Re: [asterisk-users] seeking hardware recommendation PCI versus PCI Express E1 card (te407p vs te420bf)

Kevin P. Fleming wrote: Rajesh Kumar Mallah wrote: to summarise is TE407P in anyway inferior to TE420BF Card ? does TE407P saturate the 33mhz PCI bus when all voice channels are used ? There is no effective performance difference between using the PCI-X and PCI-E versio

Re: [asterisk-users] Unable to create ZAP Channel

On Wed, Aug 13, 2008 at 10:57:28PM -0700, Jay Ray wrote: > Here it is: > > [asterisk]# asterisk -rx 'zap show channel 1' > Unable to find given channel 1 > Verbosity is at least 20 cat /proc/zaptel/* (or do you use dahdi instead?) -- Tzafrir Cohen icq#16849755 jabbe

Re: [asterisk-users] rtc issue

On Wed, Aug 13, 2008 at 04:54:12PM -0400, Jerry Geis wrote: > Jerry Geis wrote: > > Hi > > > > I am using centos 4.6 on an ebox 4300. Everything seems to be working > > except the > > /proc/interrupts rtc is always constant. On other machines the rtc > > (which ztdummy uses) is always incrementin

Re: [asterisk-users] Unable to create ZAP Channel

Output of 'zap show status' (in asterisk) Run 'ztcfg' from the command line and check that out too... PaulH Jay Ray wrote: > Here it is: > > [asterisk]# asterisk -rx 'zap show channel 1' > Unable to find given channel 1 > Verbosity is at least 20 > > > --- On *Thu, 8/14/08, Tzafrir Cohen /<[EM

Re: [asterisk-users] Unable to create ZAP Channel

Here it is: [asterisk]# asterisk -rx 'zap show channel 1' Unable to find given channel 1 Verbosity is at least 20 --- On Thu, 8/14/08, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: From: Tzafrir Cohen <[EMAIL PROTECTED]> Subject: Re: [asterisk-users] Unable to create ZAP Channel To: asterisk-users@li

Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 32

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will

Re: [asterisk-users] Unable to create ZAP Channel

On Wed, Aug 13, 2008 at 09:51:07PM -0700, Jay Ray wrote: > Her eis my call flow > > Sofphone --> Asterisk--> X100P card--> Home Phone line > > Home phone line plugged in LINE port of the FXO ccard (2 port) > > Here is the error I get > > > Aug 13 22:35:31 VERBOSE[8625]: -- Executing > 1

[asterisk-users] Unable to create ZAP Channel

Her eis my call flow Sofphone --> Asterisk--> X100P card--> Home Phone line Home phone line plugged in LINE port of the FXO ccard (2 port) Here is the error I get Aug 13 22:35:31 VERBOSE[8625]: -- Executing 1;36;40mDial0;37;40m("1;35;40mSIP/xlite1-a0fb0;37;40m", "1;35;40mZap/g1/3035061

Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Darrick Hartman wrote: > Matt Riddell wrote: >> -BEGIN PGP SIGNED MESSAGE- >> Hash: SHA1 >> >> Paul Hales wrote: >>> That's a good question - the plantronics are available with >>> interchangeable ends - which makes them easy to move between d

Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922

Matt Riddell wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Paul Hales wrote: >> That's a good question - the plantronics are available with >> interchangeable ends - which makes them easy to move between different >> phones. > > Problem is, the headset button only works for the m

Re: [asterisk-users] New GUI for Realtime Asterisk - RAGUI

Thanks to Mike and your team. I will be checking this out and will report back to the list. Good stuff. Thanks, Steve Totaro On Wed, Aug 13, 2008 at 7:32 PM, Mike Clark <[EMAIL PROTECTED]> wrote: > Brandon: > > It manages the MySQL database tables directly via RoR. It is > specifically built fo

Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?

On Wed, Aug 13, 2008 at 10:41 PM, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: > > > Steve Totaro wrote: >> On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller <[EMAIL PROTECTED]> wrote: >>> From what I can determine while troubleshooting a voice-dropping >>> issue, the Asterisk server in my orga

Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Paul Hales wrote: > That's a good question - the plantronics are available with > interchangeable ends - which makes them easy to move between different > phones. Problem is, the headset button only works for the minijack thingy so if you use the pl

Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?

