SV: [asterisk-users] trouble with * and # infront of a phonenumber
This is the entry i use to
place #31# infront of a phonenumber
exten =
_00XX,2,Dial(SIP/SIPTrunk/#31#${EXTEN},55,o)
//Michael
-Ursprungligt meddelande-
Från:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] För Alyed Tzompa
Skickat: den 9 juli 2006 07:23
Till:
asterisk-users@lists.digium.com; asterisk-users@lists.digium.com
Ämne: re: [asterisk-users] trouble
with * and # infront of a phonenumber
As
of Asterisk 1.0.X a # was recognized as a pattern not as a digit,
hence in order to use it at the begining of an extension you should use
_ before it. I guess this is still valid in 1.2.X versions.
i.e: use _#31#0046011 in your extensions.conf
Alyed
Return-Path: [EMAIL PROTECTED] Sat Jul 08
12:32:46 2006
Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
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Sat, 8 Jul 2006 12:32:46 -0700
I cant make call when
using #31#0046011
The call just
disapper and noting shows on CLI
Using asterisk 1.2.9.1
I am using cubix IAX2
softphone
And I had the same
problem when I used xlite sip softphone
My voip-provider is
Rixtelecom
//Michael Nielsen
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[asterisk-users] packet8 dta 310 power supply question
I just picked up a second hand packet8 dta310 model from a local thrift shop. It has no power supply with it. It says it needs 12v and 0.6a. I have a spare power supplies that might fit this description. What polarity does this unit require? positive tip? negative tip? If you don't understand, reply with a closeup picture so I can read all the text and symbols off the power adapter. Thanks! Scott ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suggesstion Required
Hi all, I want to setup asterisk box to do the following jobs. 1- 100 cuncurent calls 2- 1000 User Registration 3- MySQL Realtim 4- PerlAGI Here is my question could u please reply it: 1- No RTP only singnaling, Is it possible? Ans: 2- How much RAM? Ans: 3- How much bandhwidth per month with G729 Ans: 4- Proccessor? Ans: I will be appriciate for your kind of replies. Abdul, __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggesstion Required
You might want to look around the wiki (www.voip-info.org) as most of your questions have answers there. I want to setup asterisk box to do the following jobs. 1- 100 cuncurent calls 2- 1000 User Registration 3- MySQL Realtim 4- PerlAGI Here is my question could u please reply it: 1- No RTP only singnaling, Is it possible? Ans: Yes, but probably not 100% likely. If any of the 1000 users have their equipment behind firewall/nat boxes, you're likely to end up handling rtp / transcoding. Also, depending on the exact equipment used by the 1000 users, not all devices support g729. And, any local calls terminating via analog or PRI pstn lines will likely be g711 only. 2- How much RAM? Ans: Since memory is cheap, through a couple of gig at it. 3- How much bandhwidth per month with G729 Ans: Rough answer... about 30 kbps of bandwidth per g729 call. Therefore, 100 simultaneous calls times 30,000 equals 3,000,000 bps. (That's roughly two T1's, or 30% of a 10 meg ethernet, etc.) 4- Proccessor? Ans: Any... really doesn't make that big of a difference whether its AMD or Intel. Plenty of examples on the wiki. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID
On 7/9/06, Ryder Brook [EMAIL PROTECTED] wrote: I have 2 POTs line coming into Asterisk. We have callerid feature from Verizon on one of the lines. What interface are the lines connected to? I am not able to track any CallerID coming in, in the log. I am pretty green with asterisk, and it's not clear if I have to activate for CallerID in the dialplan. The voicemail keeps saying call from an unknown caller etc. For zaptel interface, the configuration is in zapata.conf and /etc/zaptel.conf. Things like usecallerid=yes are done there. As to your greenness, everyone starts there. One suggestion is to read available books that systematically take you through which config files do what, such as http://asteriskdocs.org Another is to look at the info in /usr/src/asterisk/doc/ and the sample config files which theoretically each contain all available statements for that file. Eveyone complains that the existing docs are not accurate. Asterisk being a moving target, you can only hope to get a basic understanding reading most of these, but it's well worth the trouble. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID
I have 2 POTs line coming into Asterisk. We have callerid feature from
Verizon on one of the lines.
I am not able to track any CallerID coming in, in the log. I am pretty
green with asterisk, and it's not clear if I have to activate for
CallerID in the dialplan. The voicemail keeps saying call from an
unknown caller etc.
Eventually, i would like to pass on the callerID and name to a pager, if
the call is not picked up, at the extension, after hanging up the call.
Assuming you are trying to use an X100P or Digium TDM analog card,
you'll probably want to start with zapata.conf entries something like this:
context=inbound-home
usecallerid=yes
signalling=fxs_ks
faxdetect=no
callerid=asreceived
echocancel=yes
usecallerid=yes
hidecallerid=no
echocancelwhenbridged=yes
rxgain=7
txgain=5
channel = 1
The rxgain and txgain values will need to be tweaked to your pstn
lines, and the values will be highly dependent upon exactly how far you
are from your Verizon central office. Gains that are to high or to low
can impact how well callerid info is received.
In your extensions.conf, you might want something like this:
[inbound-home]
exten = s,1,NoOp,${CALLERID(all)}
exten = s,2,Dial(${PHONE3}${PHONE4})
If you start asterisk from the command like (eg, asterisk -rvv),
you will see the incoming callerid displayed on the command line because
of the NoOP statement shown above.
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[asterisk-users] What's the story with X10*P FXO cards?
Hi It looks like the X101P clones I bought from eBay are dogs, so I'll look into buying some FXO-SIP box instead. Hopefully, I won't have the same problems with static, or caller ID and call termination not being detected. Still, considering the number of people having similar problems with those cards, I was wondering what the problem is. Is it because the hardware, no matter what is advertised, is actually not identical from card to card so the zaptel driver doesn't work reliably unless they are among the few remaining authentic cards made by Digium before it stopped manufacturing them? Because they're actually voice softmodems, and hence, very sensitive to the computers in which they're installed (voltage on the PCI slot, sharing IRQ's, etc.)? Other reasons? Thank you VD. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07/07/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the story with X10*P FXO cards?
