[Asterisk-Users] STUPID question? Tellabs echo can cards and PSTN?
I am wondering if the instructions for hard wiring a Tellabs canceler are applicable to a regular old two wire loop? Or is this only something that works for people with T1? Any comments from people that have tried this are appreciated. ___ --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] Fedora Core 3 or Fedora Core 4? yum update ornot?
On Wed, Feb 08, 2006 at 10:20:43AM +, Jens Vagelpohl wrote: On 8 Feb 2006, at 09:43, JP Carballo wrote: Alex Barnes wrote: I think the once it's working, leave it alone advice is very sound indeed :) A similar rule says If it ain't broke, don't fix it. Until you realize some script kiddie has exploited another Apache/ mod_ssl bug and is now remote-controlling your box. There are no hard and fast recipes here. Neither the automatically apply any and all updates nor the build and never look at it again- policies should be applied without taking the specific situation into account. If your box is on the internet you simply cannot forego updates. Period. If your box is completely walled off from the internet you can be lax about it (unless you have to worry about attacks from the inside). If the box does voip then it is on a network. And thus an explotable target. You should also make it not trivial for an attacker to gain root even after some successful exploit, if possible. The best policy is probably one that is halfway between the two. There are packages you only ever want to update under parental supervision, like kernels. Then there are packages where you want to grab any update you can get ASAP, like Apache, or PHP, or SSH. Yum allows you to express this in its configuration, you can exclude packages from the automatic update. But first and formost, pick a distro on which you could trust to provide relieble updates that don't break. If you can't rely on the distro for apache, PHP, SSH and the kernel, you'll end up with a broken config. I assume that this is not the only box you'll have to maintain. And that you'll have better things to do than watchig bugtraq all day long. I personally run a nightly script that uses yum to determine if there are updates. I apply them by hand. However, this is only feasible because it runs on just two machines. Not sure about other distros. On $MY_DISTRO there is a package to run that automatically. Which is kind of expected because enough people have come to rely on the updates to apply the automatically. The least you should do is to download al the updates automaically, to mak th time required for applying them minimal. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] IP Authorization
Ah that is from the CLI but still unclear about how to setup the extension.conf or etc.. Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Friday, February 10, 2006 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] IP Authorization You can use the following: switch3*CLI show function SIPCHANINFO switch3*CLI -= Info about function 'SIPCHANINFO' =- [Syntax] SIPCHANINFO(item) [Synopsis] Gets the specified SIP parameter from the current channel [Description] Valid items are: - peeripThe IP address of the peer. - recvipThe source IP address of the peer. - from The URI from the From: header. - uri The URI from the Contact: header. - useragent The useragent. - peername The name of the peer. All the info you need is there. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam Tam Sent: Thursday, February 09, 2006 9:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] IP Authorization Can you be more detail about the setup? Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson Sent: Friday, February 10, 2006 4:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IP Authorization Sam Tam wrote: I think this is a question that has been discussed before. But you see nowadays most carriers will provide thing like SIP using IP authorization rather than username and password and I am now wondering whether Asterisk can do something like that or not? In the voip channels as well as in manager you can set ACLs for the connections you define. /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update ornot?
On Wed, Feb 08, 2006 at 10:38:33AM -0500, Technical Support wrote: I think that some people try to make their asterisk box a do-everything super server. Can you image a traditional PBX with direct access via the internet, serving web pages via apache, running sendmail, etc. Our approach has been keep it simple. We lock each Asterisk PBX down has hard as possible. This includes no direct internet connection (it should sit behind a real firewall), minimal services running, etc. With this philosophy, one can treat the PBX as an appliance: don't touch it if it's working. Then I suppose your PBX does not do direct voip. All voip is proxied by the firewall (with a special voip anti-virus to keep the bad guys from exploiting you through there). This also applies to whaever other voice channels you use. And also to some overly-complicated IVRs that may allow unintended privileges escalation: you wanted to avoid a clear and simple web interface, so you opted for a complicated phone interface. If you must run host web pages, run mail servers, offer SQLnet connections, make visible to the internet, Actually if a mail/SQL server is used it is either only availble to localhost. etc. then other users are correct - you better continually patch/update ASAP. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Fedora Core 3 or Fedora Core 4? yum update ornot?
On Wed, Feb 08, 2006 at 01:03:51PM -0500, Paul wrote: Ryan Amos wrote: This is turning into a sysadmin theory flamewar, but I think the main point is that Fedora probably isn't the best thing to run on production machines for QA reasons. This is because Fedora is more or less the QA testbed for RHEL. CentOS is, for all intents and purposes (except a little bug I discovered with large block devices 2 TB) the same as RHEL without the support contract, so it is probably a better choice for a server you want to keep working for a while. Debian stable would probably work just as well (though IMO debian tends to be a bit TOO old,) as would SUSE's stable release version. Just don't use a testing release on a production machine. yum update (or up2date, or apt) is pretty safe on stable release trees, but in the testing releases you can run into problems with package dependencies, versions, slowly updated mirrors... you get the point. Debian stable is not so old. No decent distro is going to do a new stable release every time a new asterisk, openoffice, firefox, etc. is released. That's why they call it stable. There are several ways to get newer asterisk versions onto a debian stable system. The end user decides what risks to take in modifying any stable distro. Best approach for me has been to limit those changes to what I really must have. I take something like a new openoffice and try it out on a debian system running testing or unstable. If I like it enough, I find or build debian packages for the stable release. I think this sane and careful approach works with most linux distros but I have seen some distros where the testing or unstable branch was not installable at times. http://backports.org seems to be building Asterisk 1.2 relatively regularly from Unstable. We (Xorcom Rapid) also provide rather compatible Sarge backports of all things Asterisk -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Corrupt CDR records in Asterisk 1.2.x
I have an [EMAIL PROTECTED] installed and then upgraded to 1.2.4 with bristuff and i have the same problem. If you see the logs the INSERT trace has wrong values before the comand is executed. By the way, everyone of us that have this problem use HFC cards? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: jueves, 09 de febrero de 2006 18:28 Para: Jeroen Zwarts; Asterisk Users Mailing List - Non-Commercial Discussion CC: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x You are in my same situation. I thought I solved the problem (if you look at tomorrow post) but it isn't My situation is a bit different: I have the last bristuffed version of asterisk 1.2.4 (released yesterday) And I also have 2 zaphfc cards. but the behaviour is absolutely the same If you restart asterisk, you get one or two calls ok, the again the problem On the first zaphfc, the problem is almost immediate (1 or two calls) the second is stronger, and is ok for a longer period ( 1 day ??) then it also falls in problem on clid and src It seems to me some buffer overwrite problem. the clid is trasmitted ok to the internal phones. So I am not alone on this side... Andrea Jeroen Zwarts [EMAIL PROTECTED] nl To Sent by: asterisk-users@lists.digium.com asterisk-users-bo cc [EMAIL PROTECTED] m.com Subject [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x 09/02/2006 11.05 Please respond to Jeroen Zwarts [EMAIL PROTECTED] nl; Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I have a problem with CDR recording in Asterisk 1.2.x. This is the situation: An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single HFC-S ISDN BRI card. I log the call records to both the Master.csv and MySQL. The problem is that when an incoming call from the ISDN line is logged to the CDR, the src and the clid field show up as something like 'h? ' (random weird ASCII characters). This is in the MySQL table as well as the Master.csv, so my guess is that it is not a MySQL problem. Furthermore, I don't think it is a zaptel/bristuff problem, because my AGI scripts get the incoming number without problems all the time. The internal SIP calls are logged without a problem all the time. It's only ISDN calls from the outside world that are corrupt. When I stop Asterisk with stop now and restart it, the src and clid fields are OK for a while, but after a few calls, or as some time passes by (I don't know what triggers it), it goes back to the 'random ASCII weirdness'. I also tested this with Asterisk 1.2.4 (BRIstuffed-0.3.0-PRE-1h with florz) and I have the same problem. Again, when I start Asterisk, everything is OK for a while, and then suddenly, the src and clid fields are like 'ÀÜ' Anybody has a clue as where to start looking for a solution for this problem? I can't seem to find a single post, list e-mail or bug related to this problem. Thanks, Jeroen Zwarts ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users
[Asterisk-Users] Expression GotoIf - bug or personal misunderstanding?
Hi, I am using 1.2.4 of asterisk. From the console: -- Executing GotoIf(Zap/29-1, 1 0?4:3) in new stack -- Goto (macro-stdexten,s-NOANSWER,4) In my understanding the expression (1 0) should be lead to 0, but in this case it leads to 1. Can anybody explain this to me? Much thanks ___ --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] Any way to grep through fast moving consolemessages?
Yeah I do this, create 2 ssh sessions to the same box, on the first session do `script -f /tmp/astcli` `asterisk r` (and whatever other options you need on the second session `tail f /tmp/astcli | grep -i bob` (on the grep you may have to ignore control chars if you have colour at cli, I think thats the -a option) then you can modify the grep to look for any messages you want. You can also stop the script and read the /tmp/astcli as you like. Hope that helps -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop Sent: 10 February 2006 02:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Any way to grep through fast moving consolemessages? Or perhaps slow them down or pipe to a file? ___ --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] Unistim Packet Decoder
On Thu, Feb 09, 2006 at 07:37:42PM -0500, Polycom User wrote: Anyone know of one that I could use? What do you mean? There's a chan_unistim for Asterisk -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] 2wav2mp3, monitor, mixmonitor, mpg123, queues
Hello!
I'm using Asterisk for our office telephony, but we have some problems
that still we can't resolve about it. Here they are:
1) merge in/out call recording files
I also tried to use a script I found on the internet, called 2wav2mp3
In extensions.conf I added the following lines
; script to be executed when monitoring has been finished
MONITOR_EXEC=/usr/local/bin/2wav2mp3
exten = 102,1,SetVar(CALLFILENAME=${TIMESTAMP}-${EXTEN}-${CALLERID})
exten = 102,2,Monitor(wav,${CALLFILENAME},m)
;exten = 102,2,MixMonitor(${CALLFILENAME}.gsm)
;exten = 102,2,MixMonitor(test.wav,W(-3))
exten = 102,3,Ringing
exten = 102,4,Dial(Sip/giuseppedd,20,rtwW)
...but I always get two separate files.
As you can see I also tried the MixMonitor application but the resulting
files
contain one channel that is clearly audible and the other seems to be noise.
2) an alternative to mpg123 becouse it generates a lot of errors like this:
Feb 3 19:50:08 WARNING[9568]: res_musiconhold.c:488 monmp3thread:
Unable to spawn mp3player
Feb 3 19:58:28 NOTICE[9568]: res_musiconhold.c:507 monmp3thread:
Request to schedule in the past?!?!
Feb 3 19:58:28 WARNING[9568]: res_musiconhold.c:421 spawn_mp3: Found no
files in '/usr/share/asterisk/mohagents'
3) how to play different files to an agent before he picks up a call
depending on which queue the call comes from
[qlu500]
musiconhold = qlu500
announce = vm-from-phonenumber ; here is the problem
context = qlu500out
wrapuptime=15
announce-frequency = 60
...
Comments or suggestions are greatly appreciated.
Thanks a lot.
Giuseppe
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RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x
Yes, everybody of us use zaphfc. No problem at all with zap channel that I have installed in several other * Box (PRI E1, Quad Analog, Chan_capi on 1.0 * series) So the problem, I think, is the zaphfc, or the patch to the * and zaptel provided by the bristuff. I tried to post a question to [EMAIL PROTECTED] but the mail was refused with a status code of 550 5.0.0, Dial-Up IP address rejected The public ip address I am using is from a newly buyed (3 days ago) set of 8 IP Address, so maybe in the past was used for spam...by the way, junghanns is the only domain refusing my mail If somebody else could ask to junghanns. Andrea Sergio Garcia Murillo [EMAIL PROTECTED]To Sent by: 'Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion' [EMAIL PROTECTED] asterisk-users@lists.digium.com, m.com 'Jeroen Zwarts' [EMAIL PROTECTED] cc 10/02/2006 09.51 [EMAIL PROTECTED] .com Subject Please respond to RE: [Asterisk-Users] Corrupt CDR Asterisk Users records in Asterisk 1.2.x Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I have an [EMAIL PROTECTED] installed and then upgraded to 1.2.4 with bristuff and i have the same problem. If you see the logs the INSERT trace has wrong values before the comand is executed. By the way, everyone of us that have this problem use HFC cards? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: jueves, 09 de febrero de 2006 18:28 Para: Jeroen Zwarts; Asterisk Users Mailing List - Non-Commercial Discussion CC: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x You are in my same situation. I thought I solved the problem (if you look at tomorrow post) but it isn't My situation is a bit different: I have the last bristuffed version of asterisk 1.2.4 (released yesterday) And I also have 2 zaphfc cards. but the behaviour is absolutely the same If you restart asterisk, you get one or two calls ok, the again the problem On the first zaphfc, the problem is almost immediate (1 or two calls) the second is stronger, and is ok for a longer period ( 1 day ??) then it also falls in problem on clid and src It seems to me some buffer overwrite problem. the clid is trasmitted ok to the internal phones. So I am not alone on this side... Andrea Jeroen Zwarts [EMAIL PROTECTED] nl To Sent by: asterisk-users@lists.digium.com asterisk-users-bo cc [EMAIL PROTECTED] m.com Subject [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x 09/02/2006 11.05 Please respond to Jeroen Zwarts [EMAIL PROTECTED] nl; Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I have a problem with CDR recording in Asterisk 1.2.x. This is the situation: An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single HFC-S ISDN BRI card. I log the call records to both the Master.csv and MySQL. The problem is that when an incoming call from the ISDN line is logged to the CDR, the src and the clid field show up as something like 'h? ' (random weird ASCII characters). This is in the MySQL table as well as the Master.csv, so my guess is that it is not a MySQL problem. Furthermore, I don't think it is a zaptel/bristuff
RE: [Asterisk-Users] attended call transfer
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: 10 February 2006 01:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] attended call transfer this is a Normal behaviour, nevertheless i dont think is a correct behaviour. Several weeks ago other user asked the same, i suggested him to open a feature request on bugs.digium.com, check for that regards Hi Yes that was me, this is still a big issue for us. Unfortunately we only have 1.2.1 installed on our live / dev boxes at the moment and when I registered an account on the bug tracker and read the rules it said you must have tested the issue on the very latest CVS head. I have been up to my eye balls the last couple of weeks so haven't had time to do this. I didn't want to raise this as a feature request as in my opinion this has to be a defect as attended transfer is basically unusable for a commercial environment (unless there exists a business that doesn't have a problem cutting off its customers :P ) HTH Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Obtaining billsecs in the dialplan after a call?
Hi,
I'm stuck on a silly thing. I need to get the billsec CDR value after a
call. But I'm finding its always 0.
Here's my test code:
exten = *244*,1,Dial(Local/[EMAIL PROTECTED]/n,,g)
exten = *244*,n,Noop(after dial duration is ${CDR(duration)} billsec is
${CDR(billsec)})
exten = *244*,n,Hangup
[custom-tests]
exten = test,1,Answer
exten = test,n,Playback(tt-somethingwrong)
exten = test,n,Hangup
The actual CDR record that gets posted in Master.csv looks like so:
,200,*244*,default,Exten 200 200,SIP/200-94dd,Local/[EMAIL
PROTECTED],1,Hangup,,2006-02-10
11:57:42,2006-02-10 11:57:42,2006-02-10
11:57:45,3,3,ANSWERED,DOCUMENTATION
So the duration is there just fine. But ${CDR(billsec)} remains stubbonly
0.
