Re: [asterisk-users] Could Asterisk be crashing under high context switches?
On Fri, Dec 18, 2009 at 06:53, Jason Martin jmar...@metrixmatrix.comwrote: Hello! I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs. In this configuration, we have trouble maintaining stability. It may be fine for days, but soon the load slowly creeps up on the server from below 1 all the way up to 6 which is when no one can dial out and asterisk pretty much has to be killed to be stopped. We also have bandwidth.com set up as a SIP provider. If we use bandwidth.com, stability is greatly improved. I installed munin on the phone server yesterday and noticed something dramatic, I think! Asterisk became unstable 3 times yesterday. 2 of those times, the number of context switches went to almost 80k the first time, then over 70k the second. First question - is this abnormal for around 20 ongoing recorded calls? I did a little bit of searching and found this: http://wiki.sangoma.com/files/wanpipe-linux-asterisk-tutorials/How_to_Reduce_Asterisk_System_Loads.pdf It talks about zaptel/DAHDI chunk size and that directly affects system load. Second question - the document explains how to change the chunk size for Sangoma hardware. Is there a general way to do that for DAHDI? Thanks is advance! Jason Martin Metrix Matrix, Inc. 785 Elmgrove Rd, Bldg 1 Rochester, NY 14624 Office: 888-865-0065 x202 Mobile: 585-705-1400 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Jason, Indeed what you are seeing is not typical. I don't have normal number available off-hand, but a system should have no problems whatsoever with 2 or 3 R4T1s. As you can expect, Rhino has thousands and thousands of customers running with no problem, which makes this instance the exception. All Rhino cards use the same amount of bus resources (time hold the PCI bus, data copied, etc) no matter if it's an R1T1 or R4T1, or how many active calls you have. There is no need to change the CHUNKSIZE as we have chosen the optimal solution (in our testing) for keeping system load down on hardware as minimal as a few hundred megahertz. That said, there's no way you can change the CHUNKSIZE on a Rhino card, it would require a completely different firmware. In my experience, I have seen issues similar to this arise from hard disk activity hogging the bus. Whether it's simultaneous recordings or perhaps a considerable amount of other reading/writing, what ends up happening is the CPU is switching between the Rhino card's interrupt and the IDE/SATA controller interrupt. When one of those interrupts becomes more frequent and holds the bus for too long, that takes time away from the R4T1 and data has to be discarded. We last saw these issues with nVidia hardware in the 2.6.9 kernels, but it's possible some derivative is affecting you. I would suggest investigating other factors that may be affecting system load when your call load increases. Context switches are simply a symptom and you still need to find the culprit. Regards, Bryce Chidester Rhino Equipment Corp. br...@rhinoequipment.com Tel: +1 (480) 621-4000, +1 (877) RHINO-T1 FAX: +1 (480) 961-1826 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch between FXS ports
On Mon, 2005-08-15 at 13:20 -0500, Chris Wade wrote: First, FXS = handset / FXO = telco line. Ditto this. Maybe something like fax-callback; call-in, hangup, Asterisk dials back on the other channel using the CID received - a purely physical solution. Otherwise, have the telco setup a rotary hunt to go between the two lines. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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] 8 FXS in Asterisk Server
On Mon, 2005-08-15 at 23:32 +0200, Roland Zagler wrote: Hello everyone, I want to build an Asterisk Box where i need 8 FXS interfaces to connect 8 phones to. The problem is, that there is only one PCI slot available. What i have is 4 USBs 2.0 interfaces free (if this helps). So here's my question: how am i going to do this? i tried to find any PCI cards supporting 8 FXS interfaces, but without success. does anyone know such hardware? Thanks in advance, Roland Either a T1 card to channel bank with 8 FXS channels, which is expensive but allows for great expandability down the line, or you'll have to look at ATAs/gateways (networking in other words). If the computer only has 1 PCI slot, how would you expect to fit 2 brackets worth of FXS connectors if you went with an internal solution? (I'm thinking 1 PCI slot, 1 backside bracket.) I know there is a Zaptel USB module, but don't know of any hardware or compatibility information. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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] Comedian annoucment files
http://www.voip-info.org/tiki-index.php?page=Asterisk+CMD+voicemail On Fri, 2005-08-12 at 11:37 -0400, kurt x wrote: A user has their unavailable message played and once that message is over the Comedian message is played right after. Is there any way to prevent the Comedian message being played if the user's unavailable/busy message is being played. Thanks, Kurt ___ 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] Incompatible destination (88) Error Message
On Fri, 2005-08-12 at 20:09 +0300, Iraklis Zografos wrote: Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '3118' ] That seems to be fairly clear to me. As I understand it, the Avaya is rejecting the call. Check that 3118 is an acceptable input (i.e. extension) on the Avaya. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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 2.0 Stable release out now
On Thu, 2005-08-11 at 18:06 +0100, Kevin Walsh wrote: Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Justin Selleck wrote: Is asterisk 2.0 real? Running in c#? I see references to it but cannot find it anywhere. r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. r makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. I think you might have replied to the wrong article. Actually, I believe that's part of his signature, seeing as it comes after the '--' denoting a signature, and after his name. Also, I don't believe it's an exact quote, or anything at all like the wiki entry. Rather, this is just an expression of the frustration of people using the 'r' option in Dial() statements where it usually messes things up rather than make them right. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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] dialplan defenition
