Re: [asterisk-users] How to do Automatic Daylight Saving on Grandstream GXP-2000
If you have automated the configuration process, all you have to do is:1) Set option P75 (Daylight savings time) to 0 or 1 accordingly in the configuration file.2) Regenerate the compiled configuration file(s).3) Make sure they are in the TFTP or Web server.4) Reboot the phones so they read the new configuration file.- DanielOn Oct 30, 2006, at 9:21 AM, Zeeshan Zakaria wrote:Hi, I'd set the daylight saving option to yes on all the GXP-2000 phones, but apparantly it doesn't move it an hour back on last sunday of October. So now I am stuck will all the phones showing the wrong time. Isn't there an option so that it'll automatically update daylight savings? Thanks-- Zeeshan A Zakaria ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: 1.4 on mac OSX 10.4.8
You can get wget for OSX from DarwinPorts (http://wget.darwinports.com/) On Oct 17, 2006, at 4:13 PM, Martin Joseph wrote: On 2006-10-17 01:06:25 -0700, Tzafrir Cohen [EMAIL PROTECTED] said: On Tue, Oct 17, 2006 at 12:57:46AM -0700, Martin Joseph wrote: SVN Trunk doesn't currently build on OSX (10.4.8). If you're in for stability now, try branches/1.4 and *not* trunk. This will eventually become beta3, rc or 1.4.0. OK, I like that idea and didn't know that existed. So thanks for that. I did: cd /usr/src svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 LOTS of files come in... Then I did: ./configure make clean make To this point all seems well... make install This proceeds to a point and then: make -C sounds install make[1]: Entering directory `/usr/src/asterisk/sounds' /bin/sh line 2: wget: command not found make[1] ***[/var/lib/asterisk/sounds/.asterisk-core-sounds-en- gsm-1.4.3]Error 1 There is no wget as far as I see on this system... The 1.40beta 2 didn't do this, but both trunk and branch do. Is this a new change in the make file that is causing this? Thanks for your wisdom and help, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream SX2000 attended tranfer
We can do attended transfers on the GXP-2000 just fine with a single account. When you have a call on Line 1, simply press Line 2 (Line 1 will be put on hold automatically) and press SEND. Once the other party picks up, you announce the call and then press TRNSFR and then press Line 1. - Daniel On Sep 20, 2006, at 11:12 AM, Faris Raouf wrote: magnus wrote: Hi all, could anyone share how to perform attended transfers with Asterisk and Grandstream SX2000's - we are able to perform blind transfers with no problem, but attended transfers fail - is it necessary to set two line identities on the phones to be able to do this? Appreciate all input, thanks - Magnus Funny you should ask -- I was going to ask the exact same question about the GXP-2000 (is that the model you mean or is there a new similar phone?). At any rate they both seem to have the same problem: In order to do an attended transfer on the Grandstreams we have to have two accounts defined on the phone (both on separate usernames/ numbers in our case - maybe you can do it with one?), one on Line 1 and one on Line 2. Call comes in on Line 1. Put caller on hold. Dial person you want to transfer to on Line 2. Then transfer. I've tried pressing Line 2 until the identity of Line 1 comes up - i.e. reuse Line 1 - but this does not work. Instantly fails. The instruction manual gives completely different instructions but these simply do not work. And what is not clear is how the transfer works when using the strange two account situation - is the transfer going * - phone - person you are transferred to once transferred? (can reinvite = no incidentally) or is the phone This is all completely unlike the case with a Polycom where it just lets you transfer with no problems and just one line. I'm using the latest stable firmware on the Grandstreams - it has been like this for all firmware versions I've used for over a year now. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GXP2000 - Blind Transfer Hangs Up Call
I have a couple of clients with a bunch of GXP-2000. They can do attended transfers with no problems. However, there are times that the party to transfer to is simply not at their desk and the party wanting to transfer the call knows that. In these cases, they'd like to blind transfer the call so that the voicemail picks up. The problem is that when they do so, the call drops. This happens in all the phones and all three clients of mine that have these phones. They could leave with just attended transfer. The problem is that (at least on these phones), the TRFR button will not work until the party picks up and not while it's ringing. So, currently, they need to let the phone ring until the voicemail picks up before they can transfer the call. Is there a trick to blind transfer a call on these phones (or in general)? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GX-2000, doesn't send calls to free lines
You need to enable call waiting on the phone's config.- DanielOn Sep 8, 2006, at 9:35 AM, Zeeshan Zakaria wrote:First call is answered by LINE1, but if this line is still busy and a second call comes in, it doesn't go to LINE2, instead called listens asterisk message, "all lines are busy, please leave your message after the tone". I tried resetting phone to factory default setting too, but still it does the same. Same extension if configured on X-TEN, it works with no problems for all available free lines. Grandstream phone should go upto 11 lines and only for 12th call should say that lines are busy. What I need to configure in this Grandstream phone which I haven't figured out yet. I've firmware 1.1.0.16-- Zeeshan A Zakaria ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys PAP2 Ring Settings
That worked great!. I was using Ring_WaveForm and I guess it's case sensitive and the correct form should be Ring_Waveform. Thanks, Daniel On Aug 25, 2006, at 11:48 PM, Shanon Swafford wrote: This works for me on my SPA-3000 ver 3.1.10(GWd). Ring_WaveformTrapezoid/Ring_Waveform Then back to default. Ring_WaveformSinusoid/Ring_Waveform PAP2-NA shouldn't be any different. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Friday, August 25, 2006 6:27 PM To: Non-Commercial Discussion Asterisk Subject: [asterisk-users] Linksys PAP2 Ring Settings I have a few PAP2-NA that are being mass configured using the instructions on the wiki for the Sipura mass configuration. However, I need to make sure the following settings are in place as follow: Under the Regional Tab, I need the Ring Waveform to be Trapezoid instead of Sinuzoid and the Synchronized Ring to be Yes instead of No. I made an entry in the XML file for Synchronized_Ring which works just fine. However, no matter what I use for the Ring Waveform (Waveform, Ring_Waveform, Regional_Ring_Waveform), the setting is always the default (Sinuzoid). Does anyone know what the XML tag name/ settings need to be for changing the Ring Waveform? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys PAP2 Ring Settings
I have a few PAP2-NA that are being mass configured using the instructions on the wiki for the Sipura mass configuration. However, I need to make sure the following settings are in place as follow: Under the Regional Tab, I need the Ring Waveform to be Trapezoid instead of Sinuzoid and the Synchronized Ring to be Yes instead of No. I made an entry in the XML file for Synchronized_Ring which works just fine. However, no matter what I use for the Ring Waveform (Waveform, Ring_Waveform, Regional_Ring_Waveform), the setting is always the default (Sinuzoid). Does anyone know what the XML tag name/ settings need to be for changing the Ring Waveform? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GXP-2000 Call Transfer Problem
I have a client with about 24 GXP-2000. Everything seems to be working fine except one particular behavior of the blind transfer. Whenever anyone makes an outbound call, they can transfer the call between extensions either blind or attended with no problems. However, whenever an incoming call is answered, they can do attended transfers with no problem. Unfortunately, blind transfer doesn't seem to work. Whenever an incoming call is answered, they try to do a blind transfer by hitting the TRF button, dialing the extension and then hitting SEND. However, when they do that, the extension they want to transfer to does not ring at all, and they simply remain on the next line appearance (e.g. Line 2) listening to dialtone while the caller remains on hold on the line (e.g. Line 1). All the phones are running firmware 1.1.0.16 (the latest as of about 1 month ago). Does anyone have any ideas as to why this is happening and how to solve it? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoipNow 1.2.0 Beta
Is there anyway to reach ANYONE on the phone (in the US) at 4PSA? Every time you call their office, you just need to leave a message. It prompts you to enter an extension (if you know any), but whenever you press ANY digit, it simply goes straight into voicemail. Is this company for real? - Daniel On Jul 29, 2006, at 3:37 AM, Dinesh Nair wrote: On 07/29/06 02:49 Miles Scruggs said the following: http://forum.4psa.com/showthread.php?t=455 Take it for a ride around the block and tell them what you think. As powerful as the config files, and command line interface is, there is is there anywhere we can take a look at screenshots without having to download the entire package ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http:// www.openmalaysiablog.com/ +==oOO--(_)--OOo ==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | += + ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI Scripts and CDR
I have an Perl AGI script which accepts inbound calls and offers an IVR service. Depending on certain options that are selected on the IVR, the script is supposed to dial-out an external number, and therefore, basically, conference the original caller with an external number. That part is working just fine. The problem I have is that when the script executes the Dial command (somewhere before the script the Answer command was already issued), I don't see ANY records in the CDR of the outbound call. I can tell there was an outbound call because of the call duration, but that's no real reliable indicator. Is there a way to trigger a new CDR entry when the IVR script dials the call? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Scripts and CDR
That is exactly what I'm looking for. Thanks, Daniel On Jul 31, 2006, at 9:23 PM, Moises Silva wrote: may be you are looking for asterisk application ForkCDR(), more info in voip-info.org Regards On 7/31/06, Daniel Salama [EMAIL PROTECTED] wrote: I have an Perl AGI script which accepts inbound calls and offers an IVR service. Depending on certain options that are selected on the IVR, the script is supposed to dial-out an external number, and therefore, basically, conference the original caller with an external number. That part is working just fine. The problem I have is that when the script executes the Dial command (somewhere before the script the Answer command was already issued), I don't see ANY records in the CDR of the outbound call. I can tell there was an outbound call because of the call duration, but that's no real reliable indicator. Is there a way to trigger a new CDR entry when the IVR script dials the call? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http:// www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Scripts and CDR
Well, I guess I spoke too soon. The ForkCDR works great in creating a new CDR record. However, the problem is that the dst column of my CDR still shows the original dst value instead of the newly dialed number. Is there anyway to fix this? Thanks, Daniel On Aug 1, 2006, at 1:43 AM, Daniel Salama wrote: That is exactly what I'm looking for. Thanks, Daniel On Jul 31, 2006, at 9:23 PM, Moises Silva wrote: may be you are looking for asterisk application ForkCDR(), more info in voip-info.org Regards On 7/31/06, Daniel Salama [EMAIL PROTECTED] wrote: I have an Perl AGI script which accepts inbound calls and offers an IVR service. Depending on certain options that are selected on the IVR, the script is supposed to dial-out an external number, and therefore, basically, conference the original caller with an external number. That part is working just fine. The problem I have is that when the script executes the Dial command (somewhere before the script the Answer command was already issued), I don't see ANY records in the CDR of the outbound call. I can tell there was an outbound call because of the call duration, but that's no real reliable indicator. Is there a way to trigger a new CDR entry when the IVR script dials the call? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http:// www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 Softphone
Looking for a SIP or IAX softphone for a call center application that can do G729 codec. Any recommendations? Ideally it would do screen pops, meaning that it will understand the URL option of the Dial command. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Softphone
I have the eyeBeam softphone but I don't see G729 in the list of available codecs (BTW, this is the paid version not X-Lite). Any clues? Thanks, Daniel On Jul 24, 2006, at 12:00 PM, Guillermo Salas M. wrote: On Mon, 2006-07-24 at 11:41 -0400, Daniel Salama wrote: Looking for a SIP or IAX softphone for a call center application that can do G729 codec. Any recommendations? Ideally it would do screen pops, meaning that it will understand the URL option of the Dial command. Give a try to Eyebeam at www.counterpath.com , it supports video and voice with g729. BOL Siphone is freeware that supports video/voice and uses de g723.1 codec you can download it at http://www.bol2000.com/download/sipphone/ Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN Incoming Route
Check the default context defined in zapata.conf which is where incoming calls will go to. It may be going to a context that you are not aware of. - Daniel On Jul 4, 2006, at 3:46 AM, Pierre du Plessis wrote: Greetings, I have installed a new FXO card but even though there's no incoming route, it answers the line after 2 to 3 rings. If I do create an incoming route, the same happens, but it never rings the ring group or extension I enter. It's almost as if the card acts as a modem. The caller hears nothing, just silence. I have a VoIP incoming route which works perfect. Any comments will be greatly appreciated... Many thanks, P. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 to SIP Gateway