Now I don't really understand the CDR code 100% - but it looks like
billsec is only worked out then the cdr is posted. But there is no way to
force the cdr to be posted from the dialplan, is there?
Thanks,
Steve
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Re: [Asterisk-Users] Problem win Unicall
Try to change the value of protocolvariant in the
unicall.conf. Please, send us here the result.
Darlon Ferreira BortoliniRede/DesenvolvimentoBetha
SistemasFone (48) 3431-0750/Ramal 1000- Original Message
-
From:
Carlos
Chavez
To: Asterisk
Sent: Friday, February 10, 2006 1:57
AM
Subject: [Asterisk-Users] Problem win
Unicall
I am having a strange problem with an
asterisk servier using R2 Unicallin Mexico. Most calls go through
fine but some of them give me an error likethis:
-- Executing Dial("SIP/86-db41", "Unicall/g2/014448343600") in new
stack -- Called g2/014448343600Feb 9 21:44:39
WARNING[23069]: chan_unicall.c:2644 handle_uc_event: Unicall/2event
DialingFeb 9 21:44:45 WARNING[23069]: chan_unicall.c:2644
handle_uc_event: Unicall/2event Protocol failure --
Unicall/2 protocol error. Cause 32769Feb 9 21:44:45 WARNING[23069]:
app_dial.c:705 wait_for_answer: Unable toforward voiceFeb 9
21:44:45 WARNING[23069]: app_dial.c:705 wait_for_answer: Unable toforward
voice -- Hungup 'UniCall/2-1' == Everyone is
busy/congested at this time (1:0/0/1) This
particular call is national long distance. But I have seen
theproblem with some local numbers. I even had a problem dialing a
company inthe same city, their main numbers gave this error but their fax
number wentthrough without problem. I am
using Asterisk 1.2.4 (upgraded from 1.2.3 this morning), spandsp.21,
unicall 0.0.3. Any ideas? I am using a TE110P card with Zaptel
1.2.3with 10 channels from Telmex.--Carlos ChavezDirector
de TecnologíaTelecomunicaciones Abiertas de México S.A. de C.V.Tel:
+52-55-91169161 Ext
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[Asterisk-Users] QSIG error -- can somebody explain?
Hi all, I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the outside world and should forward our calls to the telco. This setup works correctly as far as I use euroisdn as the switchtype. The first problem was that it is only possible to run the * side in CPE-mode -- I wanted NET. Anyway, I configured * this way: switchtype=qsig signalling = bri_cpe facilityenable = yes My experience now is that it is possible to signal a call (both outgoing and incoming) but as soon as the callee takes off the hook the call-setup crashes. Below is the debug log of an outgoing call to a service number of the telco which tells the current time. (The point is that the called number immediately answers the call.) As you can see the Alcatel side answers to our SETUP message with a RELEASE COMPLETE and a cause number 100. This cause (taken from ECMA-143) means: Invalid information element contents , | This cause indicates that the equipment sending this cause has received an | information element which it has implemented; however, one or more of the fields | in the information element are coded in a way that has not been implemented by | the equipment sending this cause. ` Can somebody explain what the problem is? Configuration error, a bug, a problem on the Alcatel-side? Thanks in advance, Wolfgang -- Executing Dial(SIP/1993-567b, Zap/g1/006621503|55|j) in new stack 1 -- Making new call for cr 136 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (8) len=32 1 Call Ref: len= 1 (reference 8/0x8) (Originator) 1 Message type: SETUP (5) 1 [1 041 031 801 901 a31 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [1 181 011 891 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 6c1 061 211 801 311 391 391 331 ] 1 Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number not screened (0) '1993' ] 1 [1 701 0a1 c11 301 301 361 361 321 311 351 301 331 ] 1 Called Number (len=12) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '006621503' ] -- Called g1/006621503 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 8/0x8) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 811 e41 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: (null) (100), class = Protocol Error (6) ] 1 -- Making new call for cr 32776 1 -- Processing IE 8 (cs0, Cause) 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null 1 No response to SETUP message 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending 1 Protocol Discriminator: Q.931 (8) len=8 1 Call Ref: len= 1 (reference 8/0x8) (Originator) 1 Message type: DISCONNECT (69) 1 [1 081 021 811 921 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Unknown (18), class = Normal Event (1) ] -- Channel 0/1, span 1 got hangup, cause 42 -- Zap/1-1 is circuit-busy 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Request, peerstate Disconnect Indication -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Busy(SIP/1993-567b, ) in new stack == Spawn extension (dialout, 436621503, 102) exited non-zero on 'SIP/1993-567b' 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 8/0x8) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 811 d11 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Unknown (81), class = Invalid message (5) ] 1 -- Making new call for cr 32776 1 -- Processing IE 8 (cs0, Cause) 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null ___ --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] STUPID question? Tellabs echo can cards and PSTN?
Martin Joseph wrote: I am wondering if the instructions for hard wiring a Tellabs canceler are applicable to a regular old two wire loop? Or is this only something that works for people with T1? Any comments from people that have tried this are appreciated. T1 card is necessary. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] 2wav2mp3, monitor, mixmonitor, mpg123, queues
Answer for the question number 1:
Use it:
exten=,1,Macro(ramais-gravados,SIP/${EXTEN})
[macro-ramais-gravados]exten=s,1,SetVar(CALLFILENAME=i${CALLERIDNUM}-${TIMESTAMP})exten=s,2,Monitor(wav,${CALLFILENAME},m)exten=s,3,Dial(${ARG1},20,Ttr)
exten=s,4,Hangup
This script was changed
2wav2mp3
#!/bin/sh# create stereo mp3 out of two mono
wav-files# source files will be deleted## 2005 05 23 dietmar
zlabinger http://www.zlabinger.at/asterisk##
usage: 2wav2mp3 wave1 wave2 mp3# designed for
Asterisk Monitor(file,format,option) where option is "e" and# the variable
# MONITOR_EXEC/usr/bin/2wav2mp3
# location of SOX and SOXMIX# (set according to your system
settings, eg. /usr/bin)SOX=/usr/bin/soxSOXMIX=/usr/bin/soxmix#lame
is only required when sox does not support liblameLAME=/usr/bin/lame
# command line variablesLEFT="$1"RIGHT="$2"OUT="$3"
#test if input files existtest ! -r $LEFT exittest ! -r
$RIGHT exit
# convert mono to stereo, adjust balance to -1/1# left channel$SOX
-c 1 $LEFT $LEFT-tmp.wav pan -1# right channel$SOX -c 1 $RIGHT
$RIGHT-tmp.wav pan 1
# combine and compress# this requires sox to be built with
mp3-support.# To see if there is support for
Mp3 run sox -h and # look for it under the list of supported file
formats as "mp3".# $SOXMIX -v 1 $LEFT-tmp.wav -v 1 $RIGHT-tmp.wav -v 1
$OUT.mp3
# in case and old version of sox is used, the lame-encoding# can be
done afterwards$SOXMIX -v 0.5 $LEFT-tmp.wav $RIGHT-tmp.wav $OUT
echo $OUT final.datFINAL=`cat final.dat | sed 's/wav/mp3/g'`
$LAME --silent -V7 -B24 --tt $OUT --add-id3v2 $OUT $FINAL
#remove temporary filestest -w $LEFT-tmp.wav rm
$LEFT-tmp.wavtest -w $RIGHT-tmp.wav rm $RIGHT-tmp.wavtest -w
$OUT rm $OUT
#remove input files if successfull#test -r $OUT.mp3 rm $LEFT
$RIGHTtest -r $FINAL rm $LEFT $RIGHTrm -f final.dat
Darlon Ferreira BortoliniRede/DesenvolvimentoBetha
SistemasFone (48) 3431-0750/Ramal 1000- Original Message
-
From:
[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, February 10, 2006 7:13
AM
Subject: [Asterisk-Users] 2wav2mp3,
monitor, mixmonitor, mpg123, queues
Hello!I'm using Asterisk for our office
telephony, but we have some problemsthat still we can't resolve about it.
Here they are:1) merge in/out call recording filesI also tried
to use a script I found on the internet, called 2wav2mp3In extensions.conf
I added the following lines; script to be executed when monitoring has
been finishedMONITOR_EXEC=/usr/local/bin/2wav2mp3exten =
102,1,SetVar(CALLFILENAME=${TIMESTAMP}-${EXTEN}-${CALLERID})exten =
102,2,Monitor(wav,${CALLFILENAME},m);exten =
102,2,MixMonitor(${CALLFILENAME}.gsm);exten =
102,2,MixMonitor(test.wav,W(-3))exten = 102,3,Ringingexten =
102,4,Dial(Sip/giuseppedd,20,rtwW)...but I always get two separate
files.As you can see I also tried the MixMonitor application but the
resulting filescontain one channel that is clearly audible and the
other seems to be noise.2) an alternative to mpg123 becouse it
generates a lot of errors like this:Feb 3 19:50:08
WARNING[9568]: res_musiconhold.c:488 monmp3thread: Unable to spawn
mp3playerFeb 3 19:58:28 NOTICE[9568]: res_musiconhold.c:507
monmp3thread: Request to schedule in the past?!?!Feb 3 19:58:28
WARNING[9568]: res_musiconhold.c:421 spawn_mp3: Found no files in
'/usr/share/asterisk/mohagents'3) how to play different files to
an agent before he picks up a call depending on which queue the call comes
from[qlu500]musiconhold = qlu500announce =
vm-from-phonenumber ; here is the problemcontext =
qlu500outwrapuptime=15announce-frequency =
60...Comments or suggestions are greatly
appreciated.Thanks a
lot.Giuseppe___--Bandwidth
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[EMAIL PROTECTED]: Re: [EMAIL PROTECTED]: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x]]
Kapejod is working on a fix for the CDR problem in bristuff. See below - Forwarded message from [EMAIL PROTECTED] - Resent-From: [EMAIL PROTECTED] Resent-Date: Fri, 10 Feb 2006 13:01:25 +0200 Resent-Message-ID: [EMAIL PROTECTED] Resent-To: [EMAIL PROTECTED] Envelope-to: [EMAIL PROTECTED] Delivery-date: Fri, 10 Feb 2006 05:19:50 -0500 Date: Fri, 10 Feb 2006 11:19:47 +0100 (CET) Subject: Re: [EMAIL PROTECTED]: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x] From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Hi, i have had reports about the CDR corruption from various sources. It is a bug introduced by BRIstuff (shame on me) and i am currently working on a fix. The mail server rejecting Andrea's emails is one of Germany's largest ISPs which we use as a proxy. So unfortunately we dont have an influence on that. In any case people can usually find me on MSN ([EMAIL PROTECTED]) to report BRIstuff bugs (no matter if they use our hardware or not). Can you please forward this information to the users list? (I unsubscribed months ago since i couldnt handle the mail volume.) best regards Klaus As seen on asterisk-users... And generally, where should people without Junghanns hardware and systems (plain zaphfc users) turn to? - Forwarded message from [EMAIL PROTECTED] - Envelope-to: [EMAIL PROTECTED] Delivery-date: Fri, 10 Feb 2006 04:32:49 -0500 Resent-From: [EMAIL PROTECTED] Resent-Date: Fri, 10 Feb 2006 11:32:00 +0200 Resent-Message-ID: [EMAIL PROTECTED] Resent-To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x To: asterisk-users@lists.digium.com From: [EMAIL PROTECTED] Date: Fri, 10 Feb 2006 10:22:19 +0100 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Yes, everybody of us use zaphfc. No problem at all with zap channel that I have installed in several other * Box (PRI E1, Quad Analog, Chan_capi on 1.0 * series) So the problem, I think, is the zaphfc, or the patch to the * and zaptel provided by the bristuff. I tried to post a question to [EMAIL PROTECTED] but the mail was refused with a status code of 550 5.0.0, Dial-Up IP address rejected The public ip address I am using is from a newly buyed (3 days ago) set of 8 IP Address, so maybe in the past was used for spam...by the way, junghanns is the only domain refusing my mail If somebody else could ask to junghanns. Andrea Sergio Garcia Murillo [EMAIL PROTECTED] To Sent by: 'Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion' [EMAIL PROTECTED] asterisk-users@lists.digium.com, m.com 'Jeroen Zwarts' [EMAIL PROTECTED] cc 10/02/2006 09.51 [EMAIL PROTECTED] .com Subject Please respond to RE: [Asterisk-Users] Corrupt CDR Asterisk Users records in Asterisk 1.2.x Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I have an [EMAIL PROTECTED] installed and then upgraded to 1.2.4 with bristuff and i have the same problem. If you see the logs the INSERT trace has wrong values before the comand is executed. By the way, everyone of us that have this problem use HFC cards? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: jueves, 09 de febrero de 2006 18:28 Para: Jeroen Zwarts; Asterisk Users Mailing List - Non-Commercial Discussion CC: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x You are in my same situation. I thought I solved the problem (if you look at tomorrow post) but it isn't My situation is a bit different: I have the last bristuffed version of asterisk 1.2.4 (released yesterday) And I also have 2 zaphfc cards. but the behaviour is absolutely the same If you restart asterisk, you get one or two calls ok, the again the problem On the first zaphfc, the problem is almost immediate (1 or two calls) the second is stronger, and is ok for a longer period ( 1 day ??) then it also falls in problem on clid and src It seems to me some buffer overwrite problem. the clid is trasmitted ok to the internal phones. So I am not alone on this side... Andrea Jeroen Zwarts [EMAIL PROTECTED] nl To Sent by: asterisk-users@lists.digium.com asterisk-users-bo cc [EMAIL PROTECTED] m.com Subject [Asterisk-Users]
[Asterisk-Users] Sip + Cisco 7940/7960 + Panel + DND + queues
Hi all,
Running bristuffed 1.2.4 system with solely Cisco 7940/7960 phones
with SIP.
I'm using also op_panel 0.25 (snapshot).
I'm using * queues.
I want to properly implement DND via *78 and *79.
I'm using op_panel's documentation RECIPE 1 solution with astdb and dnd
variables and this is fine for FOP.
The DND works in normal cases, since I catch it with my Macro dialsip,
HOWEVER
The Queue handling does not look at my astdb DND settings... so the
Phones that are member of a queue keep ringing, even after pressing
*78 and being in DND state.
Any solution ?
The only way 'around' that I see is to Dynamically Add and Remove the
members from the queue when people set DND on/off.
I do not like this, since the 'knowledge' with which queue members are
related, is defined twice (in the Dialplan and in queues.conf)
Is there a way to put a Sip phone from the dialplan in state 'Busy' ?
(= alternative work around)
Is there a way the let 'Queue' use my Macro for Dialing the members ?
Txs
Alex
my configs (or part of it)
queues.conf:
[support]
musiconhold = default
strategy = ringall
member = Sip/301
member = Sip/311
member = Sip/302
member = Sip/310
member = Sip/315
...
