[Asterisk-Users] Asterisk as VoIP gateway
I want to interconnect 2 pbx switches from to distinct location via an internet vpn using asterisk as VoiP gateways. The problem is what interfaces i must use between asterisk servers and pbx switch (FXO or FXS), and why? thank you in advance ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bri dropping calls
Altus Snyman wrote: Where do you get this new version of bristuff,I had a look on the webpage and there's only RC3 My first action every morning is to look at the top of this page: http://www.junghanns.net/asterisk/downloads/?C=M;O=D -- Best regards Peer Oliver Schmidt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to pop up called number details using php scripts in agi scripts
Michiel van Baak wrote: On 05:14, Tue 08 Feb 05, Mazhar Hussain wrote: If this sounds usefull to you, reply so on the list and I will try to setup a clear txt doc where and how to find the sourcecode. I would like to see the information you can provide on this. Thanks, Matt -- Matt Gibson VOIP Administrator NJ Tech Solutions PSTN: 1.877.999.4678 ex. 6400 FWD: 472645 IAXTEL: 1.700.761.1828 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] bri dropping calls
Hi Michael, -Original Message- dIf you reread his email, he is stating that he has a quadbri So do we. We are seeing something similar, even on RC5. On Wed, 09 Feb 2005 07:58:38 +0100, Peer Oliver Schmidt [EMAIL PROTECTED] wrote: Altus Snyman wrote: We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3 All installed and working.BUT after 5min+ of talking it just drops the calls? Are you sure the call get dropped? We have a similar problem, but the call does not get dropped, but stays silent for a couple of seconds. If both parties don't hangup, they will be able to continue the conservation. (And yes, the latest to get is bristuff_0.0.2RC5 [RC6 seems to be for quadbri and octobri cards, only]) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] add_pppd dialout problems
Hi I am trying to get app_pppd to make an outgoing call to my ISP. Has anybody got this to work yet? Thanks Roger ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Compile Problem on Red Hat 9
I get the following error when trying to compile asterisk 1.05 on red hat 9. [EMAIL PROTECTED] asterisk]# make install *** You don't have mpg123 installed. You're going to need *** *** it if you want MusicOnHold *** ./mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -02/08/05-20:18:18\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN `ls *.c` : invalid option Usage: /bin/sh [GNU long option] [option] ... /bin/sh [GNU long option] [option] script-file ... GNU long options: --debug --dump-po-strings --dump-strings --help --init-file --login --noediting --noprofile --norc --posix --rcfile --rpm-requires --restricted --verbose --version --wordexp Shell options: -irsD or -c command or -O shopt_option (invocation only) -abefhkmnptuvxBCHP or -o option make: *** [.depend] Error 2 Any help is greatly appreciated Vince ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error compiling app_icd
On Wed, 9 Feb 2005, Stefan Gofferje wrote: I wanted to try out app_icd but... [EMAIL PROTECTED]:/opt/app_icd make === Compile: /opt/app_icd/app_icd.c (app_icd.o) app_icd.c: In function `app_icd__log_events': app_icd.c:2104: error: structure has no member named `cid' app_icd.c:2104: error: structure has no member named `cid' make: *** [app_icd.o] Error 1 [EMAIL PROTECTED]:/opt/app_icd Got it from CVS head yesterday night (CET). Machine: Athlon XP / 512MB RAM, SuSE 9.2 Icd applies to the head version of Asterisk. Perhaps you are running stable 1.0.x? The callerid handling inside Asterisk changed between stable and head. There are a few problems that we have seen that are not patched in icd cvs yet. Dialing a zap channel does not work without a patch, and likewise with the wrapup time for an agent. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk performance monitoring
I'm not sure it answers all your questions but there is ast-stats from http://areski.net/areski/index.php? option=com_contenttask=categorysectionid=5id=70Itemid=54 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hung Sip Channels
On Tue, 8 Feb 2005 14:27:28 -0500, Brian C. Fertig [EMAIL PROTECTED] wrote: Does anyone know how to get rid of these hung channels? I am getting this when I do a: show sip channels 209.82.xxx.xxx0071495217 2591218534@ 00103/1 unknow(d) 209.82.xxx.xxx0041590104 0690231739@ 00103/1 unknow(d) 209.82.xxx.xxx0070259259 3265102826@ 00103/1 unknow(d) 209.82.xxx.xxx0071948143 1927207026@ 00103/1 unknow(d) 209.82.xxx.xxx0022576786 1752809624@ 00103/1 unknow(d) 209.82.xxx.xxx0070153955 0085223171@ 00103/1 unknow(d) I have about 60 of them and growing. I have submitted a ticket with my provider to let them know of this problem but I would like to clear them out w/o restarting the asterisk binary. I have the same problem. For me it looks like this some not completed transactions. This hanged transcations don't affect anything, but I worry, how much their number will increase, when I will have bigger load on asterisk box (Currently near 20 concurent calls.) I think dumping of all signalling and analyzing it can help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem using TDM400P FXS card