I'm trying to setup an Asterisk box as an H323 to SIP gateway. Basically, I'd like to receive traffic in H323 and forward to another Asterisk box (on the same network) using either IAX2 or SIP so that the second Asterisk box communicates with other gateways using SIP. Therefore, if I receive a request from a remote H323 gateway to dial a particular number, the H323-to-SIP gateway should forward the request to the Asterisk SIP gateway, who would simply terminate the call according to whatever rules are defined in the context. Can anyone tell me how can this be done? I setup chan_oh323 on an * box and played with the configurations but have not been able to make it all work. I can place connect the two * boxes using SIP-to-SIP as well as IAX2-to-IAX2 just fine, but have not gotten the H323 to work. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Suggested Phone
If pricing is an issue, I've had very good experience with GXP-2000. Otherwise, I really like the SPA-941/2.- DanielOn Jun 28, 2006, at 3:05 PM, Forrest Beck wrote: We are looking to deploy asterisk at one of our locations that will have about 50 phones. I have been buying different phones to test there quality and feature set. So far we have a Grandstream 2000Grandstream HandyTone 488Cisco 7912Polycom SoundPoint IP And we are looking at getting a Linksys SPA-942 Anyone have a favorite? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ExternalIVR vs AGI
I have an Perl AGI script that acts as an IVR for my Asterisk box. Basically, it simply plays audio files to the caller, collecting DTMF input and logging every DTMF input into a database table, simply to document every step or option selected by the caller. One thing is that in addition to playing audio files, it also, at some point, plays SayUnitTime command. This IVR has constantly about 20 simultaneous callers 24x7. Would it be more resource efficient to migrate this to ExternalIVR? What are the pros/cons of using ExternalIVR vs using my Perl AGI. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances
Beautiful. Will test and give you comments. Nice work. - Daniel On Jun 26, 2006, at 2:55 PM, Dustin Wildes wrote: Daniel Salama wrote: Dustin, any updates on this? Thanks, Daniel Hey Daniel! Yes - just posted the link. I appologize for the delay. Here's the link to the forum as well, if anyone is interested. This should compile and run on Asterisk-1.2.4 and higher. http://www.vecsector.com/phonecall/valet/ Enjoy! Dustin Wildes VecSector, LLC 1.912.422.7082 x101 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances
Dustin, any updates on this? Thanks, Daniel On Jun 23, 2006, at 1:07 PM, Dustin Wildes wrote: shadowym wrote: That feature is called Bridged (or Shared) line appearance. That is one of the things Asterisk cannot do and nobody seems very interested in making it do that because it is apparently not easy. There has been some talk about implementing it but so far there does not seem to be any progress. http://forums.digium.com/viewtopic.php?p=23974#23974 I will be posting the code later today. --Dustin Wildes ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP 2000 - BLF and Hold/Hangup Answering
I had the same problem some time ago. Make sure call waiting is NOT disabled. This will make the phone receive more calls on the other lines. - Daniel On Jun 23, 2006, at 1:29 AM, Corporate IT Solutions - Michael Dunne wrote: I have a network of GXP 2000 phones and would like to know if there is a way to configure the phones so that if there is one person talking, and another call comes in then they can hold/hangup that call and take the incoming call. At the moment, when a call comes in and the phone is offhook, then that phone is completely unavailable for that ring session, any call coming in after that call will of course ring. Is this limited to the GXP series or does the SNOM phones fix this, etc. Any advice is appreciated of course. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 and Shared Line Appearances
I have a client with 20 GXP-2000s. Everything seems to be working fine. However, after a couple of weeks of use, the client is having a hard time adjusting to the new IP based phone systems and only misses one feature from their old Lucent system. That is, they had 8 analog lines before and all their old Lucent phones showed a button for each line. So, it was easy for anyone to say, pick up line 2 or anyone to see which lines were in use. Is it possible to use the GXP-2000 line buttons or extension buttons to show the lines in use, shared by all phones. Since the client is purchasing 8 virtual lines, I have them restricted in a call group and also with incoming and outgoing call limits. Is it possible for all the GXP-2000s to show that line 1 is in use, and so on? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Kernel 2.4 / 2.6 and timer
I've read in different places that if I want to do trunking and meetme on Asterisk I need to have a reliable timer. People have recommended that I install a Digium board, even if I don't have any circuits connected to it, just to get a reliable timer. However, I've also read that if I'm using kernel 2.6, I don't need to have a Digium board. I have a few servers that need to do trunking and meetme and I don't have ANY PSTN-type circuits. I do everything via VoIP. All my servers are running kernel 2.6. Do I really need to have these boards? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WRTG54GS Capacity
Given that the NSLU2 can't do trunking, do you think that a PIII 733Mhz, 128MB RAM will do? Thanks, Daniel On Jun 15, 2006, at 4:15 AM, Tim Panton wrote: On 15 Jun 2006, at 02:59, Daniel Salama wrote: Does anyone know how many simultaneous calls can a WRTG54GS handle? Assuming SIP phones are connected locally using G711.u codec and the WRTG54GS connects to a remote Asterisk server using IAX2 trunking using GSM codec. Very few (2 perhaps) - You will be transcoding on the WRTG54. On that sort of box you need to stick to a single codec. In your case I guess GSM. If you want to transcode, you will need a bigger cpu. If your phones support it, I'd use GSM everywhere, since your original problem was bandwidth. Do take a look at the OpenSlug on the nslu2 - The nice thing about the 'Slug' is that you can add a USB harddrive for swap and voicemail, so it is more 'expandable' than the WRTG54 I should warn you I have never tried trunking IAX on my slug, I will do at some point Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ECHO Tutorial
Is there anyone that could explain to me the phenomenon of Echo or at least point me where I can learn more? Why is this affecting the VoIP world so much and not the regular PSTN analog world? What does the PSTN industry have that they can handle such high volume of calls and there is no echo problem? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER
I have been reading about integrating Asterisk with SER to help Asterisk deal with large volume of registrations (mainly). I was planning on fronting Asterisk with SER for that purpose. Not that I have the traffic at this moment, but because I wanted to get the infrastructure in place. However, my providers are using G711 codec and I offer G711 and G729 to my clients because they don't have the best broadband service available. So, if my clients are talking G729, I suppose I will have to always keep Asterisk in the media path so as to do codec translation. Is that correct? I was also planning on using SER's nathelper, but if Asterisk _HAS_ to be in the media path, there may not be a need for SER's nathelper. Is this assumption correct? If my purpose of using SER is basically to alleviate registration load and help route (possibly load balance) traffic among multiple Asterisk servers as well as SIP providers, do I really need SER? Would you recommend it? Granted, I have been running both Asterisk and SER as separate systems for a while and they both seem very stable to me. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOS Scores and LCR
Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random calls so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOS Scores and LCR
Thanks for the lecture. Yes, I thought MOS was more of a perception type of measurement, but I can't say I know enough to opinion-ate and thus the reason for the question. Also, thanks for the links. They seem helpful. Since I have several scripts in Cacti and Nagios, I'm gonna see if I can come up with something that could create some performance data per provider. Then I'll give it a such at integrating that with Asterisk, unless someone out there has done something like it. Thanks, Daniel On Jun 17, 2006, at 2:00 AM, trixter aka Bret McDanel wrote: On Sat, 2006-06-17 at 01:26 -0400, Daniel Salama wrote: Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random calls so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? MOS (Mean Opinion Score) is generally a bunch of people sitting there listening to audio and rating it 1-5 (there is a newer method that is twice as good becuase it goes 1-10, basically all values are double). Its their opinion. This generally cant be dont automagically and still be MOS. You can try to track frame drops and other things on your end to rate call quality and try to come up with something, but that technically isnt MOS. AFAIK asterisk doesnt keep statistics of jitter, frame drops or anything else, that might be a good project for someone to take on, especially if you have multiple providers so you can rate quality in a more meaningful way. The human ear really isnt the best tool for much of this. http://searchnetworking.techtarget.com/sDefinition/ 0,,sid7_gci786677,00.html http://www.tmcnet.com/tmcnet/articles/2005/voice-quality- measurement-voip-alan-clark-telchemy.htm http://channels.lockergnome.com/it/archives/ 20050715_voipqos_mos_mean_opinion_score_explained.phtml -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WRTG54GS Capacity
It sounds nice, but, how many calls can you get on the NSLU2? Say the SIP phones are talking either G711.u or GSM only and the IAX trunk is GSM only. Thanks, Daniel On Jun 15, 2006, at 4:15 AM, Tim Panton wrote: On 15 Jun 2006, at 02:59, Daniel Salama wrote: Does anyone know how many simultaneous calls can a WRTG54GS handle? Assuming SIP phones are connected locally using G711.u codec and the WRTG54GS connects to a remote Asterisk server using IAX2 trunking using GSM codec. Very few (2 perhaps) - You will be transcoding on the WRTG54. On that sort of box you need to stick to a single codec. In your case I guess GSM. If you want to transcode, you will need a bigger cpu. If your phones support it, I'd use GSM everywhere, since your original problem was bandwidth. Do take a look at the OpenSlug on the nslu2 - The nice thing about the 'Slug' is that you can add a USB harddrive for swap and voicemail, so it is more 'expandable' than the WRTG54 I should warn you I have never tried trunking IAX on my slug, I will do at some point Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WRTG54GS Capacity
Wow! Can anyone comment on this? If this was the original suggestion, can anyone confirm that trunking DOES work on the NSLU2? Thanks, Daniel On Jun 15, 2006, at 10:47 AM, Kristian Kielhofner wrote: Daniel Salama wrote: It sounds nice, but, how many calls can you get on the NSLU2? Say the SIP phones are talking either G711.u or GSM only and the IAX trunk is GSM only. Thanks, Daniel Unless someone has ported zaptel (and ztdummy) to run on the mipsel you won't be doing any trunking... -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 Audio Quality
Wow! 22Kbps of overhead? Are you sure? That sounds like way too much overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any other suggestion? Thanks, Daniel On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote: G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for 256k upstream you should be able to handle 8 calls but this is in ideal conditions. If you were to use IAX and enable trunking then you would use 30kbps for the 1st call and 10kbps for each additional call. See http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth +iax2 On Wed, 2006-06-14 at 04:17, Daniel Salama wrote: I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 Audio Quality
That may not be such a bad idea. I've read people trying to put Asterisk on a WRTG54 or something like that. Would that be good? I guess I could do SIP in the office and trunk via IAX2 and save on bandwidth plus internal calls would be local. I tried to upgrade them to 512K but because they're borderline to the 18K feet, the best BellSouth can offer them is 256K. I'm talking to Comcast to see if they can get their broadband service which can go up to 768K. Thanks, Daniel On Jun 14, 2006, at 12:45 PM, Tim Panton wrote: Well, with 16 phones, it might be worth putting a 'satellite' asterisk in their office, have it handle local transfers, and act as a protocol converter, talking sip to the phones and (trunked) IAX2 to the outside world. An embedded low power system would do fine. You might even get away with an nslu2, but I'm not sure it has the RAM for 16 calls. A better alternative is to get them to upgrade the DSL to 512 uplink. Tim. On 14 Jun 2006, at 17:11, Daniel Salama wrote: Wow! 22Kbps of overhead? Are you sure? That sounds like way too much overhead. I can't use IAX2 because the GXP-2000 are SIP phones :( Any other suggestion? Thanks, Daniel On Jun 14, 2006, at 4:37 AM, Gareth Blades wrote: G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for 256k upstream you should be able to handle 8 calls but this is in ideal conditions. If you were to use IAX and enable trunking then you would use 30kbps for the 1st call and 10kbps for each additional call. See http://www.voip-info.org/wiki/index.php?page=Asterisk +bandwidth+iax2 On Wed, 2006-06-14 at 04:17, Daniel Salama wrote: I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)