[sales]
musiconhold = default
strategy = ringall
member = Sip/300
member = Sip/307
member = Sip/304
member = Sip/308
member = Sip/313
...
parts of extensions.conf
[macro-dialsip]
; DND
exten = s,1,Set(value=${DB(dnd/SIP/${ARG1})})
exten = s,n,GotoIf(${value}?dnd:nodnd)
exten = s,n(nodnd),NoOp
exten = s,n,Dial(Sip/${ARG1},60)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s,n(dnd),NoOp
exten = s,n,Playtones(busy)
exten = s,n,Wait(30)
exten = s,n,Hangup
exten = s-BUSY,1,Playtones(busy)
exten = s-BUSY,n,Wait(10)
exten = s-BUSY,n,Goto(mainmenu,s,1)
exten = s-NOANSWER,1,Goto(mainmenu,s,1)
exten = _s-.,1,Playtones(congestion)
exten = _s-.,n,Wait(10)
exten = _s-.,n,Goto(s-NOANSWER,1)
[apps]
...
; DND On
exten = *78,1,UserEvent(ASTDB|Family: dnd^State: On)
exten = *78,n,Set(temp=${CHANNEL})
exten = *78,n,Cut(temp=temp,,1)
exten = *78,n,Set(DB(dnd/${temp})=On)
exten = *78,n,Playback(dnd-aan)
exten = *78,n,Wait(10)
exten = *78,n,Hangup
; DND Off
exten = *79,1,UserEvent(ASTDB|Family: dnd^State: ^)
exten = *79,n,Set(temp=${CHANNEL})
exten = *79,n,Cut(temp=temp,,1)
exten = *79,n,DBDel(dnd/${temp})
exten = *79,n,Playback(dnd-uit)
exten = *79,n,Wait(10)
exten = *79,n,Hangup
...
part of the IVR:
[mainmenu]
exten = s,1,Background(hoofdmenu)
exten = s,n,Goto(s,1)
; Operator / Admin
exten = 1,1,Goto(menu-admin,s,1)
exten = 2,1,Goto(menu-support,s,1)
exten = 3,1,Goto(menu-sales,s,1)
exten = 4,1,Goto(s,1)
; invalid
exten = i,1,Playback(menu-verkeerd)
exten = i,n,Goto(s,1)
; timeout lang gewacht
exten = t,1,Goto(s,1)
[menu-admin]
; hoofdnummer
exten = s,1,Dial(Sip/300,30,t)
exten = s,n,Queue(sales|t|||180)
exten = s,n,Queue(support|t|||60)
exten = s,n,Queue(development|t|||60)
exten = s,n,Goto(mainmenu,s,1)
[menu-sales]
exten = s,1,Queue(sales|t|||180)
exten = s,n,Queue(support|t|||60)
exten = s,n,Queue(development|t|||60)
exten = s,n,Goto(mainmenu,s,1)
[menu-support]
exten = s,1,Queue(support|t|||180)
exten = s,n,Queue(development|t|||60)
exten = s,n,Queue(sales|t|||60)
exten = s,n,Goto(mainmenu,s,1)
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Re: [Asterisk-Users] TE210P + MicroITX as E1 to TDMoE appliance?
James Harper schrieb: Has anyone every attempted to set up a mini PC to achieve much the same functionality as the fonebridge box? The sort of thing I'm imagining is a micro itx board case in a completely solid state configuration (flash disk, maybe a psu fan but only if really required), with a TE210P (or equiv) card(s). The sole purpose of this would be to be a bridge between E1 lines and TDMoE. I think that is quite exactly what the fonebridge is. Christian ___ --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] Issues in Australia? Ringing, iaxy etc
There are quite a few Asterisk systems running in Australia, so you should be fine PaulH Melbourne - Original Message - From: Chris Earle (CBL) [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, February 10, 2006 3:31 AM Subject: [Asterisk-Users] Issues in Australia? Ringing, iaxy etc Hi all, getting a server going wiht a few TDM400's and some phones, and some IAXys too I haven't heard any issues about AU phones being able to RING in Australia, like the problem in the UK with ring capacitors on the BT system. Are there any problems like that? Also, with the iaxy's -- they should work (and ring) in Australia right? The only hint I'm seeing around is the use of notransfer=yes in the iax.conf for the iaxy entry Basically, just hoping for a smooth transition over to the asterisk system Cheers -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Corrupt CDR records in Asterisk 1.2.x
I will try to e-mail Junghanns with this problem. I'll put in a link to the Google Groups thread that archives this mail. In the meanwhile I have an idea for a workaround: Create an AGI script that runs in the hangup extension, reads the telephonenumber and the unique ID from the AGI variables and updates the mysql CDR table with the correct telephone number. I don't know if it's going to work, but that is what I came up with on the way to my work this morning. I'll keep you posted about the progress, and if I receive an e-mail back from junghanns. Jeroen - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, February 10, 2006 10:22 AM Subject: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x Yes, everybody of us use zaphfc. No problem at all with zap channel that I have installed in several other * Box (PRI E1, Quad Analog, Chan_capi on 1.0 * series) So the problem, I think, is the zaphfc, or the patch to the * and zaptel provided by the bristuff. I tried to post a question to [EMAIL PROTECTED] but the mail was refused with a status code of 550 5.0.0, Dial-Up IP address rejected The public ip address I am using is from a newly buyed (3 days ago) set of 8 IP Address, so maybe in the past was used for spam...by the way, junghanns is the only domain refusing my mail If somebody else could ask to junghanns. Andrea Sergio Garcia Murillo [EMAIL PROTECTED]To Sent by: 'Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion' [EMAIL PROTECTED] asterisk-users@lists.digium.com, m.com 'Jeroen Zwarts' [EMAIL PROTECTED] cc 10/02/2006 09.51 [EMAIL PROTECTED] .com Subject Please respond to RE: [Asterisk-Users] Corrupt CDR Asterisk Users records in Asterisk 1.2.x Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I have an [EMAIL PROTECTED] installed and then upgraded to 1.2.4 with bristuff and i have the same problem. If you see the logs the INSERT trace has wrong values before the comand is executed. By the way, everyone of us that have this problem use HFC cards? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: jueves, 09 de febrero de 2006 18:28 Para: Jeroen Zwarts; Asterisk Users Mailing List - Non-Commercial Discussion CC: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x You are in my same situation. I thought I solved the problem (if you look at tomorrow post) but it isn't My situation is a bit different: I have the last bristuffed version of asterisk 1.2.4 (released yesterday) And I also have 2 zaphfc cards. but the behaviour is absolutely the same If you restart asterisk, you get one or two calls ok, the again the problem On the first zaphfc, the problem is almost immediate (1 or two calls) the second is stronger, and is ok for a longer period ( 1 day ??) then it also falls in problem on clid and src It seems to me some buffer overwrite problem. the clid is trasmitted ok to the internal phones. So I am not alone on this side... Andrea Jeroen Zwarts [EMAIL PROTECTED] nl To Sent by: asterisk-users@lists.digium.com asterisk-users-bo cc [EMAIL PROTECTED] m.com Subject [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x 09/02/2006 11.05 Please respond to Jeroen Zwarts [EMAIL PROTECTED] nl; Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I have a problem with CDR recording in Asterisk 1.2.x. This is the situation: An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single HFC-S ISDN BRI card. I log the call records to both the Master.csv and MySQL. The problem is that when an incoming call from the ISDN line is logged to the CDR, the src and the
Re: [Asterisk-Users] Obtaining billsecs in the dialplan after a call?
Hi,
You can use the ANSWEREDTIME variable :
exten = *244*,n,Noop(after dial duration is ${ANSWEREDTIME})
Regards,
Olivier
-
http://www.olivier-perrin.net
Le vendredi 10 février 2006 à 12:19 +0200, [EMAIL PROTECTED] a écrit :
Hi,
I'm stuck on a silly thing. I need to get the billsec CDR value after a
call. But I'm finding its always 0.
Here's my test code:
exten = *244*,1,Dial(Local/[EMAIL PROTECTED]/n,,g)
exten = *244*,n,Noop(after dial duration is ${CDR(duration)} billsec is
${CDR(billsec)})
exten = *244*,n,Hangup
[custom-tests]
exten = test,1,Answer
exten = test,n,Playback(tt-somethingwrong)
exten = test,n,Hangup
The actual CDR record that gets posted in Master.csv looks like so:
,200,*244*,default,Exten 200 200,SIP/200-94dd,Local/[EMAIL
PROTECTED],1,Hangup,,2006-02-10
11:57:42,2006-02-10 11:57:42,2006-02-10
11:57:45,3,3,ANSWERED,DOCUMENTATION
So the duration is there just fine. But ${CDR(billsec)} remains stubbonly
0.
Now I don't really understand the CDR code 100% - but it looks like
billsec is only worked out then the cdr is posted. But there is no way to
force the cdr to be posted from the dialplan, is there?
Thanks,
Steve
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Re: [Asterisk-Users] QSIG error -- can somebody explain?
I can only guess, but I think I can remember that the creflen needs to be 2 octets for qsig. Check what the Alcatel switch sends in the setup message to *. Anyway, why do use QSIG ? Does name display work on the * implementation ? Best regards Hans P.S.: Schoene Gruesse an Kurt Krenn Wolfgang Zweimueller schrieb: Hi all, I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the outside world and should forward our calls to the telco. This setup works correctly as far as I use euroisdn as the switchtype. The first problem was that it is only possible to run the * side in CPE-mode -- I wanted NET. Anyway, I configured * this way: switchtype=qsig signalling = bri_cpe facilityenable = yes My experience now is that it is possible to signal a call (both outgoing and incoming) but as soon as the callee takes off the hook the call-setup crashes. Below is the debug log of an outgoing call to a service number of the telco which tells the current time. (The point is that the called number immediately answers the call.) As you can see the Alcatel side answers to our SETUP message with a RELEASE COMPLETE and a cause number 100. This cause (taken from ECMA-143) means: Invalid information element contents , | This cause indicates that the equipment sending this cause has received an | information element which it has implemented; however, one or more of the fields | in the information element are coded in a way that has not been implemented by | the equipment sending this cause. ` Can somebody explain what the problem is? Configuration error, a bug, a problem on the Alcatel-side? Thanks in advance, Wolfgang -- Executing Dial(SIP/1993-567b, Zap/g1/006621503|55|j) in new stack 1 -- Making new call for cr 136 -- Requested transfer capability: 0x00 - SPEECH 1 Protocol Discriminator: Q.931 (8) len=32 1 Call Ref: len= 1 (reference 8/0x8) (Originator) 1 Message type: SETUP (5) 1 [1 041 031 801 901 a31 ] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [1 181 011 891 ] 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit, Other Spare: 0, Exclusive Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [1 6c1 061 211 801 311 391 391 331 ] 1 Calling Number (len= 8) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number not screened (0) '1993' ] 1 [1 701 0a1 c11 301 301 361 361 321 311 351 301 331 ] 1 Called Number (len=12) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '006621503' ] -- Called g1/006621503 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 8/0x8) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 811 e41 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: (null) (100), class = Protocol Error (6) ] 1 -- Making new call for cr 32776 1 -- Processing IE 8 (cs0, Cause) 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null 1 No response to SETUP message 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending 1 Protocol Discriminator: Q.931 (8) len=8 1 Call Ref: len= 1 (reference 8/0x8) (Originator) 1 Message type: DISCONNECT (69) 1 [1 081 021 811 921 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Unknown (18), class = Normal Event (1) ] -- Channel 0/1, span 1 got hangup, cause 42 -- Zap/1-1 is circuit-busy 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Request, peerstate Disconnect Indication -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Busy(SIP/1993-567b, ) in new stack == Spawn extension (dialout, 436621503, 102) exited non-zero on 'SIP/1993-567b' 1 Protocol Discriminator: Q.931 (8) len=9 1 Call Ref: len= 2 (reference 8/0x8) (Terminator) 1 Message type: RELEASE COMPLETE (90) 1 [1 081 021 811 d11 ] 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 1 Ext: 1 Cause: Unknown (81), class = Invalid message (5) ] 1 -- Making new call for cr 32776 1 -- Processing IE 8 (cs0, Cause) 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null,
[Asterisk-Users] Double ring
Title: Double ring Can anyone shed any light on to why I get a double ring when calling external numbers? When calling out I hear the actually ring-ring of the called phone and the asterisk ring tone. I'm using the same config I used with 1.0.10 but have now upgraded to 1.2.4. Regards Lee ___ --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] Double ring
I was getting something very similar with my Aastra test phones until I change 'callprogress=' to 'no'. Thanks, Bob On Fri, 10 Feb 2006 12:13:47 - Lee Archer [EMAIL PROTECTED] wrote: Can anyone shed any light on to why I get a double ring when calling external numbers? When calling out I hear the actually ring-ring of the called phone and the asterisk ring tone. I'm using the same config I used with 1.0.10 but have now upgraded to 1.2.4. Regards Lee ___ --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 vs. Traditional PBX
I am sorry. I thought you wrote Dell 850. Should have looked closer. The machine should do just fine. However it would not hurt to ptu in another gig. Also see if anyone else on the list has used a 650 and what expiriences they have had. Regards, Dovid --- Nora Lavelle [EMAIL PROTECTED] wrote: Hi Dovid, Thank you for the book. I'm already reading it. I have a dell 650 server, 1Gig of memory, 1 CPU (3.07Ghz). What hardware would you recommend for the 200 users w/ about 20 concurrent calls ? As always I thank you so much for your help. Nora Lavelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 09, 2006 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk vs. Traditional PBX I think your problem is the Dell 650. What are the specs on it ? If you want a system that can support 200 users you will need to do a lot better than that. Also you will be dealing with T1's/E1's and not POTS lines. I think a good place to start (if you havent already) is the book that has come out a while back. I have it on my server at http://www.h6315.com/ast_book/ Regards, Dovid (I posted my server and not from the publisher becuase I do not know thier URL and I have email access only now.) --- Nora Lavelle [EMAIL PROTECTED] wrote: Hi everyone ! So here's my question of the day ! I need to make a decision on whether or not to go to a voip solution or configure an existing pbx (norstar) that my company has available. We are a small startup. I'm wanting a solution that will support up to about 200 people, with direct dial-in capability, up to about 30 concurrent phone calls and good voice quality. Right now I have an asterisk deployment with about 15 people on it. We have sipura 841 phones. The biggest issue currently is voice quality. lot of complaints there. I have a dell 650 poweredge (single processory system), with a digium tdm400 card and 4 analog lines plugged into it. So here are my questions: * Is asterisk a good solution for my company ? or should I just install the traditional pbx and look to move to asterisk in a couple of years ? (I personally would prefer asterisk cuz I'm a unix person not a phone person so from a manageability perspective i would love this ) * If I were to go to an asterisk solution to support about 200 people with the requirements above what hardware platform would you recommend ? I'm guessing I'd need a PRI line and a different digium card? Also would a 1cpu poweredge dell be enough ? or would that have to be upgraded too ? If anyone is running an environment similar to this that can provide help I would really appreciate this. I'm having a hard time making this decision and would love to hear anybody's experience in a real time environment. Thanks again this list ROCKS! Nora Lavelle ___ --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 __ 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 ___ --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 __ 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] Tormenta CAS signaling
Hello Can anyone know how may change(inverting) cas signaling ABCD bits at the Tormenta 2 (four E1 ports) cards My cards send idle code ABCD 0101 but my mux which use as channel bank wait ABCD 1001 Best Regards Viktor [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] Double ring
Oddly enough I'm on Aastra phones too. Doesn't happen with Grandstream phones. I've tried callprogram=yes and no to no effect. What firmware did you have, I'm on 1.3. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: 10 February 2006 12:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double ring I was getting something very similar with my Aastra test phones until I change 'callprogress=' to 'no'. Thanks, Bob On Fri, 10 Feb 2006 12:13:47 - Lee Archer [EMAIL PROTECTED] wrote: Can anyone shed any light on to why I get a double ring when calling external numbers? When calling out I hear the actually ring-ring of the called phone and the asterisk ring tone. I'm using the same config I used with 1.0.10 but have now upgraded to 1.2.4. Regards Lee ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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] Double ring
