.h for AGGRESSIVE_HCNTR=160 that says in the comments 20ms, I'm assuming this is to tell how long to suppress the other party. There is nothing on this that I have found anywhere and since we are live, I can't change until later to see how it works. We have Polycom SIPS for users, and it doesn't matter what the other party is. It seems from another thread, that the problem midcall is that the electrical properties of the line change midcall causing the echo to return. Without AGGRESSIVE_SUPPRESSOR defined the first minute or so is fine, then a click happens and the echo begins. The phones also seem to go extra sensitive then. You can then hear even keyboard clicks from typing where you don't normally. I've wondered if it's the zaptel cards or poor electricity at my place to the asterisk server. I have put in a SmartUPS 1500 to try to condition electricity there just to make sure. As far as echo and PRI, thanks for making me cry since I just knew that would solve it. On Mon, 2005-03-07 at 12:31, Andrew Kohlsmith wrote: On March 7, 2005 01:23 pm, Dennis Webb wrote: Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while they are talking. I haven't been able to find anywhere in the MEC2 source to limit when it mutes the remote party. It seems to do it with just the slightest bit of sound coming from the room. What are my options besides getting mad and ordering a PRI and a TE100? PRI and TE110 won't save you; we've had echo issues with our TE405P and a Bell Canada PRI. All the PRI does is ensure *you* are not causing echo. Now I'm curious -- What physical phones are on either side of this call? Do you have a speakerphone on? I've never heard of an echo canceller acting how you describe, but lots of speakerphones do exactly that. -A. ___ 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] Tweaking AGGRESSIVE_SUPPRESSOR
Thanks for both of the responses. I have 2 questions now, is there a better codec to use? I know if I can limit the voip delay that exists, it might help the problem some. 20ms is what is current but maybe 10 would help. What would be best, bandwidth is not an issue currently. This doesn't fix the wctdm issue obviously, which makes perfectly good sense, but might would help. Second question, what about a chennel bank? I here people talking about the adit 600 but from what I read, it seems more like it requires a T1 as it's a DSU if I'm correct. This would bypass the tdm boards and everybody says it has great echo cancellation. I'm recompiling the drivers right now and turning on the MMX optimizations hoping it might help the issue (a man can dream can't he). I'm getting desperate here because the system works great except for the echo which the users are getting irritable with it and I agree with them they should. Once again thanks. We've been running full steam with asterisk for over 2 months now. On Mon, 2005-03-07 at 18:14, Steve Underwood wrote: Steve Kann wrote: What he describes is echo suppression. Because an echo canceller can, generally, only remove some part of an echo, not the entire echo, systems are generally designed to suppress the residual echo in some circumstances. Old speakerphones had poor on no echo cancellation, so the suppression kicked in like that, because it was the only choice. In modern systems, the echo cancellation is much better, so suppression is not needed as much, and when it is used, it's probably done much more imperceptibly (with comfort-noise and stuff like this). Only a very few high end conferencing speakerphones have ever used echo cancellation. Even most expensive digital phones on PBXs merely do echo suppression in speakerphone mode. The nature of A-law/u-law limits the performance of an echo canceller across the PSTN to about 30dB of echo improvement. If you look at the behaviour of those codecs, you will see they give a roughly contant 30dB of instantaneous dynamic range, and the echo cancellation enhancement will never exceed that dynamic range. There is still enough residual echo that good quality cancellers have to perform non-linear suppression to eliminate it, and substitute comfort noise. 30dB, on top of the minimum of at least 12-15dB of echo suppression the hybrids give, means the echo should be rather quiet. It is still enough to annoy people, though, and suppression is standard practice. It is specified in G.168. Regards, Steve ___ 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] Question with email notification
set serveremail= to the address in voicemail.conf On Mon, 2005-03-07 at 18:42, J P Edmund wrote: I have been searching all over for the answer on all sources online and have come to the conclusion that it must be rudimentary or I am asking the wrong question. I cannot figure out how to configure the box to set the from address to a correct domain, as my outgoing isp will not pass mail from [EMAIL PROTECTED], as I expect it wouldn't. Any help is appreciated, even just what should I be looking for to find the correct information. J.P. Edmund If you think it's not a game, you've already lost ___ 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] Polycom SP300 questions
A quicker way to get to the missed calls list is to hit the down arrow button. Just exiting out just clears the display of missed calls and resets counter, the records are still in there. Look for the polycom remote reboot script and reboot the phones daily to clear the list for good so you don't eventually suck up all the memory in the phone. Also, update your firmware to 1.3.4 due to a bug I had where the phone would randomly stop getting audio into the headset. Only a reboot fixed this. On Mon, 2005-03-07 at 18:15, Jerry wrote: On Mar 7, 2005, at 5:52 PM, [EMAIL PROTECTED] wrote: Hi, all I have two questions regarding usage of Polycom SP300 with Asterisk. No sure if it is Astersisk or phone related, though. 1. When dialing an extension, one has to perss Dial or Send on the phone after number is entered. Is it possible to avoid this and only enter the number? Configure the digitmap under sip settings to match your dialplan 2. This is probably phone only related, but hopefully someone know the answer. If there wwas a missed call, phone shows 1 call missed. I am trying to figure out how to clear this message from the phone. There are no buttons as far as I can see to get rid of this message on the phone. Goto the Missed Calls directory and exit, the counter will clear. Thanks, Rudolf ___ 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] Asterisk@Home and VoiceMail
1. how about something like a gotoif statement that compares CALLERIDNUM = EXTEN and if they match, goto a VoicemailMain(${CALLERIDNUM}) priority.