Can anyone explain to me what this means: Jun 14 19:46:10 NOTICE[7391]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 66.175.1.1 When I try to make a call from certain IP phones, I see that message on the console. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)
Mainly GXP-2000 (with silence suppression off) and Eyebeam (with Enable microphone noise reduction off) Thanks, Daniel On Jun 14, 2006, at 7:55 PM, Mike Fedyk wrote: Comfort noise is the sound you hear from the phone to assure the user that there is still a connection to the other end. It is there to keep you from hearing no sound through the speaker and thinking you have been disconnected. Check your phone's config for comfort noise or silence suppression and turn it on or off respectively. What phone model(s) do you see this with? Daniel Salama wrote: Can anyone explain to me what this means: Jun 14 19:46:10 NOTICE[7391]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 66.175.1.1 When I try to make a call from certain IP phones, I see that message on the console. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WRTG54GS Capacity
Does anyone know how many simultaneous calls can a WRTG54GS handle? Assuming SIP phones are connected locally using G711.u codec and the WRTG54GS connects to a remote Asterisk server using IAX2 trunking using GSM codec. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
Would you mind telling me how to setup the GXP-2000's VLAN/QoS settings with the DES-1226G? I just purchased the DES-1226G and want to make sure I setup it up right. I don't have the ability to run separate wiring for the PC and the phone and that's why I need this help. Thanks, Daniel On Jun 7, 2006, at 9:52 PM, Mike Fedyk wrote: I have heard good things about the D-Link DES-1226G switch ($150 at newegg). If you can run a separate cable to the computer and phone. If you can't run the extra cables, then configure your phone to tag itself as part of the voip vlan and let the switch tag everything else as the computer vlan. I happen to have asterisk running as a router, so I use it doing QoS with tc (traffic control) and wondershaper set to prioritize based on port ranges. I sent a patch to the debian bug tracking system a while back with a few improvements -- I should check on that. It basically prioritizes smaller packets before larger packets with ~8 levels of priority and groups of sizes for the packets. Just doing that automatically handles 80% of the need for prioritization without specifying port ranges for the sip rtp packets. Mike Daniel Salama wrote: They are extremely casual web surfers. Just have their Outlook client opened checking email every minute. Email traffic is very low. They are all connected to the same switch. It's a Netopia DSL router/modem/switch for the BellSouth DSL service. The computers are connected to the PC port behind the GXP-2000. Any suggestions? Thanks, Daniel On Jun 7, 2006, at 8:49 PM, list mail wrote: What do they do on the internet? Heavy surfing, large transfers, myspace. How are these units connected to the network? Are they passing through the same switch? I don't think it is the phones... On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote: Mike, I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM: Jun 7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms) Jun 7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (66ms / 500ms) Jun 7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms) Jun 7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (68ms / 500ms) Jun 7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO LAGGED! (1114ms / 500ms) Jun 7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now REACHABLE! (90ms / 500ms) Jun 7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1077ms / 500ms) Jun 7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (72ms / 500ms) Jun 7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO LAGGED! (1077ms / 500ms) Jun 7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now REACHABLE! (73ms / 500ms) As you can see, it only happens to a couple of their phones and at random times. They're behind a DSL circuit. I don't know if it's because their DSL line is going up/down. They don't necessarily claim their Internet goes down, however, they are not constantly check it. What would you (or anyone else) suggest? Thanks, Daniel On Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote: Do you have multiple phones going down at the same time? If so, monitor them with qualify=500 in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 Audio Quality
I have a client with about 16 GXP-2000. They complain that the audio quality is terrible after 2 or 3 simultaneous conversations. They are behind DSL 1.5Mbps down and 256Kbps up. Because they are using G711.u codec, I know they upstream bandwidth is the limiting factor and they most likely won't be able to have more than 3 simultaneous conversations, and if they're surfing the net and/or checking email, things will only get worse. So, I purchased some g729 codec licenses and forced their sip peer configuration to g729 codec. We made sample test calls and were able to make 8 simultaneous calls. On the eighth call, the audio started to sound choppy. Then we dropped the eighth call and tested with 7. We could hear just fine on the GXP-2000 but the remote end heard us a bit choppy and/or with a robot-like voice. So, we kept dropping calls until they were of acceptable quality. My question is, if they were using g729 which, in theory uses 8kbps plus overhead, they should have been just fine handling eight calls. All the computers were turned off on the network, so there shouldn't have been any other traffic but VoIP. Does anyone have any ideas? How can I improve their audio quality? I requested BellSouth to upgrade their capacity but because of where they are located, the best they can get is 900Kbps/256Kbps, so the upstream continues to be the limiting factor. I purchased a Dlink-1226G switch to allow me to control QoS on the LAN. I also upgraded their Netopia DSL router to the latest firmware which allows me to configure VLANs and DiffServ. All the computers are connected to the PC port on the phone because there is no available second wiring. Can anyone suggest how to configure the QoS settings on the phones, the Dlink and the Netopia? While there was no traffic on the wire, pinging from/to the Asterisk box gave me about 47ms latency. When we went passed the 4th call, the latency started increasing significantly and when we got to 8 calls, the latency was up in the 2000ms. Obviously, if anything I did in the QoS configuration gave VoIP a priority, then ICMP packets would have the lowest priority and I could understand that to be the reason for such result. However, I'm not sure I configured QoS properly and that's why I'm asking for help. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as Wholesale
This seems to be only for prepaid calling cards. Is there something that can also handle prepaid and postpaid multi-tenant SIP phones?- DanielOn Jun 12, 2006, at 5:00 PM, William Piper wrote:www.asterisk2billing.org On 6/12/06, Wasif [EMAIL PROTECTED] wrote: Hi,I need to use Asterisk as a switch which can handle wholesale traffic withbilling. Please advice me how I can I implement this. Thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 MultiPurpose Keys
Wow! Awesome. This template is much more complete than the one on GS's download page. Thanks, Daniel On Jun 9, 2006, at 10:26 AM, Gareth Blades wrote: Yes you can as long as you have at least the 1.0.2.13 firmware. I have attached the template. The multi-purpose key settings are at the end. On Fri, 2006-06-09 at 14:41, Daniel Salama wrote: Is it possible to program the multi-purpose keys on a GXP-2000 remotely via a TFTP configuration file? If so, what are the parameters to put in the configuration file? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users gxp2000_config_1.0.2.13.txt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 MultiPurpose Keys
That's great. GS support people are great, but I had asked him how to set other parameters that I see on the web and they told me they didn't know. That I should look through the wiki or other web sources. Anyway, that's great to know. Thanks, Daniel On Jun 10, 2006, at 5:16 AM, Phil Blundell wrote: For future reference, I think the Grandstream config files can program any parameter that's included in the web interface. If you want to set something that isn't in the template, you can use view source on the web form to figure out the name of the option: the field names in the HTML are the same as the ones that go in the config file. p. On Sat, 2006-06-10 at 02:06 -0400, Daniel Salama wrote: Wow! Awesome. This template is much more complete than the one on GS's download page. Thanks, Daniel On Jun 9, 2006, at 10:26 AM, Gareth Blades wrote: Yes you can as long as you have at least the 1.0.2.13 firmware. I have attached the template. The multi-purpose key settings are at the end. On Fri, 2006-06-09 at 14:41, Daniel Salama wrote: Is it possible to program the multi-purpose keys on a GXP-2000 remotely via a TFTP configuration file? If so, what are the parameters to put in the configuration file? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users gxp2000_config_1.0.2.13.txt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 MultiPurpose Keys
Is it possible to program the multi-purpose keys on a GXP-2000 remotely via a TFTP configuration file? If so, what are the parameters to put in the configuration file? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Virtual PBX Billing and Management Software
Is there any open source software capable of managing Asterisk to offer Virtual PBX services to multiple clients, including billing? Or is there a combination of open source initiatives that offer this? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: GXP-2000 (steer clear)
While I would agree with you, the price difference between a GXP-2000 and a Polycom 430 or a Thomson ST-2030. These latter units are, at least, twice as expensive as the GXP-2000.BTW, I never heard of the Thomson ST-2030, but it looks _really_ nice.Thanks,DanielOn Jun 7, 2006, at 8:09 AM, Louis-David Mitterrand wrote:I've only had bad experiences with these phones and steer clear of them. In the same price range you can now get the Thomson ST-2030 or Polycom 430 for a much, much better user experience. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
They don't all go down at the same time, or at least, my client hasn't noticed. I just added the qualify option. Let's see how that goes.As for the SPA-841, I have a client with a few of them and he cannot stop complaining about the bad audio quality. I replace a couple with a PAP-2 and another one with the GXP-2000 and he claims the quality to be incredibly better for both the PAP2 and the GXP-2000. He hasn't complained about the problems I mentioned on the GXP-2000 - yet :)Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time? If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. Also I have heard some phones have trouble with broadcast packets (at least this has been said about the spa-841 on the wiki). You should strongly consider putting them on a separate vlan to avoid any issues like that. In the future, for phones under $100 then look at the spa-841 phones. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun 7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (66ms / 500ms)Jun 7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (68ms / 500ms)Jun 7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO LAGGED! (1114ms / 500ms)Jun 7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now REACHABLE! (90ms / 500ms)Jun 7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1077ms / 500ms)Jun 7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (72ms / 500ms)Jun 7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO LAGGED! (1077ms / 500ms)Jun 7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now REACHABLE! (73ms / 500ms)As you can see, it only happens to a couple of their phones and at random times. They're behind a DSL circuit. I don't know if it's because their DSL line is going up/down. They don't necessarily claim their Internet goes down, however, they are not constantly check it.What would you (or anyone else) suggest?Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time? If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
The complete opposite. The user complaints that either they cannot hear the remote party well or the remote party cannot hear them well. Sometimes it works and sometimes the volume is very low and that's why they cannot hear.- DanielOn Jun 7, 2006, at 1:35 PM, Mike Fedyk wrote:What specifically were the voice quality complaints about the spa-841 phones? The only thing I have noticed is calls can be louder than expected. What else have you seen? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
Latest firmware installed and problem with handset. They don't use headset nor speakerphone. Thanks, Daniel On Jun 7, 2006, at 3:14 PM, John Novack wrote: Daniel Salama wrote: snip As for the SPA-841, I have a client with a few of them and he cannot stop complaining about the bad audio quality. Latest/last firmware upgrade? Handset? speaker phone? headset? I find the handset quite acceptable Speaker phones are a can of worms, with so many issues not related to the phones the SPA-841 might as well not have a display. Is the 94x any better? seems without backlighting, any are next to useless. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
No changes whatsoever. Unplugged the spa and replaced it with a gxp. I haven't tweaked any RTP or QoS parameters for I don't have any documentation on it :( Thanks, Daniel On Jun 7, 2006, at 3:44 PM, Mike Fedyk wrote: Did you try setting the RTP packet time size to 0.020? Also I would look at the trunk, provider or internet connection before the phones I started suspecting the phones. I have had the same problems with providers, and the conversations sound great from one location to another over the internet, but once it hits a provider, the sound quality drops. That is not the fault of the phones. Are you sure you didn't change anything else when you switched from the spa-841 phones? Daniel Salama wrote: The complete opposite. The user complaints that either they cannot hear the remote party well or the remote party cannot hear them well. Sometimes it works and sometimes the volume is very low and that's why they cannot hear. - Daniel On Jun 7, 2006, at 1:35 PM, Mike Fedyk wrote: What specifically were the voice quality complaints about the spa-841 phones? The only thing I have noticed is calls can be louder than expected. What else have you seen? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