On Sun, 2006-07-09 at 10:22 +0200, Vincent Delporte wrote: Still, considering the number of people having similar problems with those cards, I was wondering what the problem is. Is it because the hardware, no matter what is advertised, is actually not identical from card to card so the zaptel driver doesn't work reliably unless they are among the few remaining authentic cards made by Digium before it stopped manufacturing them? digium didnt really make em. as to the reliability, some of that depends greatly on what chipset is on the card itself. There are a couple different ones that while cost about $5 couldnt be resold for $100 for good reason. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?
Michiel van Baak wrote: If you buy a model without the spare in it's name, you have the license to use them right ? To use them with a CCM or CCME, yes :-) How about secondhand phones you get from ebay ? Is my cisco smartnet account enough to run the phone legally ? It's not a spare model (at least that was not in the deal description) My understanding is, if you have any license at all, Cisco will probably not bother you. But it is most definitely not the way they intended :-) Florian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intel vs amd motherboards
Tzafrir, are you trying to tell me that I can realy do double on the intel becuase the second CPU will do it? On 7/7/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jul 06, 2006 at 03:32:04PM -0400, C F wrote: I have recently build 2 machines, one with an Intel Pentium Dual Core CPU, and one SATA HDD, and the other with a single AMD 64 bit CPU, and a RAID 1 w 2 SATA HDD. Both costed the same even though the AMD had 2 HDDs. Here are the show translations from both: Intel Dual Core machine: pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 517 -17 ulaw - 2 - 1 2 2 1 517 -17 alaw - 2 1 - 2 2 1 517 -17 g726 - 2 2 2 - 2 1 517 -17 adpcm - 2 2 2 2 - 1 517 -17 slin - 1 1 1 1 1 - 416 -16 lpc10 - 3 3 3 3 3 2 -18 -18 g729 - 4 4 4 4 4 3 7 - -19 speex - - - - - - - - - - - ilbc - 3 3 3 3 3 2 618 - - AMD 64 bit machine: pbx*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 313 -12 ulaw - 3 - 1 2 2 1 313 -12 alaw - 3 1 - 2 2 1 313 -12 g726 - 3 2 2 - 2 1 313 -12 adpcm - 3 2 2 2 - 1 313 -12 slin - 2 1 1 1 1 - 212 -11 lpc10 - 3 2 2 2 2 1 -13 -12 g729 - 4 3 3 3 3 2 4 - -13 speex - - - - - - - - - - - ilbc - 4 3 3 3 3 2 414 - - This shows that the AMD 64 bit is worth much more than just the price difference. It shows that the AMD CPU performs better than each of the Intel CPUs separately: each such translation is inherently a single task eprformed by a single CPU. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the story with X10*P FXO cards?
It looks like the X101P clones I bought from eBay are dogs, so I'll look into buying some FXO-SIP box instead. Hopefully, I won't have the same problems with static, or caller ID and call termination not being detected. Still, considering the number of people having similar problems with those cards, I was wondering what the problem is. Is it because the hardware, no matter what is advertised, is actually not identical from card to card so the zaptel driver doesn't work reliably unless they are among the few remaining authentic cards made by Digium before it stopped manufacturing them? Because they're actually voice softmodems, and hence, very sensitive to the computers in which they're installed (voltage on the PCI slot, sharing IRQ's, etc.)? Other reasons? The X100P-style cards were never manufactured by Digium; they were simply analog modem cards that happened to use integrated circuits with some voice encoding/decoding functionality. The card was manufactured overseas using a chipset from Silicon Labs, and were fairly popular back in the pre-broadband analog-modem days. Digium wrote the asterisk drivers to take advantage of the functionality within the chipset. The Silicon Labs chipset used on the card was designed to meet US Telephone standards, and several competing analog modem manufactures designed their cards using Motorola, Intel, and/or other chipsets. Each chipset has its own unique driver requirements in terms of initializing the chips, moving data, timing, etc. Also keep in mind the card designs were mostly completed back in the days of PCI v1 standards. Some of the cards sold on Ebay are those that use the Silicon Labs chipsets while others are obviously based on the Motorola chipset. The two are not interchangeable without modifying the asterisk drivers. As mentioned, the Silicon Labs chipset used on the card were manufactured to US telephone standards (eg, 600 ohm impedance). Other countries have different standards (guessing, maybe 30-to-50 different telephony standards), and attempting to use the x100p in those environments represents electrical mismatches resulting in echo, no callerid support, and many other objectionable or non-working conditions (depending upon exactly which country you were in). Are some of the Ebay ad's misrepresented? Probably. You're likely to become just about as frustrated with the fxo-sip gateway boxes on the market. Expensive ones have excellent echo cancellation while the cheaper ones are rather poor (or unusable). Some will accept incoming pstn calls while others basically only support outgoing calls (intended for certain failed conditions). Some are targeted for the US market (telephony standards) while others support a larger subset. Some work well on long pstn loops while lots of them don't work very well at all. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?