'callprogress', in zapata.conf: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf Thanks, Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: Friday, February 10, 2006 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double ring Oddly enough I'm on Aastra phones too. Doesn't happen with Grandstream phones. I've tried callprogram=yes and no to no effect. What firmware did you have, I'm on 1.3. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: 10 February 2006 12:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double ring I was getting something very similar with my Aastra test phones until I change 'callprogress=' to 'no'. Thanks, Bob On Fri, 10 Feb 2006 12:13:47 - Lee Archer [EMAIL PROTECTED] wrote: Can anyone shed any light on to why I get a double ring when calling external numbers? When calling out I hear the actually ring-ring of the called phone and the asterisk ring tone. I'm using the same config I used with 1.0.10 but have now upgraded to 1.2.4. Regards Lee ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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
[Asterisk-Users] cdr (again) and deadlocks
Hello, Today I had again problems with CDR. My MySQL cdr table was corrupted and thus CDR couldn't be logged. At this moment Asterisk console started to display the following message "Avoided deadlock for '0x843fa98', 10 retries!" hundreds, thousands of times (together with the table corrupted message), until it simply displayed a "Terminated" message and went down. I had to repair the MySQL table, and then restart Asterisk. The table corrupted message was useful for me to identify the corrupted table and repair it... but wouldn't it be possible that Asterisk would not "Terminate" because of this? Thank you Dov ___ --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] Tormenta CAS signaling
Viktor Tatianin wrote: Hello Can anyone know how may change(inverting) cas signaling ABCD bits at the Tormenta 2 (four E1 ports) cards My cards send idle code ABCD 0101 but my mux which use as channel bank wait ABCD 1001 The idle code is set in zapata.conf. For example: cas=1-15:1101 Sets CAS mode for channels 1 to 15, with the idle pattern 1101. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem win Unicall
Darlon wrote: Try to change the value of protocolvariant in the unicall.conf. Please, send us here the result. The answer to R2 problems is not stumbling in the dark, randomly trying variants. If you are in country X, the required variant is unlikely to be anything but X. If you are expecting 7 digits, the required number of expected digits is unlikely to be anything other than 7, etc. Regards, Steve *Darlon Ferreira Bortolini* Rede/Desenvolvimento Betha Sistemas Fone (48) 3431-0750/Ramal 1000 - Original Message - *From:* Carlos Chavez mailto:[EMAIL PROTECTED] *To:* Asterisk mailto:asterisk-users@lists.digium.com *Sent:* Friday, February 10, 2006 1:57 AM *Subject:* [Asterisk-Users] Problem win Unicall I am having a strange problem with an asterisk servier using R2 Unicall in Mexico. Most calls go through fine but some of them give me an error like this: If only a few calls do this, there may be something special happening on the line for those calls. Perhaps an unexpected category code. R2 varies quite a bit. The log you provided doesn't say much that is useful, as the logging level is too low. In unicall.conf put the line loglevel=255 somewhere before the channels are defined. Then run the system, and send me the complete log of a problem call. From that, I should be able to tell you what is going wrong. Regards, Steve -- Executing Dial(SIP/86-db41, Unicall/g2/014448343600) in new stack -- Called g2/014448343600 Feb 9 21:44:39 WARNING[23069]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Dialing Feb 9 21:44:45 WARNING[23069]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Protocol failure -- Unicall/2 protocol error. Cause 32769 Feb 9 21:44:45 WARNING[23069]: app_dial.c:705 wait_for_answer: Unable to forward voice Feb 9 21:44:45 WARNING[23069]: app_dial.c:705 wait_for_answer: Unable to forward voice -- Hungup 'UniCall/2-1' == Everyone is busy/congested at this time (1:0/0/1) This particular call is national long distance. But I have seen the problem with some local numbers. I even had a problem dialing a company in the same city, their main numbers gave this error but their fax number went through without problem. I am using Asterisk 1.2.4 (upgraded from 1.2.3 this morning), spandsp .21, unicall 0.0.3. Any ideas? I am using a TE110P card with Zaptel 1.2.3 with 10 channels from Telmex. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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] Double ring
Sorry I meant callprogress. I've tried it set to yes and no with no difference. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: 10 February 2006 13:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double ring 'callprogress', in zapata.conf: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf Thanks, Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: Friday, February 10, 2006 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double ring Oddly enough I'm on Aastra phones too. Doesn't happen with Grandstream phones. I've tried callprogram=yes and no to no effect. What firmware did you have, I'm on 1.3. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: 10 February 2006 12:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double ring I was getting something very similar with my Aastra test phones until I change 'callprogress=' to 'no'. Thanks, Bob On Fri, 10 Feb 2006 12:13:47 - Lee Archer [EMAIL PROTECTED] wrote: Can anyone shed any light on to why I get a double ring when calling external numbers? When calling out I hear the actually ring-ring of the called phone and the asterisk ring tone. I'm using the same config I used with 1.0.10 but have now upgraded to 1.2.4. Regards Lee ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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] IP Authorization
Exten = 123,1,NoOp(${SIPCHANINFO(recvip)})
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sam Tam
Sent: Friday, February 10, 2006 3:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] IP Authorization
Ah that is from the CLI but still unclear about how to setup
the extension.conf or etc..
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alexander Lopez
Sent: Friday, February 10, 2006 1:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] IP Authorization
You can use the following:
switch3*CLI show function SIPCHANINFO
switch3*CLI
-= Info about function 'SIPCHANINFO' =-
[Syntax]
SIPCHANINFO(item)
[Synopsis]
Gets the specified SIP parameter from the current channel
[Description]
Valid items are:
- peeripThe IP address of the peer.
- recvipThe source IP address of the peer.
- from The URI from the From: header.
- uri The URI from the Contact: header.
- useragent The useragent.
- peername The name of the peer.
All the info you need is there.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Sam Tam
Sent: Thursday, February 09, 2006 9:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] IP Authorization
Can you be more detail about the setup?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Olle E
Johansson
Sent: Friday, February 10, 2006 4:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IP Authorization
Sam Tam wrote:
I think this is a question that has been discussed before.
But you see nowadays most carriers will provide thing like
SIP using
IP authorization rather than username and password and I am now
wondering whether Asterisk can do something like that or not?
In the voip channels as well as in manager you can set ACLs for the
connections you define.
/O
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[Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...
Hi, I thought I had this problem licked but there still is a rights problem with ARI and Asterisk when using a non-root user (Following the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-rootdiff2=25). When I issue the following: chmod --recursive u=rwX,g=rX,o= /var/spool/asterisk The above command results in the following rights on messages: msg.gsmrwxr-x--- asterisk msg.txtrw-r- asterisk msg.wavrwxr-x--- asterisk I can transfer voicemails and play them but new messages comming in get the following: msg.gsmrwx-- asterisk msg.txtrw-r--r-- asterisk msg.wavrwx-- asterisk After changing the rights a transferred messages has the folowing rights: msg.gsmrw-r- apache msg.txtrw-r- apache msg.wavrw-r- apache New voicemail cannot be played, deleted or transferred by the ARI application. Apache is belongs to the Asterisk group. I thought I understood SUID, GUID and sticky bit now I am not so sure. What is really confussing to me is why the rights on the .txt file do not match the other 2 after running the 'chmod --recursive ...' command. Any help here would be greatly appreciated. I am using the lastest versions of Asterisk 1.2.4 and Zaptel 1.2.3, etc. Thanks ___ --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] Double ring
I have the similar problem with thomson sip voip cable modems: http://bugs.digium.com/view.php?id=6083 On Fri, 2006-02-10 at 12:13 +, Lee Archer wrote: Can anyone shed any light on to why I get a double ring when calling external numbers? When calling out I hear the actually ring-ring of the called phone and the asterisk ring tone. I'm using the same config I used with 1.0.10 but have now upgraded to 1.2.4. Regards Lee ___ --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 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] Corrupt CDR records in Asterisk 1.2.x
Hi all bristuffers... Klaus knows about this as I was hassling him about it too! He is investigating. Rgds Tim Robinson Jeroen Zwarts wrote: I will try to e-mail Junghanns with this problem. I'll put in a link to the Google Groups thread that archives this mail. In the meanwhile I have an idea for a workaround: Create an AGI script that runs in the hangup extension, reads the telephonenumber and the unique ID from the AGI variables and updates the mysql CDR table with the correct telephone number. I don't know if it's going to work, but that is what I came up with on the way to my work this morning. I'll keep you posted about the progress, and if I receive an e-mail back from junghanns. Jeroen - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, February 10, 2006 10:22 AM Subject: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x Yes, everybody of us use zaphfc. No problem at all with zap channel that I have installed in several other * Box (PRI E1, Quad Analog, Chan_capi on 1.0 * series) So the problem, I think, is the zaphfc, or the patch to the * and zaptel provided by the bristuff. I tried to post a question to [EMAIL PROTECTED] but the mail was refused with a status code of 550 5.0.0, Dial-Up IP address rejected The public ip address I am using is from a newly buyed (3 days ago) set of 8 IP Address, so maybe in the past was used for spam...by the way, junghanns is the only domain refusing my mail If somebody else could ask to junghanns. Andrea Sergio Garcia Murillo [EMAIL PROTECTED]To Sent by: 'Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion' [EMAIL PROTECTED] asterisk-users@lists.digium.com, m.com 'Jeroen Zwarts' [EMAIL PROTECTED] cc 10/02/2006 09.51 [EMAIL PROTECTED] .com Subject Please respond to RE: [Asterisk-Users] Corrupt CDR Asterisk Users records in Asterisk 1.2.x Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I have an [EMAIL PROTECTED] installed and then upgraded to 1.2.4 with bristuff and i have the same problem. If you see the logs the INSERT trace has wrong values before the comand is executed. By the way, everyone of us that have this problem use HFC cards? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: jueves, 09 de febrero de 2006 18:28 Para: Jeroen Zwarts; Asterisk Users Mailing List - Non-Commercial Discussion CC: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x You are in my same situation. I thought I solved the problem (if you look at tomorrow post) but it isn't My situation is a bit different: I have the last bristuffed version of asterisk 1.2.4 (released yesterday) And I also have 2 zaphfc cards. but the behaviour is absolutely the same If you restart asterisk, you get one or two calls ok, the again the problem On the first zaphfc, the problem is almost immediate (1 or two calls) the second is stronger, and is ok for a longer period ( 1 day ??) then it also falls in problem on clid and src It seems to me some buffer overwrite problem. the clid is trasmitted ok to the internal phones. So I am not alone on this side... Andrea Jeroen Zwarts [EMAIL PROTECTED] nl To Sent by: asterisk-users@lists.digium.com asterisk-users-bo cc [EMAIL PROTECTED] m.com Subject [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x 09/02/2006 11.05 Please respond to Jeroen Zwarts [EMAIL PROTECTED] nl; Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com I have a problem with CDR recording in Asterisk 1.2.x. This is the situation: An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single HFC-S ISDN BRI card. I log the call records to both the Master.csv and MySQL. The problem is that when an incoming call from the ISDN line is logged to the CDR, the src and the clid field show up as something like 'h? ' (random weird ASCII characters).
[Asterisk-Users] Problems with Cepstral and Asterisk
Hello, For some reason I cannot get Cepstral to work with 1.2.4. I followed all the directions here http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt When I try to load asterisk I get [app_cepstral.so]Feb 10 04:58:36 WARNING[23037]: loader.c:325 __load_resource: libswift.so.4: cannot open shared object file: No such file or directory Feb 10 04:58:36 WARNING[23037]: loader.c:554 load_modules: Loading module app_cepstral.so failed! Ouch ... error while writing audio data: : Broken pipe I put the proper path to libswift.so.4 in /etc/ld.conf so where else do I have to define where it is located, or does someone know of a fix? Thanks, Steve Totaro http://www.asteriskhelpdesk.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] odd 'digital' sound artifacts
So nobody heard these before? or did I do something stupid that anyone should know and nobody wanted to yell at me for it ;) On Wed, 2006-02-08 at 12:54 -0600, Gerard Saraber wrote: Hi, I've got some weird sound artifacts happening during calls, they're very hard to describe, so I have a 122kb recording: http://openprojects.rarcoa.com/~miztic/artifact.wav normally the artifacts are just short blips, not quite as long as the one above, but they sound the same. When using the aggressive echo suppressor, it seems like those artifacts cause a really loud buzzing sound to come out of the cisco phone, pretty much made using the aggressive canceler impossible to use, it's too bad because it worked the best out of all of them, mark3 works ok but still gives echos on at least 20% of the calls. I thought they might be caused by IRQ sharing, so I pulled one of the TDM400P cards out and made sure the remaining two were on their own IRQ, the artifacts were still there. I've also tried running a kernel with all the low-latency stuff turned on, and the same kernel with it all turned off (2.6.16-rc2) doesn't appear to make any difference either. I'm not sure what else to try, any input would be appreciated. Thanks, Gerard Saraber [EMAIL PROTECTED] hardware: AMD64 1.8Ghz 512M ram MSI nforce3 socket 754 mainboard 3 Digium TDM400P cards, 10 FXO + 2 FXS modules /proc/interrupts CPU0 0:2784232IO-APIC-edge timer 1: 8IO-APIC-edge i8042 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 71552 IO-APIC-level eth0 185: 9412 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217:5577811 IO-APIC-level wctdm, wctdm 225:2769262 IO-APIC-level wctdm lspci (for completeness): 02:07.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at ac00 [size=256] Memory at fdeff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 225 I/O ports at a800 [size=256] Memory at fdefe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at a400 [size=256] Memory at fdefd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [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] Double ring
Am I the only one with this problem? I've got Aastra phones running the 1.3 firmware. It doesn't happen on the Grandstream phones but I'd like to know if anyone else has Aastra 9133i phones with the 1.3 firmware and Asterisk 1.2.4. I'm running a TE110P Pri card. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: 10 February 2006 13:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double ring Sorry I meant callprogress. I've tried it set to yes and no with no difference. Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: 10 February 2006 13:14 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double ring 'callprogress', in zapata.conf: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf Thanks, Bob McDowell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: Friday, February 10, 2006 7:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Double ring Oddly enough I'm on Aastra phones too. Doesn't happen with Grandstream phones. I've tried callprogram=yes and no to no effect. What firmware did you have, I'm on 1.3. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell Sent: 10 February 2006 12:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double ring I was getting something very similar with my Aastra test phones until I change 'callprogress=' to 'no'. Thanks, Bob On Fri, 10 Feb 2006 12:13:47 - Lee Archer [EMAIL PROTECTED] wrote: Can anyone shed any light on to why I get a double ring when calling external numbers? When calling out I hear the actually ring-ring of the called phone and the asterisk ring tone. I'm using the same config I used with 1.0.10 but have now upgraded to 1.2.4. Regards Lee ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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] Half Solved - Fail over to Pri on VoIP connection failure
I want to say thanks to everyone for the help so far. I figured out a
way to modify some AAH code that worked for me (well sort of). The line
I modified is s,14 in macro-dialout-trunk. Then I just added a variable
and passed it from 9_outside.