Hello, everyone After having spent several time to look for any solution for my problem, I decided to write here. Here is the problem got a Digium X101 FXO card and Digium TDM400P alias freshmaker (1 fxs module on it on first port ) in my asterisk box. The X101 works perfectly. The problem comes from the TDM400, when i plug a tested working phone on it , i get no dialtone; the goal is to connect a few old non-ip phones to the PBX. The modules zaptel and wctdm compiled and are loaded successfully. Here are some infos : debian linux asterisk 1.0.5 cvs zaptel zaptel.conf fxoks=2 ; the x101 card uses channel 1 with fxsks=1 loadzone=fr default zone = fr when i do a ztcfg it succesfully says me registered zone 2 But i cant see the phone in the zttool /etc/asterisk/zapata.conf [channels] language=fr context=default signalling=fxo_ks echocancel=yes group= 1 busydetect=yes channel = 2 Asterisk loads succesfully; and a zap show channels tells me : pseudocartezapfr 2cartezapfr But the problem is : i still dont have any dialtone on the phone. Anyone has an idea ? Thanks in advance Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] incoming h323 calls, routed to SIP/H323 drop after connection
Hello, I am attempting to use Asterisk as a protocol converter. I have set up asterisk to route incoming h323 calls to a SIP termination carrier. I make a test, call is coming correctly, is rerouted to termination carrier. Call connects and phone rings. Then, I pick up the phone and it hangs up after 2 seconds. I initially thought it was a codec issue. I made sure codec is g729 in all sip.conf h323.conf parts (general context + specific contexts). Still, call drops after connects and gives error cannot bridge between X call and Y call. Is this familiar to anyone? Do you have idea what to search next? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Encrypted VOIP?
Our SIPS implementation is absolutely standard conform according to RFC3261 and our SRTP implementation follows RFC3711. Regards Nils Ohlmeier On Tuesday 08 February 2005 13:37, Remco Barende wrote: What about SIPS (Secure SIP)? I cannot find anything about it in the Wiki but the Snom phones claim support for SIPS. No clue whether this is a standard or something proprietary that Snom have developed themselves. -- snom technology AGPascalstrasse 10bD-10581 Berlin Nils Ohlmeier mailto:[EMAIL PROTECTED] http://www.snom.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on hold distorted
Yesterday I setup music on hold by downloading and installing mpg123 r Now I have music on hold but it sounds terrible - clipping, buzzing, digital distortion, and its too loud (which probably isn't helping) and I'm just running it thru the 'default' line in music onhold.conf line default = quietmp3:/var/lib/asterisk/mohmp3, with the default mp3s. This is a 1.0.3 box, running headless (no x desktop) on FC2. On a P4 2.4GHz I have listened to the music on hold from both xlite and a budgetone 102 and it sounds the same from both. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: How do I match a D? (Was: RE: [Asterisk-Users] In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
David Brodbeck wrote: -Original Message- From: David Brodbeck [mailto:[EMAIL PROTECTED] Okay, the problem appears to be that I'm tone deaf. ;) I finally thought to turn on debugging on the channel. The PBX is sending D, not *. The programmer of the previous voice mail system (whose configuration I was cribbing from) seems to have made the same mistake. Is there some trick for matching the letter tones? I added this extension: exten = D,1,Goto(bye,s,1) But it doesn't trigger, even though I see this debugging output when I hang up: [ TYPE: DTMF (1) SUBCLASS: D (68) ] [Zap/1-1] ___ I'm prbably stupid, but wont this do what you want? exten = 1,1,Goto(bye,s,1) -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SIPphone won't cooperate
When attempting to call one of the example numbers, like 17474745000, I
only get 488 Not Acceptable Here. It works fine when I configure the
softphone (Xten X-Lite) to use sipphone's server directly. Am I missing
something? Here's my relevant config sections:
sip.conf:
in [general]:
register = 17472442457:[EMAIL PROTECTED]
[sipphone]
type=friend
host=proxy01.sipphone.com
username=17472442457
secret=mypassword
fromuser=17472442457
fromdomain=proxy01.sipphone.com
;also tried it with these
;disallow=all
;allow=ulaw
;allow=alaw
;allow=g729
extensions.conf:
[default]
exten = _1747NXX,1,SetCallerID(${SIPPHONENUMBER})
exten = _1747NXX,2,SetCIDName(${NAME})
exten = _1747NXX,4,Dial(SIP/[EMAIL PROTECTED])
exten = _1747NXX,5,Playback(invalid)
exten = _1747NXX,6,Hangup