On Wed, 2005-08-10 at 15:51 +0100, Joao Pereira wrote: exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) Just an observation that you have an invalid address there; you have 1193 instead of 193 I believe. Fix this and I see no reason for your problem to remain. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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 to PSTN
On Tue, 2005-08-09 at 10:59 -0700, Edwin Lam wrote: hi folks. i'm planning to connect * to 120 POTS line. i've done some research on FXO cards but unfortunately most manufacturers only make 4 ports/card. the most i've found is 12 ports. so do i have to get 10 of these cards and setup 3 Asterisk servers (assuming each have 4 free PCI slots) link them together with some insane dialplan? or is there an easier way? any suggestions? comments? remarks? parameters? thx. You're going about things all wrong if you're looking at 120 FXO lines. In such high quantities, you want to look at either 5 FXO channel banks connected to a quad-port and a single-port T1 card[s], or, more sensibly with that many lines, order 5 T1s (24 channels each), or a T2 (96 channels) and a T1 (24), or a T3 (672 channels) from your carrier (all of which require the same number of ports.) If you can't get away from 120 copper pairs, then you'll have to go the channel bank route. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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 to PSTN
On Tue, 2005-08-09 at 12:54 -0700, Edwin Lam wrote: i guess the way to go is using channel banks to convert those to E1 then connect Asterisk that way. further research, how about using these: http://www.welltech.com.tw/product_e_03.htm will that work? Sure, that would work, all 20 of them, which considering they only support 10Mbit ethernet may get kind of crowded (depending on any other traffic and hardware). I think the channel bank route will be the best overall and quite probably cheaper and easier to maintain. My experience is that most channel banks out there are T1 interface (24 channels), but there are some E1 channel banks too which you would only require 4 of. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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] Call Recording with *
On Mon, 2005-08-08 at 17:03 -0400, Craig Bruenderman wrote: I'm attempting to set up call recording with Asterisk. Using automon = *1 ; One Touch Record in features.conf does not appear to be working. I'm using Polycom 501's but when someone dials *1 while in a call, nothing happens. I'm wondering if the phone or Asterisk is even detecting the DTMF. I suspect that is the problem but don't know how to verify or correct. Using Zaptel channels, I know it detects the DTMF (debug output says so) but nothing comes of it, or *0, *2, or any of the other feature codes. Call parking and # transfer work though, so I'm guessing they're simply not implemented yet, as of 1.0.8. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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] Call Recording with *
On Mon, 2005-08-08 at 16:33 -0500, Eric Wieling aka ManxPower wrote: Bryce Chidester wrote: On Mon, 2005-08-08 at 17:03 -0400, Craig Bruenderman wrote: I'm attempting to set up call recording with Asterisk. Using automon = *1 ; One Touch Record in features.conf does not appear to be working. I'm using Polycom 501's but when someone dials *1 while in a call, nothing happens. I'm wondering if the phone or Asterisk is even detecting the DTMF. I suspect that is the problem but don't know how to verify or correct. Using Zaptel channels, I know it detects the DTMF (debug output says so) but nothing comes of it, or *0, *2, or any of the other feature codes. Call parking and # transfer work though, so I'm guessing they're simply not implemented yet, as of 1.0.8. They will never be put into 1.0.x since 1.0.x does NOT get new features. It's bug fix only. Wasn't expecting it to make it in if it wasn't already - merely identifying what version I was using. This isn't a big feature for me, and you can simply write a manger interface to turn on and off monitoring anyways. Just confirmation that this isn't in the 1.0.x branch -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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 TDM card connected to phone lines AND fax line
On Wed, 2005-08-03 at 10:23 +, Andres Tello Abrego wrote: Assign an extension to the fax at extension.conf Create a menu. Why even bother to do that much? Just put the 3rd port/line into its own extension where s automatically dials the fax machine on 4. You can still use 1, 2, and 3 for outbound if you group them and dial with one of zaptel's grouping options. Other idea being to make sure the fax machine picks up first, but this issue's been discussed on the list before. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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] IAXY Voicemailmain problem
On Thu, 2005-21-07 at 23:08 -0500, Steve Maroney wrote: I have the original version of the IAXY. I had it laying around collecting dust, now Im actually putting it to use. When I call my voicemail extension (8500), Before I get the voice prompts from the voicemail app, I hear tones that sound like the caller id tones that are heard when montoring a phone call. While watching my Asterisk CLI, I see this error at the sound of each tone: Jul 21 23:06:03 WARNING[5111]: res_adsi.c:292 __adsi_transmit_messages: Unexpected response to ack: (retry 2) and then after a few tones I see: Jul 21 23:06:04 WARNING[5111]: res_adsi.c:296 __adsi_transmit_messages: Maximum ADSI Retries (3) exceeded and then the app conttinues : -- Playing 'vm-youhave' (language 'en') -- Playing 'digits/9' (language 'en') So Im guessing its something to do with ADSI. So far, I only have this problem when checking voicemail, not for outgoing calls to another voip--pstn gateway. Thank you, Steve Maroney Indeed, that's the Comedian Mail ADSI scripting being sent to the device as in-band FSK tones, just like CallerID. I know in zapata.conf, you can specify adsi=no, but I don't think you can do it for iax. What's more, IIRC, the ADSI scripting is hard-coded into app_voicemail.c and therefore wouldn't be affected by the previously mentioned ADSI setting. So it looks like you're stuck with it, unless either I'm wrong and you can turn off app_voicemail's ADSI functions, or you go ahead and patch up an ADSI-free version. -Bryce [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
Re: [Asterisk-Users] How to send Fax from Asterisk
On Thu, 2005-07-21 at 13:00 +0900, Vic wrote:
Dear all,
I had Tom Rymes and several others suggest how I can implement sending
fax using Asterisk. The idea is to have On-Demand-Fax.