I just have one last problem. This waits for an answer not ringing. So
if the called party has a long ring to voice mail the call is dropped
and goes out the PRI.
Does anyone know of a way to listen for ringing on an IAX2 channel?
[9_outside]
exten = _9Z.,1,Macro(dialout-trunk,trunk-number-here,${EXTEN:1},,20)
exten = _9Z.,2,Macro(dialout-trunk,trunk-number-here,${,${EXTEN:1},)
exten = _9Z.,3,Macro(outisbusy); No available circuits
[macro-dialout-trunk]
exten = s,1,GotoIf($[foo${ARG3} = foo]?3:2)) ; arg3 is pattern
password
exten = s,2,Authenticate(${ARG3})
exten = s,3,Macro(user-callerid)
exten = s,4,Macro(record-enable,${CALLERIDNUM},OUT)
exten = s,5,Macro(outbound-callerid,${ARG1})
exten = s,6,SetGroup(OUT_${ARG1})
exten = s,7,CheckGroup(${OUTMAXCHANS_${ARG1}})
; if we've used up the max channels, continue at (n+101)
exten = s,8,SetVar(DIAL_NUMBER=${ARG2})
exten = s,9,SetVar(DIAL_TRUNK=${ARG1})
exten = s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper
dial string for this trunk
exten = s,11,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ;
OUTNUM is the final dial number
exten = s,12,Cut(custom=OUT_${ARG1},:,1) ; Custom trunks are prefixed
with AMP:
exten = s,13,GotoIf($[${custom} = AMP]?16)
;exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM}) ; Regular Trunk Dial
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM},${ARG4}) ; Regular Trunk
Dial w/ timeout
exten = s,15,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
${DIALSTATUS})
exten = s,16,Goto(s-${DIALSTATUS},1)
Thanks,
___
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[Asterisk-Users] RE: ex-girlfriend (ex-boyfriend)
Hi, Is there any syntax we can apply in the extensions to use the anti-ex-girl(boy)friend technique to multiple callers without having to replicate the lines? I mean, can I write the following two lines in only one line? exten= 12345/100,1,Hangup exten= 12345/200,1,Hangup Ricardo ___ --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] Make Meetme start only when somebody puts in the admin PIN
Hi, Is there anyway to have a MeetMe conference start only when somebody (anyone, let's say I don't want to manage who is the "marked user") connects and has the admin PIN instead of the user PIN? I would have assumed this was an obvious feature, but I dont see iton the Wiki. Or I am misreading it, which is entirely possible. Mike ___ --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] Polycom dialplan restriction
On 2/9/06, Doug Lytle [EMAIL PROTECTED] wrote: Carlos Chavez wrote: Is there any way to increase the number of digits before the number is diales automatically? Yes, I don't know about the 601s, but under the 301s and the 501s you can edit the digit map via the web interface or the sip.cfg on your ftp server. You're spot on. The 601's read the digitmap settings just as the 301s and 501s do from the global sip.cfg file fed from the boot server. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] 4 TE411P in one server installation
On 2/9/06, Raymond Chen [EMAIL PROTECTED] wrote: Does anyone try to install 2 or multiple TE411 card into one server? Can it be done? What about stability? Depends what you are looking to do. I'd imagine you could get two of the new TE411P's in doing G711. If you're doing dial around back out on the same TDM trunks, you may be able to get even more than 2 boards in a machine since the newer TE411P cards know how to bridge two TDM calls on the same board on the board itself rather than bringing it in to the PCI bus and then back out again. However, if you're looking to do gateway functionality into a compression codec like g729, I think Digium recommends no more than 2 cards per high horsepower machine as your bottleneck will then become the CPU itself. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] asterisk 1.2.4,addons 1.2.1, ooh323 and freebsd
Title: Message is there a way to compile ooh323 on freebsd, I have tried many solutions, nothing works :( Any good idea is welcome. Kind regards, Olivier ___ --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] Cisco 2620 as PRI gateway
debug ccsip message Kurt ___ --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] Yuck! Asterisk Crash...
Hi,
I'm currently running CVS-HEAD 2005-09-03
I do plan to upgrade to the newest version, but need to do some
testing with it first. In the mean time... does anyone know what
these messages below are about? I've never seen it before, but when
it happened it locked Asterisk up pretty good.
Feb 10 10:16:51 DEBUG[14917] chan_zap.c: Echo cancellation already on
Feb 10 10:16:57 DEBUG[14917] chan_zap.c: Exception on 14, channel 1
Feb 10 10:16:57 DEBUG[14917] chan_zap.c: Got event Dial Complete(9) on
channel 1 (index 0)
Feb 10 10:16:57 DEBUG[14917] chan_zap.c: Echo cancellation already on
Feb 10 10:17:01 DEBUG[14917] chan_iax2.c: Peer lastms 3, historicms 3,
maxms 2000
Feb 10 10:17:01 DEBUG[14917] chan_sip.c: Assigning Replace-Call-ID
Info [EMAIL PROTECTED] to REPLACE_CALL_ID
Feb 10 10:17:01 DEBUG[14917] chan_sip.c: 202 Accepted (supervised)
Feb 10 10:17:01 DEBUG[14917] channel.c: Planning to masquerade channel
Zap/2-1 into the structure of SIP/570601-8621
Feb 10 10:17:01 DEBUG[14917] channel.c: Done planning to masquerade
channel Zap/2-1 into the structure of SIP/570601-8621
Feb 10 10:17:01 DEBUG[14917] channel.c: Got clone lock for masquerade
on 'Zap/2-1' at 0x9e0d024
Feb 10 10:17:01 DEBUG[14917] chan_sip.c: update_call_counter(570601) -
decrement call limit counter
Feb 10 10:17:01 DEBUG[14917] channel.c: Putting channel Zap/2-1 in 4/4 formats
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: New owner for channel 2 is Zap/2-1
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Updated conferencing on 2,
with 0 conference users
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Updated conferencing on 2,
with 0 conference users
Feb 10 10:17:01 DEBUG[14917] channel.c: Released clone lock on
'SIP/570601-8621ZOMBIE'
Feb 10 10:17:01 DEBUG[14917] channel.c: Done Masquerading Zap/2-1 (6)
Feb 10 10:17:01 DEBUG[14917] channel.c: Didn't get a frame from
channel: SIP/570601-8621ZOMBIE
Feb 10 10:17:01 DEBUG[14917] channel.c: Bridge stops bridging channels
SIP/570601-3ac4 and SIP/570601-8621ZOMBIE
Feb 10 10:17:01 VERBOSE[14917] logger.c: -- Attempting native
bridge of Zap/2-1 and Zap/1-1
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: master: 2, slave: 1, nothingok: 0
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Stopping tones on 2/0 talking to 1/0
Feb 10 10:17:01 DEBUG[14917] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Stopping tones on 1/0 talking to 2/0
Feb 10 10:17:01 VERBOSE[14917] logger.c: == Spawn extension
(macro-dialout-trunk, s, 12) exited non-zero on 'SIP/570601-3ac4' in
macro 'dialout-trunk'
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Making 1 slave to master 2 at 0
Feb 10 10:17:01 VERBOSE[14917] logger.c: == Spawn extension
(from-internal, 18009033637, 1) exited non-zero on 'SIP/570601-3ac4'
in macro 'dialout-trunk'
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Making 1 slave to master 2 at 0
Feb 10 10:17:01 VERBOSE[14917] logger.c: == Spawn extension
(from-internal, 18009033637, 1) exited non-zero on 'SIP/570601-3ac4'
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Added 14 to conference 9/2
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Added 15 to conference 9/1
Feb 10 10:17:01 VERBOSE[14917] logger.c: -- Executing
Macro(SIP/570601-3ac4, hangupcall) in new stack
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Updated conferencing on 2,
with 0 conference users
Feb 10 10:17:01 DEBUG[14917] chan_zap.c: Updated conferencing on 1,
with 0 conference users
Feb 10 10:17:01 VERBOSE[14917] logger.c: -- Executing
ResetCDR(SIP/570601-3ac4, w) in new stack
Feb 10 10:17:01 VERBOSE[14917] logger.c: -- Executing
NoCDR(SIP/570601-3ac4, ) in new stack
Feb 10 10:17:01 VERBOSE[14917] logger.c: -- Executing
Wait(SIP/570601-3ac4, 5) in new stack
Feb 10 10:18:02 DEBUG[14917] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Feb 10 10:18:02 DEBUG[14917] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('2006-02-10
10:14:29','18009033637','18009033637','8884664646','from-internal',
'SIP/570601-3ac4','Zap/2-1','ResetCDR','w',213,210,'ANSWERED',3,'72071')
Feb 10 10:18:02 WARNING[14917] cdr.c: CDR on channel 'SIP/570601-3ac4'
already posted
Feb 10 10:18:02 WARNING[14917] cdr.c: CDR on channel 'SIP/570601-3ac4'
already started
Feb 10 10:18:02 WARNING[14917] cdr.c: CDR on channel 'SIP/570601-3ac4'
already posted
Feb 10 10:18:02 WARNING[14917] cdr.c: CDR on channel 'unknown' lacks end
Feb 10 10:18:02 WARNING[14917] cdr.c: CDR on channel 'unknown' lacks start
Feb 10 10:18:02 DEBUG[14917] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Feb 10 10:18:02 DEBUG[14917] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode)
VALUES ('1969-12-31 19:00:00','','','','',
'','','','',0,0,'UNKNOWN',0,'')
Feb 10 10:18:02 WARNING[14917] cdr.c: CDR on channel
Re: [Asterisk-Users] How can I send DTMF from the console?
i dont think you can do it from the console unless you hack the code to be able to use http://www.voip-info.org/wiki-Asterisk+cmd+SendDTMF from the console. RegardsOn 2/9/06, Anthony Azzopardi [EMAIL PROTECTED] wrote: How can I send DTMF from the console?Anthony.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...
Hi Chuck. I had the same problem. I solved it using the externnotify parameter inside voicemail.conf. Just launch a script which changes the /var/spool/asterisk permissions. Giorgio Incantalupo Chuck Bunn wrote: Hi, I thought I had this problem licked but there still is a rights problem with ARI and Asterisk when using a non-root user (Following the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-rootdiff2=25). When I issue the following: chmod --recursive u=rwX,g=rX,o= /var/spool/asterisk The above command results in the following rights on messages: msg.gsmrwxr-x--- asterisk msg.txtrw-r- asterisk msg.wavrwxr-x--- asterisk I can transfer voicemails and play them but new messages comming in get the following: msg.gsmrwx-- asterisk msg.txtrw-r--r-- asterisk msg.wavrwx-- asterisk After changing the rights a transferred messages has the folowing rights: msg.gsmrw-r- apache msg.txtrw-r- apache msg.wavrw-r- apache New voicemail cannot be played, deleted or transferred by the ARI application. Apache is belongs to the Asterisk group. I thought I understood SUID, GUID and sticky bit now I am not so sure. What is really confussing to me is why the rights on the .txt file do not match the other 2 after running the 'chmod --recursive ...' command. Any help here would be greatly appreciated. I am using the lastest versions of Asterisk 1.2.4 and Zaptel 1.2.3, etc. Thanks ___ --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 ATA should I buy?
AFIK, fax is supported and installed with with app_txfax app_rxfax If this proves to be true why would you need the ATA? RCS [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Parcina Sent: Thursday, February 09, 2006 2:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] What ATA should I buy? I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of those ATA-s? Sipura SPA-2100 SIP-ATA 160$ Sipura SPA-1001 SIP-ATA 125$ ALL7902 IP SIP ATA Adapter / Router 106$ Grandstream HandyTone ATA486142$ Thank you for any suggestions. P.S. If this is second time you see this message, then sorry for resending, but I didn't see it on list... -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)393447 e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Forwarding any number issue
Hi everyone: I'm new to the list and forgive me if my question has been discussed million times before. I have an asterisk box up and running supporting about 10 extensions. I can setup extension to forward incoming calls to an external number when its unavailable. My question is that is it possible to have setup like this: Someone calls the number, or getts to an extension, instead of being forwarded to a preset outside number, could he get a prompt or oppurtunity to dial any number he wants to forward to? Similar senerio: if I have my asterisk at home with a VoIP number that's capable of making free international calls, when I'm outside using my cell phone, can I dial in then use my VoIP number to make long distance calls? The only charge will only be from my cell phone to my VoIP number. Thanx in advance for any hints. Leon ___ --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] Double ring
Hi, the setting progressinband=no seems to fix the problem with my Aastra phones. The Grandstreams were unaffected and still are. Thanks Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Domjan Attila Sent: 10 February 2006 14:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Double ring I have the similar problem with thomson sip voip cable modems: http://bugs.digium.com/view.php?id=6083 On Fri, 2006-02-10 at 12:13 +, Lee Archer wrote: Can anyone shed any light on to why I get a double ring when calling external numbers? When calling out I hear the actually ring-ring of the called phone and the asterisk ring tone. I'm using the same config I used with 1.0.10 but have now upgraded to 1.2.4. Regards Lee ___ --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 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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] attended call transfer
Yep, im interested in coding to solve that problem, unfortunately i havent had time. I hope to be free in 2 weeks and start looking in the code to see if i can do something. Unless some one else has done it already. Regards.On 2/10/06, Alex Barnes [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] ] On Behalf Of MoisesSilva Sent: 10 February 2006 01:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] attended call transfer this is a Normal behaviour, nevertheless i dont think is a correct behaviour. Several weeks ago other user asked the same, i suggested himto open a feature request on bugs.digium.com, check for that regardsHi Yes that was me, this is still a big issue for us.Unfortunately we only have 1.2.1 installed on our live / dev boxes atthe moment and when I registered an account on the bug tracker and readthe rules it said you must have tested the issue on the very latest CVS head.I have been up to my eye balls the last couple of weeks sohaven't had time to do this.I didn't want to raise this as a feature request as in my opinion thishas to be a defect as attended transfer is basically unusable for a commercial environment (unless there exists a business that doesn't havea problem cutting off its customers :P )HTHAlexInformation contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation.All unauthorized use, disclosure or distribution is strictly prohibited.If you are not the addressee, please notify the sender immediately and destroy all copies of this email.Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding.Thank you.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with Cepstral and Asterisk
On Friday 10 Feb 2006 15:02, Steve Totaro wrote: Hello, For some reason I cannot get Cepstral to work with 1.2.4. I followed all the directions here http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt When I try to load asterisk I get [app_cepstral.so]Feb 10 04:58:36 WARNING[23037]: loader.c:325 __load_resource: libswift.so.4: cannot open shared object file: No such file or directory Feb 10 04:58:36 WARNING[23037]: loader.c:554 load_modules: Loading module app_cepstral.so failed! Ouch ... error while writing audio data: : Broken pipe I put the proper path to libswift.so.4 in /etc/ld.conf so where else do I have to define where it is located, or does someone know of a fix? Yup, but did you run ldconfig and is the path fully readable by the user running Asterisk? B -- http://www.mailtrap.org.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] Billing inbound calls per minute
On Monday 06 February 2006 09:25, JP Carballo wrote:
Michiel van Baak wrote:
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten = 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten = 1122334455,3,Hangup
DeadAGI is for hungup channels, not for active channels.
That might be a problem.
