RE: [Asterisk-Users] Re: Problems with TDM400P card
-Original Message-
From: Adam Goryachev [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 05, 2005 10:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Problems with TDM400P card
On Thu, 2005-05-05 at 12:11 -0400, Andrew Kohlsmith wrote:
On May 5, 2005 11:13 am, Mike Mueller wrote:
Couple this with the fact that the driver now seems to pull 100% CPU
every 5 seconds or so and it didn't before and I think we have a good
case for there being something weird in the driver that is causing frame
slips or other weirdness that is generally not audible for most people
but wreaks havoc even for G3 or ECM (I think that's the term for
error-correcting fax) fax machines.
As measured with top?
No; vmstat 1, without anything (not even asterisk) running.
Driver unloaded: no spiking. Driver loaded: spiking.
Of the three machines running asterisk just now, none of them display
this 'spike' as described. This was while asterisk was running etc as
well.
{clip}
I also observe this spiking behavior with a t100p and cvs-head. I'm running
kernel 2.6.11.8 with the realtime patches applied and see the cpu time going to
the kthread servicing the interrupt that t100p is on (IRQ 9 in my case), with
or without asterisk itself running. This occurs even after raising the priority
of the IRQ 9 kthread to the highest priority process.
However, this on its own doesn't noticeably affect the audio - small timing
defects on the T1 can usually be noticed by dialing into a milliwatt() target
on the problem box across that link. That said, perhaps the jitter is too
slight to hear on a 1004hz sign wave and needs a T1 analyzer to detect properly.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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RE: [Asterisk-Users] Opinions on Cisco 7960G, Polycom IP-600, and Snom 360
On Thu, 2005-05-05 at 17:20 -0500, Daniel Bingham wrote: Reviewing the IP-600 and IP-500 further, the spec sheets have me a little confused. The IP-500 states it supports three lines, and the IP-600 six lines. What is confusing me is that on the IP-600 spec sheet (http://www.polycom.com/common/pw_item_show_doc/1,1276,1820,00.pdf), under the IETF SIP bullet, is says up to 3 dedicated lines, 2 call appearances per line. This same wording is used under the same bullet on the IP-500 spec sheet (http://www.polycom.com/common/pw_item_show_doc/1,1276,1042,00.pdf). Looks like a copy paste error :) Can someone who has used both phones clear this up? Does the IP-600 fully support 6 lines under SIP, or is the line functionality of the IP-500 and IP-600 identical? The IP 600 definitely has 6 separate lines. I have one on my desk, 4 lines to my own asterisk, 1 to another customer asterisk, and the 6th one to a third asterisk box... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I hide caller id on the fly (per each use setting) on Bristuffed * and quadbri
The command you may play with, is CallingPres. The values that did work
for me, with a zaphfc an with Swisscom (telco), are:
- 0 - hide callerID
- 32 - show callerID
There is a quite good explanation you to calculate the presentation on:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20CallingPres
example (extensions.conf):
exten = .0_,1,CallingPres(32)
exten = .0_,2,SetCallerID(123456789)
exten = .0_,3,Dial(Zap/g1/${EXTEN})
exten = .0_,4,Hangup
Jean-Christophe
Robert Rozman a écrit :
Hi,
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 ?
Thanks in advance,
regards,
Rob.
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[Asterisk-Users] sccp transfer question
Hello all, sccp transfer question! Tell at somebody it has turned out to make a transfer of a call between 3 sccp (HARD) phones. |---| |---| |---| | A | --- | B |-(#)-| C | |___| |___| |___| sccp sccp sccp And if that has turned out as? If that is possible an example of a configuration. For me does not work on 7902 :-( -- Best regards, dsv mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold
What version of mpg123 do you use ? You must have a 0.59r one, else you will ear a strange noise, and the nothing. But no warning message. For test, you can try this version: ftp://ftp.proxad.fr/pub/Distributions_Linux/Mandrakelinux/official/10.2/i586/media/main/mpg123-0.59r-23mdk.i586.rpm This is a Mandrake one, that I'm sure it's working. You can extract the binary out of the RPM with mc (Midnight Commander). You can also test the MP3Player, i.e.: exten = 999,1,MP3Player(http://my.stream:8000/stream1) Jean-Christophe Sahil Gupta a écrit : Hi, I've been trying to get music on hold going on one of our servers: Upon dialling extension 005, it plays: -- Executing WaitMusicOnHold(SIP/parssyd1-4dbe, 30) in new stack -- Started music on hold, class 'default', on SIP/parssyd1-4dbe However, no music in the background MPG123 is intalled.. musiconhold.conf shows: default = mp3:/var/lib/asterisk/mohmp3 The directory has?: [EMAIL PROTECTED]:~# ls -al /var/lib/asterisk/mohmp3 total 6589 drwxr-xr-x 2 root root 160 2005-04-21 10:25 ./ drwxr-xr-x 8 root root 216 2005-02-17 22:48 ../ -rw-r--r-- 1 root root 1939812 2005-04-21 10:25 fpm-calm-river.mp3 -rw-r--r-- 1 root root 2582496 2005-04-21 10:25 fpm-sunshine.mp3 -rw-r--r-- 1 root root 2217563 2005-04-21 10:25 fpm-world-mix.mp3 Any clues ? Seems like it actions things but isn't playing the mp3 files.. Regards, Sahil Gupta VoiceValley ___ 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] QoS for improvements
I've spent may hours to play with HTB QoS settings on the firewall, but with absolutely no effect. In fact, this is normal, because the time required to let a data packet going through the ADSL line will break the voice jitter. The only right way to handle this issue is to modify the MTU on the router. Without setting a TOS for voip, data where going through and voice was unusable. With a lowdelay (0x10) TOS set for voip, voice was going through, but data was blocked. With a lowdelay TOS and an HTB QoS on the router, data where going through slowly and voice was scambled. After many tests, an MTU of 700 did work quite well. I did loose 15% of bandwidth for data (twice more overheads), but data and voice may be used together. Those tests have been done on a 256 kbps up stream. There is a quite good explenation about this issue on Cisco's web site, and about they're LFI technology (link fragmentation and interleaving): http://www.cisco.com/warp/public/788/voice-qos/voip-mlppp.html#link_frag Jean-Chrsitophe Kumara Jayaweera a écrit : Hello! Everybody!!, I want to run VoIP in the same LAN (15 windows clients) which we use for surfing the Internet. 6-7 softphones in the same client's machines is 'the target'. My DSL is 128kbps, (I can go to 256kbps if required). So, I am told to install some QoS's in the LAN to improve the voice quality. Frankly, I don't know what it (QoS= Quality of Service) is. I hope you may help me giving Links to read and briefing me your ideas. Thanks to everybody in the list. So far my success and progress are your help. Thanks again Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: polycom phones
Me too... :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor Harrison Sent: Tuesday, April 12, 2005 11:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: polycom phones On Apr 11, 2005 11:49 PM, Greg Boehnlein [EMAIL PROTECTED] wrote: On Mon, 11 Apr 2005, Noah Miller wrote: This this may sound ridiculous, but we've had problems with this when the users did not plug the handset cord in completely. 8 out of our 12 employees made the mistake, as the plug on the IPX00's appears to be all the way in when it is actually not. Not ridiculous at all. We had the same problem. In fact, the cord will click into place when it's not really all the way in. I had the same problem.. :) aolMe too!/aol Took a few minutes to figure it out... was sweating bullets. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 600 not ringing
Erase your caller directory, happened to me because the default ring on directory was 1 (silent) Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Sent: Thursday, April 07, 2005 12:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP 600 not ringing Bugger :-( On Wed, 06 Apr 2005 20:14:06 -0500, Eric Wieling [EMAIL PROTECTED] wrote: Alex wrote: Sorry... It should reed Polycom IP 600 does not make an audible ring sound ...half a sleep :-) On Thu, 07 Apr 2005 10:07:57 +1000, Alex [EMAIL PROTECTED] wrote: Hi guys, Has anyone come across a problem when Polycom IP 600 does not make an audible ring sound, even though the call comes in? I can see it on LCD and red light flashes. When I pickup the phone, everything is fine. It only applies to SIP calls. If the call comes in from PSTN via TDM400 card, everything seems to be ok. The same scenario happens when I try to ring from IP600 other Polycom phones, again they don't ring. I am using 2.6.1 bootrom and 1.4.1 SIP firmware. Anyone can point me in the right direction? I have seen this problem on the Polycom 500. ONLY happens when I call from port 1 of my SPA-2000, works fine on port 2. I cannot see any significant difference in the way the two ports are configured (on the device or in sip.conf). I have no idea how to fix it. ___ 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 -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/ ___ 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] RED ALARM on PRI channel takes Asterisk DOWN
On 05/05/05 18:43 Peter Svensson said the following: My guess is that the Sangoma card does not switch to internal clocking when the external clocking is lost, thus depriving Asterisk of the zaptel if this is the case, the sangoma would need to fix this asap. it'd be hard explaining to a customer that your pbx just died because the telco provided E1 went the way of the dodo. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.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 | +=+ ___ 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] Put a wait in a .call file.
Correct. I'm dialing out through the infamous TDM... It picks up the correct Zap channel on the TDM, and does dial, it just leave out the first #. I'm going to look at my configs again, as suggested by William. I don't know if I will find anything though since I am using a .call file, and it does pickup the correct channel (As I can listen in using another handset that is plugged into that line somewhere else). -Ronan On Thu, 2005-05-05 at 08:03 -0500, Eric Wieling aka ManxPower wrote: Ronan Eckelberry wrote: Don't think I explained it properlyI don't have a problem with the call files. I know how to use them...The problem is, I can sit on a handset directly connected to the POTS line. When * finally picks up the line it has already dialed 1-2 #'s. I am only hearing 6-7 tones dialed...as such it never completes a call and just continues to redial over and over and over... You mist be using analog ports. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: This is a digitally signed message part ___ 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] Put a wait in a .call file.
On Fri, 2005-05-06 at 03:11 -0400, Ronan Eckelberry wrote: Correct. I'm dialing out through the infamous TDM... It picks up the correct Zap channel on the TDM, and does dial, it just leave out the first #. I'm going to look at my configs again, as suggested by William. I don't know if I will find anything though since I am using I'm sorry. I mean Karl. :P -Ronan signature.asc Description: This is a digitally signed message part ___ 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] Put a wait in a .call file.
I'll take a look. Although this is dialing directly out of the analog
port and I can hear it on another handset that is plugged into the same
line.
This is my .call file that I am testing with:
Channel: Zap/28/ww3027357
MaxRetries:
RetryTime: 5
WaitTime: 3
Context: internal
Extension: 123
Priority: 1
It looks right to me, but I am most likely wrong.
-Ronan
On Thu, 2005-05-05 at 10:23 -0400, Karl H. Putz wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ronan
Eckelberry
Sent: Thursday, May 05, 2005 9:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Put a wait in a .call file.
No go. Now, it picks up the phone, waits, and still only dials 6 of the
7 #'s. It is VERY wierd
This sounds like it might be a dialplan or zapata.conf problem.
Be sure that you do not have a stripmsd=1 in your zapata.conf (or any
value greater than 0).
In extensions.conf be sure that in the context that you are dialing from you
are not stripping digits in
the Dial command. i.e. if the Dial command passes a number to be dialed
that includes a : and a number
;exten = _41X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,r)
Above, the first 2 digits, 41, will not be part of the dialstring.
Also double check the TRUNKMSD value in extensions.conf if you are using any
of the default setup
that includes this variable. Anything other than 0 will strip digits.
Good luck,
Karl Putz
-Ronan
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[Asterisk-Users] Harald Baron/EBAROH/CH/Ascom ist nicht anwesend.