Unfortunately, I wrote down the wrong workflow: the real one is:
1. Person will call our phone number
2. He will be asked to press 1 for Office 1 map, 2 for Office 2 map
and 3 for Office 3 map.
3. User presses 1.
4. User is asked to enter his phone number.
5. User enters his phone number and hangs up.
6. Asterisk calls the number entered by user and sends a fax.
Can it be done?
Thanks,
Vic
Sure, quite easily. Setup said menus that direct through the dialplan
and terminate at something as simple as a System() that executes a
simple shell script that will then create a .call file
in /var/spool/asterisk/outgoing/, specifying the outgoing channel to use
and Application: TxFax. This is mine basically, to give you a start.
exten = x,1,System(/usr/local/bin/asterisk-sendfax ${FAXMACHINE}
${FAXFILE} ${LOCALSTATIONID}/var/spool/asterisk/outgoing/${UNIQUEID}.call)
#/usr/local/bin/asterisk-sendfax ${FAXMACHINE} ${FAXFILE}
${LOCALSTATIONID}
echo Channel: $1
MaxRetries: 0
WaitTime: 20
Context: incoming-fax
Application: TxFax
Data: $2|caller
SetVar: LOCALSTATIONID=$3
This was just a quick and dirty hack I made to try out TxFax and it
works. Just a programming note: as stated on the wiki and in various
docs, it's unwise to output straight into a call file due t timing with
when Asterisk may read it and it isn't finished being written, thus it
is always wisest to output to a temporary directory, then perform a mv
operation to place the whole file at once in the directory (not to
mention, its presence isn't written to the filesystem until all the data
is written).
Good luck!
--
-Bryce
[EMAIL PROTECTED]
NOTICE: The views expressed in this e-mail do not neccesarily reflect
those of my employer, this company, or its employees. This is a personal
e-mail and as such, the opinions expressed are my own.
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Re: [Asterisk-Users] No dialtone - iaxy
On Wed, 2005-07-20 at 08:26 -0400, Ousmane Doukara wrote: Hi, I am unable to get a dialtone from iaxy (the old model). When dial a mailbox, I can see the mailbox app reacting. iaxy gets registered. I can make call and the remote phone can hear me. No sound for iaxy user. ./iaxyprov 192.168.1.134 provinfo 01: 05: 11 d9 0d: 00000004 0f: 4546d2e7 10: 11 d9 06: 6d616c69 07: 636f756d62613738 0c 0000000d Provisioning is 44 bytes Total packet is 58 bytes Got response back from 192.168.1.134 --- dhcp codec:ulaw server:192.168.1.140 user:username pass:pass register -- Any idea ? Sounds like your IAXy is fried. This was/is a fairly common issue with the old model when left on long enough. Presumably, this was the reason for the redesign. Of course, this could be a completely different issue, but the symptoms match. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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 MRTG
On Wed, 2005-07-20 at 16:09 -0600, Scott wrote: I have tried to get MRTG to graph my Asterisk box but have run into a problem. When I run the perl script provided at: http://karlsbakk.net/asterisk/ I get the following error: [EMAIL PROTECTED] asterisk]# ./asterisk-mrtg -h myasteriskip.mydomain.com -v -1 SIP -2 IAX2 -u 109 -p Asterisk Call Manager/1.0 Action: Login Username: 109 Secret: Response: Error Message: Authentication failed My question is what username and password is this script looking for? Thanks, Scott. The one you were supposed to define in /etc/asterisk/manager.conf :-). It's stated in the docs IIRC, and I can assure you it works just fine when all configured. -- -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer, this company, or its employees. This is a personal e-mail and as such, the opinions expressed are my own. ___ 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 Asterisk ring a specific extension based on the number the outside caller dialed?
On Thu, 2005-07-14 at 13:33 -0700, Jeff Ramsey wrote: If I have six channels of a T1 dedicated to Voice, and have 24 phone numbers in a hunt group so that any of the 24 numbers will ring the next available of the six T1 channels, can Asterisk ring a certain extension when a certain number was dialed? For instance, can I dial xxx-xxx-xxx1 and get extension 1, and then dial xxx-xxx-xxx2 and get extension 2? Direct Dialing is what I am trying to accomplish. But still having an IP phone system with auto answer on nights, voicemail, and all of the other features that Asterisk brings to the table. You're looking for DID service (http://www.voip-info.org/tiki-index.php?page=DID). Contact your T1 provider to set this up. -- Bryce Chidester [EMAIL PROTECTED] Rhino Equipment Corp. ___ 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] Odd MOH problem...