Steve Totaro wrote: > On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller <[EMAIL PROTECTED]> wrote: >> From what I can determine while troubleshooting a voice-dropping >> issue, the Asterisk server in my organization has been dropping RTP >> packets between the asterisk server process and the netwo

Re: [asterisk-users] Very loud noise on TDM400

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Carlos Chavez wrote: > I am having a problem with and Asterisk installation where two ports > connected to a TDM400 card will have a very loud noise when you try to > dial. The server has an OpenVox D110P, a TDM04B and a Xorcom Astribank > 8 fxs

[asterisk-users] Asterisk 1.4 T38 UDPTL Pass Through MAX TNT and Linksys 2102

Hi All, I finally got the time to test t38 pass through with a TNT, * 1.4.21.1 and Linksys 2102: PRI><2102> ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: h

Re: [asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?

On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller <[EMAIL PROTECTED]> wrote: > From what I can determine while troubleshooting a voice-dropping > issue, the Asterisk server in my organization has been dropping RTP > packets between the asterisk server process and the network interface. > > I determi

[asterisk-users] Callwaiting and CallerID on ZAP PSTN Line

I have a standard analog POTS service attached to a TDM401 card. My zapata.conf for this line has: ; PSTN connected here ;immediate=no ;busydetect=yes ;busycount=8 ;musiconhold=default mailbox=100 mwimonitor=fsk mwilevel=512 mwimonitornotify=/usr/local/sbin/zapnotify.sh faxdetect=incoming signall

[asterisk-users] Cisco 7960

This one is a little off-topic, it's more about the phone than asterisk itself. I have a cisco 7960 configured with 2 lines to 2 different sip providers (cant get asterisk to register with the 2nd provider, but that's another story). Is there a way yo determine which direction speed-dial buttons

[asterisk-users] Asterisk might be dropping RTP packets before reaching eth int?

>From what I can determine while troubleshooting a voice-dropping issue, the Asterisk server in my organization has been dropping RTP packets between the asterisk server process and the network interface. I determined this from an RTP debug that showed packets sent to the phone and packets receive

Re: [asterisk-users] New GUI for Realtime Asterisk - RAGUI

Brandon: It manages the MySQL database tables directly via RoR. It is specifically built for using Realtime Asterisk. It uses #exec statements in the extensions.conf file to set up the context declarations. Thanks, Mike bkruse wrote: > That's pretty cool, I love ruby. What method does it use

Re: [asterisk-users] Asterisk and Radius

Ciao Salvo, That is not directly possible. But, you can integrate a GPL PERL RADIUS client with Asterisk : http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth It works good, I use it to make Asterisk work as an IVR with a billing system, that acts as a RADIUS server. Cheers, -- P

[asterisk-users] ztdummy on centos 4.6 i386

I am running centos 4.6 i386 kernel 2.6.9-67 ztdummy compiles fine, loads fine, but does not work. modprobe ztdummy debug=1 dmesg shows ztdummy: init() finished however the debug is supposed to print something every 5 seconds it does not do this. Nor does /proc/interrupts rtc value increment.

Re: [asterisk-users] New GUI for Realtime Asterisk - RAGUI

That's pretty cool, I love ruby. What method does it use to communicate to Asterisk? Does it manipulate raw config files or use manager or something similar? I am curious :) -Brandon Mike Clark wrote: > Our company, WebPoint IT Solutions has just released an open source (GPL > V2 license), Rub

Re: [asterisk-users] TDM400P FXO not seeing ringing after software update

In article <[EMAIL PROTECTED]>, Drew Gibson <[EMAIL PROTECTED]> wrote: > Kevin P. Fleming wrote: > > Tony Mountifield wrote: > > > >> My guess is that 1.2.24 will work, which was at revision 3842 of the tree > >> (rev 3741 of wctdm.c). The next change to wctdm.c (rev 4126) looks > >> innocuous