Ich werde ab 06.05.2005 nicht im Büro sein. Ich kehre zurück am 29.05.2005. Ich bin vom 6.5.05 bis 29.5.05 nicht per Email erreichbar und werde die Emails sobald als möglich bearbeiten. Dringende Anfragen bitte an Andreas Widrig/CZWIAN/CH/Ascom oder Ralf Knobel/CZKNOR/AScom machen. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to analog pbx
Well , problem solved the problem was with [EMAIL PROTECTED] i have installed an asterisk from scratch and everything works fine now .. weird ./ Thanks! El mié, 04-05-2005 a las 10:23 +0200, Julio Saura escribió: Hi i posted it this morning i guess is a [EMAIL PROTECTED] problem... installing a new OS with * from scratch it does not even call outside connecting fxo to pots :? El mié, 04-05-2005 a las 09:55 +0200, Mehdi Chouikh escribió: Hello all is right, the analog extension should ring, but maybe your dialplan is not correct or you call a bad extension in you PBX. can you post your dialplan?, to see it. regards - Original Message - From: Julio Saura [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, May 03, 2005 2:37 PM Subject: [Asterisk-Users] asterisk to analog pbx Hi there i have an asterisk box running ok, and now i am trying to integrate it with my local analog pbx So far, i have connected the fxo port of my * to an analog extension port of my analog pbx. As far as i know, if a call an extension of my analog pbx on a sip phone ( i have done the right dial plan for routing these calls to de zap channel ) the analog pbx extension should ring ... am i right? asterisk says the call is done, but the analog extension keeps in silence .. :? any clue, am i doing something wrong? Best regards. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] IAXy Firmware Upgrade
Hello, I'd like to known what I have to do to upgrade the firmware into a IAXy device. thanks stefano. ___ 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] Opinions on Cisco 7960G, Polycom IP-600, and Snom 360
On Thu, 2005-05-05 at 18:00 -0500, Daniel Bingham wrote: Apologies for asking more questions so quickly after my last one. A few more questions about the Polycom phones: Searching the list I found a few references like this: I would also like to figure out how to make the phone *ring* when you're already on another line, but haven't had a chance to seriously explore it yet. Yes, it does, whether it is in the handset/headset, or from the ringer, I can't remember, but it does definitely make some sort of audible notification. The Wiki says the IP-500 requires an additional chip to support power over ethernet. Is this true of the IP-600 as well? No, the IP500 requires an additional cable, which may have some chip or something internally. The IP300 is in the same boat. The IP600 only requires a PoE enabled standard cat5 cable (ie, the other end of the cable is inserting power). If anyone can answer any of these questions, I would really appreciate it. Yes, I can, and at least the PoE questions are answered on the polycom website and the wiki. Also, please remember: a) don't top post b) delete irrelevant text Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] IAX hint
Hi all, I want to monitor the status of the IAX2 clients connected to my Asterisk in a way similar to what I do inserting the hint special priority in extensions.conf. I mean, if I put a line like the one below in extensions.conf I get an ExtensionStatus event every time my SIP phone changes its state and its really precious for me. exten = 2000,hint,SIP/2000 The same configuration for an extension bound to a IAX2 client, doesnt seem to work. I can manually request its status with an ExtensionState action, but Id like to have it automatically the way it works with SIP. How can I do? Is there another way for IAX2? Thank you very very very much Luca Maccarini PS. My not working IAX2 conf exten = 2004,1,Dial(IAX2/2004,20,tT) exten = 2004,2,Voicemail(u2003) exten = 2004,102,Voicemail(b2003) exten = 2004,103,Hangup exten = 2004,hint,IAX2/2004 -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.0 - Release Date: 29/04/2005 ___ 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] Is such a thing as a analog (or even IP) video door entry system available?
I want to setup a video door entry system. I understand a lot of the systems on the market use proprietary technology. But ideally if the system could connect into a normal analog port or even use IP to my Asteirsk that would be a lot better. Then I could have video phones on users desks so anyone can see who is at the door. Anyone aware of any suitable products. Angus Comber [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Images
Greg Boehnlein wrote: On Thu, 5 May 2005, Charlie Watts wrote: Manjit Riat wrote: Out of curiosity what's the reason? Why would they not sell phones to asterisk users? Do they not trust asterisk or their phones to work with each other? My guess: They don't want to compete with the folks that OEM Polycom hardware. Lots of commercial phone system vendors just re-brand Polycom phones, and Polycom doesn't want to hurt their relationship with those businesses. I bet that at some point one of the Asterisk-using Polycom vendors gets the momentum to get a better Polycom-Asterisk user relationship going. The trouble is that most of the Asterisk-using hardware vendors are hardware agnostic, and don't want sign anything that says We'll only sell Polycom equipment. This is not the case. I had a long conversation with Polycom's VP of Product management at VON this year, and asked why they didn't want to support Asterisk. Their response is that Asterisk has not yet completed the Self-Certification program w/ Polycom, and that this should be a relatively easy thing to do. They recognize that they are missing opportunites w/ Asterisk, but they want to make sure that their phones work correctly with it, and that Digium will take some responsibility for certifying their code w/ Polycom. Last I heard, Polycom was supposed to be shipping a couple of cases of phones to Digium for certification w/ Asterisk Business Edition. Wonderful! ___ 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
R: [Asterisk-Users] IAX hint
Please, help :( Luca Maccarini Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Luca Maccarini Inviato: venerd 6 maggio 2005 10.27 A: Asterisk Users Oggetto: [Asterisk-Users] IAX hint Hi all, I want to monitor the status of the IAX2 clients connected to my Asterisk in a way similar to what I do inserting the hint special priority in extensions.conf. I mean, if I put a line like the one below in extensions.conf I get an ExtensionStatus event every time my SIP phone changes its state and its really precious for me. exten = 2000,hint,SIP/2000 The same configuration for an extension bound to a IAX2 client, doesnt seem to work. I can manually request its status with an ExtensionState action, but Id like to have it automatically the way it works with SIP. How can I do? Is there another way for IAX2? Thank you very very very much Luca Maccarini PS. My not working IAX2 conf exten = 2004,1,Dial(IAX2/2004,20,tT) exten = 2004,2,Voicemail(u2003) exten = 2004,102,Voicemail(b2003) exten = 2004,103,Hangup exten = 2004,hint,IAX2/2004 -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.0 - Release Date: 29/04/2005 -- Internal Virus Database is out-of-date. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.0 - Release Date: 29/04/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.5 - Release Date: 04/05/2005 ___ 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] FXO ATA?
Jon Gabrielson wrote: On Thursday 05 May 2005 05:28 pm, Joseph wrote: It has 1-FXS and one 1-Life Line (it is pass through type) I've seen the pass-through term used alot and I'm not quite for sure what that means. What is the difference between a passthrough type and a regular FXO. What can you do with one that you can't do with the other? I noticed that the wiki says that the handytone 486's lifeline FXO port is not usable via SIP, only used as a fallback for power failure. Is this considered a passthrough or are there 3 types, pass-through, lifeline, and full FXO. I just checked my dictionary and it defines pass-thru as meaning totally useless for most people. Pass-thru and lifeline seem to be different terms for the same thing. i.e. The FXO port is connected to the FXS port in the event of a power outage, but other than that it is not useful. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: 911 service
Federal Communications Commission chairman Kevin Martin has proposed requiring internet-based telephone services to offer 911 emergency services to customers by as early as the end of September, people familiar with the plan said on Wednesday. full article http://www.wired.com/news/technology/0,1282,67436,00.html It is unclear how this would be implemented (normally a dedicated T1 is connected to the swtich that the PSAP is connected to, this i a bit cumbersome for a voip network that exists largely in one physical location. There are companies that will route this traffic for you however. If this happens VoIP could take off more than it is, although all the FUD articles about VoIP insecurities are more likely to create consumer hysteria than anything real. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] MOH per User
I've tried it by changing the MOH context after I have identified the caller but I find that mpg123 doesn't properly switch over to the new context. Once it starts playing a stream it appears to be stuck with it. exten = s,1,SetMusicOnHold(default) MARK. [EMAIL PROTECTED] wrote: Is it possible to set different hold music for different users or a different context? I searched the wiki and I don't really see anything that mentions be able to do it on a per user basis or per context basis. Thanks. Peder ___ 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] Re: CDR for PSTN
hello Thanks for replying. i know duration and billsec. but i am getting wrong billsec. for example in one call billsecduration 48 55 and actually in this call phone rings 10 seconds. and accual duration on my cell phone is 35 Hi, Look at http://www.voip-info.org/wiki-Asterisk+billing duration: Total time in system, in seconds (integer), from dial to hangup What are you looking for (from my point of view) is billsec: Total time call is up, in seconds (integer), from answer to hangup -b - Original Message - From: Kamran Ahmad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, May 05, 2005 7:39 AM Subject: [Asterisk-Users] Re: CDR for PSTN hello Any help. CDR duration starts from 183 Session Progress. cdr duration should start from 200 OK when both parties are inside session. i am using Quintum gw for PSTN Calls. here is the call flow between Asterisk and QuintumGateway. ASTERISK GW 1 |-INTITE--| 2 |183 Session Progress-| cdr starts 3 |180 Ringing--| 4 |200 Ok---| 5 |ACK--| should here 6 |AUDIO Session---| any idea why call duration is starting from step 2. actually session starts from step 5. Kamran Discover Yahoo! Get on-the-go sports scores, stock quotes, news and more. Check it out! http://discover.yahoo.com/mobile.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is such a thing as a analog (or even IP) video door entry system available?
Angus Comber wrote: I want to setup a video door entry system. I understand a lot of the systems on the market use proprietary technology. But ideally if the system could connect into a normal analog port or even use IP to my Asteirsk that would be a lot better. Then I could have video phones on users desks so anyone can see who is at the door. Anyone aware of any suitable products. I don't know if that's what you mean, but you may take a look at ZoneMinder? http://www.zoneminder.com/ Tomek -- Znajdz swoja milosc na wiosne... http://link.interia.pl/f187a ___ 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] Re: Sangoma A102 cards testing FIXED
Title: Re: Sangoma A102 cards testing FIXED Hi all! I have got same problem with Sangoma A102 and didn't get how did you fix a problem? Can anybody explain? -Original Message- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vikram Rangnekar Sent: Tuesday, February 15, 2005 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: Sangoma A102 cards testing FIXED Its fixed and working great. I was working on the dma issue that nenad pointed out and when i tried to hdparm -d 1 /dev/hda my harddisk i got a permissioned denied error (I was root) So i started researching a bit more and realised that since i have a sata hdd and running it in IDE mode i cant start dma so i just recompiled the kernel with sata drivers and scsi activated then i change the bio SATA setting to SATA ENHANCED (nothing else seems to work) and changed the root=/dev/hda2 to root=/dev/sda2 in the kernel boot options and i was on my way. now when i tierd hdparm -d 1 /dev/sda i was told that scsi dosent have dma so it didnt matter. Next i started the wanpipe drivers and started asterisk i didnt get any errors so to test th config i used exten = 111,1,Dial(Zap/g1/301) exten = 111,2,Hangup and dialed that extension 111 and the extension 301 rang. i had started pri intense debugging and used show channels to make sure the call went over the e1 channels. i'm still testing it, it seems to be working great right now only error i got was a FCS BAD or somthing like that once. My motherboard is a SuperMicro P4SCI just for your information. br, dmitry ___ 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] DTMF oddity with OH323