When you restarted Asterisk, did you kill the mpg123 processes? -Bryce [EMAIL PROTECTED] NOTICE: The views expressed in this e-mail do not neccesarily reflect those of my employer. This is a personal e-mail and as such, the opinions expressed are my own. On Jul 12, 2005, at 11:52, Patrick Friedel wrote: So I decided, for the formal asterisk rollout, to change over to less commercially-infringing MOH than the prior admin had thrown on the server. (plus: it was blown out and nasty sounding over the phones. Ew.) I changed the files in /var/lib/asterisk/mohmp3 to something else (can't dig up the link, but it was from the voip- info wiki). My musiconhold.conf looks like this: ; ; Music on hold class definitions ; [classes] default = quietmp3:/var/lib/asterisk/mohmp3,-z When I put a phone on hold, I get this in the console: -- Started music on hold, class 'default', on SIP/pjf-51af And yet, the MOH is still the same old song from before when I put a caller on hold. Asterisk has restarted, the phones (snom 360s) don't have their personalized SIP MOH settings set, the offending file has been deleted from the filesystem, I can't find anything else that sets the MOH to a different class, umm. Any ideas? Out of curiousity, I tried setting up this: exten = 6101,1,Answer exten = 6101,2,MusicOnHold(default) Same results. ___ 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] Epia C3 Linux
It's a common (and commonly overlooked) problem and whenever there appears to be no logic behind irrational behavior, the RAM is the first place I look. Because the RAM is effectively changing the running program's code at the bit level, any and all actions are unpredictable, along with their results.-BryceOn Jul 6, 2005, at 08:25, Wiley Siler wrote: Rob, How in the world did you know that… I just ran the memtest86 and it is nothing but error after error….Switched out the ram and I am getting no errors on memtest86 now. I am back in the saddle. Fedora Core 3 is installing as we speak… Thank you! WileyFrom: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Rob Thomas Sent: Tuesday, July 05, 2005 6:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux Sounds to me like bad RAM. Try running memtest (your Fedora CD has it, just type ‘memtest’ at the cd boot prompt) --Rob From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Wiley Siler Sent: Wednesday, 6 July 2005 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux OK. Something is truly rotten in Denmark. I took the 2.5 inch drive out altogether and setup a regular 3.5 IDE drive with a CDROM. BIOS recognizes both. Try to install Redhat 9, it dies. Fedora Core 3 dies, kernel panic. How in Zeus’ Red Ripe Ass did you guys get this to install? Am I going to have to make a custom kernel? To recap… This is a Via Mini-ITX board 800MHz Samuel 2 Processor AKA E-Series C3 (not Eden, this one has a fan) Thanks to all, Wiley PS. AstLinux bombed too… From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Wiley Siler Sent: Tuesday, July 05, 2005 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have attempted FC3, RedHat 9, Mandriva 10, Gentoo 2005.0, Gentoo 2004.3, Debian Mini, Debian woody, and Windows XP. Nothing will install. All see the HDD. All attempt partitioning (XPO seemingly completes), none will install the OS. BIOS posts the correct HDD and all the installers see the HDD. All bomb out immediately after attempting to partition with the exception of Gentoo. The LIVECD will allow me to set a partition table but it dies when I attempt to apply filesystem ext3 to the root partition. I am officially stumped. Thanks for all the input everyone! Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 2:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux It installed directly from the FC3 dvd, no changes...no external drivers required From: Wiley Siler [mailto:[EMAIL PROTECTED]] Sent: Friday, July 01, 2005 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 LinuxDid it require any special work or did you just download the ISO for FC3 and install? Thanks,Wiley From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Michael Stahl Sent: Friday, July 01, 2005 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Epia C3 Linux I have Fedora Core 3 running great on an Epia mobo From: Wiley Siler [mailto:[EMAIL PROTECTED]] Sent: Friday, July 01, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Epia C3 LinuxAnyone know a good distro for an Epia Mobo with the C3 chip? I have been trying to get Debian and Gentoo installed (new to me) and so far having little luck. Does anyone know a good install for this processor/mobo combo? ThanksWiley ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Connect 30 phone lines to asterisk how to
Assuming you mean you have 30 analog POTS lines, the way to go about this would be with a couple channel banks and a quad-T1 (I haven't seen a two-port around, but that's all that is needed) card. For the record, 30 individual analog lines is generally inefficient and would be done more cleanly with an E1 or 2 T1s. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 Disclaimer: I work for a company that makes channel banks. I believe we make a very fine product and are the best out there, but I have kept these views to myself and in no way allowed them to influence the above advice. On Jul 6, 2005, at 12:27, Angel Diaz wrote: Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel. ___ 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] Connect 30 phone lines to asterisk how to
I choose not to acknowledge Sangoma's existence whenever possible. I've had some very poor experiences with the quality of their cards, firmware, and drivers, so I tend to /not/ recommend them. -Bryce On Jul 6, 2005, at 16:24, TC wrote: this would be with a couple channel banks and a quad-T1 (I haven't seen a two-port around, but that's all that is needed) card. not sure how hard you look :) http://www.sangoma.com/products/p_aft-et1-specs.htm 2 T1 spans with daughter board, and an adit 600 with 4 8 port fxo cards would fit the bill if you realy want analog fxo ports ___ 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] #include not working with *1.0.9