Try this:
exten = h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
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Re: [Asterisk-Users] Disabling SELinux in FC3 - good or bad
Just my opinion, but I came to the conclusion that SELinux is so convoluted and confusing (to me), that it's better to disable it. I would rather use things I understand like IPTables and PAM. On Feb 9, 2006, at 5:34 PM, Zach A wrote: Hi all, I had problem running MySQL on FC3 and what I found from googling was that SELinux should be disabled to make MySQL work n FC3. Now I am concerned about Asterisk, is it a good idea to disable SELinux. Or is there any other way to make MySQL work without disabling SELinux? Thanks, Zeeshan A Zakaria ___ --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 Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.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] calling to sip provider
Hello, I am new user of Asterisk. Yesterday I was trying to call from softphone to Asterisk, and that Asterisk routes this call to sipphone.com provider. I have found information on internet about how to register to sipphone and it seems that I have done. sip show status (or similar command) in CLI was showing me that I was registered. To call was not working, and on Asterisk's logs appeared: -- == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. -- Registered SIP '200' at 192.168.1.121 port 5060 expires 900 -- Saved useragent Linphone-1.2.0/eXosip for peer 200 -- Got SIP response 481 Subcription Does Not Exist back from 192.168.1.121 -- Executing SetCallerID(SIP/200-0e5a, Name 17476304480) in new stack -- Executing Dial(SIP/200-0e5a, SIP/[EMAIL PROTECTED]|20|r) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 500 I'm terribly sorry, server error occured (1/SL) back from 198.65.166.131 -- SIP/proxy01.sipphone.com-8a47 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/200-0e5a' status is 'CONGESTION' -- Calling using linphone or other softphones was working, so it is not circuit-busy error. I tried lot of configurations (in sip.conf and extensions.conf). Call is getting the correct route, but connection it is not working. Asterisk is behind NAT, without any redirected port. I was using externip and nat directives in configuration file. I think that I shouldn't need redirected ports because I was trying to call, not to receive calls. And NAT problem should be that I can listen but not talk (or vice-versa...) Any idea about what I can check? Any suggestion? Tahnk you very much, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --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] TDM - Analog Trunk - CallerID question
Hello list.
I have a question about how to read the incoming calls' callerid on an
FXO interface of a TDM 400 analog card; (it's one of those RED modules).
Now -may this is the complexity adding step..- I have a GSM gateway
attached to this FXO thing; incoming calls are processed as they should.
But both when peeking on the CLI, as well as in the phone display I do
not see the caller id.
Here I copy the very simple zapata.conf contents:
;
; Zapata telephony interface
;
; Configuration file
;
; Zapata configuration for Asterisk server zeta-stargate
;
[channels]
; edited by aaberga % 10.02.06
;cidsignalling=v23 ; Added for UK CLI detection
;cidstart=polarity ; Added for UK CLI detection
usecallerid=yes
signalling = fxo_ks
callerid= 2302
context=internal
channel = 1
signalling = fxo_ks
callerid= 2105
context=internal
channel = 2
signalling = fxs_ks
context = gsm_gateway
callerid=asreceived
channel = 4
I made a couple of attempts activating and moving around the two UK CLI
(this unit should work in UK; I thougt those settings might help)
settings. But nothing changed, except for the fact that the calls were
no more answered. ;-)
The usecallerid = yes and the callerid=asreceived have been added and
removed, but with no success.
This is what I see on the CLI when calling the gsm unit:
-- Starting simple switch on 'Zap/4-1'
Feb 10 16:20:02 NOTICE[7409]: chan_zap.c:5405 ss_thread: Got event 17
(Polarity Reversal)...
Feb 10 16:20:06 NOTICE[7409]: chan_zap.c:5405 ss_thread: Got event 2
(Ring/Answered)...
-- Executing NoOp(Zap/4-1, GSM Gateway Call from: ) in new stack
-- Executing Dial(Zap/4-1, Zap/1|30) in new stack
This is what I have in the relevant context of the dialplan:
[gsm_gateway]
exten = s,1,NoOp(GSM Gateway Call from: ${CALLERID})
exten = s,2,Dial(Zap/1,30)
exten = s,3,Hangup()
exten = t,1,Hangup()
exten = i,1,Hangup()
Can anybody point out what I do in a wrong way?
Thanks in advance..
Aldo
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Re: [EMAIL PROTECTED]: Re: [EMAIL PROTECTED]: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x]]
Ok Thank you very much to all people !! I will wait for the patch, and perhaps in the meantime I could try to introduce the agi workaround suggested by Jeroen, when it will be available. Andrea Tzafrir Cohen [EMAIL PROTECTED] rg.il To Sent by: Asterisk Users list asterisk-users-bo asterisk-users@lists.digium.com [EMAIL PROTECTED] cc m.com Subject [EMAIL PROTECTED]: Re: 10/02/2006 12.07 [EMAIL PROTECTED]: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x]] Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com Kapejod is working on a fix for the CDR problem in bristuff. See below - Forwarded message from [EMAIL PROTECTED] - Resent-From: [EMAIL PROTECTED] Resent-Date: Fri, 10 Feb 2006 13:01:25 +0200 Resent-Message-ID: [EMAIL PROTECTED] Resent-To: [EMAIL PROTECTED] Envelope-to: [EMAIL PROTECTED] Delivery-date: Fri, 10 Feb 2006 05:19:50 -0500 Date: Fri, 10 Feb 2006 11:19:47 +0100 (CET) Subject: Re: [EMAIL PROTECTED]: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x] From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Hi, i have had reports about the CDR corruption from various sources. It is a bug introduced by BRIstuff (shame on me) and i am currently working on a fix. The mail server rejecting Andrea's emails is one of Germany's largest ISPs which we use as a proxy. So unfortunately we dont have an influence on that. In any case people can usually find me on MSN ([EMAIL PROTECTED]) to report BRIstuff bugs (no matter if they use our hardware or not). Can you please forward this information to the users list? (I unsubscribed months ago since i couldnt handle the mail volume.) best regards Klaus As seen on asterisk-users... And generally, where should people without Junghanns hardware and systems (plain zaphfc users) turn to? - Forwarded message from [EMAIL PROTECTED] - Envelope-to: [EMAIL PROTECTED] Delivery-date: Fri, 10 Feb 2006 04:32:49 -0500 Resent-From: [EMAIL PROTECTED] Resent-Date: Fri, 10 Feb 2006 11:32:00 +0200 Resent-Message-ID: [EMAIL PROTECTED] Resent-To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Corrupt CDR records in Asterisk 1.2.x To: asterisk-users@lists.digium.com From: [EMAIL PROTECTED] Date: Fri, 10 Feb 2006 10:22:19 +0100 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Yes, everybody of us use zaphfc. No problem at all with zap channel that I have installed in several other * Box (PRI E1, Quad Analog, Chan_capi on 1.0 * series) So the problem, I think, is the zaphfc, or the patch to the * and zaptel provided by the bristuff. I tried to post a question to [EMAIL PROTECTED] but the mail was refused with a status code of 550 5.0.0, Dial-Up IP address rejected The public ip address I am using is from a newly buyed (3 days ago) set of 8 IP Address, so maybe in the past was used for spam...by the way, junghanns is the only domain refusing my mail If somebody else could ask to junghanns. Andrea Sergio Garcia Murillo [EMAIL PROTECTED] To Sent by: 'Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion' [EMAIL PROTECTED] asterisk-users@lists.digium.com, m.com 'Jeroen Zwarts' [EMAIL PROTECTED] cc 10/02/2006 09.51 [EMAIL PROTECTED] .com Subject Please respond to RE: [Asterisk-Users] Corrupt CDR Asterisk Users records in Asterisk 1.2.x Mailing List - Non-Commercial
[Asterisk-Users] ZapRas
Hi all, could ZapRas work on system with a HFC isdn card? Tnaks Giordano ___ --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] T1 Channel splitting PRI/data not working
I'll keep asking weird questions here: I have two T1 WAN circuits originating at my asterisk box and terminating at Asterisk servers at remote sites. Currently I use zaptel ppp assigned to all 24 channels for data. Now I would like to split off a few of these channels for voice trunks (I should use VOIP, but at the moment I do not have fast enough servers for the codecs) At one site the split work perfectly, with 8 channels for voice and 16 for data. Oddly enough the second site, using the exact same asterisk configuration (I just imaged the disks and am using the same hardware), doesn't work if the channels are split (the PRI and the PPP fail to come up) but does work if all 24 channels are assigned to PRI or data. I've always suspected this T1 line has some problem (large data transfers occasionally timeout, and I sometimes catch packet loss under high load), but I don't have any good ways of testing it. The phone company tests to their smartjacks and claims everything is great, but the smartjack is a long way from the equipment and I don't trust the guy that did the wiring too much. This line was once used for a frame relay circuit and repurposed, whereas the working line was installed as a bare point to point circuit from the start. Two questions, is there a way using the zaptel hardware to view T1 line errors and the like? Or is there anytime else that could cause it to work as a whole connection but not channelized? (Something weird like channels not having the same number on each end..I wouldn't think that was possible) Only other idea I have is to find a T1 BERT and test the line. --/etc/zaptel.conf-- #Span 1 T1 to Building 1 (This way works) span=1,0,0,esf,b8zs clear=1-24 #Span 1 T1 to Building 1 (This doesn't work) #span=1,0,0,esf,b8zs #clear=1-8 #bchan=7-23 #dchan=24 #Span 2 T1 to Building 2 (This works) span=2,0,0,esf,b8zs clear=25-32 bchan=33-47 dchan=48 --snip-- Thanks Mark ___ --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] Some articles
A bit old but a couple of interesting articles in here: http://www.acmqueue.com/modules.php?name=Contentpa=list_pages_issuesissue_id=16 ___ --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: Some articles
Sorry about the non asterisk thread, sent it to the wrong list On 2/10/06, Shidan [EMAIL PROTECTED] wrote: A bit old but a couple of interesting articles in here: http://www.acmqueue.com/modules.php?name=Contentpa=list_pages_issuesissue_id=16 ___ --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] Polycom dialplan restriction
As an example, here is my custom digitmap: digitmap dialplan.digitmap=9,911|9,411|0T|00T|1xx|9,011x.T|9,1[2-9]xx[2-9]xx|9,[2-9]xx|*7x|7x|*1xx|*8 The | is used to separate different entries. The comma means that it'll keep providing the dial tone after hitting 9. If you see a T after one of the entries, that means it'll wait till the timeout expires, if theres no T, then it'll dial that number as soon as you hit a match. Extensions in this office are 1xx. The reason i have 7x is for the park extensions. *8 is for call pickup (pickupgroup, callgroup) *7x is for various things such as call waiting, etc 0T and 00T - i don't think I even have dialplans setup for these... The rest are pretty self explanatory, for local, long distance, and international dialing. The international dialing has a T at the end, because different countries will have different numbers of digits. Any questions, let me know. Roman Carlos Chavez wrote: I am having a problem with some Polycom 601 phones. If I dial without picking up the handset or selecting the speaker I can dial numbers that are any lenght. But if I pick up the handset or are using the speaker I can only dial numbers that are 8 digits. When I dial the 8th digit it dials immediately. Obviously this creates problems when I am dialing long distance numbers or anything that needs more than 8 digits. Is there any way to increase the number of digits before the number is diales automatically? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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] Remapping Polycom IP501 buttons
Henry Kwan wrote: Just started using an asterisk-based PBX with Polycom IP501 phones. Am Fairly satisfied and am starting to get into FTP setup of the phones. Have figured out most things except for how button remapping works. In sip.cfg, I have this entry: keys key.IP_500.31.function.prim=DoNotDisturb/keys This works as expected but if I try to change the remapping to any other value like MyStatus, SpeedDialMenu, or BuddyStatus, it doesn't work. I got the list of values from Polycom's admin guide. Why does DoNotDisturb work and no other values that I've tried? This is the big question as far as I'm concerned with using Polycomm phones, I have about 30 501's running in our office and I like everything about them except the button remapping problem. If someone can figure that I would be totally psyched. Like you, I have been able to do simple remaps, like setting the Transfer button to issue a #, but anything more complex than that just doesn't work. Matt ___ --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: Remapping Polycom IP501 buttons
Noah Miller wrote: Just started using an asterisk-based PBX with Polycom IP501 phones. Am Fairly satisfied and am starting to get into FTP setup of the phones. Have figured out most things except for how button remapping works. In sip.cfg, I have this entry: keys key.IP_500.31.function.prim=DoNotDisturb/keys This works as expected but if I try to change the remapping to any other value like MyStatus, SpeedDialMenu, or BuddyStatus, it doesn't work. I got the list of values from Polycom's admin guide. Why does DoNotDisturb work and no other values that I've tried? You've run into the same problem a lot of other people have had. Remapping hard keys works fine, but remapping soft keys does not. In fact, trying to remap the soft keys results in some pretty weird behavior. The Polycom manual is a little misleading in that it doesn't mention this at all. My best guess is that the softkeys don't work because they can mean different things depending on what the phone is doing at the time. Polycom, if you're reading this, this would be another great feature to have! Who has hard buttons remapped for anything but the simplest of actions? If you do, I would very much like to hear about it. Can you post some details? Matt ___ --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: Remapping Polycom IP501 buttons
Henry Kwan wrote: Hi Noah, You've run into the same problem a lot of other people have had. Remapping hard keys works fine, but remapping soft keys does not. In fact, trying to remap the soft keys results in some pretty weird behavior. The Polycom manual is a little misleading in that it doesn't mention this at all. My best guess is that the softkeys don't work because they can mean different things depending on what the phone is doing at the time. Polycom, if you're reading this, this would be another great feature to have! Thanks for the info. That would explain a lot. The manual clearly states that SpeedDial should work though. On page 114 of the admin guide, it says that key.x.y.subPoint.prim will Sets the sub-identifier for key functions with a secondary array identifier such as SpeedDial. But when I try to set it: keys key.scrolling.timeout=1 key.IP_500.31.function.prim=SpeedDial key.IP_500.31.subPoint.prim=11/keys I get a volume-up action instead. So I guess it's a bug that they haven't gotten to fixing yet? I have had the same exact problem a button remapping gone wrong that results in volume up. I don't know if it's a bug, or bad documentation or what, but it's very frustrating. Matt ___ --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] Disabling SELinux in FC3 - good or bad
Zach A wrote: Hi all, I had problem running MySQL on FC3 and what I found from googling was that SELinux should be disabled to make MySQL work n FC3. Now I am concerned about Asterisk, is it a good idea to disable SELinux. Or is there any other way to make MySQL work without disabling SELinux? Thanks, Zeeshan A Zakaria Zach, I've been running Asterisk on FC3 without SELinux for a few months with no problems. As long as security is not a major concern, I think it may just be complicating things for you. If security is a concern, try hitting the Fedora users list http://www.redhat.com/mailman/listinfo/fedora-list. Someone there should be able to help you get MySQL running under SELinux. Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Cisco 79XX firmware 7.5
Just noted the above firmware on the Cisco site. Appears to be several (Two pages worth) bug fixes. Doug ___ --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] more cpu intensive echo cancellers ?