And some sip debug output:
Sip read:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 142.59.15.250:5060;branch=z9hG4bK0b1543f4
From: Christopher sip:[EMAIL PROTECTED];tag=as5fc6d603
To: sip:[EMAIL PROTECTED];tag=19E038902017
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
Server: Sip EXpress router (0.8.14-2 (i386/linux))
Content-Length: 0
Warning: 392 198.65.166.131:5060 Noisy feedback tells: pid=17299
req_src_ip=142.59.15.250 req_src_port=5060
in_uri=sip:[EMAIL PROTECTED]
out_uri=sip:[EMAIL PROTECTED] via_cnt==1
9 headers, 0 lines
Sip read:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 142.59.15.250:5060;branch=z9hG4bK0b1543f4
From: Christopher sip:[EMAIL PROTECTED];tag=as5fc6d603
To: sip:[EMAIL PROTECTED];tag=19E038902017
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
Contact: sip:[EMAIL PROTECTED]
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Content-Length: 0
Warning: 392 198.65.166.130:5060 Noisy feedback tells: pid=28177
req_src_ip=198.65.166.131 req_src_port=5060
in_uri=sip:[EMAIL PROTECTED]
out_uri=sip:[EMAIL PROTECTED] via_cnt==0
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Re: [Asterisk-Users] add_pppd dialout problems
On Wed, 2005-02-09 at 10:20 +0200, Roger Wrethman wrote: Hi I am trying to get app_pppd to make an outgoing call to my ISP. Has anybody got this to work yet? Any reason you can't use a .call file to initiate the call? And just a simple reminder, it has to be ISDN. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID Question
I just discontinued my BV service, however CID was working just fine from a 303 line on cvs head. If you post the relavent sections of exentensions.conf and sip.conf, we might be able to suggest a couple of things. Outbound callerid via BV to the pstn will not show anything more then the number assigned to your servcie. When queried on that, BV indicated that setting the callerid name was a legal issue. Not sure why they believe that, but its their words not mine. Matt, I have the same issue and someone told me on IRC that broadvoice did not allow the caller ID to be changed like you want. If you do get this to work please share with the group so we can benefit. Thanks! Randy Matt Schwartz wrote: How do I get the incoming caller id to work correctly? I have a broadvoice line going into my asterisk box. My dial plan then routes the call to extension 1000. However, instead of the caller id from the incoming call, I see the caller id number 1000 from the extension? How do I correct this. In other words, I want to see XXX-XXX- from the incoming call and not the number of my extension on my caller display. Thanks much! Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] calling problem in cvs verison on fedora core2
hello any one using cvs version of asterisk(realtime addons). i have defined two users 2000 and 3000 in sip.conf. after that when i try to call 2000 from 3000 or try to call 3000 from 2000 it is giving me 404 Not Found error. Found user '2000' Looking for 3000 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.147;branch=z9hG4bKc0a800930131c9b1420a015c4a64c541000a From: rootsip:[EMAIL PROTECTED];tag=4145368953696567536 To: sip:[EMAIL PROTECTED];tag=as3521c065 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.147:5060 __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem using TDM400P FXS card
After having spent several time to look for any solution for my problem, I decided to write here. Here is the problem got a Digium X101 FXO card and Digium TDM400P alias freshmaker (1 fxs module on it on first port ) in my asterisk box. The X101 works perfectly. The problem comes from the TDM400, when i plug a tested working phone on it , i get no dialtone; the goal is to connect a few old non-ip phones to the PBX. The modules zaptel and wctdm compiled and are loaded successfully. Here are some infos : debian linux asterisk 1.0.5 cvs zaptel zaptel.conf fxoks=2 ; the x101 card uses channel 1 with fxsks=1 loadzone=fr default zone = fr Your /etc/zaptel.conf file should have fxoks=1 ; for the tdm card with a fxs module for telephone use fxsks=2 ; for the x100p card connected to a pstn line You might have to reverse the 1 and 2 in the above depending upon whether the x100p card or the tdm card is discovered/listed first. Then do a ztcfg - to see what's registered, etc. when i do a ztcfg it succesfully says me registered zone 2 But i cant see the phone in the zttool /etc/asterisk/zapata.conf [channels] language=fr context=default signalling=fxo_ks echocancel=yes group= 1 busydetect=yes channel = 2 Asterisk loads succesfully; and a zap show channels tells me : pseudocartezapfr 2cartezapfr But the problem is : i still dont have any dialtone on the phone. Anyone has an idea ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] limit iax calls