Just a thought, but I seem to recall that in the dialplan, inlcude and other similar statements are not prefixed by the hash character (#). Try include = . -Bryce On Jul 4, 2005, at 00:05, [EMAIL PROTECTED] wrote: We are running * V1.0.9 on a demo box. We have set up everything in our dialplan and we have a directory where we store individual extension settings. That directory is called extensions- phones.d and it contains a number of .conf files. In my extensions.conf file I have put a #include extensions-phones.d/*.conf in my [globals] context If we reload and restart *, and then try to dial one of the defined extensions in the included directory, nothing...just Service Unavailable. If I copy and paste a few of the extensions that are in the .conf files directly into the [default] context of the extensions.conf file, the extensions work. So it seems to me that the include statement no longer works in 1.0.9 I figure this is the case, because we were running 1.0.5 and the same config file worked fine. Anybody know what's going on? Brent ___ 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] Buy IP address
Sell 'em quick, 'cause here comes IPv6 and something tells me the market's going to be saturated. Hmmm... what to do with four and a quarter billion (round numbers) addresses... I know, porn sites! -Bryce On Jul 3, 2005, at 11:08, Mark Charlton wrote: On 7/3/05, Jerry Glomph Black [EMAIL PROTECTED] wrote: On Sun, 3 Jul 2005, chawki hammoud wrote: Hi: I have my Asterisk server behind a nat and I want to buy a static IP. Is there a company that sell IP and forward it to IAX file as in the DID service. Any reference or recommendations please? I have a good quantity of IP numbers for sale. Meet me at the Brooklyn Bridge tomorrow at 14:30. Oh yeah, you can have the bridge too, make an offer. 14:30 GMT / BST / EST or MNT? Can I get mine in black?? I've made a fortune on my LiveVoip shares I need to invest wisely. BTW - You'll be able to spot me easily - I'll be the one in trousers and a t-shirt. ___ 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] Connecting two servers - dial string
You're leaving the :1 in the dial expression, which cuts off the
first digit so what's really being dialed to the server is only 88.
-Bryce
On Jul 3, 2005, at 19:01, Joseph wrote:
On Mon, 2005-07-04 at 01:09 +0200, Roland Zagler wrote:
Hi Joseph,
here is how i did it:
iax.conf of server1:
[server2]
type=friend
auth=md5
username=server1
secret=secret
context=default
host=dynamic
defaultip=public ip of server2
deny=0.0.0.0/0.0.0.0
permit=public ip of server2/255.255.255.255
disallow=all
allow=g729,gsm
iax.conf of server2:
[server1]
type=friend
auth=md5
username=server2
secret=secret
context=default
host=dynamic
defaultip=public ip of server1
deny=0.0.0.0/0.0.0.0
permit=public ip of server1/255.255.255.255
disallow=all
allow=g729,gsm
extensions.conf of server1:
exten = _2X.,1,Dial(IAX2/server2/${EXTEN:1},30)
extensions.conf of server2:
exten = _1X.,1,Dial(IAX2/server1/${EXTEN:1},30)
That worked for a while but it seems to me I can not transfer the
call to any extension.
I have a context incoming in extension.conf in incoming server2
[incoming]
exten = 888,1,Goto..
And I dial from server1:
exten = 888,1,Dial(IAX2/server2/${EXTEN,1},30,r)
It keeps telling on the server2 (that accept the call) that request
'@incoming doesn't exist.
When I try without any extension
exten = 888,1,Dial(IAX2/server2,30,r)
and forward the call to context that start with s for example
'office-open'
it keeps telling me that request '[EMAIL PROTECTED]' doesn't exist.
I'm using asterisk 1.0.7.
Why the extension is not being passed to the dialing server?
--
#Joseph
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Re: [Asterisk-Users] New Setup with Analog Phone lines
I think the easiest way to accomplish that would be through a channel bank which could rackmount as well, and interface via T1 to the computer, thus only one PCI card and room to grow to 24 channels, FXO and even FXS. That or go with a 2U rackmount computer. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 On Jun 30, 2005, at 14:53, Chris Gamble wrote: I am looking for hardware recommendations for a new asterisk setup. I want to connect 8 analog phone lines (I think the term used is POTS?), to an asterisk server similar to the way the Wildcard TDM400 with FXO cards would. However, The Asterisk server is rack- mount lacking the needed PCI ports, so I need something that is standalone. What is a good solution for this? Thanks, and please forgive my uncertainty of terminoligy. I am very new to the VOIP / PBX worlds. I thought elves made the phone ring! ___ 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 Setup with Analog Phone lines
Well, I purposely didn't drop names due to what might be perceived as conflict of interest, but I would have to recommend the Rhino Channel Bank with a Digium T1 card. Aside from making the Rhino Channel Bank, this exact configuration works great for us. I'll leave any further selling to off-list e-mails, but definitely check us out. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 On Jun 30, 2005, at 15:26, Chris Gamble wrote: Any chance you could drop some product names for me to further research on the T1/channel bank solution? Preferably something I can get from a vendor in the Texas, USA area -- or is this off-topic? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bryce Chidester Sent: Thursday, June 30, 2005 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] New Setup with Analog Phone lines I think the easiest way to accomplish that would be through a channel bank which could rackmount as well, and interface via T1 to the computer, thus only one PCI card and room to grow to 24 channels, FXO and even FXS. That or go with a 2U rackmount computer. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 On Jun 30, 2005, at 14:53, Chris Gamble wrote: I am looking for hardware recommendations for a new asterisk setup. I want to connect 8 analog phone lines (I think the term used is POTS?), to an asterisk server similar to the way the Wildcard TDM400 with FXO cards would. However, The Asterisk server is rack- mount lacking the needed PCI ports, so I need something that is standalone. What is a good solution for this? Thanks, and please forgive my uncertainty of terminoligy. I am very new to the VOIP / PBX worlds. I thought elves made the phone ring! ___ 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] Setting Caller ID after Dial