Gerard Saraber wrote: Thanks! testing it now, on my test calls it appears to start out with less echo then the Mark3 canceler, but it trains slower, seems like it took a long time for the echo to completely disappear, the real test will be seeing what the people at my company have to say. Feb 9 14:47:51 [kernel] Zapata Telephony Interface Registered on major 196 Feb 9 14:47:51 [kernel] Zaptel Version: SVN-trunk-r934M Echo Canceller: MG2 I've had really good luck with the echocan preload patch that was posted on the asterisk dev list a while back as well, and I've been recommending it to people as well. This has really helped minimize the echo problems to a minimal level, although I don't know about recommending this system to our customers. I still think a lot of my audio quality problems are being caused by my phones (not echo, but clicks and pops and various overmodulation problems). We're getting there, but I'm still nervous with trying to sell an * system to someone who is used to the quality of a traditional PBX or key system. Clint ___ --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: ex-girlfriend (ex-boyfriend)
On 15:08, Fri 10 Feb 06, Ricardo Monteiro wrote: Hi, Is there any syntax we can apply in the extensions to use the anti-ex-girl(boy)friend technique to multiple callers without having to replicate the lines? I mean, can I write the following two lines in only one line? exten= 12345/100,1,Hangup exten= 12345/200,1,Hangup Hi, I use the lookupblacklist for this http://www.voip-info.org/wiki/view/Asterisk+cmd+LookupBlacklist Works like a charm ___ --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] MixMonitor ControlPlayback of g729 files
Hi Should Mixmonitor ControlPlayback suppport file recordings in g279 format (I have enough licenses). call is alaw to alaw but would like to store the calls in g729 format instead of gsm. Thanks __ 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] SIP compact headers
Hi Anyone know if parsing of SIP compact headers is slower than full headers? like if there is an extra lookup step - mapping short to long? or should it be faster or about the same? Thanks __ 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] SIP Aliases
Is it possible with asterisk to setup aliases for SIP? For example, direct [EMAIL PROTECTED] to [EMAIL PROTECTED] If this isn't possible directly with asterisk, does SER offer anything along those lines? A search of the usual sites didn't turn up anything conclusive. Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] attended call transfer
Hi! I got this answer from the digium support: You may wish to use the attended transfer by either using hold or flashhook instead of the # features.conf attended transfer option. From a phone connected via a Zap channel, you would need to hit flash. Then enter the extension which you wish to transfer to. You can either hangup once it begins ringing or wait until the remote end answers. Once the remote ends answers you may announce the caller then hangup. This method works better than the attended transfer option available in the features.conf file. You must have your dial plan configured properly to allow for transfers. Dial plan configuration also falls under our Express Technical Support Service. Regards, Chris Hozian That means, that an attended transfer is possible as it would be liked in this mailing-list-thread. I tried to make call transfers with the flash button, but it doesnt work. threewaycalling and transfer is set to yes in my zapata.conf. But when I hit the flash-button - nothing happens. All incoming calls triggers a Dial() on all extensions with the Dial-Parameter t - so a call transfer should be possible. (Are here further configurations necesseary in my dialplan?) What am I doing wrong? Tom ___ --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: Asterisk-Users Digest, Vol 19, Issue 72
[EMAIL PROTECTED] is believed to have said:
Hi,
Is there any syntax we can apply in the extensions to use
the anti-ex-girl(boy)friend technique to multiple callers without having
to replicate the lines?
I mean, can I write the following two lines in only one line?
exten= 12345/100,1,Hangup
exten= 12345/200,1,Hangup
One way that even makes it possible to add new girls to your list
without touching the dialplan would be the following:
exten = 12345,1,DBGet(ex-girlfriend=disposedGirlfriend/{CALLERIDNUM})
exten = 12345,2,Hangup()
; jump to here only if CALLERIDNUM was NOT found in the list
exten = 12345,102,Dial(SIP/Myself)
Regards,
Aldo
___
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I: [Asterisk-Users] ZapRas
And about app_pppd, could it work with bristuff ? Thanks Giordano Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Giordano GrandisInviato: venerdì 10 febbraio 2006 17.44A: Asterisk Users Mailing List - Non-Commercial DiscussionOggetto: [Asterisk-Users] ZapRas Hi all, could ZapRas work on system with a HFC isdn card? Tnaks Giordano ___ --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] More Polycom IP501 questions
I am starting to get the hang of this, I think. These are more implementation questions; is this a proper/good way of using/doing this kind of questions. The IP501 has three line appearances. I have learned that shared line appearances cannot place calls, only receive them. They're indicated by the half telephone icon beside the button. Private line appearances can both place and take calls, and they show up as a full telephone icon. (where in the world is the manual that describes this stuff?) So, I figure that for a typical business setup you want to have two shared appearances (for the main #, for example) and then a private appearance so you can actually place calls. It seems kind of silly to waste 33% of my line appearances for my own extension, so that is the first question: Is a private line appearance required in order to place calls? Or do you simply not use the shared appearances for this, and let Asterisk handle it through ringing groups and pickup groups? I've set up the first two buttons to be the shared appearance for the Main line, and then the third for my own extension. However... When I go to use the live keypad to dial, I can enter the number and hit the Dial soft button, but the phone picks the shared appearance. Since the shared line appearance can't place calls, it fails. However, if I dial the number and hit the private line appearance it dials out just fine. This is telling me one of two things. Either the phone's kind of dumb because it is choosing the first available line even though it can't place a call out of it (unlikely) or I'm just doing this in a dumb way (far more likely). How do all of y'all out in asterisk-users land set these phones up, and why did you choose to do it the way you did? Were there nifty features you discovered through your particular configuration, are they set up to specifically avoid problems, or is it a mix of the two? I haven't even started to play with the mini browser; that looks like it is going to have some serious potential, too. Now for a side note: kxmleditor *rocks* for editing these damn Polycom XML configuration files! It almost makes me feel a little queasy, like I'm editing the Windows Registry. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem win Unicall
On Fri, 2006-02-10 at 08:38 -0200, Darlon wrote: Try to change the value of protocolvariant in the unicall.conf. Please, send us here the result. I am using mx,10,4 in the protocol variant of unicall.conf. What seemed to solve the problem is a very old tip that said I should change the DEFAULT_T1 value of mfcr2.c fomr 5000 to something like 2. I also included a bit timeout of 120 seconds in the dial command. For the moment every call is going through although I still have some testing to do. -- Carlos Chavez Director de Tecnologa Telecomunicaciones Abiertas de Mxico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 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] More Polycom IP501 questions
On 2/10/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: I am starting to get the hang of this, I think. These are more implementation questions; is this a proper/good way of using/doing this kind of questions. The IP501 has three line appearances. I have learned that shared line appearances cannot place calls, only receive them. They're indicated by the half telephone icon beside the button. Private line appearances can both place and take calls, and they show up as a full telephone icon. (where in the world is the manual that describes this stuff?) So, I figure that for a typical business setup you want to have two shared appearances (for the main #, for example) and then a private appearance so you can actually place calls. It seems kind of silly to waste 33% of my line appearances for my own extension, so that is the first question: Is a private line appearance required in order to place calls? Or do you simply not use the shared appearances for this, and let Asterisk handle it through ringing groups and pickup groups? I've set up the first two buttons to be the shared appearance for the Main line, and then the third for my own extension. However... When I go to use the live keypad to dial, I can enter the number and hit the Dial soft button, but the phone picks the shared appearance. Since the shared line appearance can't place calls, it fails. However, if I dial the number and hit the private line appearance it dials out just fine. This is telling me one of two things. Either the phone's kind of dumb because it is choosing the first available line even though it can't place a call out of it (unlikely) or I'm just doing this in a dumb way (far more likely). How do all of y'all out in asterisk-users land set these phones up, and why did you choose to do it the way you did? Were there nifty features you discovered through your particular configuration, are they set up to specifically avoid problems, or is it a mix of the two? I haven't even started to play with the mini browser; that looks like it is going to have some serious potential, too. Now for a side note: kxmleditor *rocks* for editing these damn Polycom XML configuration files! It almost makes me feel a little queasy, like I'm editing the Windows Registry. :-) Shared line appearances can make calls once the SIP you're transacting with responds positively to the request for access to the shared line resource to make a call. chan_sip in Asterisk doesn't presently know how to do deal with this. For the integration with Asterisk, you don't want to enable your line appearances as shared for the time being, but rather they should all be private. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] More Polycom IP501 questions
On Feb 10, 2006, at 12:15 PM, Andrew Kohlsmith wrote: I am starting to get the hang of this, I think. These are more implementation questions; is this a proper/good way of using/doing this kind of questions. The IP501 has three line appearances. I have learned that shared line appearances cannot place calls, only receive them. They're indicated by the half telephone icon beside the button. Private line appearances can both place and take calls, and they show up as a full telephone icon. (where in the world is the manual that describes this stuff?) So, I figure that for a typical business setup you want to have two shared appearances (for the main #, for example) and then a private appearance so you can actually place calls. It seems kind of silly to waste 33% of my line appearances for my own extension, so that is the first question: Is a private line appearance required in order to place calls? Or do you simply not use the shared appearances for this, and let Asterisk handle it through ringing groups and pickup groups? Do not use shared. Yes let asterisk dial multiple phones, works great. Monitoring a LINE can be problematic though. Remeber asterisk is a pbx, not a key system. I've set up the first two buttons to be the shared appearance for the Main line, and then the third for my own extension. However... When I go to use the live keypad to dial, I can enter the number and hit the Dial soft button, but the phone picks the shared appearance. Since the shared line appearance can't place calls, it fails. However, if I dial the number and hit the private line appearance it dials out just fine. This is telling me one of two things. Either the phone's kind of dumb because it is choosing the first available line even though it can't place a call out of it (unlikely) or I'm just doing this in a dumb way (far more likely). How do all of y'all out in asterisk-users land set these phones up, and why did you choose to do it the way you did? Were there nifty features you discovered through your particular configuration, are they set up to specifically avoid problems, or is it a mix of the two? I haven't even started to play with the mini browser; that looks like it is going to have some serious potential, too. Now for a side note: kxmleditor *rocks* for editing these damn Polycom XML configuration files! It almost makes me feel a little queasy, like I'm editing the Windows Registry. :-) -A. ___ --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] odd 'digital' sound artifacts
Your output looks like you have 3 cards, two of which are sharing interrupts - or am I missing something? On Feb 10, 2006, at 7:04 AM, Gerard Saraber wrote: So nobody heard these before? or did I do something stupid that anyone should know and nobody wanted to yell at me for it ;) On Wed, 2006-02-08 at 12:54 -0600, Gerard Saraber wrote: Hi, I've got some weird sound artifacts happening during calls, they're very hard to describe, so I have a 122kb recording: http://openprojects.rarcoa.com/~miztic/artifact.wav normally the artifacts are just short blips, not quite as long as the one above, but they sound the same. When using the aggressive echo suppressor, it seems like those artifacts cause a really loud buzzing sound to come out of the cisco phone, pretty much made using the aggressive canceler impossible to use, it's too bad because it worked the best out of all of them, mark3 works ok but still gives echos on at least 20% of the calls. I thought they might be caused by IRQ sharing, so I pulled one of the TDM400P cards out and made sure the remaining two were on their own IRQ, the artifacts were still there. I've also tried running a kernel with all the low-latency stuff turned on, and the same kernel with it all turned off (2.6.16-rc2) doesn't appear to make any difference either. I'm not sure what else to try, any input would be appreciated. Thanks, Gerard Saraber [EMAIL PROTECTED] hardware: AMD64 1.8Ghz 512M ram MSI nforce3 socket 754 mainboard 3 Digium TDM400P cards, 10 FXO + 2 FXS modules /proc/interrupts CPU0 0: 2784232 IO-APIC-edge timer 1: 8 IO-APIC-edge i8042 8: 0 IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 71552 IO-APIC-level eth0 185: 9412 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217: 5577811 IO-APIC-level wctdm, wctdm 225: 2769262 IO-APIC-level wctdm lspci (for completeness): 02:07.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at ac00 [size=256] Memory at fdeff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 225 I/O ports at a800 [size=256] Memory at fdefe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 02:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0001 Flags: bus master, medium devsel, latency 32, IRQ 217 I/O ports at a400 [size=256] Memory at fdefd000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] more cpu intensive echo cancellers ?
On Fri, 2006-02-10 at 11:01 -0600, Clint Sharp wrote: Gerard Saraber wrote: Thanks! testing it now, on my test calls it appears to start out with less echo then the Mark3 canceler, but it trains slower, seems like it took a long time for the echo to completely disappear, the real test will be seeing what the people at my company have to say. Feb 9 14:47:51 [kernel] Zapata Telephony Interface Registered on major 196 Feb 9 14:47:51 [kernel] Zaptel Version: SVN-trunk-r934M Echo Canceller: MG2 I've had really good luck with the echocan preload patch that was posted on the asterisk dev list a while back as well, and I've been recommending it to people as well. This has really helped minimize the echo problems to a minimal level, although I don't know about recommending this system to our customers. I still think a lot of my audio quality problems are being caused by my phones (not echo, but clicks and pops and various overmodulation problems). We're getting there, but I'm still nervous with trying to sell an * system to someone who is used to the quality of a traditional PBX or key system. Clint Found it, going to go test it right now :) thanks! So far reports have been positive on the echo, but its a slow day ;) We're using cisco 7960 phones, they're pricy, but they work great and sound good, if it wasn't for the echo issue, I would have been able to roll the whole setup out already. Actually that's not quite true, I still have to make the 7914 addon module work with the 7960 phone, but that's not a show stopper. Either way, so far big thumbs up for the MG2 echo can, and if any developers read this, feel free to add a compile flag to make it more cpu intensive ;) and do more canceling. -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: More Polycom IP501 questions
Hi Andrew - How do all of y'all out in asterisk-users land set these phones up, and why did you choose to do it the way you did? Were there nifty features you discovered through your particular configuration, are they set up to specifically avoid problems, or is it a mix of the two? We've never bothered with Shared line appearances. I've pretty much assumed they are only good for emulating a key system. For us it would be more of a hindrance than a feature, as only our receptionists are ever interested in taking calls from the main line (everyone else is DID). We just use our multiple line appearances to allow our users to take multiple calls. Sort of an advanced, visually-driven call-waiting. I haven't even started to play with the mini browser; that looks like it is going to have some serious potential, too. Does the minibrowser work on the 500's? I though it was only on the 600 series. Now for a side note: kxmleditor *rocks* for editing these damn Polycom XML configuration files! It almost makes me feel a little queasy, like I'm editing the Windows Registry. :-) Cool. Thanks for the tip. I've always hated editing those stupid XML files with pico. - Noah ___ --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] Multiple Asterisk Server Question
I'm not exactly sure how to phrase this, but I have two offices that I want to each have an * server. One office has two POTS lines coming in, the other has five. I'll call the two line office A and the other office B I want office A to be able to extension dial office B, which I think is straightforward enough because I'll just add a line in extensions.conf that points a particular extension (i.e., 101) to the other server [EMAIL PROTECTED] The problem now comes up how to dial local numbers. If the two offices are in different area codes, I would want office A to dial all local numbers using the local * server and all long distance using B's * server. How do you set that up? I.E., office A is area code 303, office B is 490. If you dial 303 from either office it will run through office A's * box, and vice versa for 490 calls. Also, would there be a way to use perhaps 8+number for the local * box and 9+number to use the remote box? Thanks for any advice! ___ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.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] odd 'digital' sound artifacts
On Fri, 2006-02-10 at 11:20 -0800, Anthony Rodgers wrote: Your output looks like you have 3 cards, two of which are sharing interrupts - or am I missing something? That is correct, the thing is, I pulled one of the cards out (as stated in my first email), and made sure each was on their own IRQ, and I *still* got the same artifacts, so I'm not sure that IRQ sharing is the problem. /proc/interrupts CPU0 0:2784232IO-APIC-edge timer 1: 8IO-APIC-edge i8042 8: 0IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 177: 71552 IO-APIC-level eth0 185: 9412 IO-APIC-level libata, NVidia CK8S 193: 0 IO-APIC-level ehci_hcd:usb1 201: 0 IO-APIC-level ohci_hcd:usb2 209: 0 IO-APIC-level ohci_hcd:usb3 217:5577811 IO-APIC-level wctdm, wctdm 225:2769262 IO-APIC-level wctdm -- Regards, Gerard Saraber Network Admin, Rarcoa, Inc. (630) 654-2580 x11 [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] Re: More Polycom IP501 questions
On Friday 10 February 2006 14:22, Noah Miller wrote: We've never bothered with Shared line appearances. I've pretty much assumed they are only good for emulating a key system. For us it would be more of a hindrance than a feature, as only our receptionists are ever interested in taking calls from the main line (everyone else is DID). We just use our multiple line appearances to allow our users to take multiple calls. Sort of an advanced, visually-driven call-waiting. *nods* -- and based on what others are saying, Asterisk's SIP stack does not really work with shared line appearances anyway. A classic case of me making things more complicated than they need to be. :-) I haven't even started to play with the mini browser; that looks like it is going to have some serious potential, too. Does the minibrowser work on the 500's? I though it was only on the 600 series. Hmm... that explains why the one set of config files I have has the home URL defined as http://dont.you.wish.you.had.an.ip601.hehe/;. Thanks for the input, everyone. I need to digest this information into a Quick and Easy setup of a Polycom IP501 for the wiki. -A. ___ --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] Cisco 79XX firmware 7.5
Yeah, that's been on there since July according to the download page... That firmware messes up re-registration when the proxy dies... we had about 60 phones with that firmware just stop working after we had to restart the asterisk service. Aaron Doug Lytle wrote: Just noted the above firmware on the Cisco site. Appears to be several (Two pages worth) bug fixes. Doug ___ --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] Repeating Zap Message
What would cause the message: == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up To keep appearing on CLI about once every second? If I do a zap show status: Description Alarms IRQbpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2OK 0 0 0 T4XXP (PCI) Card 0 Span 3UNCONFIGUR 0 0 0 T4XXP (PCI) Card 0 Span 4UNCONFIGUR 0 0 0 Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --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] GotoIf number exists in file. How can i do this?