They are extremely casual web surfers. Just have their Outlook client opened checking email every minute. Email traffic is very low.They are all connected to the same switch. It's a Netopia DSL router/modem/switch for the BellSouth DSL service. The computers are connected to the PC port behind the GXP-2000.Any suggestions?Thanks,DanielOn Jun 7, 2006, at 8:49 PM, list mail wrote:What do they do on the internet? Heavy surfing, large transfers, myspace. How are these units connected to the network? Are they passing through the same switch?I don't think it is the phones...On Jun 7, 2006, at 12:32 PM, Daniel Salama wrote:Mike,I added a qualify=500 on those phones. My client has peers 100218 thru 100222 (a total of 5 phones). Below is the messages log since I activated it this morning at 8:30AM:Jun 7 10:59:21 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 10:59:31 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (66ms / 500ms)Jun 7 11:02:32 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1075ms / 500ms)Jun 7 11:02:42 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (68ms / 500ms)Jun 7 11:35:15 NOTICE[3648] chan_sip.c: Peer '100222' is now TOO LAGGED! (1114ms / 500ms)Jun 7 11:35:25 NOTICE[3648] chan_sip.c: Peer '100222' is now REACHABLE! (90ms / 500ms)Jun 7 11:48:20 NOTICE[3648] chan_sip.c: Peer '100219' is now TOO LAGGED! (1077ms / 500ms)Jun 7 11:48:30 NOTICE[3648] chan_sip.c: Peer '100219' is now REACHABLE! (72ms / 500ms)Jun 7 12:24:51 NOTICE[3648] chan_sip.c: Peer '100221' is now TOO LAGGED! (1077ms / 500ms)Jun 7 12:25:01 NOTICE[3648] chan_sip.c: Peer '100221' is now REACHABLE! (73ms / 500ms)As you can see, it only happens to a couple of their phones and at random times. They're behind a DSL circuit. I don't know if it's because their DSL line is going up/down. They don't necessarily claim their Internet goes down, however, they are not constantly check it.What would you (or anyone else) suggest?Thanks,DanielOn Jun 7, 2006, at 8:07 AM, Mike Fedyk wrote:Do you have multiple phones going down at the same time? If so, monitor them with "qualify=500" in sip.conf to see if they hit that limit. If you see more than one go down within a short period of time, you have network problems. Check the quality of the network switches they have. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000
I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I see Got SIP response 486 Busy back on the console. Is there a way to make the phone receive calls on all 4 lines? 2) Is there any more documentation as to the tftp configuration file? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
I enabled call-waiting from the tftp configuration and it now works. What firmware are you using and where can I get it? My client complaints that the phone stops working every once in a while with no explanation. My client says that he could be using the phone with no problem and a few minutes later, when he wants to make a call, the phone will always give a fast busy after pressing the fourth digit. My workaround to him was to reboot the phone. That seems to solve the problem, however, it's not practical to have that problem in an office environment with 18 GXP-2000. Any ideas what the problem could be? Thanks, Daniel On Jun 6, 2006, at 6:26 PM, Mike wrote: I can't say why you're having this problem, but I can tell you that my phone can receive (and make) multiple calls easily. It might have more to do with Asterisk than the GXP2000. I am using the latest release firmware, not a beta. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: June 6, 2006 4:12 PM To: Non-Commercial Discussion Asterisk Subject: [Asterisk-Users] GXP-2000 I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I see Got SIP response 486 Busy back on the console. Is there a way to make the phone receive calls on all 4 lines? 2) Is there any more documentation as to the tftp configuration file? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000
Well, these are encouraging words :)You're basically telling me that I should tell my client to buy other phones. I agree that you cannot compare these phones with Cisco or Polycom. After all, like you said, what do you expect for under $90. However, the fact is that my client just recently invested in these and it will be hard, if not impossible, for me to tell my client to swap them for Polycoms or something else at a much higher cost.I have heard complaints from my client about the speakerphone and they are now, I guess, getting used to picking up the handset :). I have heard any echo problems so far. What bothers me the most is that the phone stops working often (multiple times per day). By this I mean that my client won't be able to dial anything successfully. As soon as 3 or 4 digits are entered, they get a fast busy. To solve it, they need to reboot it. It sounds as if these phones were running Windows instead of Linux :)Anyway, what firmware did you use that solved so many of your problems?Thanks,DanielOn Jun 6, 2006, at 10:31 PM, Erick Baum wrote:We setup a company with 50 of these phones and had my client not been as understanding as they were, that could have put me out of business. What an unbelievable nightmare. This was about 8 months ago when the firmware was so bad the phone was a better paper weight than anything else. Since then, they've fixed a lot of problems and made a lot of the features work like they're supposed to. But we still have issues with them quite frequently. From phones that need to be rebooted occationally, to ones that just drop calls, or do nothing when you pickup the receiver... lots of little qwerks. We even experience their poor grounding problem every once in a while when you get a small static shock from the phone which cases it to reboot. I don't think there's any firmware that can fix that. We had to get several phones RMA'd because they just plain died. The worst ongoing issue has been the echo and the really crappy speakerphone. The customer is pretty much used to it now. But we're slowly replacing them with Polycom's as new people come on and as others just get fed up. Unfortunately one of the phones met it's doom by way of a hammer. But I guess, what do you expect for under a hundred bucks. Erick On 6/6/06, Daniel Salama [EMAIL PROTECTED] wrote: I enabled call-waiting from the tftp configuration and it now works.What firmware are you using and where can I get it? My client complaints that the phone stops working every once in awhile with no explanation. My client says that he could be using thephone with no problem and a few minutes later, when he wants to make a call, the phone will always give a fast busy after pressing thefourth digit. My workaround to him was to reboot the phone. Thatseems to solve the problem, however, it's not practical to have thatproblem in an office environment with 18 GXP-2000. Any ideas what the problem could be?Thanks,DanielOn Jun 6, 2006, at 6:26 PM, Mike wrote: I can't say why you're having this problem, but I can tell you that my phone can receive (and make) multiple calls easily. It might have more to do with Asterisk than the GXP2000. I am using the latest release firmware, not a beta. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Daniel Salama Sent: June 6, 2006 4:12 PM To: Non-Commercial Discussion Asterisk Subject: [Asterisk-Users] GXP-2000 I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I see Got SIP response 486 "Busy" back on the console. Is there a way to make the phone receive calls on all 4 lines? 2) Is there any more documentation as to the tftp configuration file? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aste
Re: [Asterisk-Users] Attended call transfer with GXP-2000
Lacy,I am in a similar situation, except that my users are extensions aware. However, I'd love to know how you solved your problem since call transfer seems a bit complicated at the moment.Thanks,DanielOn Jun 2, 2006, at 6:51 AM, Lacy Moore - Aspendora wrote:Kerry, so to park a call, you would put the line you are on on hold, hit line 2, dial 700 (or whatever your park ext is) listen to find out the number, then hit TRNF and hit line 1. That's a lot of work to park a call. I just realized this might be a problem. I'm about to put 4 phones in an open office (all users are in the same office area with no walls or cubicles separating them). They will be answering the phone and then having to put people on hold for someone else. They haven't grasped the concept of extensions yet (this will be a complete shock to them). Blind transfer with the speeddial may be a better option. I was thinking of using parking. I may need to look at pickup groups. On 3/16/06, Kerry Garrison [EMAIL PROTECTED] wrote: If you have Line 1 on hold, and you on a call on Line 2, then hitting TRNFand hitting Line 1 will transfer Line 2 to Line 1. Same concept as Conference. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Mimmus Sent: Thursday, March 16, 2006 7:30 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Attended call transfer with GXP-2000 Can someone explain me attended transfer with Grandstream GXP-2000? Hitting TRNF button, I get: Dial number (BLIND) or Select line (ATTENDED) What's the exact meaning of 'Select line'? Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FreePBX virtualization
Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreePBX virtualization
Not really looking to give the client web access. Just trying to make my life easier :) Thanks, Daniel On May 25, 2006, at 2:07 PM, Kerry Garrison wrote: You can by creating different contexts and using the Administrators function allow them to modify some of the settings themselves. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, May 25, 2006 10:42 AM To: Non-Commercial Discussion Asterisk Subject: [Asterisk-Users] FreePBX virtualization Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreePBX virtualization
Any alternate open-source solutions? On May 25, 2006, at 2:17 PM, Douglas Garstang wrote: Yes, but it fast becomes a provisioning and management nightmare. -Original Message- From: Kerry Garrison [mailto:[EMAIL PROTECTED] Sent: Thursday, May 25, 2006 12:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FreePBX virtualization You can by creating different contexts and using the Administrators function allow them to modify some of the settings themselves. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, May 25, 2006 10:42 AM To: Non-Commercial Discussion Asterisk Subject: [Asterisk-Users] FreePBX virtualization Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreePBX virtualization
I'm listening - daniel On May 25, 2006, at 2:34 PM, Shane Burrell wrote: We have a revision of this that we use in house. We are interested in working with others on a version 2 skipping some of the mistakes of our first version and using a better model. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, May 25, 2006 2:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FreePBX virtualization Any alternate open-source solutions? On May 25, 2006, at 2:17 PM, Douglas Garstang wrote: Yes, but it fast becomes a provisioning and management nightmare. -Original Message- From: Kerry Garrison [mailto:[EMAIL PROTECTED] Sent: Thursday, May 25, 2006 12:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FreePBX virtualization You can by creating different contexts and using the Administrators function allow them to modify some of the settings themselves. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, May 25, 2006 10:42 AM To: Non-Commercial Discussion Asterisk Subject: [Asterisk-Users] FreePBX virtualization Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with MeetMe
What I have discovered is that my motherboard only supports usb-ohci and not usb-uhci. Reading on the wiki, it says that ztdummy requires usb-uhci. To make things worse, I slapped in a TDM22B just to get timer support, only to discover that the machine kept crashing because of a hardware conflict with my RAID controller. Really weird! Anyway, my only three other options are: 1) Compile kernel 2.6, which I'd hate to do 2) Replace either the motherboard or the RAID controller, which is worse than option 1 3) Setup a separate machine where I can install the TDM22B and dedicate it just for MeetMe and may be a couple of other things. I may give this a shot. I still need to figure out how to do this, so if you guys can provide any sample configs I'd appreciate it. Any other suggestions you guys may have? Thanks, Daniel On May 12, 2005, at 10:08 AM, Chris wrote: It sounds like you don't have USB support compiled in the kernel. Chris - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2005 11:55 PM Subject: Re: [Asterisk-Users] Problem with MeetMe Chris/BJ, I am running REL3 with kernel 2.4.21-27.0.4.ELsmp. I enabled USB devices in the BIOS. Here are the problems I'm seeing: [EMAIL PROTECTED]: ~ modprobe zaptel [EMAIL PROTECTED]: ~ modprobe ztdummy /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod ztdummy failed [EMAIL PROTECTED]: ~ lsmod Module Size Used byNot tainted zaptel183104 0 (unused) soundcore 7044 0 (autoclean) iptable_filter 2412 0 (autoclean) (unused) ip_tables 16544 1 [iptable_filter] e1000 77884 2 floppy 57552 0 (autoclean) sg 37388 0 (autoclean) usbcore81152 1 ext3 89992 2 jbd55156 2 [ext3] 3w-9xxx 570016 3 sd_mod 13936 6 scsi_mod 115240 2 [sg 3w-9xxx sd_mod] [EMAIL PROTECTED]: ~ modprobe -r zaptel [EMAIL PROTECTED]: ~ modprobe usb-uhci /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod usb-uhci failed [EMAIL PROTECTED]: ~ modprobe usb-ohci [EMAIL PROTECTED]: ~ modprobe zaptel [EMAIL PROTECTED]: ~ modprobe ztdummy /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod ztdummy failed [EMAIL PROTECTED]: ~ lsmod Module Size Used byNot tainted zaptel183104 0 (unused) usb-ohci 23176 0 (unused) soundcore 7044 0 (autoclean) iptable_filter 2412 0 (autoclean) (unused) ip_tables 16544 1 [iptable_filter] e1000 77884 2 floppy 57552 0 (autoclean) sg 37388 0 (autoclean) usbcore81152 1 [usb-ohci] ext3 89992 2 jbd55156 2 [ext3] 3w-9xxx 570016 3 sd_mod 13936 6 scsi_mod 115240 2 [sg 3w-9xxx sd_mod] It still won't load ztdummy. I can't get usb-uhci to work. I read on the wiki that ztdummy requires uhci. What's the difference between ohci and uhci? Thanks, Daniel On May 11, 2005, at 9:16 PM, Chris wrote: I forgot because I haven't moved to a 2.6 kernel. Chris - Original Message - From: BJ Weschke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2005 6:56 PM Subject: Re: [Asterisk-Users] Problem