Florian Overkamp wrote: Michiel van Baak wrote: If you buy a model without the spare in it's name, you have the license to use them right ? To use them with a CCM or CCME, yes :-) How about secondhand phones you get from ebay ? Is my cisco smartnet account enough to run the phone legally ? It's not a spare model (at least that was not in the deal description) My understanding is, if you have any license at all, Cisco will probably not bother you. But it is most definitely not the way they intended :-) The bottom line are the words in the Cisco license agreement that essentially says none of their firmware licenses are transferable. A purchase from Ebay would be an attempt to transfer the license and therefore illegal. There are some Cisco authorized resellers around that do sell used (or in Cisco terms, reconditioned) phones. Some apparently have the capability to bundle smartnet contracts with the phones, but they are not living up to the actual words in the license. The flip side of this... how many people would it take for Cisco to inventory and enforce their written licenses throughout the world? Another side issue... the Cisco phones have no way to remove the installed firmware. Therefore, there is no way to legally sell a used Cisco phone, period. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freeware sip/iax client windows mobile
Attilla De Groot wrote: Hi all, I have two pda's and I want to be able to make calls, but I need a client for this. The only problem is Windows Mobile 5.0, I can't find a freeware client for this, the only one is Sjphone. But this one is still beta for windows mobile and it just doesn't work good. Does anyone have an alternative ? I'm using ppciax -- Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intel vs amd motherboards
On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote: Tzafrir, are you trying to tell me that I can realy do double on the intel becuase the second CPU will do it? In the ideal case you'll get double performance with two CPUs. In theory. A case of many concurrent calls is basically something that can be easily parallelized. So in theory nothing stops you from getting something closer to double performance. I don't know how close reality is to that nice theory. I only remarked that 'show translations' totally ignores the second CPU. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk with ISDN Fritz PCI card
Hi, Thank you for the suggestion. I tried to use mISDN first, then CAPI and now I'm trying I4L. As I'm using Debian, I can not load the FRITZ drivers. I got the source from the official site and recompiled it, but there is a strange message in the log and the capi drivers are not loaded. The problem is more linked to drivers that Asterisk. If you have any tips to get this up and running, I would be very happy as my search on the Internet didn't allowed me to solve that issue. Bests regards, Guy. At 11:25 09.07.2006 +1000, you wrote: What are you using (misdn, capi, something else?) and what problems are you having? I submitted a patch recently to mISDN which should have fixed a problem on hangup, if that's the problem you are having then try the latest cvs mqueue branch of mISDN. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Guy Corbaz Sent: Saturday, 8 July 2006 23:59 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk with ISDN Fritz PCI card Dear all, I'm desperately trying to get Asterisk working with a FRITZ PCI card on Debian with kernel 2.6.15. I'm wondering if anybody has such a working installation. Thank you for your help, Guy. Guy Corbaz ch. du Châtaignier 2 1052 Le Mont Switzerland phone:+41 21 652 26 05 mobile: +41 79 420 26 06 e-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Guy Corbaz ch. du Châtaignier 2 1052 Le Mont Switzerland phone:+41 21 652 26 05 mobile: +41 79 420 26 06 e-mail: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone Ring
Olivier Saulnier wrote: Hello, Do you know where i can download some rings for a PA1688 based Phone? All rings on this link are not very nice...: http://www.aredfox.com/edownloadsring.htm Best regards, Look at the technical documentation on the site, iirc the ringfiles are encoded in a common codec type that you can sox whatever you want to generate. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tired of fax calls... :-/
Olivier wrote: 2006/7/6, Maxim Vexler [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: NVFaxDetect does just that ;) Do you think NVFaxDetect is reliable ? Could you use it along a voicemail (I mean : someone having a single extension for voice and fax call, forward all incoming calls to its voicemail when leaving the office) Cheers If you use NVBackgroundDetect(not-here-greeting) then VoiceMail(sBOX) (s means no announcement), It appears to work. You will need a [fax] context to handle the fax. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zap and fax
Hi,My situation is : I need to send fax from sip device attached fax over zap channel. Using G711, fax send ok, but is it posible to use t.38 protocol. Maybe someone can suggest me what software to use? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can one SIP extension be used for two phones?
Picture this: Exten = 100 #My Phone Exten = 200 #MythPhone Call comes in. Dialplan calls both extensions. MythPhone is an add-on for MythTV,so when i receive a call,the CallerID is flashed up on my TV. I want to add another MythPhone to my other MythTV box upstairs. Do i have to make a third extension and get the dialplan to call all three extensions or can the second MythPhone instance also log into Asterisk using the same extension? Thanks, Alun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing MSNs and chan_misdn
Marco Mouta wrote: in your init-misdn.conf (or misdn.conf, not sure now...) you can choose the MSNs for your incoming Ports or Outgoing ports, msns=3223242,3223243,3223244 for example. Then in your calls, just set the outgoing callerid for your trunk, to one of them. Be aware that as far as i know you must own the MSN you r going to set otherwise you are spoofing MSN Please give some feedback. I've got it working now -- thank you! I notice that you can also use msns=* as well as setting the individual numbers. Once I'd entered that into misdn.conf, and used a command of the form Set(CALLERID(num)=234567) in the dialplan, it now works as I want it to. Gary H -- Gary Hawkins MBCS [EMAIL PROTECTED] PGP: 0x6D4E5C77 (expires 31 Dec 2006) Web: http://www.garyhawkins.me.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can one SIP extension be used for two phones?
Picture this: Exten = 100 #My Phone Exten = 200 #MythPhone Call comes in. Dialplan calls both extensions. MythPhone is an add-on for MythTV,so when i receive a call,the CallerID is flashed up on my TV. I want to add another MythPhone to my other MythTV box upstairs. Do i have to make a third extension and get the dialplan to call all three extensions or can the second MythPhone instance also log into Asterisk using the same extension? Each phone needs its extension But, you can have the dialplan ring more than one phone and the first to pickup the call is the lucky one :) like : Dial(SIP/100SIP/200SIP/300) hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intel vs amd motherboards
Fyi, Double Intel Xeon 3Ghz performance below g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 4 10 29 14 ulaw - 2 - 1 2 2 1 4 10 29 14 alaw - 2 1 - 2 2 1 4 10 29 14 g726 - 2 2 2 - 2 1 4 10 29 14 adpcm - 2 2 2 2 - 1 4 10 29 14 slin - 1 1 1 1 1 - 3 9 28 13 lpc10 - 3 3 3 3 3 2 - 11 30 15 g729 - 3 3 3 3 3 2 5 - 30 15 speex - 3 3 3 3 3 2 5 11 - 15 ilbc - 3 3 3 3 3 2 5 11 30 - Olivier Tzafrir Cohen a crit: On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote: Tzafrir, are you trying to tell me that I can realy do double on the intel becuase the second CPU will do it? In the ideal case you'll get double performance with two CPUs. In theory. A case of many concurrent calls is basically something that can be easily parallelized. So in theory nothing stops you from getting something closer to double performance. I don't know how close reality is to that nice theory. I only remarked that 'show translations' totally ignores the second CPU. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can one SIP extension be used for two phones?