Good day all We have 2 asterisk servers,connected with iax2 and the phone via SIP They dont have a very big line so I want to restrict the call limet to 3 iax2 calls at a time,and for instance it the 4th call is made it will say something like all lines are being use try later Please help thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web based Asterisk management tool
Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbie questions
I've got one freaky budgetone that wont work using dhcp assign ip address via mac code. Basically I need to assign it an ip address using the phones internal web server. Maybe this was your problem as well. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Panco Sent: Tuesday, February 08, 2005 1:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] newbie questions i installed it the other day but from some reason can only get one of my budgetone 100's to register...any thoughts? I have tried upgrading firmare but that didn't seem to work. thanks in advance, ken Steve Rawlings wrote: Why not try [EMAIL PROTECTED], it only takes about an hour to install and be up and running with softphones like x-lite. This takes care of the os and asterisk in one cd. Steve - Original Message - From: Shaoul Jacobson - TELLINK [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 08, 2005 4:44 PM Subject: [Asterisk-Users] newbie questions Hi, I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards) 1. the distro I downloaded a free mandrake 10.0 - 3 CD's) but some packages seem missing (some C or C++ or python ...) (buy the full version ) maybe the latest fedora is more complete ? or easier to complete with rpmfind (I am green to linux too, but I open my windows gates to the tux) (bsd, debian are a bit too tech for me yet, no flaming please.) I prefer ready made rpm's or alike than compile AT THIS TIME. (I promise to improve over time) 2. download any rpm ? or I must download sources and 'make install' ? (I found one iso, but it seemed to require a pstn card) (RTFM a second / third time could is always a good option) 3. pure VoIP is it ok to use it in pure VoIP mode without any 'phone cards' ? all (most) settings samples I see include such cards. Needed or not ? 4. g729 not free. It seems that requires some licensing to digium. Can that be without using any 'card' (just VoIP) ? How to control the licenses then ? (I e-mailed them the question, but got no answer) accounting, cdr's, ... that's for later (first I have to be able to phone) regards, Shaoul Jacobson ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: calling problem in cvs verison on fedora core2
hello any one using cvs version of asterisk with realtime mysql addons. i am having a problem with it. i have defined two users 3000 and 2000. when i try to call 3000 from 2000 it is giving me '404 Not Found' and saying Found user '2000' and Looking for '3000' but when i try to call 2000 from 3000 it is saying Found user '3000' and Looking for 2000 in default and transmitting 404. what is the actual problem can any one guide me. thanks in advance -- Found user '2000' Looking for 3000 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.147;branch=z9hG4bKc0a800930131c9b1420a077500c267790023 From: rootsip:[EMAIL PROTECTED];tag=41469299352041650756 To: sip:[EMAIL PROTECTED];tag=as77991101 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -- Found user '3000' Looking for 2000 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.141;branch=z9hG4bKc0a8008d0131c9b1420ac2816358019a From: advcommsip:[EMAIL PROTECTED];tag=443815139 To: sip:[EMAIL PROTECTED];tag=as6fb2bbad Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -- __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk connected to pbx
David J Carter wrote: How do you want Switch to appear to Asterisk. 1. As an extension. Then use an FXS connection to a CO line input. The extension interface at the PBX will be supplying battery and dial tone. Therefore, you would want to use the FXO (red) daughter board on your TDM400P card. 2. As a CO line. Then use an FXO connection to an Extension output. The trunk interface at the PBX will be receiving battery and dial tone. Therefore, you would want to use the FXS (green) daughter board on your TDM400P. Having said that... If yours is an enterprise application then I would certainly investigate whether to use a T-1 interface between the two systems. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as VoIP gateway
I want to interconnect 2 pbx switches from to distinct location via an internet vpn using asterisk as VoiP gateways. The problem is what interfaces i must use between asterisk servers and pbx switch (FXO or FXS), and why? You must use FXS ports on *, then plug these in you PBX as phone lines. Then you can route calls thru thoses lines To your PBX, it will be just more lines available FXO (Foreign eXchange Office) is for connecting phone lines FXS (Foreign eXchance Station) is for connecting phones I hope I have the right terms in the definition, I just woke up and didn't finish my first coffee hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP jitter?