On Wed, 2006-02-08 at 14:37 +0100, Arne Morten Johansen wrote:
Hi there.
I currently have a GotoIf statement that goes to a special
extension priority if the CID match with one of the numbers in
my “list” of CIDs. The way I’ve done it now is by multiple OR
operators. There must be a better way. Anyone got some
suggestions?
This is basicly what I want. “If CID Exists in $File, goto
s,10”. So when I want to add a new CID I just add a new line
in a txt file.
Or, maybe you can use the existence of a file rather then the content of
it?
exten = s,1,System(test -e /var/lib/asterisk/callerids/${CALLERID})
exten = s,2,NoOp(Normal caller)
exten = s,102,NoOp(special caller)
this way you can add callerids by simply
touch /var/lib/asterisk/callerids/phonenumber
does that help?
Conrad
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[Asterisk-Users] Agent supervisor configuration
Hi everyone. I have the follow problem: I need to configure an Agent (Supervisor) for monitoring and intercept calls regarding to different Queue, Any help is appreciated. Regards. Cristian. _ Express yourself instantly with MSN Messenger! Download today - it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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] Echo PSTN [EMAIL PROTECTED] 2.0 Digium TDM11B DSL
My problem is solved by doing the following. Many thanks to Mark Spencer for clueing me in...search for the following string in wctdm.c:/* Enable on-hook line monitor */After the wctdm_setreg function add the following lines: /* Apply negative Tx gain of 4.5db to DAA */wctdm_setreg(wc, card, 38, 0x14);/* 4db */wctdm_setreg(wc, card, 40, 0x15);/* 0.5db *//* Apply negative Rx gain of 4.5db to DAA */wctdm_setreg(wc, card, 39, 0x14);/* 4db */wctdm_setreg(wc, card, 41, 0x15);/* 0.5db */Save the file, and recompile zaptel On 12/10/05, David K Parker [EMAIL PROTECTED] wrote: I hadn't heard of fxotune. I'll check it out. I had a little bit better luck today after replacing a 25 ft cable on the pstn side with a 7 ft cable. I'm beginning to wonder about ztmonitor though. Before replacing the cable I had to adjust Rx to 22 and Tx to - 7.5. Ztmonitor still showed the Tx gain to be hot. If I went below -7.5 I couldn't complete a call. Now Rx is at 2.5 and Tx is at 8. This is adjusted when caling the CO. When I call anyone, the sound is low. I can adjust Rx to 4.5 and its better. The other party hears me fine. I changed the echo canceller to ECHO_CAN_MG2 in Zaptel and the beginning of the call isn't as bad. If I unplug Asterisk from the PSTN and use the analog direct to the telco the quality is fine. Asterisk is poor in comparison on PSTN. Any calls with Asterisk to my Teliax ld trunk are fine. On 12/9/05, Matthew Fredrickson [EMAIL PROTECTED] wrote: On Dec 9, 2005, at 9:48 AM, David K Parker wrote: I have a Digium TDM11B, I'm fighting an issue with with echo on the PSTN side. I run [EMAIL PROTECTED] 2.0. I have an analog phone on the FXS channel 1 and Telco on the FXS channel 4. I also have a coupe of softphones, 1 iax2, the other sip, and a LinkSys Sipura 941. I use a VOIP provider for long distance. I'm experiencing echo on all calls on any phone for calls going out over the PSTN, but no echo at all on Long Distance calls with my VOIP provider or Internal calls. I think its safe to say that echo is occuring on the PSTN side on channel 4. I've followed the trouble shooting provedures on voip-info.org for echo cancellation, even calling the local CO using ztmonitor to adjust rx tx gain. The only thing I haven't tried yet is installing shielded cable. I use Verizon DSL for Internet and have the appropriate filter for my PSTN on channel 4. I'm beginning to wonder if the problem is due to DSL. Has anyone else had this experience.Have you tried running fxotune on the card?It's possible that your echo problem is related to line impedance mismatch.For more details,see README.fxotune in the zaptel package.Matthew Fredrickson___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk vs. Traditional PBX
Thanks everyone ! One more question if I do go with a PRI line. Will my existing TDM card from digium work or do I need to purchase a different card to handle this ? Thanks Nora Lavelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Friday, February 10, 2006 4:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk vs. Traditional PBX I am sorry. I thought you wrote Dell 850. Should have looked closer. The machine should do just fine. However it would not hurt to ptu in another gig. Also see if anyone else on the list has used a 650 and what expiriences they have had. Regards, Dovid --- Nora Lavelle [EMAIL PROTECTED] wrote: Hi Dovid, Thank you for the book. I'm already reading it. I have a dell 650 server, 1Gig of memory, 1 CPU (3.07Ghz). What hardware would you recommend for the 200 users w/ about 20 concurrent calls ? As always I thank you so much for your help. Nora Lavelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 09, 2006 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk vs. Traditional PBX I think your problem is the Dell 650. What are the specs on it ? If you want a system that can support 200 users you will need to do a lot better than that. Also you will be dealing with T1's/E1's and not POTS lines. I think a good place to start (if you havent already) is the book that has come out a while back. I have it on my server at http://www.h6315.com/ast_book/ Regards, Dovid (I posted my server and not from the publisher becuase I do not know thier URL and I have email access only now.) --- Nora Lavelle [EMAIL PROTECTED] wrote: Hi everyone ! So here's my question of the day ! I need to make a decision on whether or not to go to a voip solution or configure an existing pbx (norstar) that my company has available. We are a small startup. I'm wanting a solution that will support up to about 200 people, with direct dial-in capability, up to about 30 concurrent phone calls and good voice quality. Right now I have an asterisk deployment with about 15 people on it. We have sipura 841 phones. The biggest issue currently is voice quality. lot of complaints there. I have a dell 650 poweredge (single processory system), with a digium tdm400 card and 4 analog lines plugged into it. So here are my questions: * Is asterisk a good solution for my company ? or should I just install the traditional pbx and look to move to asterisk in a couple of years ? (I personally would prefer asterisk cuz I'm a unix person not a phone person so from a manageability perspective i would love this ) * If I were to go to an asterisk solution to support about 200 people with the requirements above what hardware platform would you recommend ? I'm guessing I'd need a PRI line and a different digium card? Also would a 1cpu poweredge dell be enough ? or would that have to be upgraded too ? If anyone is running an environment similar to this that can provide help I would really appreciate this. I'm having a hard time making this decision and would love to hear anybody's experience in a real time environment. Thanks again this list ROCKS! Nora Lavelle ___ --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 __ 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 ___ --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 __ 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] More Polycom IP501 questions
Sorry, no minibrowser on the 501s :( Andrew Kohlsmith wrote: I am starting to get the hang of this, I think. These are more implementation questions; is this a proper/good way of using/doing this kind of questions. The IP501 has three line appearances. I have learned that shared line appearances cannot place calls, only receive them. They're indicated by the half telephone icon beside the button. Private line appearances can both place and take calls, and they show up as a full telephone icon. (where in the world is the manual that describes this stuff?) So, I figure that for a typical business setup you want to have two shared appearances (for the main #, for example) and then a private appearance so you can actually place calls. It seems kind of silly to waste 33% of my line appearances for my own extension, so that is the first question: Is a private line appearance required in order to place calls? Or do you simply not use the shared appearances for this, and let Asterisk handle it through ringing groups and pickup groups? I've set up the first two buttons to be the shared appearance for the Main line, and then the third for my own extension. However... When I go to use the live keypad to dial, I can enter the number and hit the Dial soft button, but the phone picks the shared appearance. Since the shared line appearance can't place calls, it fails. However, if I dial the number and hit the private line appearance it dials out just fine. This is telling me one of two things. Either the phone's kind of dumb because it is choosing the first available line even though it can't place a call out of it (unlikely) or I'm just doing this in a dumb way (far more likely). How do all of y'all out in asterisk-users land set these phones up, and why did you choose to do it the way you did? Were there nifty features you discovered through your particular configuration, are they set up to specifically avoid problems, or is it a mix of the two? I haven't even started to play with the mini browser; that looks like it is going to have some serious potential, too. Now for a side note: kxmleditor *rocks* for editing these damn Polycom XML configuration files! It almost makes me feel a little queasy, like I'm editing the Windows Registry. :-) -A. ___ --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 -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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] Setting up Polycom 501 with 2 Different Extensions
I'm having some difficulty setting up a Polycom 501 with two different extensions. I've set up in the phone's xml file the proper reg.1with lineKeys = 2, and reg.3 with lineKeys = 1. I have also set up the sip.conf file on the Asterisk box correctly. However, the phone is not making any sort of registration request to the Asterisk server. Is there something else that needs to be done to have the phone register with Asterisk with the second extension? The first extension registers just fine Thanks for any help,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] Setting up Polycom 501 with 2 Different Extensions
On Friday 10 February 2006 15:58, Andrew Berman wrote: registration request to the Asterisk server. Is there something else that needs to be done to have the phone register with Asterisk with the second extension? The first extension registers just fine Did you fill out the server information for the second line appearance? I have two different extensions registering just fine (to my knowledge) :-). -A. ___ --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 vs. Traditional PBX
TDM cards are for terminating regular POTS lines (anything you could plug a regular phone into and have it work.) For a PRI you would need one of Digium's T1 interface cards. The TE110P is a great card if you need less than 23 channels. I can't recommend any other brands because I've never used them. _ RYAN AMOS System Administrator FINETOOTH THE CONTRACT INTELLIGENCE COMPANY phone 512.637.3530fax 512.637.3501 mobile 512.484.6577 email [EMAIL PROTECTED] WWW.FINETOOTH.COM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nora Lavelle Sent: Friday, February 10, 2006 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk vs. Traditional PBX Thanks everyone ! One more question if I do go with a PRI line. Will my existing TDM card from digium work or do I need to purchase a different card to handle this ? Thanks Nora Lavelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Friday, February 10, 2006 4:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk vs. Traditional PBX I am sorry. I thought you wrote Dell 850. Should have looked closer. The machine should do just fine. However it would not hurt to ptu in another gig. Also see if anyone else on the list has used a 650 and what expiriences they have had. Regards, Dovid --- Nora Lavelle [EMAIL PROTECTED] wrote: Hi Dovid, Thank you for the book. I'm already reading it. I have a dell 650 server, 1Gig of memory, 1 CPU (3.07Ghz). What hardware would you recommend for the 200 users w/ about 20 concurrent calls ? As always I thank you so much for your help. Nora Lavelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 09, 2006 2:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk vs. Traditional PBX I think your problem is the Dell 650. What are the specs on it ? If you want a system that can support 200 users you will need to do a lot better than that. Also you will be dealing with T1's/E1's and not POTS lines. I think a good place to start (if you havent already) is the book that has come out a while back. I have it on my server at http://www.h6315.com/ast_book/ Regards, Dovid (I posted my server and not from the publisher becuase I do not know thier URL and I have email access only now.) --- Nora Lavelle [EMAIL PROTECTED] wrote: Hi everyone ! So here's my question of the day ! I need to make a decision on whether or not to go to a voip solution or configure an existing pbx (norstar) that my company has available. We are a small startup. I'm wanting a solution that will support up to about 200 people, with direct dial-in capability, up to about 30 concurrent phone calls and good voice quality. Right now I have an asterisk deployment with about 15 people on it. We have sipura 841 phones. The biggest issue currently is voice quality. lot of complaints there. I have a dell 650 poweredge (single processory system), with a digium tdm400 card and 4 analog lines plugged into it. So here are my questions: * Is asterisk a good solution for my company ? or should I just install the traditional pbx and look to move to asterisk in a couple of years ? (I personally would prefer asterisk cuz I'm a unix person not a phone person so from a manageability perspective i would love this ) * If I were to go to an asterisk solution to support about 200 people with the requirements above what hardware platform would you recommend ? I'm guessing I'd need a PRI line and a different digium card? Also would a 1cpu poweredge dell be enough ? or would that have to be upgraded too ? If anyone is running an environment similar to this that can provide help I would really appreciate this. I'm having a hard time making this decision and would love to hear anybody's experience in a real time environment. Thanks again this list ROCKS! Nora Lavelle ___ --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 __ 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] Setting up Polycom 501 with 2 Different Extensions
I did not because reg.3.server.1 should have the exact same info as in the sip.conf file. In the manual it says, Note: If the reg.x.server.y.address parameter is non-Null, all of the reg.x.server.y.xxx parameters will override the parameters specified in sip.cfg.I figured that I do not want to override the server 1 settings in sip.confWhich attributes did you fill out? All of them? Thanks,AndrewOn 2/10/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Friday 10 February 2006 15:58, Andrew Berman wrote: registration request to the Asterisk server.Is there something else that needs to be done to have the phone register with Asterisk with the second extension?The first extension registers just fineDid you fill out the server information for the second line appearance?Ihave two different extensions registering just fine (to my knowledge) :-). -A.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