On Sun, Jul 09, 2006 at 11:02:44AM -0400, Time Bandit wrote: Picture this: Exten = 100 #My Phone Exten = 200 #MythPhone Call comes in. Dialplan calls both extensions. MythPhone is an add-on for MythTV,so when i receive a call,the CallerID is flashed up on my TV. I want to add another MythPhone to my other MythTV box upstairs. Do i have to make a third extension and get the dialplan to call all three extensions or can the second MythPhone instance also log into Asterisk using the same extension? Each phone needs its extension But, you can have the dialplan ring more than one phone and the first to pickup the call is the lucky one :) like : Dial(SIP/100SIP/200SIP/300) hth Yeah, I have the dialplan goto voicemail if unanswered so mythphone never actually picks up. I use mythphone purely as a visual indication of who is calling. Thanks for replying. Alun. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 Handsets, Same extension
Is there a Way I can have 2 phones, on the same extension (as in Dial(SIP/phone1SIP/Phone2) ), whereby if one phone is in use, then the other one will not be rung if called? This is useful to me, in the situation where the phone system has a queue and an agent will very often not be at his desk and has access to a wireless handset. If it helps, both handsets are SIP devices. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choppy MOH (Cisco gateway)
Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not connected, separate PBXs) using ulaw all have issues with music on hold being choppy. Normal voice and SIP (taking a call from the PRI, placing a call or extension to extension calls) conversations are _perfect_ with no drop outs so its not a problem with the PRI or the 3660 talking to the Asterisk boxes. If I call from my Polycom into an extension that immediately starts MusicOnHold its perfect as well. However, calling into the box via the PRI and being placed on hold the music is choppy. Also, calling into an extension that spawns MusicOnHold immediately is choppy when it comes in via the Cisco. This happens with mpg123, madplay and I tried using the Asterisk 1.2 native mode in musiconhold.conf: [default] mode = files directory = /var/lib/asterisk/mohmp3 random = yes Same problem with all 3. Tried converting MP3s to a pcm or ulaw file, same problem (using lame and sox to do the conversions) It seems that this is common issue with no clear resolution. Machines are Pentium 4s 512MB or 1GB RAM. I would be the only call on the box, no load, etc. Using ztdummy (or without, same behavior) Asterisk ver 1.2.4 on all Normal voice, IVR, play back voicemail, etc are all 100% perfect only on MusicOnHold has this issue Polycom SIP phones or using X-Lite to test (used to make the call into MusicOnHold or answer the call coming in via the PRI and placing on hold) Calling in from landline or cell phone no difference Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?
snip Another side issue... the Cisco phones have no way to remove the installed firmware. Therefore, there is no way to legally sell a used Cisco phone, period. As a non Cisco user this whole discussion is enough to steer me away from there VOIP products for good. I hate the idea that you need special permission from them to use hardware you bought legally. You bought it you own it, perhaps there should be some sort of open sourced firmware project for these phones? Dunno, but I hate whoever though up this approach. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: What's the story with X10*P FXO cards?
At 12:00 09/07/2006 -0700, Rich Adamson [EMAIL PROTECTED] wrote: Are some of the Ebay ad's misrepresented? Probably. Thanks a lot for the info on the history of the FXO cards. Obviously, the ones I bought aren't the good ones :-) You're likely to become just about as frustrated with the fxo-sip gateway boxes on the market. Mmm... So, what would you recommend? Getting an exensive FXO/SIP box, or a Digium card? The one with a single FXO port sells for about 150E over here. Cheers Vincent. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07/07/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?
Good idea. Let's write some open-source firmware for Cisco phones. What will you be contributing? Martin Joseph wrote: snip Another side issue... the Cisco phones have no way to remove the installed firmware. Therefore, there is no way to legally sell a used Cisco phone, period. As a non Cisco user this whole discussion is enough to steer me away from there VOIP products for good. I hate the idea that you need special permission from them to use hardware you bought legally. You bought it you own it, perhaps there should be some sort of open sourced firmware project for these phones? Dunno, but I hate whoever though up this approach. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do you need a licence to connect a Ciscohardphone to Asterisk ?
On Jul 9, 2006, at 12:11 PM, Jay Milk wrote: Good idea. Let's write some open-source firmware for Cisco phones. What will you be contributing? Well, it looks like I already contributed the idea didn't I ;~) I don't have any Cisco phones, so if you want to send be a couple I'll take a look at them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Handsets, Same extension
It looks like a good starting point would be here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup On 7/9/06, Thomas Kenyon [EMAIL PROTECTED] wrote: Is there a Way I can have 2 phones, on the same extension (as inDial(SIP/phone1SIP/Phone2) ), whereby if one phone is in use, then the other one will not be rung if called?This is useful to me, in the situation where the phone system has aqueue and an agent will very often not be at his desk and has access toa wireless handset. If it helps, both handsets are SIP devices.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Global variables and AGI
Hi everyone, I know that functions like set_variable and get_variable (using php with phpagi) only apply to the channel variable. What I need to do is reset a global variable I have in our system. I have a script that is going to determine when this will happen, but I just have to make it happen. Assuming that I cannot update the variable via the script, it is there a way I can make a call to the system, such as a call file, and place it in the context of the dialplan that I need to change the variable? If so, is there anything special I need in the call file for that to work? Or is there a easier/better way to do this that I haven't thought of. Any suggestions would be helpful. Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Global variables and AGI
Hi everyone, I know that functions like set_variable and get_variable (using php with phpagi) only apply to the channel variable. What I need to do is reset a global variable I have in our system. I have a script that is going to determine when this will happen, but I just have to make it happen. Assuming that I cannot update the variable via the script, it is there a way I can make a call to the system, such as a call file, and place it in the context of the dialplan that I need to change the variable? If so, is there anything special I need in the call file for that to work? Or is there a easier/better way to do this that I haven't thought of. Reading this : http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set you need to use the g option If you can't use it from phpagi, you could, at worst, set a local variable from your AGI, then in the dialplan take the value of that one and apply it to the global one using option g hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intel vs amd motherboards