[asterisk-users] Asterisk stops sending RTP packets to ethernet interface

>From what I can determine while troubleshooting a voice-dropping issue, the Asterisk server in my organization has been dropping RTP packets between the asterisk server process and the network interface. I determined this from an RTP debug that showed packets sent to the phone and packets receive

[asterisk-users] New GUI for Realtime Asterisk - RAGUI

Our company, WebPoint IT Solutions has just released an open source (GPL V2 license), Ruby on Rails based gui manager for Realtime Asterisk called RAGUI. RAGUI is definitely a work in progress and has rough edges, but we expect to polish it up in the upcoming weeks and months. All comments, co

Re: [asterisk-users] rtc issue

Jerry Geis wrote: > Hi > > I am using centos 4.6 on an ebox 4300. Everything seems to be working > except the > /proc/interrupts rtc is always constant. On other machines the rtc > (which ztdummy uses) is always incrementing. > the uhci_hcd and ehci_hcd are both running. > > What dont I have righ

Re: [asterisk-users] TDM400P FXO not seeing ringing after software update

Drew Gibson wrote: > Did anyone find a fix for this (besides down-grading to 1.2.24)? It has been fixed in Subversion already, but a new Zaptel release hasn't been made. That will be done in the next few days. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Ast

Re: [asterisk-users] TDM400P FXO not seeing ringing after software update

Kevin P. Fleming wrote: > Tony Mountifield wrote: > > >> My guess is that 1.2.24 will work, which was at revision 3842 of the tree >> (rev 3741 of wctdm.c). The next change to wctdm.c (rev 4126) looks innocuous >> enough, but the follwing two (revs 4128 and 4132) look likely culprits, >> from lo

Re: [asterisk-users] seeking hardware recommendation PCI versus PCI Express E1 card (te407p vs te420bf)

Rajesh Kumar Mallah wrote: > to summarise is TE407P in anyway inferior to TE420BF Card ? > does TE407P saturate the 33mhz PCI bus when all voice channels are used ? There is no effective performance difference between using the PCI-X and PCI-E versions of these cards; the cards are essentially id

[asterisk-users] rtc issue

Hi I am using centos 4.6 on an ebox 4300. Everything seems to be working except the /proc/interrupts rtc is always constant. On other machines the rtc (which ztdummy uses) is always incrementing. the uhci_hcd and ehci_hcd are both running. What dont I have right on the system so rtc increments?

[asterisk-users] seeking hardware recommendation PCI versus PCI Express E1 card (te407p vs te420bf)

Dear List , I have to make a choice between TE407P (5.0V PCI slot) and TE420BF (3.3V PCI express). I have a IBM x220 with 2 x 1.2 GHz PIII CPU and OnBoard SCSI Ultra160 Drives but it does not have PCI Express slot. So i cannot use TE420BF with it. The system i am willing to build should suppor

[asterisk-users] call forward spa 841 and asterisk 1.4.21

i am install asterisk with asterisk-gui , the clients have spa841, but I am call number outsite celular and I try to transfer the call to client sip, not show XTRAN in the phone. any idea?? configuration bad ??? ___ -- Bandwidth and Colocation Provided b

Re: [asterisk-users] BLF functionality

Sean Dennis wrote: >> I find it amazing how often I find myself stuck on a problem and then >> someone else posts a question about it to the list. I am in the same >> boat with the OP (although I never thought to test incoming calls until >> I read his message). If I call a phone it will show bu

Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 31

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will

Re: [asterisk-users] OT: Asterisk on fitPC

Don't you think the USB ones would be amazing considering how the ATA is another extra piece of equipment? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Drew Gibson Sent: August 13, 2008 10:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussio

Re: [asterisk-users] help...i cant do more...

David Thomas wrote: > On Fri, Apr 25, 2008 at 4:38 AM, Bruno Pereira > <[EMAIL PROTECTED]> wrote: > >> Thanks for the answers. >> I need to say that this command is executed from another machine, with the >> command ssh >> because in ocalhost is all ok, with sudo or with root. >> >> I will t

Re: [asterisk-users] FC2 and Zaptel

1.4 from CVS..  Last nite...I corrected the below error by change the check in Makefile for HOTPLUG="yes to a "no" and it does not try to download and install that tar.gz file anymore..Not sure if it will cause a prob.. Later all teh install steps worked fine until I ran ztcfg which errored wit

Re: [asterisk-users] help...i cant do more...