I've recently started using OH323 channels, having previously used IAX. When receiving DTMF input from a caller (I'm using WAIT FOR DATA in AGI), the OH323 channel is supplying a space digit between each real digit. This didn't happen with IAX. Here is the full log with verbose=9 and debug=9 and agi debug: AGI Rx GET DATA announcement 4000 4 May 6 11:24:53 DEBUG[17236]: Set channel OH323/R22348 to write format slin May 6 11:24:53 DEBUG[17236]: OH323/R22348: Created smoother (size=320, format=8). May 6 11:24:53 DEBUG[17236]: Scheduling timer at 160 sample intervals May 6 11:24:53 VERBOSE[17236]: -- Playing 'announcement' (language 'en') May 6 11:24:55 DEBUG[17237]: ENTER exception_h323_connection. May 6 11:24:55 DEBUG[17237]: Call 'ip$xxx.xx.xxx.x:41161/22348' has exception USER_INPUT_TONE. May 6 11:24:55 DEBUG[17237]: LEAVE exception_h323_connection. May 6 11:24:55 DEBUG[17236]: OH323/R22348: Channel has generated an exception. May 6 11:24:55 DEBUG[17236]: ENTER oh323_exception. May 6 11:24:55 DEBUG[17236]: OH323/R22348: Got DTMF 1. May 6 11:24:55 DEBUG[17236]: LEAVE oh323_exception. May 6 11:24:55 DEBUG[17236]: Scheduling timer at 0 sample intervals May 6 11:24:55 DEBUG[17236]: Set channel OH323/R22348 to write format alaw May 6 11:24:56 DEBUG[17237]: ENTER exception_h323_connection. May 6 11:24:56 DEBUG[17237]: Call 'ip$xxx.xx.xxx.x:41161/22348' has exception USER_INPUT_TONE. May 6 11:24:56 DEBUG[17237]: LEAVE exception_h323_connection. May 6 11:24:56 DEBUG[17236]: OH323/R22348: Channel has generated an exception. May 6 11:24:56 DEBUG[17236]: ENTER oh323_exception. May 6 11:24:56 DEBUG[17236]: OH323/R22348: Got DTMF . May 6 11:24:56 DEBUG[17236]: LEAVE oh323_exception. May 6 11:24:56 DEBUG[17237]: ENTER exception_h323_connection. May 6 11:24:56 DEBUG[17237]: Call 'ip$xxx.xx.xxx.x:41161/22348' has exception USER_INPUT_TONE. May 6 11:24:56 DEBUG[17237]: LEAVE exception_h323_connection. May 6 11:24:56 DEBUG[17236]: OH323/R22348: Channel has generated an exception. May 6 11:24:56 DEBUG[17236]: ENTER oh323_exception. May 6 11:24:56 DEBUG[17236]: OH323/R22348: Got DTMF 3. May 6 11:24:56 DEBUG[17236]: LEAVE oh323_exception. May 6 11:24:56 DEBUG[17237]: ENTER exception_h323_connection. May 6 11:24:56 DEBUG[17237]: Call 'ip$xxx.xx.xxx.x:41161/22348' has exception USER_INPUT_TONE. May 6 11:24:56 DEBUG[17237]: LEAVE exception_h323_connection. May 6 11:24:56 DEBUG[17236]: OH323/R22348: Channel has generated an exception. May 6 11:24:56 DEBUG[17236]: ENTER oh323_exception. May 6 11:24:56 DEBUG[17236]: OH323/R22348: Got DTMF . May 6 11:24:56 DEBUG[17236]: LEAVE oh323_exception. May 6 11:24:56 VERBOSE[17236]: AGI Tx 200 result=1 3 May 6 11:24:56 VERBOSE[17236]: AGI Tx 200 result=1 3 AGI Rx WAIT FOR DIGIT 4000 May 6 11:24:56 DEBUG[17237]: ENTER exception_h323_connection. May 6 11:24:56 DEBUG[17237]: Call 'ip$xxx.xx.xxx.x:41161/22348' has exception USER_INPUT_TONE. May 6 11:24:56 DEBUG[17237]: LEAVE exception_h323_connection. May 6 11:24:56 DEBUG[17236]: OH323/R22348: Channel has generated an exception. May 6 11:24:56 DEBUG[17236]: ENTER oh323_exception. May 6 11:24:56 DEBUG[17236]: OH323/R22348: Got DTMF 2. May 6 11:24:56 DEBUG[17236]: LEAVE oh323_exception. May 6 11:24:56 VERBOSE[17236]: AGI Tx 200 result=50 May 6 11:24:56 VERBOSE[17236]: AGI Tx 200 result=50 AGI Rx WAIT FOR DIGIT 4000 May 6 11:24:56 DEBUG[17237]: ENTER exception_h323_connection. May 6 11:24:56 DEBUG[17237]: Call 'ip$xxx.xx.xxx.x:41161/22348' has exception USER_INPUT_TONE. May 6 11:24:56 DEBUG[17237]: LEAVE exception_h323_connection. May 6 11:24:56 DEBUG[17236]: OH323/R22348: Channel has generated an exception. May 6 11:24:56 DEBUG[17236]: ENTER oh323_exception. May 6 11:24:56 DEBUG[17236]: OH323/R22348: Got DTMF . May 6 11:24:56 DEBUG[17236]: LEAVE oh323_exception. May 6 11:24:56 VERBOSE[17236]: AGI Tx 200 result=32 May 6 11:24:56 VERBOSE[17236]: AGI Tx 200 result=32 AGI Rx WAIT FOR DIGIT 4000 May 6 11:24:57 DEBUG[17237]: ENTER exception_h323_connection. May 6 11:24:57 DEBUG[17237]: Call 'ip$xxx.xx.xxx.x:41161/22348' has exception USER_INPUT_TONE. May 6 11:24:57 DEBUG[17237]: LEAVE exception_h323_connection. May 6 11:24:57 DEBUG[17236]: OH323/R22348: Channel has generated an exception. May 6 11:24:57 DEBUG[17236]: ENTER oh323_exception. May 6 11:24:57 DEBUG[17236]: OH323/R22348: Got DTMF 2. May 6 11:24:57 DEBUG[17236]: LEAVE oh323_exception. May 6 11:24:57 VERBOSE[17236]: AGI Tx 200 result=50 May 6 11:24:57 VERBOSE[17236]: AGI Tx 200 result=50 [...etc...] Notice that GET DATA for 4 digits is returning with 1 3 when I tried to enter 1322. I then have further calls to WAIT FOR DIGIT which alternately return digit and space. Is this an error in chan_oh323, or an upstream setting? My guess is that the space tones are indicating key-up, i.e. the end of the tone. I would be inclined to ignore the space exception in the function oh323_exception(), as follows:
[Asterisk-Users] Re: DTMF oddity with OH323
I wrote:
I would be inclined to ignore the space exception in the function
oh323_exception(), as follows:
--- chan_oh323.c2005-04-18 21:20:30.0 +0100
+++ chan_oh323.c.new2005-05-06 11:36:03.0 +0100
@@ -1164,7 +1164,7 @@
p-fr.subclass = AST_CONTROL_FLASH;
if (option_debug)
ast_log(LOG_DEBUG, %s: Got flash hook.\n,
c-name);
- } else {
+ } else if (p-except_struct.data[0] != ' ') {
p-fr.frametype = AST_FRAME_DTMF;
p-fr.subclass = p-except_struct.data[0];
if (option_debug)
I'm using v0.6.5 of chan_oh323 with Asterisk CVS-STABLE.
Just for the record, I've now tried the above change, and it completely
fixes the problem I was having.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Problems With MusicOnHold
Hi: I recently reinstalled Asterisk 10.1 and I am experiencing a new problem. I can't play muxiconhold. On the Asterisk Console, I get the following message when MusicOnHold attempts te be executed and I don't hear anything. Executing MusicOnHold(OSS/dsp, ) in new stack -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Thanks for any suggestions. Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html ___ 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] FXO ATA?
Why not go with Multitech? They are expensive, but great units. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Friday, May 06, 2005 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXO ATA? I've been looking at something similar, but with more ports. Something to handle the incoming (FXO) analog lines, but without the investment in a channel bank and T1 card because we only need 4-8 FXOs and no FXS. I've looked at the AudioCodes MP104 which looks like it will take the FXOs and turn them into SIP channels. Anyone have experience with these? Maybe my lack of experience is causing incorrect expectations. While they are pricey (~$1,000US), they are still cheaper than a T1 card and a channel bank I think. On Thu, May 05, 2005 at 08:07:14AM -0400, Chris Mason (Lists) wrote: Is the Sipura 3000 the only way to interface a remote pstn line and connect incoming calls to Asterisk? I have a location connected by network that has a phone line, when the room is occupied I want the line ti ring there as normal, but when the employee is travelling I want the line to be conencted to a ATA that then feeds it as an incoming pstn line to the pbx located at my office so it can follow her. It sounds like the Sipura 3000 would be perfect, what else would do it? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ 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] oh323 compile problem in FreeBSD
Hi, I'm trying to compile asterisk-oh323-0.7.1 in FreeBSD 5.3. I tried to use gmake but it exits with too many errors. Did somebody compile before oh323 in FreeBSD? How should I compile it under FreeBSD? thanks, Ganbold ___ 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] CAPI on ptp with variable length digits in phone number
Hi! we have a german PtP PRI connection here. our old telephone system was programmed to accept digits by variable length. so our MSN, assigned my telco is, lets say: 123 and we can use first digit from 0-4. and every further digit like we want. means: 123-1 123-2345 123-44 till 123-499 but not 123-5... I'm using an Eicon diva server PRI 23M with chan_capi. my problem is dialing IN. block mode works perfect. (when the whole number is sent as a block) I just add exten = 123114,1,Dial(SIP/blahblah) works. but if someone dials digit-wise, lets say 123114, asterisk starts scanning the dialplan after 1231. doesnt find an extension and exits. even using exten = s,1,DigitTimeout,4 as first line in the dialin-extension won't help. so, I need to find a way that asterisk collects the digits until it has a matching one. p.ex. wait scanning the extensions till the caller typed 123114. I can live with fixed length extensions. means, always wait for 3 digits. thanks for help.. Sebastian ___ 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] Zapata.Conf Sanity Check
Hello, Haven't had a chance to test this configuration yet, so I can't really answer my own question, but wanted to get a couple of eyeballs to look at it and verify that I'm going to get the behavior that I expect out of it. Basically, I have a TE405P w/ 3 PRI to the Telco and one going to a legacy PBX, all speaking NI-2 (National ISDN 2). I need to use pri_cpe w/ the Telco and pri_net for the PBX, and I want them in different groups. I hacked the following up last night, while falling asleep, and I'm pretty sure it will do what I want, but I'd like a second opinion. ; Zapata telephony interface ; ; Configuration file [channels] ; Inbound PRI from Telco context=inbound switchtype=national signalling=pri_cpe group=1 channel = 1-23,25-47,49-71 ; Outbound PRI to PBX context=pbx switchtype = national signalling = pri_net group = 2 channel = 73-95 Thanks.. -- 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
Re: [Asterisk-Users] CAPI on ptp with variable length digits in phone number
hi back, we had this sort of problems (and some other ones... sigh) with an eicon diva chan_capi in austria. rant mode=on unfortunately we never got around to fix this spaghetti of a code in chan_capi.c to work as intended (it takes the first session_setup it gets from the line and ignores, that per specification of isdn you can send digits for DID _after_ the setup of the connection too. there should be a timer with timeout for waiting for those digits. we tried to implement a separate thread into chan_capi to handle this timeout, but the code and the variable-naming is so obscure, we never got arround to finding let alone fixing a invalid pointer not freed bug in our hack of this hack). rant mode=off as a last desperate try we got a sirrix-card (search on the wiki for sirrix) and i had some very good calls with sirrix' development department, who fixed all problems in the isdn-layer and the asterisk-channel (chan_sirrix) for us. works perfectly. sorry i have no immediate solution, but i dumped the avm and eicon cards completely in favor of the sirrix ones. Am Freitag, 6. Mai 2005 13:04 schrieb Sebastian Buntin: Hi! we have a german PtP PRI connection here. our old telephone system was programmed to accept digits by variable length. so our MSN, assigned my telco is, lets say: 123 and we can use first digit from 0-4. and every further digit like we want. means: 123-1 123-2345 123-44 till 123-499 but not 123-5... I'm using an Eicon diva server PRI 23M with chan_capi. my problem is dialing IN. block mode works perfect. (when the whole number is sent as a block) I just add exten = 123114,1,Dial(SIP/blahblah) works. but if someone dials digit-wise, lets say 123114, asterisk starts scanning the dialplan after 1231. doesnt find an extension and exits. even using exten = s,1,DigitTimeout,4 as first line in the dialin-extension won't help. so, I need to find a way that asterisk collects the digits until it has a matching one. p.ex. wait scanning the extensions till the caller typed 123114. I can live with fixed length extensions. means, always wait for 3 digits. thanks for help.. Sebastian ___ 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 pgpB1mKAGhmIe.pgp Description: PGP signature ___ 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] CAPI on ptp with variable length digits in phone number
hi again, just ignore my mentioning of the sirrix-cards, just realised you have a PRI, i overread it and thought you had a BRI. so i think your last hope is a zaptel-card. regards, Am Freitag, 6. Mai 2005 13:04 schrieb Sebastian Buntin: Hi! we have a german PtP PRI connection here. our old telephone system was programmed to accept digits by variable length. so our MSN, assigned my telco is, lets say: 123 and we can use first digit from 0-4. and every further digit like we want. means: 123-1 123-2345 123-44 till 123-499 but not 123-5... I'm using an Eicon diva server PRI 23M with chan_capi. my problem is dialing IN. block mode works perfect. (when the whole number is sent as a block) I just add exten = 123114,1,Dial(SIP/blahblah) works. but if someone dials digit-wise, lets say 123114, asterisk starts scanning the dialplan after 1231. doesnt find an extension and exits. even using exten = s,1,DigitTimeout,4 as first line in the dialin-extension won't help. so, I need to find a way that asterisk collects the digits until it has a matching one. p.ex. wait scanning the extensions till the caller typed 123114. I can live with fixed length extensions. means, always wait for 3 digits. thanks for help.. Sebastian ___ 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 pgpNP7ftIvREg.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: CDR for PSTN
Tried ForkCDR. Kritikus. From: Kamran Ahmad [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: CDR for PSTN Date: Fri, 6 May 2005 03:26:30 -0700 (PDT) hello Thanks for replying. i know duration and billsec. but i am getting wrong billsec. for example in one call billsecduration 48 55 and actually in this call phone rings 10 seconds. and accual duration on my cell phone is 35 Hi, Look at http://www.voip-info.org/wiki-Asterisk+billing duration: Total time in system, in seconds (integer), from dial to hangup What are you looking for (from my point of view) is billsec: Total time call is up, in seconds (integer), from answer to hangup -b - Original Message - From: Kamran Ahmad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, May 05, 2005 7:39 AM Subject: [Asterisk-Users] Re: CDR for PSTN hello Any help. CDR duration starts from 183 Session Progress. cdr duration should start from 200 OK when both parties are inside session. i am using Quintum gw for PSTN Calls. here is the call flow between Asterisk and QuintumGateway. ASTERISK GW 1 |-INTITE--| 2 |183 Session Progress-| cdr starts 3 |180 Ringing--| 4 |200 Ok---| 5 |ACK--| should here 6 |AUDIO Session---| any idea why call duration is starting from step 2. actually session starts from step 5. Kamran Discover Yahoo! Get on-the-go sports scores, stock quotes, news and more. Check it out! http://discover.yahoo.com/mobile.html ___ 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 _ FREE pop-up blocking with the new MSN Toolbar get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ ___ 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] 3 x TDM400P in one PC ??
Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? Ciao Joerg -- _ Don't PANIC ___ 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] FXO ATA?
Why not go with Multitech? They are expensive, but great units. For the same cost I could get a T1 card and a channel bank on Ebay and have change left over. These are exepense units. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 ___ 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] Chan_misdn - cannot get the channel driver to load...