[Asterisk-Users] switch in extensions.conf
Can anyone provide more information on switch or point me to where I can find more about it? The only I've been able to find on the wiki is: http://www.voip-info.org/tiki-index.php?page=Asterisk+-+dual+servers and towards the bottom of (section Forwarding to another Asterisk): http://www.voip-info.org/wiki-Asterisk+config+extensions.conf Some of the questions I have are: 1) If I have an asterisk machine being used only as a VoIP gateway to the PSTN, and I have multiple asterisk machines behind it waiting to make or receive calls through/from the gateway, can I have multiple switch statements in the same context, so that if the gateway tries to contact asterisk machine 1 and it's not available, try the next one and so on? 2) How to configure the other asterisk machines to use the VoIP gateway for all outbound calls? I read somewhere that you cannot have circular references using switch, but I'm not sure if it refers to what I'd like to do 3) Assuming the VoIP gateway cannot contact any of the other asterisk machines, can the voip gateway put calls on hold and continue trying and only disconnect or play busy tone after some pre-defined period of time? Thanks, Daniel ___ 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] Problem with MeetMe
Adam, See my comments below: On May 12, 2005, at 10:12 PM, Adam Goryachev wrote: On Thu, 2005-05-12 at 10:35 -0400, Daniel Salama wrote: What I have discovered is that my motherboard only supports usb-ohci and not usb-uhci. Reading on the wiki, it says that ztdummy requires usb-uhci. To make things worse, I slapped in a TDM22B just to get timer support, only to discover that the machine kept crashing because of a hardware conflict with my RAID controller. Really weird! This is always an interesting question, but when you say the machine kept crashing is that a bios issue, or a linux (and modules) issue? ie, does it not bootup, does it work fine then after x minutes you get a oops, or a what?? You don't provide *ANY* details did you try putting the various cards in different slots? What type are the slots you are using? etc... Sorry for being vague. Yes, I tried different slots 32-bit and 64-bit PCI slots. The machine boots fine. After a few minutes of working, there was a constantly scrolling error message on the console about a TDM interrupt conflict. At that point, the machine was unresponsive, other than the error message continuously scrolling. I had to hard boot the machine. In one occasion, I was able to see a message that was 3ware RAID controller stopped working. I couldn't really diagnose it so well because this was during work hours and the machine had to be in production, so I had to remove the TDM22B board and let the machine continue working without it (and without MeetMe either). Anyway, my only three other options are: 1) Compile kernel 2.6, which I'd hate to do Why, I don't see what it is about 2.6.x kernel that is so bad compared to 2.4.x kernel??? 2.6.x is the current stable linux kernel version IMHO, it just works better I agree. I also prefer 2.6.x. That's what I run on my Debian machines. However, REL3 does not support 2.6.x. They don't even have an RPM for it. I would have to get one from kernel.org and that would void any support Red Hat may provide. 2) Replace either the motherboard or the RAID controller, which is worse than option 1 Depends, if you end up with a better machine in the end? though I don't see how (1) is bad at all, so yes, this would be worse :) It's a pretty good machine I'm using (dual Xeon). The only thing that would make it better would be to make MeetMe work :) Just my 0.02c worth... Regards, Adam Thanks, Daniel ___ 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] Inbound Calls Codec
I'm noticing by watching the CLI that my inbound calls coming via T1s on a TE410P are using GSM codec. Why wouldn't it use ULAW as default? How can I make it use ULAW as default? Thanks, Daniel ___ 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] Problem with MeetMe
I'm trying to configure some meet me rooms in asterisk 1.0.2 and I'm getting the following problem: -- Executing MeetMe(SIP/3210-38a9, 0224|qM) in new stack == Parsing '/etc/asterisk/meetme.conf': Found May 11 14:05:46 WARNING[20050]: chan_zap.c:757 zt_open: Unable to open '/dev/zap/pseudo': No such device or address May 11 14:05:46 ERROR[20050]: chan_zap.c:6687 chandup: Unable to dup channel: No such device or address May 11 14:05:46 WARNING[20050]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device May 11 14:05:46 WARNING[20050]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') I have the following in meetme.conf [rooms] conf = 0224 What could be happening? I don't have any digium cards on the machine. lsmod shows: Module Size Used byNot tainted zaptel182080 0 The other modules are not related to zaptel or asterisk. ls -l /dev/zap/ps* shows: crw-r--r--1 root root 196, 255 May 11 10:07 /dev/zap/pseudo Any ideas? Thanks, Daniel ___ 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] Problem with MeetMe
I have a feeling it has something to do with that as well. I would really hate to having to install a digium card just for the timer source. As I just posted on a earlier email, I even tried building zaprtc but could not succeed. I think it may have something to do with the fact that my stock kernel has RTC support built in. At the same time, I'd rather stay away from building custom kernels. Any other suggestions? Thanks, Daniel On May 11, 2005, at 3:30 PM, John Dunham wrote: I believe it has to do with the fact that you may not have a Digum card in your * server. It uses it for timing. We have several * boxes and only one of them has the Digum card and therefore is where the conference rooms all go to. We use IAX between the boxes and works across that connection, but the destination has to be on the unit with the Digium card. John Dunham -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Daniel Salama Sent: Wednesday, May 11, 2005 1:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Problem with MeetMe I'm trying to configure some meet me rooms in asterisk 1.0.2 and I'm getting the following problem: -- Executing MeetMe(SIP/3210-38a9, 0224|qM) in new stack == Parsing '/etc/asterisk/meetme.conf': Found May 11 14:05:46 WARNING[20050]: chan_zap.c:757 zt_open: Unable to open '/dev/zap/pseudo': No such device or address May 11 14:05:46 ERROR[20050]: chan_zap.c:6687 chandup: Unable to dup channel: No such device or address May 11 14:05:46 WARNING[20050]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device May 11 14:05:46 WARNING[20050]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') I have the following in meetme.conf [rooms] conf = 0224 What could be happening? I don't have any digium cards on the machine. lsmod shows: Module Size Used byNot tainted zaptel182080 0 The other modules are not related to zaptel or asterisk. ls -l /dev/zap/ps* shows: crw-r--r--1 root root 196, 255 May 11 10:07 /dev/zap/ pseudo Any ideas? Thanks, Daniel ___ 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] Problem with MeetMe
I don't have any of those modules loaded. I don't have any digium cards so wcfxo and wcfxs won't load. When I do make clean; make install in /usr/src/zaptel, it doesn't build ztdummy. If I manually build ztdummy by running make zdummy.o and then try to insmod, it fails with a bunch of unresolved symbols. I think it may have to do with the fact that I don't have uhci support. I tried building zaprtc but when I do make load, it fails. dmesg shows rtc: I/O port 112 is not free. Any other ideas? Thanks, Daniel On May 11, 2005, at 3:41 PM, Chris wrote: No timing?Do you have the wcfxo, wcfxs, or ztdummy loaded? - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2005 1:09 PM Subject: [Asterisk-Users] Problem with MeetMe I'm trying to configure some meet me rooms in asterisk 1.0.2 and I'm getting the following problem: -- Executing MeetMe(SIP/3210-38a9, 0224|qM) in new stack == Parsing '/etc/asterisk/meetme.conf': Found May 11 14:05:46 WARNING[20050]: chan_zap.c:757 zt_open: Unable to open '/dev/zap/pseudo': No such device or address May 11 14:05:46 ERROR[20050]: chan_zap.c:6687 chandup: Unable to dup channel: No such device or address May 11 14:05:46 WARNING[20050]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device May 11 14:05:46 WARNING[20050]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') I have the following in meetme.conf [rooms] conf = 0224 What could be happening? I don't have any digium cards on the machine. lsmod shows: Module Size Used byNot tainted zaptel182080 0 The other modules are not related to zaptel or asterisk. ls -l /dev/zap/ps* shows: crw-r--r--1 root root 196, 255 May 11 10:07 /dev/zap/ pseudo Any ideas? Thanks, Daniel ___ 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] TDMoE vs IAX2
I just came across http://www.voip-info.org/tiki-index.php? page=Asterisk%20TDMoE and seemed very interesting. It prompted me to question whether it would be more efficient to do TDMoE or IAX2. The application is very simple. I have two asterisk boxes. One is strictly a gateway to the PSTN using multiple T1s. The second is where all my extensions and dialplan are defined (the workhorse for the office). Currently, I have them connected using IAX2. Would it be more efficient if I connect them using TDMoE? Since the second asterisk box does not have any digium cards, I think I'm having timing problems (see my previous posts about MeetMe). By doing TDMoE, will it solve the dependency of having to have a local timing source on the asterisk box? Thanks, Daniel ___ 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] Problem with MeetMe
Chris/BJ, I am running REL3 with kernel 2.4.21-27.0.4.ELsmp. I enabled USB devices in the BIOS. Here are the problems I'm seeing: [EMAIL PROTECTED]: ~ modprobe zaptel [EMAIL PROTECTED]: ~ modprobe ztdummy /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod ztdummy failed [EMAIL PROTECTED]: ~ lsmod Module Size Used byNot tainted zaptel183104 0 (unused) soundcore 7044 0 (autoclean) iptable_filter 2412 0 (autoclean) (unused) ip_tables 16544 1 [iptable_filter] e1000 77884 2 floppy 57552 0 (autoclean) sg 37388 0 (autoclean) usbcore81152 1 ext3 89992 2 jbd55156 2 [ext3] 3w-9xxx 570016 3 sd_mod 13936 6 scsi_mod 115240 2 [sg 3w-9xxx sd_mod] [EMAIL PROTECTED]: ~ modprobe -r zaptel [EMAIL PROTECTED]: ~ modprobe usb-uhci /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod usb-uhci failed [EMAIL PROTECTED]: ~ modprobe usb-ohci [EMAIL PROTECTED]: ~ modprobe zaptel [EMAIL PROTECTED]: ~ modprobe ztdummy /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb- uhci.o failed /lib/modules/2.4.21-27.0.4.ELsmp/kernel/drivers/usb/host/usb-uhci.o: insmod ztdummy failed [EMAIL PROTECTED]: ~ lsmod Module Size Used byNot tainted zaptel183104 0 (unused) usb-ohci 23176 0 (unused) soundcore 7044 0 (autoclean) iptable_filter 2412 0 (autoclean) (unused) ip_tables 16544 1 [iptable_filter] e1000 77884 2 floppy 57552 0 (autoclean) sg 37388 0 (autoclean) usbcore81152 1 [usb-ohci] ext3 89992 2 jbd55156 2 [ext3] 3w-9xxx 570016 3 sd_mod 13936 6 scsi_mod 115240 2 [sg 3w-9xxx sd_mod] It still won't load ztdummy. I can't get usb-uhci to work. I read on the wiki that ztdummy requires uhci. What's the difference between ohci and uhci? Thanks, Daniel On May 11, 2005, at 9:16 PM, Chris wrote: I forgot because I haven't moved to a 2.6 kernel. Chris - Original Message - From: BJ Weschke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2005 6:56 PM Subject: Re: [Asterisk-Users] Problem with MeetMe Only if you're on a 2.4 kernel. A 2.6 kernel doesn't require USB for it's timing source. On 5/11/05, Chris [EMAIL PROTECTED] wrote: Edit the Makefile for the zaptel drivers. You will see two commented lines that say ztdummy. Uncomment them and rebuild. Once you install the rebuild, do a modprobe ztdummy and you should be good to go. BTW, you do need an active USB for ztdummy to load. Chris - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2005 3:28 PM Subject: Re: [Asterisk-Users] Problem with MeetMe I don't have any of those modules loaded. I don't have any digium cards so wcfxo and wcfxs won't load. When I do make clean; make install in /usr/src/zaptel, it doesn't build ztdummy. If I manually build ztdummy by running make zdummy.o and then try to insmod, it fails with a bunch of unresolved symbols. I think it may have to do with the fact that I don't have uhci support. I tried building zaprtc but when I do make load, it fails. dmesg shows rtc: I/O port 112 is not free. Any other ideas? Thanks, Daniel
[Asterisk-Users] Zaptel problems on Debian
I just installed a TE410P on a Debian Sarge system running kernel 2.6.11-1-686-smp. Zaptel and Asterisk seem to be working fine. However, I have a couple of problems with the TE410P and Zaptel. First, the TE410P is showing me red alarms on 2 of the 4 T1s. This board (the TE410P) was just moved from another machine running REL3 and all 4 T1s were working there. I don't know why only two T1s are being recognized on the Debian system. Secondly, I no longer have zttool, which I used to have on REL3. How can I query from Zaptel the status of my T1s? Thanks, Daniel ___ 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] Setting Variables
Is it possible to set a variable for an IAX device in iax.conf that can be read from the dial plan (extensions.conf)? If so, can you explain? Thanks, Daniel ___ 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 Variables
Thanks. May be I wasn't clear enough.