On Fri, Apr 25, 2008 at 4:38 AM, Bruno Pereira <[EMAIL PROTECTED]> wrote: > Thanks for the answers. > I need to say that this command is executed from another machine, with the > command ssh > because in ocalhost is all ok, with sudo or with root. > > I will try that trace to see if it helps me

Re: [asterisk-users] BLF functionality

> > I find it amazing how often I find myself stuck on a problem and then > someone else posts a question about it to the list. I am in the same > boat with the OP (although I never thought to test incoming calls until > I read his message). If I call a phone it will show busy, however if I >

[asterisk-users] ENUM lookup

Hi All, For a 1.4 version asterisk, whats the recommended mechanism for dialling with ENUM lookup? At the moment I user SIPbroker, but am getting tired of it hanging on certain numbers, so I was thinking about implementing it myself. I've seen various vo-ip.info pages (http://www.voip-info

Re: [asterisk-users] Decline issue

post the response of the command bellow dialplan show [EMAIL PROTECTED] att. Felippe Silvestre From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of michel freiha Sent: Wednesday, August 13, 2008 11:30

Re: [asterisk-users] LNP Problems

Thanks everyone for the input. > A CSR is nothing more than a listing of the numbers by your current > provider on some sort of letterhead to indicate you actually are the > subscriber who these numbers belong to (ie, you pay the bill for > them). > > Is it necessary for the actual LNP process - n

[asterisk-users] Asterisk and Radius

Hello, I would like to know, if in Asterisk version 1.6.0-beta9 is it possible to use Radius as an external authentication server. Thank you for your help Ciao Salvo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 200

Re: [asterisk-users] OT: Asterisk on fitPC

You can use an external ATA. I have a Linksys SPA3102 which has 1 FXO (Bell line) port, 1 FXS (phone) port and one Ethernet port to connect via SIP to Asterisk. Alternatively, there are USB connected adapters eg. Xorcom.com but I haven't used them. regards, Drew Mark Hamilton wrote: > I lo

[asterisk-users] Decline issue

Dear Sir, Please find attached the log file that I took from the asterik server during a call...Please check the SIP packets exchanged between OpenSer that send an Invite SIP packet to theasterisk server and the asterisk Server and let me know wat this DECLINEd message means... Regards Asterisk L

Re: [asterisk-users] asterisk-users Digest, Vol 49, Issue 30

Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED], altrimenti vi risponderò al mio rientro. Dimitri Osler I will

Re: [asterisk-users] CDR accuracy

On Wed, Aug 13, 2008 at 02:37:58PM +0200, Klaus Darilion wrote: > Thanks for the detailed information. What about the following scenario: > ANSWER and HANGUP happens in the same second. Thus, the call duration > will be 0 seconds. How are such use cases usually handled in the billing > system? Ar

Re: [asterisk-users] Sending Set Asynchronous Balanced Mode Extended

On Wed, Aug 13, 2008 at 9:56 AM, Jon Weisman <[EMAIL PROTECTED]> wrote: > So we've got a TE410P configured as E-1. The PRI is showing up as normal, I > have green lights, but d channel doesnt seem to come up and i keep getting > this error if i do a "pri intense debug" > > The carrier swears up and

Re: [asterisk-users] cmdRecord issue related to "iax2 received mini frame before first full voice frame"?

On Wednesday 13 August 2008 04:50:24 Novak Joe wrote: > [May 21 20:31:52] WARNING[27346]: chan_iax2.c:8020 socket_process: > Received mini frame before first full voice frame > > A more detailed explanation of what the above warning means/implies, > and how or why it might be preventing recording

Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser

Doesn't Queuemetrics run on a license basis? Anything else that's probably open source and free? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Faraz Khan Sent: August 13, 2008 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: R

Re: [asterisk-users] OT: Asterisk on fitPC

I love the thin client stuff. It probably looks as big as the Samsung SWA-4000. But in terms of hardware, don't I need a PCI card to get it working? How would that work? Sorry, I have no idea about Asterisk working for home, but just SIP related stuff. :( -Original Message- From: [EMAIL P