The CallerID that is seen by others on calls originating from your PRI is set by your PRI provider; you have no control from Asterisk about this as it gets overridden by the provider. You must contact your carrier and ask them to set the CallerID for all PRI lines to the desired name/number. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 On Jun 29, 2005, at 08:33, Chee Foong Chiew wrote: Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number (which is one of the number selected from the 200 DID numbers). This I can be achieved in asterisk by calling SetCallerID before Dial command. However in the CDR, the caller id number of the number that i set using SetCallerID is always logged and there is no trace of which sip extension is making the call since the caller is always the same. This has become a serious trouble for billing. I have been searching around and could not seems to get a solution. I have tried DIAL_STATUS variable (only work if call is not answered), using 'g' option in Dial command (does not work if calling party hangup first), etc. Is there a solution or work around for this? Thanks in advance CCF ___ How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos http://uk.photos.yahoo.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] Changing Caller ID
The callerid on outside lines is set by your carrier. Talk to them. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 On Jun 26, 2005, at 14:25, Jeff Glassman wrote: I have two X100P clone cards working perfectly in my asterisk box, these lines are off an analog extension from a PRI. They each have DID # assigned to them and I can call the DID and receive calls. When I make an outgoing call using the Zap trunk the caller ID is of the PRI line. Is there any way to change the caller ID to the DID assigned to the line? Thanks in advance, Jeff ___ 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 FXS ports :)
if you're looking for a less expensive, cinch to configure channel bank, I would look at the Rhino Channel Bank (http:// www.channelbanks.com). I must admit that I work for them, but I guarantee Asterisk compatibility and good, personal technical support should you need it. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 On Jun 24, 2005, at 20:33, Carlos wrote: You can always go with like a mediatrix 1124 - 24-port FXS access device configuration is a nightmare with these things they make you use the horrible app that they make you install. And you don't even get the software free to be able to $2500 device that you just bought. Anyways I have 2 of these devices and am very un happy with them ended switching them out with asterisk with a t1 and a channel bank and it works perfect. Carlos Alcantar Race Technologies, Inc. 101 Haskins Way South San Francisco, CA 94080 P: 650.246.8900 F: 650.246.8901 E: carlos at race.com -Original Message- From: Andrew Latham [mailto:[EMAIL PROTECTED] Sent: Thursday, June 23, 2005 10:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] so many FXS ports :) The idea is that the channel bank breaks a T1/E1/J1 in to channels. There are newer channel type devices that are just gateways. I am sure there are benefits of doing it each way. On 6/23/05, Seamus Abshere [EMAIL PROTECTED] wrote: That's what I'm confused about: * two 4 port FXS cards * one 24 port FXS channel bank both, neither, and if both -- why do you need the dual digium cards? shouldn't your channel bank just take MGCP or SIP or something? What am I missing? [EMAIL PROTECTED] said: Shawn guessed correctly; Most likely a channel bank with 24FXS. We have 2 cards each with 4 ports. 1 Zap/23-1 2 Zap/12-1 4 Zap/20-1 Seamus said: this is perhaps a silly question, but how do you have so many zaptel FXS's? do you have six TDM400 cards with four FXS's each? or what am I missing? -- Seamus Abshere Isthmus Group ___ 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 -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ 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] Console ALSA Sound
modules.conf: noload = chan_alsa.so Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 On Jun 17, 2005, at 13:34, Conrad Beckert wrote: Hi ... probably one of those RTFM kind of questions (while I'd be happy to know where a good reference FM is :-) ) Has anyone an idea on how to disable the console sound driver. My problem is that a running asterisk is muting my speakers. Thank you in advance for your help Conrad ___ 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] Changing caller ID on a Zap channel
On Jun 15, 2005, at 14:05, Jeff R Glassman wrote: I have asterisk with two zap channels which are analog ports off a T1. They each have a inward DID number If they are used for outgoing they show the T1 main number not the DID's number. Is there any way to send caller ID of the inward DID number not the main number Jeff Talk to the T1 provider - they're the ones that set it. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 ___ 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 to setup a test number to know my extension number
I used to use the following but Festival is such a load hog I just
NoOp the same info and read off the console.
exten = 789,1,Festival('You are currently calling into context: $
{CONTEXT} as name: ${CALLERIDNAME}. number: ${CALLERIDNUM}.
channel: ${CHANNEL}. This is extension: ${EXTEN}.')
exten = 789,2,Hangup
Regards,
Bryce Chidester
Rhino Equipment Corp.
[EMAIL PROTECTED]SIP: [EMAIL PROTECTED]
+1 (480) 940-1826 x305IAX:
[EMAIL PROTECTED]/305
On Jun 14, 2005, at 06:00, Ronald Wiplinger wrote:
I would like to setup a test number, that speaks back my phone number.