Title: Chan_misdn - cannot get the channel driver to load... Hi. The problem I'm having is trying to get the chan_misdn channel module to load when * is started for isdn functionality. I have downloaded the chan_misdn-beta-0.0.3-rc6 cvs and compiled it. Also I have copied the chan_misdn.so file into the ./asterisk/modules/ directory. I am using the AVM Fritz PCI card and avmfritz driver. Here is the error message I receive when I issue the command #asterisk -vgc to start asterisk: [chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables the asterisk to use hardware which is supported by the new ) debug_init: using stdout for debug log debug_init: using stderr for warning log debug_init: using stderr for error log debug_init: debug_mask = 0 Locking Config Mutex UnLocking Config Mutex cannot request MGR_NEWENTITY from mISDN: Illegal seek flax pbxuser # Warning, flexible rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Here is some configuration: misdn.conf [general] context=Dial-In-Service debug=3 language=en [globals] ports=1 context=Dial-In-Service modules.conf [modules] autoload=yes noload = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so noload = chan_modem.so noload = chan_oss.so noload = chan_skinny.so noload = chan_modem_i4l.so noload = chan_modem_bestdata.so noload = chan_modem_aopen.so noload = app_groupcount.so noload = chan_agent.so noload = app_txtcidname.so noload = app_controlplayback.so load = res_features.so load = res_musiconhold.so load = chan_capi.so noload = chan_zap.so load = chan_misdn.so [global] chan_capi.so=yes flax pbxuser # lsmod Module Size Used by avmfritz 21450 1 mISDN_x25dte 32956 1 mISDN_capi 104768 1 mISDN_l2 45280 1 mISDN_l1 10824 1 mISDN_isac 14976 1 avmfritz mISDN_core 76708 6 avmfritz,mISDN_x25dte,mISDN_capi,mISDN_l2,mISDN_l1,mISDN_isac tor2 88224 0 wcfxo 11200 0 wcfxs 123680 0 zaptel 220420 3 tor2,wcfxo,wcfxs crc_ccitt 1920 1 zaptel b1pcmcia 5184 0 b1isa 8528 0 b1pci 7744 0 c4 18052 0 t1isa 21984 0 t1pci 5344 0 b1dma 14916 2 b1pci,t1pci b1 22368 7 b1pcmcia,b1isa,b1pci,c4,t1isa,t1pci,b1dma fcpci 500856 0 capi 15680 0 capifs 4136 2 capi kernelcapi 44864 11 mISDN_capi,b1pcmcia,b1isa,b1pci,c4,t1isa,t1pci,b1dma,b1,fcpci,capi e100 31840 0 Any help would be much appreciated. Phil. ___ 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] 3 x TDM400P in one PC ??
Joerg Wleklik wrote: Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? If you need 12 ports then you should use a T-1 card and a Channel Bank. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Sangoma A102 cards testing FIXED
Title: Re: Sangoma A102 cards testing FIXED I have got same problem with Sangoma A102 and didn't get how did you fix a problem? Can anybody explain? I have been working with Sangoma on a A101 installation and the fixes they came up with will be rolled into a new beta they are releasing in the next day or two. One thing I found out was you must stop the Zaptel modules loading, wanrouter will load zaptel at startup. Chris Masonwww.anguillaguide.comTel: (305) 704-7249 Fax: (815)301-9759 ___ 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] Re: Zapata.Conf Sanity Check
In article [EMAIL PROTECTED], Greg Boehnlein [EMAIL PROTECTED] wrote: Hello, Haven't had a chance to test this configuration yet, so I can't really answer my own question, but wanted to get a couple of eyeballs to look at it and verify that I'm going to get the behavior that I expect out of it. Basically, I have a TE405P w/ 3 PRI to the Telco and one going to a legacy PBX, all speaking NI-2 (National ISDN 2). I need to use pri_cpe w/ the Telco and pri_net for the PBX, and I want them in different groups. I hacked the following up last night, while falling asleep, and I'm pretty sure it will do what I want, but I'd like a second opinion. ; Zapata telephony interface ; ; Configuration file [channels] ; Inbound PRI from Telco context=inbound switchtype=national signalling=pri_cpe group=1 channel = 1-23,25-47,49-71 ; Outbound PRI to PBX context=pbx switchtype = national signalling = pri_net group = 2 channel = 73-95 Thanks.. Looks fine to me too. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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] Transparently Routing German pri through Asterisk
Hi, at the moment we have in Avaya Integral PBX with german pri (30 lines). We want to smouthly migrate to an Asterisk server. For this reason: Is it possible to route the external german pri (E1) through Asterisk server to that Avaya PBX? I think at first we need a Digium e1 card 4-Port. But how do we have to configure the routing of the whole PRI? I really would appreciate any sample config. Thanks, Kib ___ 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] 3 x TDM400P in one PC ??
On Fri, 6 May 2005 22:10, Eric Wieling aka ManxPower wrote: Joerg Wleklik wrote: Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? If you need 12 ports then you should use a T-1 card and a Channel Bank. That would be easy, but.. I have 8 analogue lines incoming right now and changing the phone number is not an option (costs for advertising). This lines go right now into an analogue PBX. A new building will get IP-Phones connected to an asterisk box. The idea is to take the incoming calls in the asterisk, route to the new building via IP and serve the old PBX with 4 analogue lines. Ciao Joerg -- _ Don't PANIC pgpXevJAxDKcy.pgp Description: PGP signature ___ 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] 3 x TDM400P in one PC ??
Joerg Wleklik wrote: On Fri, 6 May 2005 22:10, Eric Wieling aka ManxPower wrote: Joerg Wleklik wrote: Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? If you need 12 ports then you should use a T-1 card and a Channel Bank. That would be easy, but.. I have 8 analogue lines incoming right now and changing the phone number is not an option (costs for advertising). This lines go right now into an analogue PBX. A new building will get IP-Phones connected to an asterisk box. The idea is to take the incoming calls in the asterisk, route to the new building via IP and serve the old PBX with 4 analogue lines. A channel bank provides analog ports and allows you to interface them to Asterisk via a T-1 card. Asterisk(T-1 card)-(T-1 interface)Channel Bank(analog interface) ___ 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] snom mass deployment (probably off topic)
Hello,
to prevent further rumores and wrong facts about our phones:
All our phones, and this includes the 360 as well, do store their settings on
the flash. After the settings are stored once, you can leave the phone as
long as you want without power, and it will come up with old settings
whenever you restart it. If this is not the case the person which is facing
the problem should contact the snom support to get this sorted out.
Regarding the real topic: did you already read our white-papers about
mass-deployment and setting up snom phones?
http://www.snom.com/white_papers.html
You do not have to mess around with faked DNS responses. If the phone is
getting a settings server via DHCP, it will never contact snom.com (expcept
you say so). The phone just falls back to snom.com as a default setting, in
case it cant find a setting server locally. But in case: you can even turn
off that the phone tries to load any settings from any server.
Best regards
Nils Ohlmeier
On Friday 06 May 2005 05:40, Daniel Bingham wrote:
Hi David,
First, thanks for the reply to my questions about the Snom 360. I may have
a few followup questions when I get a little more time.
As for the 360 getting the configuration directly from Snom's servers, I
find that very backwards. What if your phones have no gateway to the
internet? It sounds like they are working around not having any flash
memory, but it's a poor workaround. Your idea of using DNS to fool it into
going to your servers is a good one. I assume you'll just put in a mapping
for provisioning.snom.com or just snom.com in your DNS server to the IP of
your web server.
If I understand correctly, you will be able to create an
/snom360/snom360.php script on your web server, which you would then like
to redirect to the static html files in the /snom directory. Assuming the
Snom supports redirects, the PHP code is as simple as:
?php header(Location: /snom/snom360-$_GET[mac].html) ?
If the phone doesn't support redirects, it gets a little complex, in that
the script will need to open the file from the filesystem and return it
directly.
If I misunderstood or I didn't make sense, I'll be happy to try again.
Thanks,
Daniel Bingham
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] ] On Behalf Of David John
Walsh Sent: Thursday, May 05, 2005 8:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] snom mass deployment (probably off topic)
Hello
Although not stictly a asterisk issue, any help would be apreciated.
Firstly a few notes on the snom 360, which I have had on a test bed for the
last week. Its a great phone, with a good user interface, both physically
and its web based one.
At its lastest firmware it does have a few quirks, with regards to the way
it handles usernames and passwords on the physical interface. These have
been passed back, and hopefully will be addressed.
Its worst feature as I see it is twofold, with regards to its power fail
features. If it loses power for more than a few minuites it loses its
settings - not the best thing in a world where routers and firewalls can be
given power back days later and be fine.
It has an interesting configuration mode, it tries to contact snom, who
then (if told about it) goes to their national distrubtor who then either
has your config or passes it on again
The settings file is well documented, and you can pull them direct from
phone in a ready to go way.
---
I now have my configs in the file name format of snom360-{mac}.htm (where
{mac} is the MAC address of the phone in question)
The phone initally tries to goto
provisioning.snom.com/snom360/snom360.html this sends it onto
http://snom.com/snom360/snom360.php?mac=
http://snom.com/snom360/snom360.php?mac= {mac}
Assuming that I perform some creative dns records on my dns server, would
someone be kind enough to write some sample php code to take the url
http://snom.com/snom360/snom360.php?mac=
http://snom.com/snom360/snom360.php?mac= {mac}
and provide the url http://asterisk-demo/snom/snom360-
http://asterisk-demo/snom/snom360- {mac}.html
The code the url needs to go in is as follows:
# Redirect all phones to the php script
setting_server: http://asterisk-demo/snom/snom360-
http://asterisk-demo/snom/snom360- {mac}.html
I'm useless with php and most launguages, so thank you to any help this
request generates
David
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[Asterisk-Users] Web GUI
Hello all, I just installed Asterisk 1.0.7 and astguiclient so you can say Im very new at this. How can I manage my Asterisk using the web or somehow, since there are too many configuration files and too many variables Thanks, Marc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how do I register my Asterisk with oh323 on gatekeeper?
I use Asterisk CVS-HEAD version with oh323-0.7.2. I have a H323-ID and e.164 numbers. But I don't know how to register my Asterisk on gatekeeper. Please let me know how to config oh323.conf. I will appreciate your help. ___ 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] Web GUI
Look at [EMAIL PROTECTED] project. Its better to starting with it... -b - Original Message - From: Marc Khayat To: asterisk-users@lists.digium.com Sent: Friday, May 06, 2005 3:46 PM Subject: [Asterisk-Users] Web GUI Hello all, I just installed Asterisk 1.0.7 and astguiclient so you can say Im very new at this. How can I manage my Asterisk using the web or somehow, since there are too many configuration files and too many variables Thanks, Marc ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mitel SX200 integration
I am changing one a resort over from a SX200 with digital phones to Asterisk with Polycom phones. The office can be changed over immediately but the guest's villas can't until we do some rewiring, and that will be villa by villa as they are available. How can I merge Asterisk into the existing installation? For example, I would like to have extensions 7XX be on Asterisk while 1XX, 2XX, and 3XX are on the mitel. Any ideas? Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 ___ 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] good bri card not junghanns
On Fri, 2005-05-06 at 14:29 +0200, Eugenio De Vena wrote: Hi there, will someone suggest me a good and * combatible isdn card ( 1 , 2 , 4 , 8 channels ). I am currently working with but can not stand their complete lack of support. In all fairness to Junghanns, my current release Asterisk CVS-D2005.05.02.22.00.00-05/04/05-18:22:14 - is rather cool Someone was busy over the May 1st long weekend. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how do I register my Asterisk with oh323 on gatekeeper?
Hi Kim. could you compile oh323 with current cvs head? i cannot. what pwlib and openh323 version you use? where can i download this? i use h323 asterisk native driver with stable release 1.0.7 and all is ok, but in cvs head, the channel appear to be broken, it compile, but is not functional. Best regardws. César García. Director de Sistemas, IdecNet S.A. Centro de Gestión de Red. Edificio IdecNet. C/Juan XXIII 44. E-35004, Las Palmas de Gran Canaria, Islas Canarias - España. Tfn: +34 828 111 000 Ext: 340 Kim Daeyong escribió: I use Asterisk CVS-HEAD version with oh323-0.7.2. I have a H323-ID and e.164 numbers. But I don't know how to register my Asterisk on gatekeeper. Please let me know how to config oh323.conf. I will appreciate your help. ___ 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] Web GUI
Marc Khayat wrote: Hello all, I just installed Asterisk 1.0.7 and astguiclient so you can say Im very new at this. How can I manage my Asterisk using the web or somehow, since there are too many configuration files and too many variables You may take a look at [EMAIL PROTECTED] - it includes AMP (Asterisk Management Portal) and other tools. But it will be hard at the beginning, anyway :) Tomek -- Znajdz swoja milosc na wiosne... http://link.interia.pl/f187a ___ 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] 3 x TDM400P in one PC ??