Olivier can you please do a cat /proc/cpuinfo and post it here? I think you have a 64 bit cpu. On 7/9/06, olivier.taylor [EMAIL PROTECTED] wrote: Fyi, Double Intel Xeon 3Ghz performance below g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 2 2 2 2 1 4102914 ulaw - 2 - 1 2 2 1 4102914 alaw - 2 1 - 2 2 1 4102914 g726 - 2 2 2 - 2 1 4102914 adpcm - 2 2 2 2 - 1 4102914 slin - 1 1 1 1 1 - 3 92813 lpc10 - 3 3 3 3 3 2 -113015 g729 - 3 3 3 3 3 2 5 -3015 speex - 3 3 3 3 3 2 511 -15 ilbc - 3 3 3 3 3 2 51130 - Olivier Tzafrir Cohen a écrit : On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote: Tzafrir, are you trying to tell me that I can realy do double on the intel becuase the second CPU will do it? In the ideal case you'll get double performance with two CPUs. In theory. A case of many concurrent calls is basically something that can be easily parallelized. So in theory nothing stops you from getting something closer to double performance. I don't know how close reality is to that nice theory. I only remarked that 'show translations' totally ignores the second CPU. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] intel vs amd motherboards
Thanks for that Tzafrir. Why does it ignore the secend CPU? BTW, on a side note on this topic, how can one calculate simultaneous transcoded channels using show transalation? In the case where it tells me 17 ms for encoding and 4 for decoding, that gives me 21ms per channel, in what time frame can I squeeze in how many channels before the calls start becoming intolerable? In other words should I aim for a 200ms time frame which means that I will get around 10 channels? or can I aim for a full second? which will give me around 50 channels? Thank You On 7/9/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Jul 09, 2006 at 05:07:16AM -0400, C F wrote: Tzafrir, are you trying to tell me that I can realy do double on the intel becuase the second CPU will do it? In the ideal case you'll get double performance with two CPUs. In theory. A case of many concurrent calls is basically something that can be easily parallelized. So in theory nothing stops you from getting something closer to double performance. I don't know how close reality is to that nice theory. I only remarked that 'show translations' totally ignores the second CPU. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Handsets, Same extension
On Sun, 09 Jul 2006 19:17:36 +0100, Thomas Kenyon wrote: Is there a Way I can have 2 phones, on the same extension (as in Dial(SIP/phone1SIP/Phone2) ), whereby if one phone is in use, then the other one will not be rung if called? This is useful to me, in the situation where the phone system has a queue and an agent will very often not be at his desk and has access to a wireless handset. If it helps, both handsets are SIP devices. I do this throught using Astra 480i CT phones. Each desk phone comes with an associated cordless handset. Calls ring on both with only one registration into the server. Costs about the same as a decent Polycom phones. Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: What's the story with X10*P FXO cards?
Skip local FXOs altogether. Setup an account with somone who provides DIDs via IP. Call forward your analog line to the IP based number. It will be absolutely painless compared to the troubles of small FXO interfaces. Michael On Sun, 09 Jul 2006 21:09:40 +0200, The Masked Cucumber wrote: At 12:00 09/07/2006 -0700, Rich Adamson [EMAIL PROTECTED] wrote: Are some of the Ebay ad's misrepresented? Probably. Thanks a lot for the info on the history of the FXO cards. Obviously, the ones I bought aren't the good ones :-) You're likely to become just about as frustrated with the fxo-sip gateway boxes on the market. Mmm... So, what would you recommend? Getting an exensive FXO/SIP box, or a Digium card? The one with a single FXO port sells for about 150E over here. Cheers Vincent. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07/07/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] NuFone suggests to use Vonage!!!!
Part of a conversation with NuFone. It is untrue, that they do not answer, but if than: Quote: 3. change your attitude towards customers!! No, if you don't like it, go use Vonage. End of quote! I had always problems with these people. bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NuFone suggests to use Vonage!!!!
Ronald Wiplinger wrote: Part of a conversation with NuFone. It is untrue, that they do not answer, but if than: Quote: 3. change your attitude towards customers!! No, if you don't like it, go use Vonage. End of quote! I had always problems with these people. bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users To some extent I see your point and have been on the receiving end of one of Jeremy's tirades. I've since decided that NuFone is an interesting study in whether your business can survive with only clueful customers. Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy MOH (Cisco gateway)
i had a similar issue with the first branch of asterisk 1.2 and cheap phones (tip-100 from tatung) i'll suggest you to upgrade your asterisk box are you using bristuff ? try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1 lemme know .mike On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote: Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not connected, separate PBXs) using ulaw all have issues with music on hold being choppy. Normal voice and SIP (taking a call from the PRI, placing a call or extension to extension calls) conversations are _perfect_ with no drop outs so it’s not a problem with the PRI or the 3660 talking to the Asterisk boxes. If I call from my Polycom into an extension that immediately starts MusicOnHold it’s perfect as well. However, calling into the box via the PRI and being placed on hold the music is choppy. Also, calling into an extension that spawns MusicOnHold immediately is choppy when it comes in via the Cisco. This happens with mpg123, madplay and I tried using the Asterisk 1.2 native mode in musiconhold.conf: [default] mode = files directory = /var/lib/asterisk/mohmp3 random = yes Same problem with all 3. Tried converting MP3s to a pcm or ulaw file, same problem (using lame and sox to do the conversions) It seems that this is common issue with no clear resolution. Machines are Pentium 4s 512MB or 1GB RAM. I would be the only call on the box, no load, etc. Using ztdummy (or without, same behavior) Asterisk ver 1.2.4 on all Normal voice, IVR, play back voicemail, etc are all 100% perfect only on MusicOnHold has this issue Polycom SIP phones or using X-Lite to test (used to make the call into MusicOnHold or answer the call coming in via the PRI and placing on hold) Calling in from landline or cell phone – no difference Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NuFone suggests to use Vonage!!!!