2. In phoneXXX.cfg, set your MWI settings. You can set the msg.mwi.1.callback to be your check vm extension, in my case 299. Also, you might want to set bypassinstantmessage=1 in there also.
On Mon, 2005-03-07 at 16:01, Don Murray wrote:
Hi all,
Yeah, another newbie here but I'm getting down to just the nice to
have problems not the need to have problems so life with Asterisk is
good :)
I have installed [EMAIL PROTECTED] 0.6. I have 2 Polycom IP500's plugged
into the network, no outside connection yet but I have ordered a TE110P
card from Digium.
I have a voicemail related questions. Voicemail is working but is there
a way to
(1) have dialing one's own phone automatically put you into voicemail?
If I connect to my own extension I get to leave a message at my
voicemail box (because I am on the phone at the moment) but no amount of
key banging (*98 specifically) seems to get me into accessing my
voicemail-mode, which is the behaviour I would prefer.
(2) Reconfigure Polycom phones to remap the call made when you push the
voicemail buttton, to a voicemail extension, for example. I have a
voicemail extension (111) working so if I could reprogram this button it
would solve (1). The thing is that at the moment pushing the voicemail
button on the phone dials the phone's own extension by default. I konw
if the voicemail button doesn't work I can look forward to having to
explain why to each and everyone person with one of these phones,
probably twice each :)
I have successfully configured a voicemail extension that behaves
exactly the way I would like the voicemail button to work.
Thanks,
Don
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Re: [Asterisk-Users] Recommended Phone for beginner
I use Polycom 300's at my office and they are about $130 and work fine. I also hear that the sipura's are nice. They have an $85 model. On Mon, 2005-03-07 at 20:32, Ryan Burke wrote: Hello everyone, I've been watching this list for a while, but it is the first time I've posted. I'ved decided to setup a * server for my house and will need 3 phones (one main, one for my wife, and one for my office). I was wondering if there was a particular brand that people reommended? I'd like ot get an actual SIP phone, instead of an adapter like the Sipura SPA-2000. I've been looking at the Grandstream BudgetTone 100 series but after looking at the Wiki for setting up * with that phone it looks like it might be more trouble than its worth. Of course I would love a Cisco 79* but I'd like to keep the cost at a minimum but get a good amount of flexibility in tersm of features. Hopefully once I get over the learning hump I can start contributing to this list. Any input would be appreciated. Thanks, Ryan ___ 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] Park Call timeout
I tried this but the problem is that on a blind transfer from an outside call, the caller id comes through as the PSTN Callerid and not the transferring extensions. I want the callerid to stay that way, so I guess I'm out of luck at the moment. On Fri, 2005-02-25 at 02:41, ST wrote: This is a bug in asterisk. Caller's exten is saved nowhere, so park cannot call back when timeouts. What you have to do is copy caller's username, as long as it is its extension, to parkee's callee number. I gave this from SIP's point of view. ___ 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] Polycom Call Parking
I tried getting this to work and gave up when the polycom admin guide said the park button didn't work in SIP or MGCP applications. Don't know why it's there. On Thu, 2005-02-24 at 09:54, Johann wrote: Has anyone gotten the call parking soft button on the polycom phones(specifically the IP 600) to work with call parking that asterisk provides? If so what configuration changes were needed? --johann ___ 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 Dennis Webb Delta Express #3 Afco Road West Memphis, AR 72301 (901)508-9711 ___ 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] Park Call timeout
I have searched the lists and voip-info and having trouble with a call parking issue. When I park a call and it times out, it seems to immediately tries to goto exten s in whatever context the person who parked the call is in. Voip-info under config features.conf that it will ringback to the original extension. Is the original extension the person who parked the call, or the original extension the call came in on. I would like that if a parked call times out, it ring back to the person who parked the call and after like 10 seconds if the original parker doesn't answer, go to the operators. Any help would be appreciated. ___ 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] transfer ringback
Not everybody here has voicemail and we have more than one person answering the phones. I would like to know that if Person A transfers to Person B, that after a predefined timeout, it would then transfer the call back to Person A. Right now when a new call comes in, it rings 3 phones. When Person A transfers to Person B, it tries for 15 seconds, then transfers the call back to all 3 people via the Dial command. It's a little confusing to write out, so if more info is needed, I will be happy to elaborate. ___ 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] Asterisk and #
make sure you have tT when the incoming call comes in. I've been studying up on parking today and saw this a few times. On Thu, 2005-02-24 at 11:51, Marco Ziglioli wrote: Hi ml, I have a problem related to call parking. When on my X-Lite try to parking a call dialing #700 I don't obtain anything. I can only ear dtmf tones during conversation but not other happens. I also read in some post that only pressing # should place call in hold state but this doesn't happen on my system. Can someone help me? Thanks. Marco ___ 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] Can you set up a phone via MAC address?