How can I set this up?
bye
Ronald
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Re: [Asterisk-Users] Making Asterisk NOT Pickup a Line when Ringing?
On Jun 14, 2005, at 09:04, Matt wrote: Hi, What do I need to do to get asterisk to NOT pickup a Zap channel when it rings? The channel in question is used for outbound calls only, and all incoming calls are answered by an analog phone elsewhere in the building that does not run through asterisk... so.. either make it not answer.. or make it delay for like 90 seconds.. I've tried wait's.. but it still seems to pickup the channel (even without an answer!) Simply not defining the channel in zaptel/zapata (either or both, but primarily zapata) should do it. If Asterisk doesn't know about the channel, then it won't know it's ringing. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 ___ 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-INFO down?
On Jun 14, 2005, at 08:08, Damon Estep wrote: What is the deal with voip-info.org, is it a commercial agreement or a donation that has worn out its welcome? Needs more bandwidth or a faster (load balanced) server! -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, June 14, 2005 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] VOIP-INFO down? Second day in a row... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Marcel van Kaam, Fonetica Sent: Tuesday, June 14, 2005 8:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VOIP-INFO down? Hi all, Is VOIP-info down? Marcel van Kaam Fonetica Teleservices According the last e-mail from its maintainer, he's in the process of upgrading the software. He may have run into some snags that are taking awhile to resolve. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 From: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VOIP-INFO Date: June 9, 2005 22:45:14 MST To: asterisk-users@lists.digium.com Reply-To: asterisk-users@lists.digium.com Voip-info is back up -- in-spite of Murphy's law. This was phase I (install latest version of O/S) of an upgrade to improve performance and functionality. Hopefully with Phase II we will see much better performance and new functions. For those that asked, the primary voip-info-org sponsor: www.commpartners.us provides a dedicated server, bandwidth and hosting in their Las Vegas data center. Its slow not for any lack of resoruces, but because the software used is rather resource intensive. Jim James H. Thompson [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
Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
On Jun 14, 2005, at 12:39, Wiley Siler wrote: Which then presumably leads to higher overselling in the home market since use is presumed lower. Also there are often restriction on the line like no Ips given for servers and no servers allowed. I doubt they really care if we can afford it persay... I think it is just a matter of what pricepoint to what feature set. W There's also the fact that a lot of companies charge LESS for home access than for a business, under the assumption that the business will utilize it more, and/or can afford the higher price. It's no so much can afford it but they're more willing to pay a higher price. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 ___ 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
[OT] Why not use both? WAS: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Just a thought but, why not leverage both simultaneously and boost the overall speed? Clients will only notice a slowdown when one or the other goes down, but QOS will usually be better. At least you wouldn't be wasting $80/$60 a month on something not used. Even better is that no intervention would be necessary - both connections were live to begin with so you're simply dropped to 50%. I looked into doing this myself with WiFi and dialup connections in addition to cable, but seeing as my neighbours have cable connects and I don't have a dedicated line for dialup (not to mention the intolerably slow speeds for even one computer) I haven't implemented it. However, I have met quite a few that spread their connects across various DSL, cable, T1, and other frame-relays so it certainly is doable. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 On Jun 11, 2005, at 10:42, Joseph wrote: Another alternative is to get another connection in addition to DSL for example Cable Connection. That is what we have, our main connection is DSL and we have a backup Cable connection, if one connection goes down you switch to another. It had happened to us in a past DSL went down, 10min. and we were on Cable High Speed. So price wise it is a good arrangement as well: DSL 60CAD Cable Hight Speed (7MB down / 1Mb up) at 80CAD Not to mention the down is limited to restarting your eth0 on your server and update you DNS to new IP if you are running web-server. -- #Joseph On Fri, 2005-06-10 at 23:39 -0400, Peter A. Solomon wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barton Fisher Sent: Friday, June 10, 2005 9:27 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart ** If your looking at wanting to use QOS or Multiprotocol Label Switching on the same line, then a T is the way to go. You don't mention the equipments though so it's hard to answer your question. How many calls, Data VOIP, Protocol? Tier One ISP? You get what you pay for, it all depends up what you need. ___ 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] VOIP-INFO
Same here. There goes my quick reference... at least Google has cached copies and a better search! On Thu, 2005-06-09 at 14:33 -0700, Chris Coulthurst wrote: Anyone else unable to get to www.voip-info.org? Site is returning 'connection refused' here. Chris Coulthurst [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] Windows IAX Softphone
You might try IAXComm. It's a bit immature but works fairly well on Windows, and is cross-platform as well. However, I've found SIP clients to be better generally and better supported. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305 IAX: [EMAIL PROTECTED]/305 On May 23, 2005, at 09:08, Jeromy Grimmett wrote: Is there a softphone for windows that supports IAX? I can't seem to find anything out there...maybe im looking in the wrong places... Jeromy Grimmett VoipEmpire.com [EMAIL PROTECTED]___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] MusicOnHold Loudness/Distortion