Well lets do the math: 1 * TDM40B: TDM400P + 4 PORT FXS Bundle = $305.00 http://store.yahoo.com/asteriskpbx/witd4pofxsbu.html 2 * TDM04B: TDM400P + 4 PORT FXO Bundle = $674.00 ($337.00 ea) http://store.yahoo.com/asteriskpbx/newitd4pofxo.html Total = $979.00 Vs this: 1 * Wildcard TE110P = $595 http://store.yahoo.com/asteriskpbx/wildcarde100p.html 1 * CAC Adit 600 (new) with 3 FXS and 1 FXO = $955 Buy it now option http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemcategory=51271item=5771094378rd=1 Total = $1550 and for that you get 24 FXS (well only total of 24 ports is usaable since you only have a single span T1), and 8 FXO What you really need is only 1 FXS card and 1 FXO card with the Adit 600. If you really go shopping on ebay you shoud be able to get it all for around $500.00 which makes a total of $1095.00. Do your self a favor and go this route. You will save your self tons of trouble. On 5/6/05, Joerg Wleklik [EMAIL PROTECTED] wrote: Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? Ciao Joerg -- _ Don't PANIC ___ 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] FXO ATA?
On Fri, May 06, 2005 at 04:24:32AM -0500, Eric Wieling aka ManxPower said: Jon Gabrielson wrote: On Thursday 05 May 2005 05:28 pm, Joseph wrote: It has 1-FXS and one 1-Life Line (it is pass through type) I've seen the pass-through term used alot and I'm not quite for sure what that means. What is the I just checked my dictionary and it defines pass-thru as meaning totally useless for most people. Pass-thru and lifeline seem to be different terms for the same thing. i.e. The FXO port is connected to the FXS port in the event of a power outage, but other than that it is not useful. Not quite. A pure life-line FXO that is not voip accessable is useless to *. Usually this means that an extension on the FXS port uses the PSTN on the FXO during powerfailure / 911 calls. Some ATA's have this kind of port. The SPA-3000's FXO CAN pass through in life-line mode automatically for power faliures and if it is configured to do so via the dial-plan. The dial plan on the 3000 allows lots of flexibility here. From a VoIP standpoint, the FXS and FXO ports can be configured to be totally separate devices, where if you want to make a call via the PSTN, the call is looped through *. Pass through can also be used in terms of how the FXO interfaces with *. The standard config of the SPA-3000 for example answers the call and THEN forwards to * - acting more like a full gateway than a dumb FXO. It can also be configured (kludged) to pass through call info to * BEFORE the call is answered (which is frequently more desirable in many situations.) Hope this helps. ___ 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] FXO ATA?
On Fri, 2005-05-06 at 04:24 -0500, Eric Wieling aka ManxPower wrote: I just checked my dictionary and it defines pass-thru as meaning totally useless for most people. Pass-thru and lifeline seem to be different terms for the same thing. i.e. The FXO port is connected to the FXS port in the event of a power outage, but other than that it is not useful. Well, I didn't check any dictionary :-) but I've checked what other people saying/posting about the AG-168 on their board at: http://en.atcom.com.cn/bbs/ AG-168V Series From their conversation/questions it seems to me that that unit can receive and make call to/from PSTN line. -- #Joseph ___ 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] Operator Monitoring...flash operator panel?
Anyone using FOP or another solution to watch 200 + lines at once? I don't think I can cram the buttons down small enough in FOP to do it! Currently they are used to the Avaya Definity panel with the whole decade setup. I may need to deploy over 200 phones and have them all visible to the operator. We were thinking of adding a second monitor for our receptionist with a panel of extensions. Thanks for any direction...planning planning planning. -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI on ptp with variable length digits in phonenumber: SOLUTION for EICON
Hello! I finally found a working solution. calling divactrl with the parameter -n [0..20] gives the DID-length means, if you wanna have 123-XXX in digit-wise mode, then call divactrl load -c 1 -n 3 -f ETSI and the card will wait for n digits. regards, Sebastian -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von bladerunner Gesendet: Freitag, 6. Mai 2005 14:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] CAPI on ptp with variable length digits in phonenumber hi again, just ignore my mentioning of the sirrix-cards, just realised you have a PRI, i overread it and thought you had a BRI. so i think your last hope is a zaptel-card. regards, Am Freitag, 6. Mai 2005 13:04 schrieb Sebastian Buntin: Hi! we have a german PtP PRI connection here. our old telephone system was programmed to accept digits by variable length. so our MSN, assigned my telco is, lets say: 123 and we can use first digit from 0-4. and every further digit like we want. means: 123-1 123-2345 123-44 till 123-499 but not 123-5... I'm using an Eicon diva server PRI 23M with chan_capi. my problem is dialing IN. block mode works perfect. (when the whole number is sent as a block) I just add exten = 123114,1,Dial(SIP/blahblah) works. but if someone dials digit-wise, lets say 123114, asterisk starts scanning the dialplan after 1231. doesnt find an extension and exits. even using exten = s,1,DigitTimeout,4 as first line in the dialin-extension won't help. so, I need to find a way that asterisk collects the digits until it has a matching one. p.ex. wait scanning the extensions till the caller typed 123114. I can live with fixed length extensions. means, always wait for 3 digits. thanks for help.. Sebastian ___ 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] Mitel SX200 integration
Yes the most efficient way would be to put a t1 between your asterisk system and the sx200 and set up DID 2Way Trunk lines between the two on those channels. In asterisk you would set up the dial plan to send 1XX, 2XX, 3XX to the DID trunks and redial the EXTENSION as the DID number on trunks assigned as DID's to the SX200. On the SX200 use the ARS system to use 7 as an access number for the two way trunk group to the astrerisk system, and have it redial out the number, and asterisk would just process the incoming digits as normal. Other than that you could set up analog DID lines for calls into the SX200 from the asterisk system, and just regular analog lines to do the same thing. -- From: Chris Mason (Lists)[SMTP:[EMAIL PROTECTED] Reply To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, May 06, 2005 8:53 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject:[Asterisk-Users] Mitel SX200 integration I am changing one a resort over from a SX200 with digital phones to Asterisk with Polycom phones. The office can be changed over immediately but the guest's villas can't until we do some rewiring, and that will be villa by villa as they are available. How can I merge Asterisk into the existing installation? For example, I would like to have extensions 7XX be on Asterisk while 1XX, 2XX, and 3XX are on the mitel. Any ideas? Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 ___ 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] PRI timing problems: Fax Voice
On May 5, 2005 09:52 pm, Greg Boehnlein wrote: This is very confusing. Most clocking language in the industry refers to either Internal or Recovered clocking. Basically, the span can either use it's own clock, or attempt to pull it from the line. clock = 0 means internal clock = 1 means recovered from this span clock = 2 means recovered from this span if the span with clock=1 is down etc. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 x TDM400P in one PC ??
On Fri, May 06, 2005 at 10:25:24PM +1000, Joerg Wleklik said: On Fri, 6 May 2005 22:10, Eric Wieling aka ManxPower wrote: Joerg Wleklik wrote: Does anybody have experiences with plugging 3 TDM400P cards in one PC?? If you need 12 ports then you should use a T-1 card and a Channel Bank. That would be easy, but.. I have 8 analogue lines incoming right now and changing the phone number is not an option (costs for advertising). This lines go right now into an analogue PBX. A new building will get IP-Phones connected to an asterisk box. The idea is to take the incoming calls in the asterisk, route to the new building via IP and serve the old PBX with 4 analogue lines. That's what the channel bank does. The other reason you want to use a T1 card over multiple TDM400 cards is that the 3 TDM cards will generate 3 times the number of interrups, and likely have interrupt sharing problems. Good channel banks also are going to be much less prone to have echo problems. You also will have room for expansion. BTW, if you went with a new T1/PRI to the telco, you can probably have your old numbers forwareded / migrated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: CDR for PSTN
On 5/6/05, Kamran Ahmad [EMAIL PROTECTED] wrote: hello Thanks for replying. i know duration and billsec. but i am getting wrong billsec. for example in one call billsecduration 48 55 It doesn't make sense what you are saying. If it only rings 10 seconds, and you were on the phone for only 35 seconds, then how is the total duration 55? In any case what type of device are you using to connect to the cell phone? ZAP/FXO will start billing as soon as it is answered, which means as soon as the dialing starts (before the ring). and actually in this call phone rings 10 seconds. and accual duration on my cell phone is 35 Hi, Look at ___ 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] Operator Monitoring...flash operator panel?
Anyone using FOP or another solution to watch 200 + lines at once? I don't think I can cram the buttons down small enough in FOP to do it! I have a similar problem ... managed to size down the buttons to fit 120 extensions but that's it ... Ivan ___ 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] snom mass deployment (probably off topic)
Nils
Firstly to the flash issue, if you are on a DHCP server this seems to
work. If you are not then my particular handset does indeed lose its
config - or rather it invokes the start up wizard which does not
have the last settings to confirm, so to all intents it apears to have
lost its settings (although I now freely admit if you give it an IP
address and go into the web interface, all is not lost) - therefor my
appologies on that, however I would prefer to see the last settings in
the setup wizard.
With regards to my second issue, regarding creative dns records
locally. You state in your white paper, that you need to invoke
options 66 and 67 for the snom device to check an update server other
than snom's its self.
Prehaps again it is my error, however looking at ethereal packet
traces, the phone is supplied the dhcp options, but still attemps to
goto snom.com - would you be kind enough to supply sample config for a
dhcpd server, so I can be sure of the setup
Other than those two minor observations - I do indeed think that the
phone is one of the best in the market place, and I have bench tested
most user agents to date.
Thank you for your time on this matter.
David
On 5/6/05, Nils Ohlmeier [EMAIL PROTECTED] wrote:
Hello,
to prevent further rumores and wrong facts about our phones:
All our phones, and this includes the 360 as well, do store their settings on
the flash. After the settings are stored once, you can leave the phone as
long as you want without power, and it will come up with old settings
whenever you restart it. If this is not the case the person which is facing
the problem should contact the snom support to get this sorted out.
Regarding the real topic: did you already read our white-papers about
mass-deployment and setting up snom phones?
http://www.snom.com/white_papers.html
You do not have to mess around with faked DNS responses. If the phone is
getting a settings server via DHCP, it will never contact snom.com (expcept
you say so). The phone just falls back to snom.com as a default setting, in
case it cant find a setting server locally. But in case: you can even turn
off that the phone tries to load any settings from any server.
Best regards
Nils Ohlmeier
On Friday 06 May 2005 05:40, Daniel Bingham wrote:
Hi David,
First, thanks for the reply to my questions about the Snom 360. I may have
a few followup questions when I get a little more time.
As for the 360 getting the configuration directly from Snom's servers, I
find that very backwards. What if your phones have no gateway to the
internet? It sounds like they are working around not having any flash
memory, but it's a poor workaround. Your idea of using DNS to fool it into
going to your servers is a good one. I assume you'll just put in a mapping
for provisioning.snom.com or just snom.com in your DNS server to the IP of
your web server.
If I understand correctly, you will be able to create an
/snom360/snom360.php script on your web server, which you would then like
to redirect to the static html files in the /snom directory. Assuming the
Snom supports redirects, the PHP code is as simple as:
?php header(Location: /snom/snom360-$_GET[mac].html) ?
If the phone doesn't support redirects, it gets a little complex, in that
the script will need to open the file from the filesystem and return it
directly.
If I misunderstood or I didn't make sense, I'll be happy to try again.
Thanks,
Daniel Bingham
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] ] On Behalf Of David John
Walsh Sent: Thursday, May 05, 2005 8:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] snom mass deployment (probably off topic)
Hello
Although not stictly a asterisk issue, any help would be apreciated.
Firstly a few notes on the snom 360, which I have had on a test bed for the
last week. Its a great phone, with a good user interface, both physically
and its web based one.
At its lastest firmware it does have a few quirks, with regards to the way
it handles usernames and passwords on the physical interface. These have
been passed back, and hopefully will be addressed.
Its worst feature as I see it is twofold, with regards to its power fail
features. If it loses power for more than a few minuites it loses its
settings - not the best thing in a world where routers and firewalls can be
given power back days later and be fine.
It has an interesting configuration mode, it tries to contact snom, who
then (if told about it) goes to their national distrubtor who then either
has your config or passes it on again
The settings file is well documented, and you can pull them direct from
phone in a ready to go way.