[asterisk-users] Sending Set Asynchronous Balanced Mode Extended

So we've got a TE410P configured as E-1. The PRI is showing up as normal, I have green lights, but d channel doesnt seem to come up and i keep getting this error if i do a "pri intense debug" The carrier swears up and down that there are no issues on their end. Any thoughts? localhost*CLI> >

Re: [asterisk-users] BLF functionality

Johansson Olle E wrote: > > 13 aug 2008 kl. 00.45 skrev Dan Peters: > >> We have had Asterisk up and running for a while now and it works very >> well. Recently we tried to integrate a Linsys SPA962 with the >> associated SPA932 console. We can get the BLF lights to blink when a >> phone is

Re: [asterisk-users] CDR accuracy

I think that the most appropriate answer for this would be it depends on your setup and requirements. Some of our customers bill all answered calls for the entire minimum duration/increment (even if duration is 0) while others have configured a rule not to bill all calls whose duration is less tha

Re: [asterisk-users] CDR accuracy

Steve Murphy wrote: > On Tue, 2008-08-12 at 16:39 +0200, Klaus Darilion wrote: >> Hi! >> >> I wonder how Asterisk measures the call duration. The CDR files have a >> accuracy of seconds. Thus, what happens if the call duration is 0.3 >> seconds. What will Asterisk report? 0 seconds? 1 second? >>

Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser

queuemetrics Lee, John (Sydney) wrote: > I am trying to look for a software (open source or proprietory) that > could do reporting on both queue and CDR in Asterisk 1.4.* > > Could someone give me some suggestions? > > >

Re: [asterisk-users] Off Topic: Disable Polycom Soundpoint DoNotDisturb Feature

Roi Stork wrote: > However, the problem is that there is still no ringing sound so the user > can't hear it. Is there a way to make the ringing tone audible? You can remap the DND key to do something else (or nothing). It may still be possible for the user to set DND status via the menus, though.

Re: [asterisk-users] FWD $30 membership-fee

I stand corrected, I finally received a few of these yesterday. They're not unclear about the process; The $30 yearly charge is mandatory. The messages do state that you can link as many accounts to one payment as you'd like though. -- Alex Robar [EMAIL PROTECTED] On Thu, Aug 7, 2008 at 3:05 PM,

[asterisk-users] Newbie: Queue and CDR Reporter and Analyser

I am trying to look for a software (open source or proprietory) that could do reporting on both queue and CDR in Asterisk 1.4.* Could someone give me some suggestions? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon

[asterisk-users] cmdRecord issue related to "iax2 received mini frame before first full voice frame"?

Hi, I tried sending this message a few months back but never lucked into a response. I thought I'd try one more time, juicing up the subject heading a bit, as I am still seeing this behavior intermittently. I'm running several asterisk servers in combination with dundi. The servers are in di

Re: [asterisk-users] BLF functionality

13 aug 2008 kl. 00.45 skrev Dan Peters: We have had Asterisk up and running for a while now and it works very well. Recently we tried to integrate a Linsys SPA962 with the associated SPA932 console. We can get the BLF lights to blink when a phone is ringing and we can get the BLF lights

Re: [asterisk-users] VICIDial error

Solved! You have to get to the end of the "scratch install directions" to find the database setup. This information SHOULD be in the standard vicidial install instructions. Classic case of stupid flippn' administrator combined with poor documentation. Install the database. du! --- On Fri

Re: [asterisk-users] OT: Asterisk on fitPC

On Tue, 12 Aug 2008, mail-lists wrote: Hm. $300 in the US and the UK disty is selling them for just short of £240, so they can go stuff themselves, low-power or not. (I buy 1GHz > systems with 1GB of RAM, running at 15W for half that. No drive though) Gordon, If you don't mind my asking: What

Re: [asterisk-users] I used to use an Asterisk server, but now it is overkill, ...

Ronald Wiplinger wrote: > I had installed in the office an Asterisk server, but the company is > gone and I could keep the server. > > However, for my family with three members and two phone lines this > server is overkill. I am looking for a compact solution, which is more > suitable for me. > >