I have local IAX users as well as remote IAX users (local means to
the local network and remote means off net). When I define their
profile in iax.conf, I would like to set a variable like LOCAL=1 or
LOCAL=0 so that in extensions.conf, I could access them as ${LOCAL},
but LOCAL is never defined in extensions.conf. Instead it is defined
in iax.conf.
I haven't seen any facilities to define variables in iax.conf, so
that's why I asked.
Thanks,
Daniel
On May 10, 2005, at 7:14 PM, Alfredo Manrique wrote:
In extensions.conf in the general section you can create globals like:
JOHN=IAX2/1234
And then use ${JOHN} in your dial plan to use that device.
Alfredo.
On 5/10/05, Daniel Salama [EMAIL PROTECTED] wrote:
Is it possible to set a variable for an IAX device in iax.conf that
can be read from the dial plan (extensions.conf)? If so, can you
explain?
Thanks,
Daniel
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Re: [Asterisk-Users] Setting Variables
This is a good solution. However, I have a regarding this approach: Does this mean that ANY incoming directed to that agent will fall into that context? I have calls coming into the system and being put in a Queue. When the agent is available, the call will be de-queued to the agent. When is the context used in extensions.conf? By the time the call is going to be directed to the agent, the call is already in the system and has gone through several contexts and queued by AppQueue. Thanks, Daniel On May 10, 2005, at 9:11 PM, Moises Silva wrote: at the moment i think is not possible. A workaround can be sending local and remote iax to different context in iax.conf like this [iaxremote] blah... context=remoteiax [iaxlocal] blah context=localiax so in extensions.conf you can do [localiax] exten = s,1,SetVar(local=1) blah... [remoteiax] exten = s,1,SetVar(local=0) blah Best Regards. - moy -- Su nombre es GNU/Linux, no solamente Linux, mas info en http:// www.gnu.org ___ 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] Zaptel problems on Debian
Sorry for the confusion. On May 10, 2005, at 10:45 PM, Tzafrir Cohen wrote: On Tue, May 10, 2005 at 02:52:39PM -0400, Daniel Salama wrote: I just installed a TE410P on a Debian Sarge system running kernel 2.6.11-1-686-smp. Which is a kernel from sid, actually. What version is it exactly? dpkg -l kernel-image-`uname -r` kernel from Sarge would not operate correctly my e1000 NIC, so I upgraded the kernel to sid's and it now works just fine. Zaptel and Asterisk seem to be working fine. However, I have a couple of problems with the TE410P and Zaptel. First, the TE410P is showing me red alarms on 2 of the 4 T1s. This board (the TE410P) was just moved from another machine running REL3 and all 4 T1s were working there. I don't know why only two T1s are being recognized on the Debian system. Secondly, I no longer have zttool, which I used to have on REL3. How can I query from Zaptel the status of my T1s? Do you use zaptel from Debian? What version exactly? If not: what asterisk/zaptel version? I downloaded latest CVS version from today for all (zaptel, libpri, and asterisk). Thanks, Daniel -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] Central Asterisk Server and Asterisk VoIP Gateway
I'm setting up a demo for two asterisk machines. One will be a
central Asterisk server which will handle everything already in VoIP
(office-like functions plus agents functionality). The second
Asterisk box will be used strictly as a VoIP gateway to the first
server.
The gateway server will have 4 T1s connected to it and what I was
thinking on doing was the following:
in /etc/asterisk/zapata.conf
switchtype = dms100
signalling = em_w
group = 1
context = inbound
channel = 1-96
What I'm trying to accomplish here is to create one huge trunk group
where ALL incoming calls will be directed to the inbound context.
On the same machine, /etc/asterisk/extensions.conf
[general]
MAIN_SERVER = IAX2/inbound:[EMAIL PROTECTED]
[inbound]
exten = _N.,1,Dial(${MAIN_SERVER}/${EXTEN})
exten = _N.,2,Congestion
exten = _N.,3,HangUp
[outbound]
exten = _N.,1,Dial(Zap/g1/${EXTEN})
exten = _N.,2,Congestion
exten = _N.,3,HangUp
On the main Asterisk server, I would have in /etc/asterisk/
extensions.conf:
[general]
GATEWAY = IAX2/outbound:[EMAIL PROTECTED]
[inbound]
exten = 1234,1,Dial(SIP/100,20,t)
exten = 1234,2,VoiceMail(uSIP/100)
exten = 1234,102,VoiceMail(bSIP/100)
exten = _N.,1,Dial(SIP/500,20,t)
exten = _N.,2,VoiceMail(uSIP/500)
exten = _N.,102,VoiceMail(bSIP/100)
[outbound]
exten = _N.,1,Dial(${GATEWAY}/${EXTEN})
exten = _N.,2,Congestion
exten = _N.,3,HangUp
Basically, what I'm trying to accomplish is that the gateway server
will forward to the main asterisk server all incoming calls into the
[inbound] context so that the main asterisk server will have all the
necessary logic to process all corresponding DIDs or what have you.
At the same time, whenever the main asterisk server wants to make ANY
outbound call, it will simply send it to the [outbound] context of
the gateway server for it to place the actual call(s).
Does this make sense to you guys? Am I missing anything? Is there
anything I should be concerned with or that I should watch out for?
Thanks,
Daniel
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Re: [Asterisk-Users] IAXy Firmware Upgrade
And how/where can you download the latest firmware onto /var/lib/ asterisk/firmware/iax? Thanks, daniel On May 7, 2005, at 9:13 AM, Time Bandit wrote: I'd like to known what I have to do to upgrade the firmware into a IAXy device. It does it automagically when it connect to Asterisk if a newer version is available. Look in /var/lib/asterisk/firmware/iax and you will see iaxy.bin. hth ___ 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 variable for a context for all extensions?
Not entirely sure, but, I wonder what would happen if you define RING
in the [globals] section first, and the use SetVar or SetGlobalVar in
the other contexts to override its value.
- Daniel
On May 8, 2005, at 2:48 AM, Mark Wormgoor wrote:
Hi,
Is it possible to set a variable for a context for all extensions?
I haven't been able to find it. I want something like this in
extensions.conf:
[from-iaxfwd]
exten = .,1,RING=r3
exten = 123456,1,Goto(from-pstn,s,1)
[from-internal]
exten = .,1,RING=r2
include = ext-local
[ext-local]
exten = 1,1,Dial(Zap/1,${LONGTIMEOUT})
exten = 2,1,Dial(SIP/2,${LONGTIMEOUT})
But how do I do this? I want to change the ring depending on the
starting context. The above doesn't work.