---
I now have my configs in the file name format of snom360-{mac}.htm (where
{mac} is the MAC address
Re: [Asterisk-Users] CAPI on ptp with variable length digits in phonenumber: SOLUTION for EICON
but with this solution you will not be able to receive calls with less than two DID-digits (like call to 123-0 where 123 is head number). it will wait for exactly two digits before answering (at least in the last version of the firmware and chan_capi i tried). regards, Am Freitag, 6. Mai 2005 15:17 schrieb Sebastian Buntin: Hello! I finally found a working solution. calling divactrl with the parameter -n [0..20] gives the DID-length means, if you wanna have 123-XXX in digit-wise mode, then call divactrl load -c 1 -n 3 -f ETSI and the card will wait for n digits. regards, Sebastian -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von bladerunner Gesendet: Freitag, 6. Mai 2005 14:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] CAPI on ptp with variable length digits in phonenumber hi again, just ignore my mentioning of the sirrix-cards, just realised you have a PRI, i overread it and thought you had a BRI. so i think your last hope is a zaptel-card. regards, Am Freitag, 6. Mai 2005 13:04 schrieb Sebastian Buntin: Hi! we have a german PtP PRI connection here. our old telephone system was programmed to accept digits by variable length. so our MSN, assigned my telco is, lets say: 123 and we can use first digit from 0-4. and every further digit like we want. means: 123-1 123-2345 123-44 till 123-499 but not 123-5... I'm using an Eicon diva server PRI 23M with chan_capi. my problem is dialing IN. block mode works perfect. (when the whole number is sent as a block) I just add exten = 123114,1,Dial(SIP/blahblah) works. but if someone dials digit-wise, lets say 123114, asterisk starts scanning the dialplan after 1231. doesnt find an extension and exits. even using exten = s,1,DigitTimeout,4 as first line in the dialin-extension won't help. so, I need to find a way that asterisk collects the digits until it has a matching one. p.ex. wait scanning the extensions till the caller typed 123114. I can live with fixed length extensions. means, always wait for 3 digits. thanks for help.. Sebastian ___ 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 pgppedy6biZ82.pgp Description: PGP signature ___ 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] Polycom 600 rollover
I have the Polycom 500 and 600 phones. Rather than put an entry in for each line appearance, I would like to use the feature that shares one extension for the lines, so that I will get the call on the enxt available button. How do I configure that? Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] can't create Zap channel
- /etc/zaptel.conf
span=1,1,0,esf,b8zs
span=2,1,0,esf,b8zs
span=3,1,0,esf,b8zs
span=4,1,0,esf,b8zs
bchan=1-23,25-47,49-71,73-95
dchan=24,48,72,96
loadzone = us
defaultzone=us
- /etc/asterisk/zapata.conf
[trunkgroups]
[channels]
language=en
context=all-incomming
switchtype=national
pridialplan=national
signalling=pri_cpe
rxwink=300
usecallerid=yes
cidsignalling=bell
cidstart=ring
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=0.0
txgain=0.0
immediate=no
group = 1
channel = 1-23
group = 2
channel = 25-47
group = 3
channel = 49-71
group = 4
channel = 73-95
group = 5
channel = 1-23,25-47
group = 6
channel = 1-23,25-47,49-71
group = 7
channel = 1-23,25-47,49-71,73-95
- the construction of the extension you are using
[all-incomming]
exten = _18X,1,Dial(Zap/R1d/${EXTEN},60) ;
Toll Free - Terminate PRI Span 1
exten = _[287]X,1,Dial(Zap/R4d/${EXTEN},30);
Local - Terminate PRI Span 4
exten = _XX,1,AGI(lcr.php) ;
Long Distance - Run lcr.php for LCR
exten = _011.,1,AGI(lcr.php) ;
International - Run lcr.php for LCR
exten = _700999,1,SetCallerID(Matthew 2814494000|a) ;
Special Cases - Do whatever
exten = _700999,2,Dial(Zap/R2d/15124512424,30)
exten = t,1,Congestion()
exten = i,1,Congestion()
-Matthew
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Re: [Asterisk-Users] Attended Transfer using wrong Context
David Phelan wrote: Can you post the context for cytel-outgoing... From what it sounds like..asterisk is picking the # as a blind transfer then 9 which means you are trying to transfer to an outside number an ddepending on your dial plan, that may not work. I do realise that you are trying to use attended transfer so maybe change the attended transfer sequence so that it doesn't use 9. Dave That is wierd. I changed it from #9 to #6 and now I can dial properly. Shouldn't asterisk ignore the 9 as it is part of the # sequence? -Matthew ___ 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] RE:how do I register my Asterisk with oh323 on gatekeeper?
Hi My oh323.conf is: listenaddress=0.0.0.0 listenport=1720 tcpStart=1 tcpend=2 udpStart=1 udpEnd=2 faststart=no h245Tunnelling=no h245inSetup=no gatekeeper=DISCOVER accountCode=H323 context= (ex.default) alias= (ex.gw) Codec= (ex.G711A) Frame= (ex.20) Ale ___ 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] Operator Monitoring...flash operator panel?
Ivan Meic (Vox Mundi) wrote: Anyone using FOP or another solution to watch 200 + lines at once? I don't think I can cram the buttons down small enough in FOP to do it! I have a similar problem ... managed to size down the buttons to fit 120 extensions but that's it ... Ivan Use FOP's ability to make Panel_contexts and put 100 on 1 page and 100 on another. Problem is, you can't transfer between the two pages. But you could put all your incomming lines on both pages.. -Matthew ___ 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] CNAM lookup: new method for Caller ID Name delivery
John Todd wrote:
[cross-posted to -biz and -users since it could fall into either
category]
Interesting new product that has been introduced that I think some
would be interested in here (at least, those users in the United
States and perhaps Canada): CNAM delivery via IP lookup.
The problem: inbound calls on many PRI connections, and also over many
VoIP providers, do not include caller name. This means that all you
see is the caller ID number, but no name. Most PSTN lines these days
(if they are enabled with Caller ID) will also include a caller ID
name. So, you'd think that a well-configured Asterisk server should
somehow be able to deliver the same data, right?
A company called Accudata has come up with an IP-based CNAM lookup
tool. It's an HTTPS delivery method, with what I assume is XML as the
specification language. The nice part is that it really doesn't
matter what the backend looks like - Accudata has built app_getcnam
that automatically takes the 10 digit NANP number and spits back a 15
character caller name from within the Asterisk dialplan. You get the
caller ID from an inbound call (IP or PRI or any channel type as long
as it has an e.164 number associated with it) and then hand off the
${CALLERIDNUM} to this application, and get back a string with the
name. I don't have exact details on the system (see disadvantages
below) but it seems to be an interesting product.
Pricing:
At the low volume end of the scale (probably under 2000 queries
per month, but I didn't ask), the price is $0.0156 per lookup, which
is reasonable enough. I'm sure better price breaks come with volume.
Upsides:
1) They have direct Asterisk integration, using app_getcname.c as a
data method.
2) They at least are willing to talk to smaller customers who aren't
pushing millions of calls a month.
3) It's all IP - no unwarranted complexity of SS7 or other signalling.
Downsides:
1) They want you to sign an NDA before they'll discuss the methods
with you. I was not willing to sign an NDA to have an XML schema
example transmitted to me, so that was a non-starter. This really
angers me, actually - does anyone actually have a clue how many
lawyers need to get involved in an NDA, and what is it exactly that
the NDA is trying to do? NDAs are used in the USA for the most
frivolous and inane reasons. As if your competition didn't know what
you were doing? Please, let's be realistic here.
2) They have a $100 monthly minimum charge. If you only have a
billing volume of under $100, then you'll pay $100. So, if you have
under 6400 queries per month, you're paying for the honor of being
billed. This isn't that big a deal if you're an ITSP, but makes this
almost impossible for a smaller user to afford. (good opportunity for
a small reseller, especially if you are smart with caching.) I can't
say I disagree with them on this model to start, but I spent some time
doing the math for small-time usage, and at a $2 minimum and 50
included queries a month (and $.02 afterwards) this would make a very
nice market for a few thousand iPBX systems. Payment via Credit Card
or Paypal would be perfect; set it up once, forget about it. However,
that's not the model they chose, since they're not shooting for the
lower end of the market.
3) There may be hidden problems with the application; I haven't run
it, so I can't vouch for it.
Other notes:
The clever integrator of this application will save themselves some
lookup $ by caching the responses from the database into their own
database, along with a datestamp. Perhaps if an entry is 90 days
old, the system will re-lookup the entry in the Accudata database but
otherwise will present the memorized answer. (Hint: the caller ID's
of your inbound call pool is probably 80% redundant)
Contact information:
http://www.accudatatech.com
Tracy Glick [EMAIL PROTECTED] [sales contact]
Kevin Nguyen [EMAIL PROTECTED] [tech contact]
If anyone else has heard of an easy-to-use method for obtaining this
data via free or commercial methods, please follow-up to this post for
the archives. I don't speak for Accudata, nor am I a user of their
services, but it seems interesting so I'll pass it along to the group.
JT
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If it isn't agiast there agement, I would happy setup a resale server
for this just as you said, and probly at the prces you listed, I will
look into this abit more later today.
Only thing I use my asterisk server for, for the most part is a few
select hpones (very low usages), but my exist customers (who have a
higher volune), or people just wanting todo CNAM look ups could befit
from this.
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Re: [Asterisk-Users] TDM users: modified zttest.c for testing
On Thu, May 05, 2005 at 09:22:24PM -0600, Rich Adamson wrote: P3 1Ghz under Tao Linux 1.0 (2.4 Kenrnel) cvs-stable w/ X101P --- Results after 66 passes --- Best: 1.024461 -- Worst: 1.024420 -- Average: 1.024447 And on our new gateway box... P4 3.0 Ghz under Tao Linux 4.0 (2.6 Kernel) cvs-stable w/ TE405P --- Results after 106 passes --- Best: 1.023967 -- Worst: 1.023953 -- Average: 1.023960 Have you tried to use spandsp at all? How about trying a fax or modem on lines on the two machines? -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SEND TEXT to an extension?
Hi, I understand SendText() sends text on the current channel. Is there a way to manipulate this feature to SendText toward another SIP device? I use Polycom IP600's. Local sendtext works fine. Would be nice to drop an instant message on another user's phone. thanks! Mark ___ 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] FXO ATA?
Folks! Let me clarify this for you all. ATCOM's ATA does not have an FXO port. The Lifeline port is not an FXO Port. It is an FXS Passthrough port. It does not have any of the FXO features that you are looking for. You cannot do a modprobe on this - nor can you pass your peer traffic to this port. Imagine this to be like an FXS Port with the Handset offhook and ready for you to dial a number to call out using your existing analog line. That's all it does. This helps you make calls using your existing analog line in case of a failure in your IP network. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Gabrielson Sent: Thursday, May 05, 2005 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXO ATA? The AG-168E has an FXO port? The only seller I can find seems to think it is just a single FXS port. http://www.iaxtalk.com/product_info.php?products_id=30 You wouldn't happen to have another link with more info would you? Thanks, Jon. On Thursday 05 May 2005 01:33 pm, Joseph wrote: Indeed SPA-3000 as a lot of features, maybe too many :-). My asterisk is controlling everything so most of these features just complicate the setup. I've one SPA-3000 and have on order AG-168VE from ATCOM. The AG-168 supports IAX2 and the FXO port is pass though type. The difference is that SPA-3000 answer the phone and rings asterisk (the phone at this moment has been answered the ringing party is incurring the charges before asterisk answered the phone), the AG-168 is ringing the asterisk directly, so I think the pass through port is a benefit in this case for asterisk users. NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk-h323
Hi
I've this plattaform
sip sjphone - - - asterisk- - -gatekeeper- - -ohphone- - -phonejack card-
- analog phone
Asterisk is registered with Gatekeeper
Ohphone is registered with Gatekeeper
Phonejack is installed successfully and get a dial tone
When i try to call phonejack with sip phone i have this message :
call with mysjphone ('alias/IP' of asterisk) completed duration 0:00
and analog phone don't ring
when i try to call mysjphone with analog phone i've this message:
Speed Dial 3231 not defined ,trying gatekeeper..
phonejack is calling host 3231
'alias' 'ip' of asterisk is busy duration 0:01
Have you suggestions?
Thanks Ale
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Re: [Asterisk-Users] can't create Zap channel
On May 6, 2005 10:06 am, Matthew Boehm wrote: - /etc/zaptel.conf span=1,1,0,esf,b8zs span=2,1,0,esf,b8zs span=3,1,0,esf,b8zs span=4,1,0,esf,b8zs You can't do that... only one span can have clocking of '1'; the others should have '2, 3, 4' or 0. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7777 (simulate incoming call) not working
I don't know if this is related, but the last two mornings I've come in, the newer AAH 1.0 computer has been locked-up. The Caps Lock and Scroll Lock lights on the keyboard are flashing (apparently in a specific pattern). The computer is a HP 7960 w/ ASUS mobo, P4, 1.3Ghz, 256MB RDram. Not being a Linux person, I don't know if this is a Linux issue and/or a hardware issue. Is there a specific log I can look at that might tell me what happened? Thanks, Doug At 03:19 PM 5/5/2005, you wrote: I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the new box, I've installed a generic ebay X100P. I don't have my livevoip or voicepulse accounts set up yet on the new box (can both boxes be registered at the same time?). I've set up one IP phone (SPA841) with the new box. I have my SBC POTS line plugged into the fxo card. I set up everything in AMP. I can make out going calls. The problem I'm having now is the digital receptionist greeting (aa_1). If I set it to automatically forward to an extension it works. But, if I have it play a message (press 200 for Joe, etc), you can't here the message at all. I can dial the extension number and * will accept and forward me to that extension. I can see on the CLI that it is suppose to be playing the message. If I dial (simulate incoming call), I get the same thing, can't hear voice but can dial extensions. I've adjusted the txgain and rxgain in zapata. This only increased echo. I have googled this list and SF, I can't find anything else to try yet, or I'm using the wrong search terms. Probably unrelated, but when I stop gracefully and then restart *, I get the following error: [app_zapbarge.so] = (Barge in on Zap channel application) == Registered application 'ZapBarge' [app_zapscan.so] = (Scan Zap channels application) == Registered application 'ZapScan' [EMAIL PROTECTED] root]# Ouch ... error while writing audio data: : Broken pipe The only thing I can do at this point is reboot the machine. I don't see any failures on the boot up. My search for this error appears to be related to mpg123. But, I never found where somebody had a solution for it. I have tried to install fax capability (install-pdf), but that doesn't work either. I get this error: There is a pretty long delay after the Server: CentOS-3 - Addons line Thanks, Doug ___ 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] 3 x TDM400P in one PC ??