What phone? I use polycoms and configure via ftp. They pickup their config based on their mac address. They will register fine and asterisk will know the ip based on registration information. On Thu, 2005-02-24 at 12:01, Philip Trauring wrote: Is it possible to configure a sip phone by MAC address instead of by IP address? Obviously if you're using DHCP it's not a good idea to set up phones by IP address - so how can someone set up a sip phone by MAC address? Thanks, Philip ___ 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] CallerID problem
Your phones should have a timezone or GMT offset setting. It appears your in the Central zone so set it to -6. Also, setup a NTP server and have your phones sync with it. I find it works fine to use the * server for the NTP server. On Thu, 2005-02-24 at 12:55, Anton Krall wrote: Guys... Ive been having problems with my callerid and I have no more clues as to what I could be.. dates and times stamped on voicemail and info received on the phones display are off by +6 hours and also the date for example today is Jan 02 :) What can I do to modify this? __ Anton Krall ___ 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] Making two * servers share same dial plan?
check here http://www.voip-info.org/wiki-Asterisk+-+dual+servers On Thu, 2005-02-24 at 14:28, Me wrote: Can someone point me to some docs that explain this or give me a direction to go in. I have seen docs on this in the past but can't seem to dig em up now when I need them. Basically I want one Asterisk server to be the traffic cop and send some calls directly to ATA's and some calls to another Asterisk server, the other Asterisk server will then direct the calls to the end users ATA on their desk or to vmail etc. Thanks.. -- Start Your Own ISP! http://www.YourOwnISP.com ___ 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] Creating extension groups
in extensions.conf, create a context for your internal extensions. In the context for outgoing calls, add include = internalextensions. Then in zapata.conf, for each extension put context=internalextensions for people with no outgoing access, and put the others in the context of the outgoing calls. On Wed, 2005-02-23 at 09:27, Kanishka Somaratne wrote: Hi I want to create 2 groups of extensions, for example group 1 cant make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of theextensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please explain how to do this in detailed. I want to know how to route all outgoing calls through our SIP server and how to stop some of the extensions from taking outgoing calls Thank you, Kanishka ___ 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 Dennis Webb Delta Express #3 Afco Road West Memphis, AR 72301 (901)508-9711 ___ 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] Asterisk manager
Title: [Asterisk-Users] Asterisk manager I use gastman. I haven't looked on WIKI for anything newer/better lately. On Wed, 2005-02-23 at 11:56, Kanishka Somaratne wrote: What is the best Asterisk manager to use, i do not mind web based or GUI. Thank You Kanishka ___ 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 Dennis Webb Delta Express #3 Afco Road West Memphis, AR 72301 (901)508-9711 ___ 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] TFTP Server
Title: [Asterisk-Users] TFTP Server Check under /etc/xinetd.d/tftp. There's a server_args variable that should read like -c -s /path/to/files This is a suse 9.1 box but should be about the same On Tue, 2005-02-22 at 16:10, Gary G. Hendershot wrote: On my server (ES3) the TFTPBOOT folder is where I put my Cisco image loader files -Original Message- From: Ferguson, Michael [mailto:[EMAIL PROTECTED]] Sent: Tuesday, February 22, 2005 1:25 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] TFTP Server G'Day All, Can anyone give me some direction in setting up the TFTP server on my RadHat ES3 box? I did quite a bit of reading, but I think I am more unsure now than before. I found the information nebulous. TFTP is already installed. I am trying to determine where the root directory for the tftp services is located so I can copy the CISCO 7960 firmware files onto it. Thanks Ferg ___ 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 Dennis Webb Delta Express #3 Afco Road West Memphis, AR 72301 (901)508-9711 ___ 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