Kind regards,
Mark
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Re: [Asterisk-Users] Can I hide caller id on the fly (per each usesetting) on Bristuffed * and quadbri
Question: can you override the caller id at all for outbound calls (whether statically or on-the-fly)? Remember that you need to have the carrier give you the ability to override the caller id. This is normally called digit manipulation and is normally regulated by the carriers so as to avoid people spam calling and hiding their caller id, specially with the DND list. - Daniel On May 5, 2005, at 4:10 AM, Robert Rozman wrote: - Original Message - From: Peer Oliver Schmidt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 04, 2005 11:14 PM Subject: Re: [Asterisk-Users] Can I hide caller id on the fly (per each usesetting) on Bristuffed * and quadbri Robert Rozman wrote: I wonder if I can hide caller id for just certain users. Can I override caller id setting for show or hide on the fly from dialplan ? Did you try setcallerid()? -- I tried but this will work if calling internal line. I'm after dynamically hiding caller id on QuadBRI outgoing ISDN calls... I guess this is possible with settings in zapata.conf, but only per channel - I wonder if it is possible to set this up by user or do it from dialplan with some command Thanks in advance, Rob. Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ 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] Realtime and Asterisk Database
I've read on the wiki that the Asterisk Database is mainly (or only) used for manipulating the dial plan (e.g. extensions.conf). I know that RealTime can be used for much more than that (i.e. sip.conf, iax.conf, etc). My question is: if I had two management applications, one that uses RealTime and one that uses Asterisk Database, and all I wanted is to manipulate the dial plan, which would be more efficient/recommended to use - RealTime or Asterisk Database? Thanks, Daniel ___ 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] Hardware Capacity/Configuration
I have most agents using Alaw and/or ulaw and a handful of agents using gsm. Thanks, - Daniel On May 5, 2005, at 1:12 PM, Mehdi Chouikh wrote: If use Alaw or ulaw as codec, i think it's enough. But if you need to make transcoding to a hard codec like g729, g723, you have to look other cpu. regards - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 04, 2005 4:21 AM Subject: [Asterisk-Users] Hardware Capacity/Configuration I know this is a frequent topic on the list. Sorry if this creates more bandwidth but I couldn't get my specific answer from neither the wiki nor searching the list. I've read that a P4 3GHz+ should be sufficient to handle a 4 T1s on a single CPU machine. I am setting up a proof of concept machine but I was only able to get a P4 1.6GHz machine. If this machine is only going to be forwarding the calls to another, much more powerful, Asterisk machine which will handle more demanding call processing rules, scripts, Monitoring, etc, do you think this CPU will be able to handle the 4 T1s? Will it handle 3? 2? 1? Efficiently, of course. The idea is to setup a basic VoIP gateway whose only intelligence will be to forward ALL incoming calls to another Asterisk box using IAX as well as placing outbound calls through the T1s from other Asterisk boxes communicating using IAX. Comments? Thanks, Daniel ___ 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] TE410P does not fit in motherboard
I'm trying to install a TE410P on a Gigabyte motherboard (8S661FX) and I'm just noticing that the TE410P does not fit in the PCI slot. It seems as if the little opening in the PCI is on the wrong side. Has anyone else seen this or is it just me and I'm too stupid to do something as basic as this? Thanks, Daniel ___ 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] TE410P does not fit in motherboard
Thank you all for pointing this out to me. I wasn't aware of the physical difference between the 3.3V and the 5V PCI slots. I guess it's time to change the motherboard. Thanks, Daniel On May 4, 2005, at 9:32 PM, Rob Thomas wrote: Well, no, it looks like the 8S661FX (which is actually called 8S661FXM-RZ or FXM_P_ for Socket 478) has 5v PCI slots. http://www.giga-byte.com/Motherboard/FileList/ProductImage/ photo_8s661fx m_rz_big.jpg and http://www.giga-byte.com/Motherboard/FileList/ProductImage/ photo_8s661fx mp-rz_big.jpg Is that what your motherboard looks like? Coz those are *definitely* 5v slots. --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, May 05, 2005 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] TE410P does not fit in motherboard I'm trying to install a TE410P on a Gigabyte motherboard (8S661FX) and I'm just noticing that the TE410P does not fit in the PCI slot. It seems as if the little opening in the PCI is on the wrong side. Has anyone else seen this or is it just me and I'm too stupid to do something as basic as this? Thanks, Daniel ___ 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] Multi-tenant Setup
I'm trying to setup a multi-tenant configuration of * and have the following question: In extensions.conf, there is a [global] section that I would normally use to define global variables for my single tenant setups. Now, is there a way to have something like global variables on a per tenant basis, so that I could define something like operator = SIP/123 for tenant A and operator = SIP/456 for tenant B? I read about SetGlobalVar, what I think that would make the variable available to all contexts (in my case tenants). Thanks, Daniel ___ 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] SIP over IAX2
Ok. I've been trying to make this work to no avail. I think I have something screwed up in my original post, so I'm going to try to rephrase what I'm trying to do. I have a bunch of agents connected to an * box using IAX (A1). I have a separate * box (A2) that is running an IVR (AGI) script that the agents need to get to. What I'm trying to do is create a extension in A1 so that when dialed by the agents, they will be connected to the IVR script in A2 using SIP (not IAX). I'm currently doing this using IAX, but I have to implement it using SIP, for the IVR machine is going to be SIP only in the near future. Here is what I have in A1: extensions.conf exten = 1234,1,Dial(IAX2/ivr_script:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = 1234,2,HangUp In A2, I have: extensions.conf [ivr_script] exten = s,1,Answer exten = s,2,AGI(play_ivr.pl) exten = h,1,HangUp If I simply change IAX2 to SIP in A1, it won't work. If I replace it with Dial(SIP/[EMAIL PROTECTED]) it shows on the console: Got SIP response 404 Not Found back from 192.168.1.1 which I could understand because there is no such SIP peer defined in sip.conf. Any suggestions would be greatly appreciated. Thanks, Daniel On Apr 30, 2005, at 8:00 PM, Tim Connolly wrote: Asterisk Box 2 (agents register) extensions.conf [agents-context] exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]) exten = 1234,2,Hangup Asterisk Box 1 Sip.conf [ab1] type=friend host=ip of ab2 context=incoming canreinvite=yes qualify=yes extension.conf [incoming] Exten = 1234etc... -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Saturday, April 30, 2005 6:50 PM To: Tim Connolly Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP over IAX2 I understand and I guess I know how to do that within a single box. If I have the following: Asterisk Box 1 (no agents) extensions.conf [test-ivr] exten = s,1,AGI(play_ivr) exten = s,2,Hangup Asterisk Box 2 (agents register) extensions.conf [agents-context] exten = 1234,1,Dial(?) exten = 1234,2,Hangup Question is, when the agents dial 1234, how do I tell the application to connect to the agent with context test-ivr of Asterisk_1? Thanks, Daniel On Apr 30, 2005, at 7:12 PM, Tim Connolly wrote: Maybe I'm missing something, but as long as you have the entension defined on the agent box to dial the extension on the IVR, you should be okay. Just make sure the default SIP context on the IVR has that extension defined, or define the IVR box as a SIP peer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Saturday, April 30, 2005 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP over IAX2 I have two asterisk boxes. I'm running an IVR script in one of them and I have agents registered on the second box. I wish to create an extension on the * box where the agents are registered, so that when dialed, it will connect the agent to the IVR script on the other * box. However, I'd like for the connection to be done using SIP instead of IAX. Can anyone help me, if at all possible, write this configuration? Thanks, Daniel ___ 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
Fwd: [Asterisk-Users] SIP over IAX2
Sorry if this is posted twice. I sent this about an hour ago and haven't seen it in the list yet. Thanks, Daniel Begin forwarded message: From: Daniel Salama [EMAIL PROTECTED]> Date: May 3, 2005 1:12:51 PM EDT To: Tim Connolly [EMAIL PROTECTED]> Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] SIP over IAX2 Ok. I've been trying to make this work to no avail. I think I have something screwed up in my original post, so I'm going to try to rephrase what I'm trying to do. I have a bunch of agents connected to an * box using IAX (A1). I have a separate * box (A2) that is running an IVR (AGI) script that the agents need to get to. What I'm trying to do is create a extension in A1 so that when dialed by the agents, they will be connected to the IVR script in A2 using SIP (not IAX). I'm currently doing this using IAX, but I have to implement it using SIP, for the IVR machine is going to be SIP only in the near future. Here is what I have in A1: extensions.conf exten => 1234,1,Dial(IAX2/ivr_script:[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten => 1234,2,HangUp In A2, I have: extensions.conf [ivr_script] exten => s,1,Answer exten => s,2,AGI(play_ivr.pl) exten => h,1,HangUp If I simply change IAX2 to SIP in A1, it won't work. If I replace it with Dial(SIP/[EMAIL PROTECTED]) it shows on the console: Got SIP response 404 Not Found back from 192.168.1.1 which I could understand because there is no such SIP peer defined in sip.conf. Any suggestions would be greatly appreciated. Thanks, Daniel On Apr 30, 2005, at 8:00 PM, Tim Connolly wrote: Asterisk Box 2 (agents register) extensions.conf [agents-context] exten => 1234,1,Dial(SIP/[EMAIL PROTECTED]) exten => 1234,2,Hangup Asterisk Box 1 Sip.conf [ab1] type=friend host=ip of ab2> context=incoming canreinvite=yes qualify=yes extension.conf [incoming] Exten => 1234etc... -----Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Saturday, April 30, 2005 6:50 PM To: Tim Connolly Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP over IAX2 I understand and I guess I know how to do that within a single box. If I have the following: Asterisk Box 1 (no agents) extensions.conf [test-ivr] exten => s,1,AGI(play_ivr) exten => s,2,Hangup Asterisk Box 2 (agents register) extensions.conf [agents-context] exten => 1234,1,Dial(?) exten => 1234,2,Hangup Question is, when the agents dial 1234, how do I tell the application to connect to the agent with context test-ivr of Asterisk_1? Thanks, Daniel On Apr 30, 2005, at 7:12 PM, Tim Connolly wrote: Maybe I'm missing something, but as long as you have the entension defined on the agent box to dial the extension on the IVR, you should be okay. Just make sure the default SIP context on the IVR has that extension defined, or define the IVR box as a SIP peer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Saturday, April 30, 2005 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP over IAX2 I have two asterisk boxes. I'm running an IVR script in one of them and I have agents registered on the second box. I wish to create an extension on the * box where the agents are registered, so that when dialed, it will connect the agent to the IVR script on the other * box. However, I'd like for the connection to be done using SIP instead of IAX. Can anyone help me, if at all possible, write this configuration? Thanks, Daniel ___ 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] TE4XXP and /etc/zaptel.conf
I'm trying to configure 4 T1s into this board. The T1s work just fine. However, I have a question about setting up the clock source properly. 3 T1s are from the same carrier and the remaining T1 is from another. I have a configuration similar to: /etc/zaptel.conf span=1,1,0,esf,b8zs em=1,24 span=2,1,0,esf,b8zs em=25-48 span=3,1,0,esf,b8zs em=49-72 span=4,1,0,esf,b8zs em=73-96 My question is: Should all the T1s be defined as primary sync source? I could understand that the T1 from one provider should not share the timing from the other provider. But, what about multiple T1s from the same provider? If they should share their timing setting, what would be the right syntax to specify it? Thanks, Daniel ___ 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] Hardware Capacity/Configuration
I know this is a frequent topic on the list. Sorry if this creates more bandwidth but I couldn't get my specific answer from neither the wiki nor searching the list. I've read that a P4 3GHz+ should be sufficient to handle a 4 T1s on a single CPU machine. I am setting up a proof of concept machine but I was only able to get a P4 1.6GHz machine. If this machine is only going to be forwarding the calls to another, much more powerful, Asterisk machine which will handle more demanding call processing rules, scripts, Monitoring, etc, do you think this CPU will be able to handle the 4 T1s? Will it handle 3? 2? 1? Efficiently, of course. The idea is to setup a basic VoIP gateway whose only intelligence will be to forward ALL incoming calls to another Asterisk box using IAX as well as placing outbound calls through the T1s from other Asterisk boxes communicating using IAX. Comments? Thanks, Daniel ___ 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] SNMP Monitoring
I've read on the wiki how you can SNMP monitor an Asterisk machine and from what I read, you're pretty much monitoring the availability of Asterisk. I'm looking for a way to be able to monitor the availability of individual T1 circuits of my TE410P card. During the storm season, some of our T1s tend to flap and I'd like to be able to monitor that. Is there something that can do this? Thanks, Daniel ___ 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] mp3 problems
Hi, I recompiled Asterisk 1.0.3 on a machine which I upgraded the kernel. I also recompiled zaptel and libpri. After doing this, I am realizing that I'm having some problems playing mp3 files. However, and very strangely, music on hold is working playing mp3 files. I have an AGI script that was working just fine. You would select a recording ID and it would go out and fetch it and then play the file. Now, it's doing everything as it should, but it's not playing the actual media. I turned on agi debug on and here is the relevant portion of what I saw: AGI Rx EXEC MP3Player /var/spool/asterisk/monitor/archive/4082-20050426-143915 -- AGI Script Executing Application: (MP3Player) Options: (/var/spool/asterisk/monitor/archive/4082-20050426-143915) May 2 19:15:46 NOTICE[2110]: app_mp3.c:91 timed_read: Poll timed out/errored out with 0 AGI Tx 200 result=0 Any clues? Thanks, Daniel ___ 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] Queue Event
Is there a way to configure asterisk to execute an AGI script upon the transferring of a call to an extension from the Queue? For example, once the call is put in the queue and the extension becomes available, the Queue app will send the call to that extension. Is there a way for me to manually execute a command that will give me the extension it was transferred to? Thanks, Daniel ___ 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] BSD Compatability
Anyone know if Digium cards, especifically TE410P, are compatible with BSD (FreeBSD or NetBSD)? How does * run on BSD? Thanks, Daniel ___ 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] BSD Compatability
It's just that this statement from the wiki confused me: Asterisk is known to run on many OS platforms. However, Linux is the main platform for development and Digium hardware support. If you are running VoIP only, or if you are comfortable with using external media gateways to connect conventional telephone equipment, then you have more systems to choose from, like FreeBSD, Mac OS X and Solaris It sounds as if BSD-like OS are good to run asterisk without the digium boards. Thanks, - Daniel On May 2, 2005, at 12:06 AM, skamp wrote: asterisk runs great on BSD if you follow the sirections, and the card i believe does work On Tue, 2005-05-03 at 00:01 -0400, Daniel Salama wrote: Anyone know if Digium cards, especifically TE410P, are compatible with BSD (FreeBSD or NetBSD)? How does * run on BSD? Thanks, Daniel ___ 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 -- skamp [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: RE : [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?