Why the channel bank if he will be routing extensions to ip phones? The T-1 card should suffice if he isn't serving analog extensions. Thats $600 (t1) instead of ~ $1000.(3 x 3?? tdm400p) line cost wise 12 channels on a t1 should be cheaper than 8 pots. Walt Reed wrote: On Fri, May 06, 2005 at 10:25:24PM +1000, Joerg Wleklik said: On Fri, 6 May 2005 22:10, Eric Wieling aka ManxPower wrote: Joerg Wleklik wrote: Does anybody have experiences with plugging 3 TDM400P cards in one PC?? If you need 12 ports then you should use a T-1 card and a Channel Bank. That would be easy, but.. I have 8 analogue lines incoming right now and changing the phone number is not an option (costs for advertising). This lines go right now into an analogue PBX. A new building will get IP-Phones connected to an asterisk box. The idea is to take the incoming calls in the asterisk, route to the new building via IP and serve the old PBX with 4 analogue lines. That's what the channel bank does. The other reason you want to use a T1 card over multiple TDM400 cards is that the 3 TDM cards will generate 3 times the number of interrups, and likely have interrupt sharing problems. Good channel banks also are going to be much less prone to have echo problems. You also will have room for expansion. BTW, if you went with a new T1/PRI to the telco, you can probably have your old numbers forwareded / migrated. ___ 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] QoS for improvements
Sometimes this all sounds so complicatedbut it needn't be. I suppose it can vary with the size of your installation. I use a m0n0wall router on a Covad DSL line. Using m0n0's traffic shaping feature I establish inbound and outbound pipes which are bandwidth restricted to just less than my mesured average DSL rate. I then break my traffic into three priority ques in each direction; highest priority, medium priority, low priority. I assign all IAX traffic in/out to the highest priority que, and map all IAX ports to the * server inside the LAN. In fact, I just used the Magic Shaper Wizard in m0n0, then added IAX specific entries to give it highest priority. The whole process took about a half hour. Just as easy as the Linksys BEFSR-81 that I had before, but more reliable and more controllable. Now to be fair, I only have about 6 phones. I use IAX2 to all my ITSPs and SIP in-house only. My DSL is 3M down / 768k up. Michael On Fri, 06 May 2005 08:37:20 +0200, Jean-Christophe Heger wrote: I've spent may hours to play with HTB QoS settings on the firewall, but with absolutely no effect. In fact, this is normal, because the time required to let a data packet going through the ADSL line will break the voice jitter. The only right way to handle this issue is to modify the MTU on the router. Without setting a TOS for voip, data where going through and voice was unusable. With a lowdelay (0x10) TOS set for voip, voice was going through, but data was blocked. With a lowdelay TOS and an HTB QoS on the router, data where going through slowly and voice was scambled. After many tests, an MTU of 700 did work quite well. I did loose 15% of bandwidth for data (twice more overheads), but data and voice may be used together. Those tests have been done on a 256 kbps up stream. There is a quite good explenation about this issue on Cisco's web site, and about they're LFI technology (link fragmentation and interleaving): http://www.cisco.com/warp/public/788/voice-qos/voip-mlppp.html#link_frag Jean-Chrsitophe Kumara Jayaweera a écrit : Hello! Everybody!!, I want to run VoIP in the same LAN (15 windows clients) which we use for surfing the Internet. 6-7 softphones in the same client's machines is 'the target'. My DSL is 128kbps, (I can go to 256kbps if required). So, I am told to install some QoS's in the LAN to improve the voice quality. Frankly, I don't know what it (QoS= Quality of Service) is. I hope you may help me giving Links to read and briefing me your ideas. Thanks to everybody in the list. So far my success and progress are your help. Thanks again Kumara ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: NAT traversal with SIP Paper
http://corp.deltathree.com/technology/nattraversalinsip.pdf Very good read for those of us who need pictures and dumbed down versions of technical info. -Matthew -- Matthew Boehm, IT DirectorCypress Telecommunications [EMAIL PROTECTED] 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. ___ 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] My Sangoma Experience in Asterisk: Followup
My Sangoma Experience in Asterisk: Followup 2005-05-06 original review can be found at: http://astguiclient.sourceforge.net/Sangoma_experience.txt It's now been a month since I finished my last round of tests with the Sangoma A104u board in Asterisk. I have had a lot of conversations with the guys at Sangoma about how my system runs so that they could set up their test systems to try to replicate my problems. They seem to take it very personally that their card did not work well for me and have been working on all sorts of fixes and improvements to the wanpipe drivers and the firmware for the boards in the last several weeks. Here's a summary of what they've done: - D4/AMI circuits now work fully with Sangoma cards - RBS EM Wink start circuits now work under full load - Created a new Hardware HDLC PRI D-channel implementation that runs more efficiently - For the A104 cards they have created a TDMV driver to streamline the voice data path to offer better scalability The first two made me very happy because I was now able to fully run the Sangoma card under full production load with D4/AMI and Wink start T1s. To upgrade to the new software I had to first upgrade the firmware on the A104 card. Sangoma includes the firmware as well as a loader with the wanpipe drivers. Upgrading the board firmware was actually very easy, just start the firmware loader script(wan_aftup), select the board, pick the firmware file to update to and it's done in a couple minutes, and you don't even have to reboot. As for the driver software, Sangoma had changed a few things in the last month(including a new installation README for asterisk). First, for the optimized HDLC to work you now have to re-compile zaptel after you finish your wanpipe installation(not a big deal). Second, you now have to configure your wanpipe spans differently depending on whether it is a PRI circuit or a RBS circuit(by selecting the DCHAN in the wancfg utility per span). Third, now the wancfg utility can setup the startup order of the spans, taking away an extra step that you previously had to do. After finishing the installation and getting Asterisk back up and running I tried both a D4/AMI circuit and an EM Wink start circuit and found they both worked well. Then I put some test traffic through all four T1 ports and again everything went well. Now it was time to put the server into production and unlike last time, it ran without any problems all day. The performance results ended up as I expected: For our production environment a 30-50% reduced system load leading to higher capacity on the server than was possible with the Digium TE405P board. Another piece of news that came out in the last month was that Digium will be shipping the TE406p in May 2005(a 405 with an echo-canceller daughtercard that will retail for $2,195). I would very much like to get my hands on one of these to test it's performance against the Sangoma A104 to see what kind of impact off-loading the echo-canceller and DTMF detection really has for one of these cards. In the last month I have learned a little more about Sangoma and Asterisk. Sangoma has given money and has been donating code to the Asterisk community for some time now. They also have given money to other fledgling telephony projects such as Yate (http://yate.null.ro/pmwiki/). In light of the improvements made in the last month and the reliability I've seen in the system I've been running for the last month with a Sangoma card in it, I would now recommend Sangoma cards for just about anyone except for Linux/Asterisk novices. The configuration might be a bit confusing for a newbie especially when compared to the ease of popping in a Digium single T1 board and going right to the Asterisk install. Other than that, Sangoma boards are now to the point where they can be used by just about any T1/E1 user in all types of environments. This has been another step forward by Sangoma in the battle of the Asterisk telco boards, but Digium has also been busy on another front recently, developing their new channelized DS3 card. It will be interesting to see how this card and the new TE406p perform as well as to see what new cards come out from Sangoma in the next year. And as always I say, competition is a good thing. MATT--- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] anyone experiencing half connections
I am running head from apr-6-2005. I am using Broadvoice (may or may not be an issue) On some calls I will have like 10 or 15 seconds when I cannot hear the other party, they can hear me, then it comes back in again and is fine. Just wondering if I dont have something setup correctly (and what that might be) or if others are having the same issue. Jerry ___ 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] ADTRAN Total Access 624 Work???
Anyone using this device? I would love to see some Zapata/zaptel/extensions.conf configs that you have used to get it to work with asterisk. Also Any input for setting up the 624 would be great. Thanks ~ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AAH lockup
Sounds like a kernel lock up. After you've rebooted, check out /var/log/messages to see what happened. - Dan [EMAIL PROTECTED] wrote: I don't know if this is related, but the last two mornings I've come in, the newer AAH 1.0 computer has been locked-up. The Caps Lock and Scroll Lock lights on the keyboard are flashing (apparently in a specific pattern). The computer is a HP 7960 w/ ASUS mobo, P4, 1.3Ghz, 256MB RDram. Not being a Linux person, I don't know if this is a Linux issue and/or a hardware issue. Is there a specific log I can look at that might tell me what happened? Thanks, Doug ___ 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 NOTIFY retries exceeded.
Hello, I get warnings in my asterisk log: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call. I've used sip debugging to figure out the cause. It's my D-link DVG-1120S that don't understand message-summary events that asterisk sends out for MWI indication to the client. Is there any way to disable this in asterisk for this particular client? Tanks in advance, Magnus ___ 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] 3 x TDM400P in one PC ??
Andres Paglayan wrote: line cost wise 12 channels on a t1 should be cheaper than 8 pots. Not from any provider I've ever dealt with. ___ 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] Re: Polycom 600 rollover
Hi Chris - I have the Polycom 500 and 600 phones. Rather than put an entry in for each line appearance, I would like to use the feature that shares one extension for the lines, so that I will get the call on the enxt available button. How do I configure that? You can register each line on a Polycom to the same username/password in sip.conf. If you do this, though, you will not be able to bypass Polycom's call waiting feature. This means that if you're on Line1 and a call comes in, it won't ring to Line2, it will ring to the call-waiting of Line1. To answer the second call, you have to press the down arrow on the phone, and then press the answer softkey. If you have two active calls on Line1, the third call will ring to Line2, etc. This setup works fine, if you can train your users on it. My experience is that it does not make sense to most users, and that it is too slow for our receptionists. If you want the phone to behave like - Call1 goes to Line1, Call2 goes to Line2, Call3 goes to Line3, etc, you'll have to do multiple registrations in sip.conf, and then use SetGroup and CheckGroup in the dialplan to limit the number of calls that can go to a single registration. If you need more info, search the archives of this list. If that doesn't make sense, let me know. - Noah ___ 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] Re: Opinions on Cisco 7960G, Polycom IP-600, and Snom 360
Hi Dan - I would also like to figure out how to make the phone *ring* when you're already on another line, but haven't had a chance to seriously explore it yet. Is this still a problem in the latest firmware? This could sink my hopes of going with a Polycom phone if there isn't a way to have them give an audible alert that another line is ringing while you're already on the phone. The phone will not ring for a second call when you're already on a line, but with the right settings it will give you a call waiting beep in the earpiece. So there is, at least, some audible indication of another call. It only beeps once, though. On the IP500's the LED on the top of the phone will flash, and on the IP600's the LED for the line of the new call will flash. The Wiki says the IP-500 requires an additional chip to support power over ethernet. Is this true of the IP-600 as well? The IP600 supports PoE natively. Use any ethernet cable, and you can get power from a PoE switch. The IP500 and IP300 need a proprietary cable that has the PoE chip embedded in it. They run for about $35 apiece. - Noah ___ 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] Digium MOH
I've got 28 mp3 files from Russian artists performing classical music that we use for our Music-On-Hold. They were grabbed from mp3.com before its demise and were listed as public domain by the artists (I don't think Beethoven, et. al. care any more either). Anyway, the 28 files are a total of 187MB and I'll gladly post them somewhere. I don't have the bandwidth to support downloads from my site but if someone is willing to host I'm willing to post. Here's what I've got: 05/14/2001 04:51 PM 5,446,844 beethoven__violin_sonata_n.mp3 05/14/2001 04:54 PM 6,322,051 beethoven_violin_sonata_n7.mp3 05/14/2001 04:57 PM 7,482,723 brahms__violin_sonata__2__.mp3 05/11/2001 03:56 PM 5,398,070 brahms__violin_sonata__2_1.mp3 05/14/2001 05:01 PM 4,379,376 brahms__violin_sonata__2_2.mp3 05/11/2001 04:18 PM 3,701,003 brahms_piano_quartet_n3_24.mp3 05/14/2001 05:06 PM 8,956,865 brahms_piano_quartet_n3_34.mp3 05/14/2001 05:09 PM 6,880,862 brahms_piano_quartet_n3_44.mp3 05/15/2001 09:32 AM 5,071,099 brahms_scherzo.mp3 05/15/2001 09:39 AM 7,513,234 brahms_violin_sonata_n3_14.mp3 05/15/2001 09:48 AM 3,808,444 brahms_violin_sonata_n3_24.mp3 05/15/2001 10:03 AM 4,849,162 brahms_violin_sonata_n3_44.mp3 05/11/2001 04:00 PM11,003,321 mahler__piano_quartet.mp3 05/15/2001 11:22 AM 6,419,435 mendelssohn__piano_trio_i1.mp3 05/15/2001 11:52 AM 3,247,961 mendelssohn__piano_trio_i2.mp3 05/15/2001 04:52 PM 7,719,706 mendelssohn__piano_trio_i3.mp3 05/15/2001 12:36 PM 8,485,825 mendelssohn__piano_trio_in.mp3 05/15/2001 01:22 PM 9,902,289 mendelssohn_violinpiano_1.mp3 05/15/2001 03:59 PM17,886,981 mendelssohn_violinpiano_2.mp3 05/15/2001 05:09 PM 8,218,331 mendelssohn_violinpiano_c.mp3 05/16/2001 10:02 AM 6,033,659 mozart__violin_sonata_e_m1.mp3 05/16/2001 10:13 AM 6,940,212 mozart__violin_sonata_e_mi.mp3 05/16/2001 11:37 AM 4,136,542 mozart__violin_sonata_g_m1.mp3 05/16/2001 04:02 PM 7,815,837 mozart__violin_sonata_g_ma.mp3 05/16/2001 04:25 PM 6,401,881 piazzolla__verano_porteno.mp3 05/16/2001 04:31 PM 2,921,117 schubert_violin_sonatina_3.mp3 05/17/2001 02:22 PM 5,909,525 schubert_violin_sonatina_4.mp3 05/17/2001 04:59 PM 4,982,073 schubert_violin_sonatina_o.mp3 28 File(s)187,834,428 bytes Greg -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of programming dept Sent: Thursday, May 05, 2005 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Digium MOH on 5/3/05 21:46, Matt Riddell at [EMAIL PROTECTED] wrote: Would there be any chance of creating a GPL exception for them if we donated them? I have rather a few songs, mostly in the trance/psytrance genre but also dub and DnB. Ideas? What we do is find local artists from our nearby radio stations. We get the artists permission to put their music on our MOH. They add trailers to say where their music can be purchased, web address and/or the sort...10 second trailer after each song is a good trade. The artists love it and they become customers of ours too. ...we don't violate copyrights and we help out the industry at the same time. -- Stu ___ 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] FXO ATA?