[Cross-posted and re-sent; it really sounds bad and needs resolution ASAP] For whatever reason, the music on hold is extremely distorted and loud. It didn't used to be this way and I haven't changed anything, yet it persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can anyone help with this, or has anyone seen this? The mp3s play fine on any computer and haven't changed since they did work. Those wishing to hear for themselves, feel free to call extension 8800 at the number/addresses below. Thank you, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305 IAX: [EMAIL PROTECTED]/305 ___ 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 IAXy from Digium
I wonder if once the new ones come out, I can purchase the old in large enough quantities such that when they fail, popping in a replacement would be no sweat and relatively cheap too? I too, of course, would be interested in anyone's results with the new model. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305 IAX: [EMAIL PROTECTED]/305 On May 19, 2005, at 07:47, Robert Webb wrote: I was just browsing Digium's web site and noticed they are taking orders for the new IAXy. Has anyone purchased and tested one of these yet?? I have thought about buying one for testing, but want to make sure it isn't going to be a flop like the last one. Robert ___ 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] MusicOnHold Loudness/Distortion
For whatever reason, the music on hold is extremely distorted and loud. It didn't used to be this way and I haven't changed anything, yet it persists. This is on all the channels we use (SIP, IAX2, Zap, ALSA). Can anyone help with this, or has anyone seen this? The mp3s play fine on any computer and haven't changed since they did work.Those wishing to hear for themselves, feel free to call extension 8800 at the number/addresses below.Thank you,Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305 IAX: [EMAIL PROTECTED]/305 ___ 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] Static on TDM Zaptel FXO
That type of echo is usually caused by incorrectly (or not at all) tuned gain settings in zapata.conf. I don't know what kind of phones you're using, but for Asterisk to even be able to detect DTMF tones on our Sayson / Aastra 390s and 480s, our FXS channels are set to -5.0 on both rx and txgain. If you're using externally-powered phones (as in not your ordinary joe-schmoe analog phone), I have found that they're usually pretty hot (loud) and Asterisk can't understand what is said. Good luck! Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 On May 17, 2005, at 19:00, Gregory Wiktor - ADCom Corp. wrote: On a recompile of the kernel I now get a 99.98 average. Static is gone, although quality so far seems not quite there yet. I am also experiencing an odd local echo. I can hear a slight echo locally, but the other end sounds fine, and the other end does not get echo. Even with the pots disconnected, you can hear it. The static would be on all calls. Hooking up a normal phone was ok. The sipsip phones are perfect too, it was only happening on the zap channel... Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Parr Sent: Monday, May 16, 2005 8:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Static on TDM Zaptel FXO On 5/16/05, Gregory Wiktor - ADCom Corp. [EMAIL PROTECTED] wrote: Hello All, I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy static. Even with the pots line disconnected, if I do a dial I still get static. This way I know it's not the line, but rather something on the card. I tried alternate pci slots. This card has a power connector, does anyone know what the power requirements are? The unit is in a small case with a 2.4ghz p-4 and 512mb ram, on an intel board with 533fsb. All other functions are fine. I am using the latest CVS on Debian 2.6test Anyone experience this? Have you tried a different phone? Does the static appear immediately when you pick up the phone? Or on the second or third time? ___ 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 - fax - spandsp
On May 15, 2005, at 00:38, Jean-Yves Avenard wrote:HelloOn 15/05/2005, at 4:40 PM, Steve Underwood wrote:span=1,1,0,ccs,hdb3,crc4 The second parameter now says "treat this E1 as the first priority as the clock source". Your box should lock itself to the PSTN's clock. If that makes no sense, the bottom line is "this is good". :-) Hum.. an interesting side effect to using the E1 as a primary clock source, is that I can't remove the wcte11xp kernel module anymore...it hangs the machine if I do so. Any ideas on how to do that?I have the same trouble with wct1xxp. I'd just chalked it up to a PCI bug or other low-level hardware problem with the Digium card. I've simply learned not to, though it would be nice to not have to learn work-arounds.Regards,Bryce ChidesterRhino Equipment Corp.[EMAIL PROTECTED] SIP: [EMAIL PROTECTED]+1 (480) 940-1826 x305 IAX: [EMAIL PROTECTED]/305___ 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 - fax - spandsp
On May 15, 2005, at 00:55, Jean-Yves Avenard wrote: Hi On 15/05/2005, at 5:51 PM, Bryce Chidester wrote: I have the same trouble with wct1xxp. I'd just chalked it up to a PCI bug or other low-level hardware problem with the Digium card. I've simply learned not to, though it would be nice to not have to learn work-arounds. This only happen if I have something like: span=1,1,0,ccs,hdb3,crc4 If I have: span=1,1,0,ccs,hdb3,crc4 I'm actually using a T1 to a ChannelBank so my span line looks like: span=1,0,0,esf,b8zs then it doesn't hang... Doubt it's a PCI bug Seems rather fishy that no matter medium or timing source, rmmod'ing the module could hang the machine. Maybe I'll look into this more when I get the logic analyzer hooked up. Luckily I have an electronic switch I can control remotely to turn the machine off/on ! Mine's in the closet at the end of the office... so far to walk, 30ft/ 10m :-P. Well, that and fortunately I don't find myself doing it often - once a month average so far. Regards, Bryce Chidester Rhino Equipment Corp. [EMAIL PROTECTED]SIP: [EMAIL PROTECTED] +1 (480) 940-1826 x305IAX: [EMAIL PROTECTED]/305 ___ 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