Along the same lines, is there some sort of capacity chart that maps capacity based on translations/transcoding? - Daniel On May 1, 2005, at 2:45 PM, Greg Boehnlein wrote: On Sun, 1 May 2005, David John Walsh wrote: what sort of level of PC is required for 300 concurrent calls? Doing what? Ulaw? That could probably be done on a single P4 2.8 Ghz. If you want to transcode from Ulaw to something else, you need to scale the hardware appropriately. Every case is different. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] SIP over IAX2
I have two asterisk boxes. I'm running an IVR script in one of them and I have agents registered on the second box. I wish to create an extension on the * box where the agents are registered, so that when dialed, it will connect the agent to the IVR script on the other * box. However, I'd like for the connection to be done using SIP instead of IAX. Can anyone help me, if at all possible, write this configuration? Thanks, Daniel ___ 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] Kernel 2.4 or 2.6
I was reading on the wiki about the supported kernels and I __THINK__ the main issues with the kernel versions have more to do with Zaptel driver and not necessarily Asterisk itself. Is this correct? Thanks, Daniel ___ 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] SIP over IAX2
I understand and I guess I know how to do that within a single box. If I have the following: Asterisk Box 1 (no agents) extensions.conf [test-ivr] exten = s,1,AGI(play_ivr) exten = s,2,Hangup Asterisk Box 2 (agents register) extensions.conf [agents-context] exten = 1234,1,Dial(?) exten = 1234,2,Hangup Question is, when the agents dial 1234, how do I tell the application to connect to the agent with context test-ivr of Asterisk_1? Thanks, Daniel On Apr 30, 2005, at 7:12 PM, Tim Connolly wrote: Maybe I'm missing something, but as long as you have the entension defined on the agent box to dial the extension on the IVR, you should be okay. Just make sure the default SIP context on the IVR has that extension defined, or define the IVR box as a SIP peer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Saturday, April 30, 2005 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP over IAX2 I have two asterisk boxes. I'm running an IVR script in one of them and I have agents registered on the second box. I wish to create an extension on the * box where the agents are registered, so that when dialed, it will connect the agent to the IVR script on the other * box. However, I'd like for the connection to be done using SIP instead of IAX. Can anyone help me, if at all possible, write this configuration? Thanks, Daniel ___ 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] SIP over IAX2
Thanks. That's what I needed. - Daniel On Apr 30, 2005, at 8:00 PM, Tim Connolly wrote: Asterisk Box 2 (agents register) extensions.conf [agents-context] exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]) exten = 1234,2,Hangup Asterisk Box 1 Sip.conf [ab1] type=friend host=ip of ab2 context=incoming canreinvite=yes qualify=yes extension.conf [incoming] Exten = 1234etc... -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Saturday, April 30, 2005 6:50 PM To: Tim Connolly Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP over IAX2 I understand and I guess I know how to do that within a single box. If I have the following: Asterisk Box 1 (no agents) extensions.conf [test-ivr] exten = s,1,AGI(play_ivr) exten = s,2,Hangup Asterisk Box 2 (agents register) extensions.conf [agents-context] exten = 1234,1,Dial(?) exten = 1234,2,Hangup Question is, when the agents dial 1234, how do I tell the application to connect to the agent with context test-ivr of Asterisk_1? Thanks, Daniel On Apr 30, 2005, at 7:12 PM, Tim Connolly wrote: Maybe I'm missing something, but as long as you have the entension defined on the agent box to dial the extension on the IVR, you should be okay. Just make sure the default SIP context on the IVR has that extension defined, or define the IVR box as a SIP peer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Saturday, April 30, 2005 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP over IAX2 I have two asterisk boxes. I'm running an IVR script in one of them and I have agents registered on the second box. I wish to create an extension on the * box where the agents are registered, so that when dialed, it will connect the agent to the IVR script on the other * box. However, I'd like for the connection to be done using SIP instead of IAX. Can anyone help me, if at all possible, write this configuration? Thanks, Daniel ___ 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 Daniel Salama [EMAIL PROTECTED] Voice: (954) 655-8051 Fax : (954) 252-3988 This e-mail contains information which may be confidential and privileged. Unless you are the addressee (or authorized to receive for the addressee), you may not use, copy or disclose to anyone the message or any information contained in the message. If you have received the message in error, please advise the sender by reply e-mail to [EMAIL PROTECTED] or tel. +1-954-655-8051 and delete the material from any computer. ___ 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 Hardware Recommendation
Sure. I setup a small lab on a machine with 4 T1s and 36 agents logged in. The system was configured to Monitor all outbound calls as well as monitor all calls distributed by Queue app (monitor-format setting in queues.conf). When recording to local disk, everything was working fine. Agents were busy 99.5% and there were at least 30 calls waiting in Queue to be distributed. Average call conversation length was about 7.5 minutes. Then I mounted /var/spool/asterisk/monitor via NFS using 10/100 Fast-E. The moment we pushed the load on the Asterisk machine, everything worked for about 40 seconds. Then call quality started suffering significantly. Chopped audio. Bad audio. No audio. Good audio. You could imagine. So we stopped the test. Then we unmounted the NFS drive and repeated the test again. Everything worked fine again. The machine we tested asterisk on is a dual Xeon 3 GHz with 2G RAM. During all tests, CPU utilization was about 55% on the average (for each CPU). Memory usage was under 1G. I would say I need to try more troubleshooting. Maybe there was congestion on the Fast-E, although preliminary analysis indicates there were no CRC errors, collisions, or packet loss. The NFS machine was completely idle. Last, we repeated the test over a 1 hour period. This time, Monitor was recording on local drives and we were copying files every 15 minutes with a background process (perl script) to NFS mount point. Everything worked fine as well. I don't know if these tests are conclusive yet. However, from the results so far, I would recommend staying away from recording to NFS mounted point. I will continue running simulations to see if anything else can be identified. Thanks, Daniel On Apr 28, 2005, at 7:26 PM, Matt Roth wrote: Daniel, Could you expand upon your experience recording to an NFS mounted drive. We are looking to use a TDM-VoIP gateway to route 16+ spans to a single Asterisk server. We were hoping to Monitor using the following scheme: - Monitor application executed on Asterisk server (no 'm' flag) - Pick a codec that the Monitor application can record in natively so that no transcoding is done on the leg files (can this be done?) - Record the leg files to an NFS mounted drive on a remote machine - Have soxmix take care of mixing and transcoding the leg files into the desired format on the remote machine According to you this now looks like a VERY BAD IDEA. Does anyone out there have any experience using monitor to digitally record large numbers of spans (16+) on a single Asterisk server using a VoIP gateway instead of TDM cards? Is it feasible? We are trying to keep the Asterisk server as slim as possible, but would like to stick to one box so that we can have centralized queuing, configuration, and reporting. Has anyone had any luck using Monitor to record to an NFS mounted drive? Are there any other options to remove the overhead of the disk subsystem when recording calls? Thanks, Matthew Roth http://voip-info.org/tiki-index.php? page=Running%20Asterisk%20on%20Debian Daniel Salama wrote: Thank you again. I will definitely do that. By cheaper asterisk servers, do you mean single-CPU machines that can handle Quad T1s and still do the call monitoring? BTW, I tried the monitoring without the 'm' option and mounted the audio directory via NFS. Big NO NO for everyone. Just do what Matt says: copy the -in and -out to archive server separately several times a day :) - don't record to NFS mounted drive. Thanks, Daniel On Apr 28, 2005, at 6:42 PM, mattf wrote: I have never been able to do more than 50 concurrent recordings with Zap - SIP phone calls without the audio skipping and/or breaking up. Also, if you are using Digium TE4XXP and want to do a lot of recording I would recommend against a SCSI RAID card because of the interrupt conflicts that you will run into over time. I would recommend a couple of cheaper Asterisk servers with a dual T1 or Quad T1 board in them and SATA drives, with a nice big archive server that the audio will be copied to several times a day. Also, do not record(Monitor) with the 'm' flag on because this will also lead to more disk read-write while you are already trying to write another 100 or so streams. Offload the -in and -out files to the archive server and let it soxmix them together instead. This is the method that we have settled on for our 12 Asterisk servers and it works rather well for us. MATT--- -Original Message- From: Daniel Salama [mailto:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Hardware Recommendation Hi, I've been reading on the wiki as well as on this list, different suggestions of what to look for when designing an asterisk server with a lot of traffic. By a lot of traffic, I mean a box with a a TE4XXP, that will be hit to full
Re: [Asterisk-Users] Asterisk Hardware Recommendation
Well, I don't think I'm ready to spend that much money :) I understand your point regarding that load depends on usage. SIP_Agents are simply agents answering calls. Average call length would be about 8 minutes. During some of these calls (maybe 25%), agents will conference the call (PSTN channel) with internal IVR script. I like Scenario 6. Will look into that further. However, if the above information gives you more grounds to make additional comments, please do so :) Thanks, Daniel On Apr 29, 2005, at 10:21 AM, mattf wrote: If price would truly not an option just get one of the Signate Telephony 5000 servers(http://www.signate.com/pbx.php) They are about $18,000 and allow you to have upto 5000 SIP streams go through it. You could have that be your gateway and do the SIP-IAX through that machine and scale upto 100 T1s if you want. But that is a bit steep. So on to your choices. I would really say that the setup you choose will depend on what kind of users you have as well as how often you need to change/add users to the system and how the users are using the system at what times. Any of them that you listed could work depending on how they are used, but in some cases you may not want to use some of the scenarios listed because they would either be incapable of meeting your needs or overly complex to manage. The easiest and cheapest one would actually not be listed: Scenario 6: Direct SIP-Zap on two separate servers half SIP users on each server PSTN --2xT1-- A1 SIP_Agents PSTN --2xT1-- A2 SIP_Agents There is really no reason to have another 2 servers running IAX to the T1 servers, and this is simple and easy to set up and involves only 2 machines. The next setup I would recommend would be Scenario 4, although you will have to get a machine with a fast/wide BUS(like an Apple G5) to handle ever increasing numbers of SIP-IAX streams as the system would grow. If you can explain more about what kind of use this system will have I can give a better recommendation. 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
Re: [Asterisk-Users] Asterisk Hardware Architecture Group
I think that would be a great idea. The only problem I see is that Asterisk is growing its feature set and maturing at such a dynamic rate, that I don't know in many cases, where to point the finger at. Sometimes it's stability of the CVS version, sometimes it's stability of Digium or whose ever interfaces, and yet sometimes it's issues with actual hardware architecture. I wouldn't mind participating in such an effort, but that may just create parallel lists or problem reports that may be so tightly related that one list would take away knowledge from the other. Comments? - Daniel On Apr 29, 2005, at 2:42 PM, Matt Roth wrote: List members, Does anyone have an interest in forming a hardware architecture group? It seems that Asterisk is so tightly linked to specialized hardware and its corresponding architecture that developing the software alone is insufficient for its adoption to large scale applications. Thank you, Matthew Roth http://voip-info.org/tiki-index.php? page=Running%20Asterisk%20on%20Debian ___ 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 Hardware Recommendation
This is an interesting question. I haven't tested it but would love to know if it works or not. Anyone? - Daniel On Apr 29, 2005, at 3:38 AM, Michael Welter wrote: I haven't seen this before--can an agent log into a queue on a remote (i.e. over IAX) Asterisk server? ___ 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 Hardware Architecture Group
Does anyone have any experience with servers from siliconmechanics.com? Are they reliable? How does * run on them? Thanks - Daniel On Apr 29, 2005, at 4:22 PM, snacktime wrote: Personally I would buy an * box from someone like asaservers.com. At least companies like that really know their hardware, and if you tell them the common issues with * they could probably put together a rock solid system. Chris ___ 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] Caller-ID Block
Question: how can I block someone from calling us? Sometimes we get crank calls into our office. We'd like to build a list of callers to be blocked. When they call, they should hear busy and then we hang up. We have about 100 DIDs routed to different contexts and I wouldn't want to have to manually edit all contexts. Is there a way to do something global to create something like a black list of caller IDs to block? Thanks, Daniel ___ 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