On Thursday 05 May 2005 10:27 pm, Tim Connolly wrote: Pass through has the same functionality as a modem with a line and a phone connection. Line is where you plug in the dialtone, the dial passes through the phone connection unless the card picks up (like a modem does). I have a X100P clone that is setup as a passthrough. I've never seen a pass through on a FXS, but then I've only messed with ATA-186's recently. That is not correct or at best not completely correct. That is what I would have believed it to be, but some passthrus (like the handytones) allow you to dial *00 or some other combination to dial out of the PSTN directly. I guess it is possible that the *00 turns off the FXS to allow the straight passthru, but even if that is so, the X100p doesn't have an equivalent functionality. ___ 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] RE: Connecting 2 * Together-Pulling hair out
http://www.voip-info.org/tiki-index.php?page=Asterisk+Connect+2+servers I have posted a doc on this to the wiki. Fist time poster. I couldn't figure out how to escape square brackets and tables looked like I would be there all day. Be nice :-) dbc. David Cook ___ 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] 3 x TDM400P in one PC ??
Reformatted top-posting... On Fri, May 06, 2005 at 08:30:52AM -0600, Andres Paglayan said: Walt Reed wrote: On Fri, May 06, 2005 at 10:25:24PM +1000, Joerg Wleklik said: On Fri, 6 May 2005 22:10, Eric Wieling aka ManxPower wrote: Joerg Wleklik wrote: Does anybody have experiences with plugging 3 TDM400P cards in one PC?? If you need 12 ports then you should use a T-1 card and a Channel Bank. That would be easy, but.. I have 8 analogue lines incoming right now and changing the phone number is not an option (costs for advertising). This lines go right now into an analogue PBX. A new building will get IP-Phones connected to an asterisk box. The idea is to take the incoming calls in the asterisk, route to the new building via IP and serve the old PBX with 4 analogue lines. That's what the channel bank does. The other reason you want to use a T1 card over multiple TDM400 cards is that the 3 TDM cards will generate 3 times the number of interrups, and likely have interrupt sharing problems. Good channel banks also are going to be much less prone to have echo problems. You also will have room for expansion. BTW, if you went with a new T1/PRI to the telco, you can probably have your old numbers forwareded / migrated. Why the channel bank if he will be routing extensions to ip phones? The T-1 card should suffice if he isn't serving analog extensions. Thats $600 (t1) instead of ~ $1000.(3 x 3?? tdm400p) line cost wise 12 channels on a t1 should be cheaper than 8 pots. Because he STILL needs analog lines for the legacy PBX (read above), and he does NOT have a T1 now - just POTS PSTN lines. That's 12 analog ports needed. Now if he were to convert to a T1 (or E1 / PRI) for his PSTN connection (as I mentioned,) then he could get a couple cheap Sipura's for the legacy PBX and forget the channel bank. Depends on what he can get for T1/E1/PRI pricing (8 channels is usually not very cost effective.) ___ 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] Newbie *@home + Xten.
I have d/l the iso ([EMAIL PROTECTED] 0.9) , built the * box and followed the directions in the * handbook and http://www.geekgazette.com/index.php?option=com_contenttask=viewid=2Itemid=26. I created extension 200 and verified that * was running fine. Loaded Xten lite, setup the proxy for local ip (10.0.0.201) per the handbook. After turning off the Norton Firewall protection, I am able to start Xten and it says Logged in. I wanted to start with the easiest thing, so I just wanted to SIP to my local server (same net, no firewall/router issues). I cant even get *43 or 1234 to work. Here is the Xten log for the 1234 call: SEND TIME: 15532804 SEND 10.0.0.201:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.250:5060;rport;branch=z9hG4bK96200694E8A848F9808AD84E829CA819 From: rdelite sip:[EMAIL PROTECTED];tag=3097086592 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 6629 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 242 v=0 o=200 15532664 15532804 IN IP4 10.0.0.250 s=X-Lite c=IN IP4 10.0.0.250 t=0 0 m=audio 8000 RTP/AVP 3 97 110 101 a=rtpmap:3 gsm/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 15532855 RECEIVE 10.0.0.201:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.250:5060;branch=z9hG4bK96200694E8A848F9808AD84E829CA819 From: rdelite sip:[EMAIL PROTECTED];tag=3097086592 To: sip:[EMAIL PROTECTED];tag=as7a1af4a2 Call-ID: [EMAIL PROTECTED] CSeq: 6629 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=68e1ffea Content-Length: 0 SEND TIME: 15532855 SEND 10.0.0.201:5060 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.250:5060;rport;branch=z9hG4bK96200694E8A848F9808AD84E829CA819 From: rdelite sip:[EMAIL PROTECTED];tag=3097086592 To: sip:[EMAIL PROTECTED];tag=as7a1af4a2 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 6629 ACK Max-Forwards: 70 Content-Length: 0 SEND TIME: 15532865 SEND 10.0.0.201:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.250:5060;rport;branch=z9hG4bKECBD700714F4480D8E32F24791BB75C0 From: rdelite sip:[EMAIL PROTECTED];tag=3097086592 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 6630 INVITE Proxy-Authorization: Digest username=200,realm=asterisk,nonce=68e1ffea,response=e4d2c26ee42a232380884715ea76ac71,uri=sip:[EMAIL PROTECTED] Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 242 v=0 o=200 15532664 15532804 IN IP4 10.0.0.250 s=X-Lite c=IN IP4 10.0.0.250 t=0 0 m=audio 8000 RTP/AVP 3 97 110 101 a=rtpmap:3 gsm/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 15532865 RECEIVE 10.0.0.201:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.0.0.250:5060;branch=z9hG4bKECBD700714F4480D8E32F24791BB75C0 From: rdelite sip:[EMAIL PROTECTED];tag=3097086592 To: sip:[EMAIL PROTECTED];tag=as7a1af4a2 Call-ID: [EMAIL PROTECTED] CSeq: 6630 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Larry Richardson ___ 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] 3 x TDM400P in one PC ??
We use a Cybermesa (local Santa Fe company) T1 $80 for the transport, $240 for 12 lines $12 for 100 dids $190 tax total = 522 8 comercial lines * ~40 = 200 So you are right. Eric Wieling aka ManxPower wrote: Andres Paglayan wrote: line cost wise 12 channels on a t1 should be cheaper than 8 pots. Not from any provider I've ever dealt with. ___ 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] HINT
Guys, what does hint do in a dialplan and how do you use it? ___ 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] IPS version 0.114
IPSwitchBoard Version 0.114 - 6. may 2005. * IPS will now delete recordings on the Asterisk server after it's transferred to your PC. * You can now configure call Rates as X of a Currency Unit. If your Currency is Dollars then the smallest Call Rate is Dollar/X. X would typically be 1000. * IPS will now save the last position and size and set that the next time your start IPS * bug fixes Download: http://ipswitchboard.thorben.dk ___ 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] To receive faxes on a dedicated extention and to forward them to a dedicated e-mail
Hi list, has someone a clear complete and systematic overview over what libs, compiler versions, etc. are really necessary to enable fax receiving and forwarding via e-mail with asterisk and spandsp? As a non-kernel-hacker I'm running a bit confused and find it all a bit unclear and not too systematic?! %) It is remarkable how many and various errors can be produced by different combinations of asterisk versions, compiler versions, library versions and what ever else versions. 8( By the way: Currently I'm testing on a debian distro (credativ) which basically was not too up to date and now lost more or less it's feeling for it's age. :) Best regards Manny ___ 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] HINT
I believe hint is used by phones such as snom to show DSS BLF info. - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, May 06, 2005 12:12 PM Subject: [Asterisk-Users] HINT Guys, what does hint do in a dialplan and how do you use it? ___ 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] Re: snom mass deployment - settings via DHCP
Hello, On Fri, May 06, 2005 at 02:13:02PM +0200, Nils Ohlmeier wrote: Regarding the real topic: did you already read our white-papers about mass-deployment and setting up snom phones? http://www.snom.com/white_papers.html My Snom 190 gets several options via dhcp, but defining the setting server URL does not work (in my environment) .. can be set ... automatically via DHCP (options 66 and 67) ... /etc/dhcp/dhcpd.conf 66: tftp-server-name 67: bootfile-name Is this correct? If not, do you have a working example? -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie *@home + Xten.
This is NOT tha AAH mailing list, please check out the sf fourm. On Fri, 6 May 2005, Larry Richardson wrote: I have d/l the iso ([EMAIL PROTECTED] 0.9) , built the * box and followed the directions in the * handbook and http://www.geekgazette.com/index.php?option=com_contenttask=viewid=2Itemi d=26. I created extension 200 and verified that * was running fine. Loaded Xten lite, setup the proxy for local ip (10.0.0.201) per the handbook. After turning off the Norton Firewall protection, I am able to start Xten and it says Logged in. I wanted to start with the easiest thing, so I just wanted to SIP to my local server (same net, no firewall/router issues). I can't even get *43 or 1234 to work. Here is the Xten log for the 1234 call: SEND TIME: 15532804 SEND 10.0.0.201:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.250:5060;rport;branch=z9hG4bK96200694E8A848F9808AD84E829CA819 From: rdelite sip:[EMAIL PROTECTED];tag=3097086592 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 6629 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 242 v=0 o=200 15532664 15532804 IN IP4 10.0.0.250 s=X-Lite c=IN IP4 10.0.0.250 t=0 0 m=audio 8000 RTP/AVP 3 97 110 101 a=rtpmap:3 gsm/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 15532855 RECEIVE 10.0.0.201:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.250:5060;branch=z9hG4bK96200694E8A848F9808AD84E829CA819 From: rdelite sip:[EMAIL PROTECTED];tag=3097086592 To: sip:[EMAIL PROTECTED];tag=as7a1af4a2 Call-ID: [EMAIL PROTECTED] CSeq: 6629 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=68e1ffea Content-Length: 0 SEND TIME: 15532855 SEND 10.0.0.201:5060 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.250:5060;rport;branch=z9hG4bK96200694E8A848F9808AD84E829CA819 From: rdelite sip:[EMAIL PROTECTED];tag=3097086592 To: sip:[EMAIL PROTECTED];tag=as7a1af4a2 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 6629 ACK Max-Forwards: 70 Content-Length: 0 SEND TIME: 15532865 SEND 10.0.0.201:5060 INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.0.250:5060;rport;branch=z9hG4bKECBD700714F4480D8E32F24791BB75C0 From: rdelite sip:[EMAIL PROTECTED];tag=3097086592 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 6630 INVITE Proxy-Authorization: Digest username=200,realm=asterisk,nonce=68e1ffea,response=e4d2c26ee42a23238 0884715ea76ac71,uri=sip:[EMAIL PROTECTED] Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 242 v=0 o=200 15532664 15532804 IN IP4 10.0.0.250 s=X-Lite c=IN IP4 10.0.0.250 t=0 0 m=audio 8000 RTP/AVP 3 97 110 101 a=rtpmap:3 gsm/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 15532865 RECEIVE 10.0.0.201:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.0.0.250:5060;branch=z9hG4bKECBD700714F4480D8E32F24791BB75C0 From: rdelite sip:[EMAIL PROTECTED];tag=3097086592 To: sip:[EMAIL PROTECTED];tag=as7a1af4a2 Call-ID: [EMAIL PROTECTED] CSeq: 6630 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Larry Richardson ___ 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