On Sun, 9 Jul 2006, Andrew D Kirch wrote: To some extent I see your point and have been on the receiving end of one of Jeremy's tirades. I've since decided that NuFone is an interesting study in whether your business can survive with only clueful customers. Some people are into SM I guess. We have used NuFone. No problems during that period. If you know what you doing it's not bad. regards joe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] audio session start delay
On Thu, 2006-07-06 at 23:22 -0300, Fabio wrote: are you using SIP reinvite ? Proably not as I'm using t in Dial()s for call transfer. post a bit more information (sip.conf) [general] context=sip allowguest=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no domain=mydomain.com domain=1.2.3.4 allowexternalinvites=no language=it relaxdtmf=yes [authentication] [as5350] ; My PSTN gateway type=peer qualify=200 host=1.2.3.5 fromdomain=1.2.3.5 insecure=very [ser] ; My SIP proxy type=peer qualify=200 host=1.2.3.6 fromdomain=1.2.3.6 insecure=very [01]; Extension example callerid=My Name 01 nat=yes type=friend username=01 secret=mypass host=dynamic dtmfmode=rfc2833 context=uffici canreinvite=no callgroup=1 pickupgroup=1 qualify=no Thanks -- Luca Corti PGP Key ID 1F38C091 Adesso dico: L'usignolo chiuso in gabbia smette di cantare. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to transfer other sessions
Hi all. I am looking for a method to transfer a caller on an existing session to another extension. For example, SIP/200 is currently talking with SIP/300. I want to force user SIP/200 to be transfered to SIP/400. (without myself being SIP/200. I am just an admin at the asterisk console) Then SIP/300 left alone, can be hung up. Is there any way this can be done from the Asterisk console or any other Dial plan commands or AGI commands ? Any advise or pointers I will truly appreciate. Thank you. -- David Shimamoto ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Choppy MOH (Cisco gateway)
I upgraded one of the boxes to 1.2.9.1 and using native MOH I still get it. I made sure to upgrade zaptel, etc as well. I do have something of interest to note... Placing the call on hold then taking it off hold and back on the music is ok (doing that once it gets choppy) of course this is not practical since the person using hold won't know if it's choppy. It then gets choppy again if you wait 15-20 secs. I have 2 ways of making outbound calls from all of the boxes, and I did the following via 1.2.9.1 and 1.2.4 1) Send the outbound call to the Cisco and send out via the PRI (sip phone ulaw to Cisco ulaw out the PRI) 2) Dial long distance to a provider using g729 (Polycom to Asterisk ulaw, Asterisk transcoding to g729 to provider) If I call from a sip phone OUT to my cell via the long distance provider I get no choppiness. I am not able to get inbound calls from the provider so I can only test one way. So I then switched talking to my Cisco via g729 (letting asterisk transcode ulaw to g729 and also g729 all the way through) and voice is fine but MOH is still choppy. So it must be something with the Cisco maybe? IOS version is Cisco IOS Software, 3600 Software (C3660-IS-M), Version 12.3(14)T6, RELEASE SOFTWARE (fc2) I have setup for the codecs: voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 incoming dial-peer: dial-peer voice 1 pots description Match all incoming calls, set DID incoming called-number .T direct-inward-dial forward-digits extra dial-peer voice 16 voip description to the asterisk server destination-pattern phone# voice-class codec 1 session protocol sipv2 session target ipv4:ip dtmf-relay sip-notify rtp-nte and outbound: dial-peer voice 1 pots description Outbound via PRI destination-pattern .T port 1/0:23 forward-digits all Could this have something to do with the Cisco suppressing the stream using silence suppression...I read somewhere that Asterisk relies on Sip packets for MOH??? There is not a bandwidth issue, the 3660 and boxes are on the same switch VLAN w/ DSCP enabled. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mike Sent: Monday, July 10, 2006 2:51 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) i had a similar issue with the first branch of asterisk 1.2 and cheap phones (tip-100 from tatung) i'll suggest you to upgrade your asterisk box are you using bristuff ? try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1 lemme know .mike On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote: Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not connected, separate PBXs) using ulaw all have issues with music on hold being choppy. Normal voice and SIP (taking a call from the PRI, placing a call or extension to extension calls) conversations are _perfect_ with no drop outs so it's not a problem with the PRI or the 3660 talking to the Asterisk boxes. If I call from my Polycom into an extension that immediately starts MusicOnHold it's perfect as well. However, calling into the box via the PRI and being placed on hold the music is choppy. Also, calling into an extension that spawns MusicOnHold immediately is choppy when it comes in via the Cisco. This happens with mpg123, madplay and I tried using the Asterisk 1.2 native mode in musiconhold.conf: [default] mode = files directory = /var/lib/asterisk/mohmp3 random = yes Same problem with all 3. Tried converting MP3s to a pcm or ulaw file, same problem (using lame and sox to do the conversions) It seems that this is common issue with no clear resolution. Machines are Pentium 4s 512MB or 1GB RAM. I would be the only call on the box, no load, etc. Using ztdummy (or without, same behavior) Asterisk ver 1.2.4 on all Normal voice, IVR, play back voicemail, etc are all 100% perfect only on MusicOnHold has this issue Polycom SIP phones or using X-Lite to test (used to make the call into MusicOnHold or answer the call coming in via the PRI and placing on hold) Calling in from landline or cell phone - no difference Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Choppy MOH (Cisco gateway)
You will also want to add no vad to your dial-peer config to disable voice activity detection. I do not think it will resolve your issue, but worth a shot. -John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bill Gibbs Sent: Sunday, July 09, 2006 7:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Choppy MOH (Cisco gateway) I upgraded one of the boxes to 1.2.9.1 and using native MOH I still get it. I made sure to upgrade zaptel, etc as well. I do have something of interest to note... Placing the call on hold then taking it off hold and back on the music is ok (doing that once it gets choppy) of course this is not practical since the person using hold won't know if it's choppy. It then gets choppy again if you wait 15-20 secs. I have 2 ways of making outbound calls from all of the boxes, and I did the following via 1.2.9.1 and 1.2.4 1) Send the outbound call to the Cisco and send out via the PRI (sip phone ulaw to Cisco ulaw out the PRI) 2) Dial long distance to a provider using g729 (Polycom to Asterisk ulaw, Asterisk transcoding to g729 to provider) If I call from a sip phone OUT to my cell via the long distance provider I get no choppiness. I am not able to get inbound calls from the provider so I can only test one way. So I then switched talking to my Cisco via g729 (letting asterisk transcode ulaw to g729 and also g729 all the way through) and voice is fine but MOH is still choppy. So it must be something with the Cisco maybe? IOS version is Cisco IOS Software, 3600 Software (C3660-IS-M), Version 12.3(14)T6, RELEASE SOFTWARE (fc2) I have setup for the codecs: voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 incoming dial-peer: dial-peer voice 1 pots description Match all incoming calls, set DID incoming called-number .T direct-inward-dial forward-digits extra dial-peer voice 16 voip description to the asterisk server destination-pattern phone# voice-class codec 1 session protocol sipv2 session target ipv4:ip dtmf-relay sip-notify rtp-nte and outbound: dial-peer voice 1 pots description Outbound via PRI destination-pattern .T port 1/0:23 forward-digits all Could this have something to do with the Cisco suppressing the stream using silence suppression...I read somewhere that Asterisk relies on Sip packets for MOH??? There is not a bandwidth issue, the 3660 and boxes are on the same switch VLAN w/ DSCP enabled. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mike Sent: Monday, July 10, 2006 2:51 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Choppy MOH (Cisco gateway) i had a similar issue with the first branch of asterisk 1.2 and cheap phones (tip-100 from tatung) i'll suggest you to upgrade your asterisk box are you using bristuff ? try bristuff-0.3.0-PRE-1q which comes with asterisk 1.2.9.1 lemme know .mike On Sun, 2006-07-09 at 14:49 -0400, Bill Gibbs wrote: Cisco 3660 with a PRI talking SIP to various Asterisk boxes (not connected, separate PBXs) using ulaw all have issues with music on hold being choppy. Normal voice and SIP (taking a call from the PRI, placing a call or extension to extension calls) conversations are _perfect_ with no drop outs so it's not a problem with the PRI or the 3660 talking to the Asterisk boxes. If I call from my Polycom into an extension that immediately starts MusicOnHold it's perfect as well. However, calling into the box via the PRI and being placed on hold the music is choppy. Also, calling into an extension that spawns MusicOnHold immediately is choppy when it comes in via the Cisco. This happens with mpg123, madplay and I tried using the Asterisk 1.2 native mode in musiconhold.conf: [default] mode = files directory = /var/lib/asterisk/mohmp3 random = yes Same problem with all 3. Tried converting MP3s to a pcm or ulaw file, same problem (using lame and sox to do the conversions) It seems that this is common issue with no clear resolution. Machines are Pentium 4s 512MB or 1GB RAM. I would be the only call on the box, no load, etc. Using ztdummy (or without, same behavior) Asterisk ver 1.2.4 on all Normal voice, IVR, play back voicemail, etc are all 100% perfect only on MusicOnHold has this issue Polycom SIP phones or using X-Lite to test (used to make the call into MusicOnHold or answer the call coming in via the PRI and placing on hold) Calling in from landline or cell phone - no difference Any ideas? Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users
Re: [asterisk-users] CallerID
Thanks to everyone.I had everything right except that rxgain and txgain were set to 0.I am actually embarassed to say that I spent most of Saturday getting lost and learning a lot and the stupid mistake was that the telephone that I was calling from has caller id blocked. Well, the only satisfaction is that I now know how the whole thing is configured. I also learnt that you have to start with a clean slate for dialplans, and throw away what's left as *.conf files after [EMAIL PROTECTED] installation. Also, avoid deeply nested includes.-bramanRyder Brook PediatricsP.O.Box 608Morrisville, VT 05661 __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] spandsp and app_*fax.c
Greetings, Does anyone know where I can get app_txfax and app_rxfax thatll work with spandsp 0.0.3pre22? Ive tried the source in the snapshots/test-apps directory. Ive also tried the files that came with t38bits.tgz. This is what I get out of make: make[1]: *** No rule to make target `/usr/local/include/spandsp/alaw_ulaw.h', needed by `app_rxfax.o'. Stop. make[1]: Leaving directory `/usr/src/asterisk-1.2.9.1/apps' make: *** [subdirs] Error 1 My real issue is that I cant get txfax to send a fax without a core dump using 0.0.2pre26 with Asterisk 1.2.9.1. Ive tried many versions of libtiff; 3.6.0, 3.6.1, 3.5.7, 3.8.2 and others - always a core dump. Ive tried the same thing with a brand new trixbox install core dump. I did get a step further by commenting lines in t4.c but I know this is not a real solution. Short of trying to debug this (which Im going to try while waiting for responses to this) what should I be trying to do? Any advice would be very much appreciated. Heres a backtrace from gdb if this is at all helpful. (gdb) frame #0 0x00930540 in _int_malloc () from /lib/tls/libc.so.6 (gdb) bt #0 0x00930540 in _int_malloc () from /lib/tls/libc.so.6 #1 0x009320b1 in malloc () from /lib/tls/libc.so.6 #2 0x00932869 in realloc () from /lib/tls/libc.so.6 #3 0x00738f55 in t4_encode_eol (s=0x9f5820) at t4.c:404 #4 0x0073af7e in t4_encode_row (s=0xb7b163d0, bp=0x8c4b348 'ÿ' repeats 200 times...) at t4.c:1570 #5 0x0073b76c in t4_tx_start_page (s=0xb7b163d0) at t4.c:1854 #6 0x0073ee4e in hdlc_accept (user_data=0xb7b163d0, ok=1, msg=0xb7b16644 ÿ\023\204ê}, len=3) at t30.c:1135 #7 0x007328a5 in hdlc_rx_bit (s=0xb7b1661c, new_bit=0) at hdlc.c:300 #8 0x0072f861 in fsk_rx (s=0xb7b169d8, amp=0x8c1a6fc, len=160) at fsk.c:344 #9 0x0073f7bd in fax_rx (s=0xb7b163d0, buf=0x8c1a6fc, len=160) at t30.c:2712 #10 0x00896427 in txfax_exec (chan=0x8bfc770, data=""> at app_txfax.c:230 #11 0x0808eac2 in ast_pbx_outgoing_app ( type=0x8c4ba4c 'ÿ' repeats 200 times..., format=64, data=""> timeout=2, app=0x8c4bc4c 'ÿ' repeats 200 times..., appdata=0x8c4bd4c 'ÿ' repeats 200 times..., reason=0xb7b18438, sync=2, cid_num=0x8c4c050 'ÿ' repeats 200 times..., cid_name=0x8c4c150 'ÿ' repeats 200 times..., vars=0x0, account=0x8c4c250 'ÿ' repeats 172 times, locked_channel=0x0) at pbx.c:553 #12 0x002e736d in attempt_thread (data="" at pbx_spool.c:262 #13 0x00baa371 in start_thread () from /lib/tls/libpthread.so.0 #14 0x009959be in clone () from /lib/tls/libc.so.6 (gdb) Cheers, Warrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Global variables and AGI
Kevin Smith wrote: Hi everyone, I know that functions like set_variable and get_variable (using php with phpagi) only apply to the channel variable. What I need to do is reset a global variable I have in our system. I have a script that is going to determine when this will happen, but I just have to make it happen. Assuming that I cannot update the variable via the script, it is there a way I can make a call to the system, such as a call file, and place it in the context of the dialplan that I need to change the variable? If so, is there anything special I need in the call file for that to work? Or is there a easier/better way to do this that I haven't thought of. Any suggestions would be helpful. Thanks, Kevin As Timebandit pointed out -- http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Set or SetGlobalVar in 1.0.x If most of the interaction with that variable occurs through agi, you might also want to consider storing it outside of Asterisk. I've stored a good number of values in mysql for an asterisk application before. If most of the interaction occurs within the dialplan and/or you're trying to avoid agi, you could also use the asterisk database directly with DBPut and DBGet. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: What's the story with X10*P FXO cards?
The Masked Cucumber wrote: At 12:00 09/07/2006 -0700, Rich Adamson [EMAIL PROTECTED] wrote: Are some of the Ebay ad's misrepresented? Probably. Thanks a lot for the info on the history of the FXO cards. Obviously, the ones I bought aren't the good ones :-) You're likely to become just about as frustrated with the fxo-sip gateway boxes on the market. Mmm... So, what would you recommend? Getting an exensive FXO/SIP box, or a Digium card? The one with a single FXO port sells for about 150E over here. As mentioned previously, recommending a specific device basically assumes one understands exactly what type of pstn connection you've got, how far you are from the central office, etc. Some folks have had good luck with the spa3000 (and some not). The TDM400 card from digium works pretty good, but can be less then acceptable in some cases. The Sangoma A200D (with hardware EC) works very well on all pstn lines that I've tested, but is rather expensive. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI Random Disconnected
Dear Group, I am having some problem with PRI, my calls randomly get disconnected and after I am running Debug, I got the out from CLi screen... Cli messages, -- Executing Dial(Zap/31-1, zap/g1/100||rTt) in new stack -- Making new call for cr 32809 -- Requested transfer capability: 0x10 - 3K1AUDIO Protocol Discriminator: Q.931 (8) len=30 Call Ref: len= 2 (reference 41/0x29) (Originator) Message type: SETUP (5) [04 03 90 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 03 21 81 20] Calling Number (len= 5) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) ' ' ] [70 04 c1 31 30 30] Called Number (len= 6) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '100' ] -- Called g1/100 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 41/0x29) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) -- Zap/1-1 is proceeding passing it to Zap/31-1 Protocol Discriminator: Q.931 (8) len=12 Call Ref: len= 2 (reference 41/0x29) (Terminator) Message type: STATUS (125) [08 02 81 e4] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (100), class = Protocol Error (6) ] [14 01 09] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Incoming Call Proceeding (9) -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 41/0x29) (Terminator) Message type: ALERTING (1) -- Zap/1-1 is ringing Protocol Discriminator: Q.931 (8) len=12 Call Ref: len= 2 (reference 41/0x29) (Terminator) Message type: CONNECT (7) [4c 05 01 80 31 30 30] Connected Number (len= 7) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Ext: 1 Presentation: Presentation permitted, user number not screened (0) '100' ] -- Processing IE 76 (cs0, Connected Number) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 41/0x29) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/1-1 answered Zap/31-1 -- Attempting native bridge of Zap/31-1 and Zap/1-1 ---Extension 100 pickup the call, and get disconnected Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 41/0x29) (Terminator) Message type: DISCONNECT (69) [08 02 85 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the remote user (5) Ext: 1 Cause: Unknown (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup request /. Zapata.conf file [channels] context=from-alcatel switchtype=national overlapdial=yes signalling=pri_net pridialplan=local resetinterval=never facilityenable = yes priindication=outofband group=1 usecallerid=yes hidecallerid=no threewaycalling=yes transfer=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=-3.5 txgain=-3.5 busydetect=yes ;Busydetect I did set to no, and calls does not hangup so I turn to yes. relaxdtmf=yes immediate=no channel=1-15 channel=17-31 Equipment Topologies Connection: Analog Trunk line PSTN (Alcatel Ompcx Office) -- Asterisk (E1 as PRI_NET) Incoming goes into Alcatel and Alcatel Divert it to Asterisk via E1 Asterisk as a Auto-Attendance System. Asterisk R1.2.5 / Zaptel 1.2.5 / Libpri 1.2.2 .CentOS 4.3 final (Kernel 2.6.9-34.01EL) Appreciate if some one can help. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: What's the story with X10*P FXO cards?
At 22:36 09/07/2006 -0700, Michael Graves [EMAIL PROTECTED] wrote: Skip local FXOs altogether. Setup an account with somone who provides DIDs via IP. Call forward your analog line to the IP based number. It will be absolutely painless compared to the troubles of small FXO interfaces. I'll look into this, although 1. I'm not sure VoIP providers do this here yet 2. While the POTS is very reliable, I can't say the same for ADSL. I'm a bit scared to depend on the Net for incoming calls. At 22:36 09/07/2006 -0700, Rich Adamson [EMAIL PROTECTED] wrote: Some folks have had good luck with the spa3000 (and some not). The TDM400 card from digium works pretty good, but can be less then acceptable in some cases. The Sangoma A200D (with hardware EC) works very well on all pstn lines that I've tested, but is rather expensive. Thanks for the info. This little experiment is getting expensive ;-) VD. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.9.10/383 - Release Date: 07/07/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