I've discovered that one of the pitfalls of wanting to try out the new jitter buffer is that you have to move to CVS head... Which isn't a biggie unless you've been using mysql without odbc. Am I dreaming or is the old type of non-odbc sql support eliminated from cvs head? Anyhow, just thought I'd put that out there as a warning for anyone considering patching head for the new jitter buffer. -mark On Feb 8, 2005, at 8:25 PM, Andrew Kohlsmith wrote: On February 8, 2005 07:53 pm, Steve Kann wrote: Glad it's working for you, Peter.. Seems to be working for me too; I'm using both 2532 and 3400. Your iax2 test pktloss patch moved my build to /opt/asterisk/vCVS which caused me some consternation but it's all good now. :-) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 Problem
On Mon, Feb 07, 2005 at 11:16:05PM -0600, Eric Rees wrote: Has anyone seen this message trying to install an TDM400.. spurious 8259A interrupt: IRQ7 Not sure what to has to do with your system, but I read somewhere that it is related to how the original interrupt controllers worked. If a card signalled an interrupt but then withdrew it before the host processor got around to reading the interrupt register, it would register as IRQ7. The kernel here is just pointing out that it got an IRQ7 but wasn't expecting it and it has now disabled it. If a module wants it it needs to register it. It's a harmless message... I'm fairly sure that IO-APIC only systems don't have this problem. -- Martijn van Oosterhout ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] giving up on x100p in Australia
OK, I've spent way more time than I wanted to on getting an x100p clone to work in Australia. I'm happy to consider other (more functional) options. Does anyone have an opinion on both the Sipura 3000 and other Digium cards (like the TDM400P)? I need something that works with no much fuzz. I know the Sipura 3000 is cheaper the the TDM400P card. All I need is to channel my POTS line into Asterisk. Nothing else! And I only have one line! The spa3k works just fine but it also has a few little quirks that may require you to devote some time to resolve. For example, it has a very large number of configurable options that allow you to do just about anything that you'd like, but the documentation on how to use some of the config options aren't very clear to someone that does not have extensive experience with the product. I also experience some small amount of what I believe is talk off where the spa3k intermitantly reacts to speech. It has not dropped the call or anything, but once in awhile will generate dtmf tones during a conversation as though it detected fax tones or something like that. It also seems to flip-flop between full-duplex and half- duplex audio occasionally. It seems female voices tend to be more objectionable then male voices in some cases. (I've played with two different spa's, about four different firmware versions, and have seen the same issues with each. But, maybe I'm being a little hyper-critical of voice quality issues.) Support for the spa products is basically limited to user-lists unless you're large enough to be considered a reseller of their products. They too are very secretive about existing problems. If you've watched the list for the last several months, you've probably noticed the TDM card has some issues as well. The most disturbing is the card goes out to lunch about every two weeks, which has required users to stop asterisk and reload the TDM drivers. Not everyone with the TDM card experience this problem, but those of us that do, have had the problem since the card originally came out. Digium is supposedly working at diagnosing/resolving the problem, but they appear to be highly secretive of that effort. It certainly does not help that the problem is so infrequent, making it extremely difficult for _anyone_ to try to diagnose the problem. (I've made some modifications to my TDM card to help diagnose/resolve the issue, but its way too early to tell if those mod's have actually impacted the problem.) The TDM card and the digium-sold x100p card use the same basic code and zaptel drivers, etc, so whatever issues (eg, echo cancellation) one might experience with the x100p aren't any better with the TDM card. (I swapped two x100p cards for a TDM card with four fxo pstn lines attached, and as of right now, it works fairly well except for the intermitant two-week failures.) I might also add that several of the digium cards have an issue with recording playing back voicemail messages. For whatever reason, there is a 10db loss in transmission levels when one attempts to play back voicemail messages via the pstn network (bug #2023). That bug has been around for about six/seven months and has never been addressed. The 10db loss has been measured (very professionally) but everyone at digium refuses to believe its a bug. Regardless of what its called, the 10db loss makes it almost impossible to listen to voicemails remotely unless you're in a very very very quiet room. Multiple * implementors have complained about this problem, so its not a simple parameter/configuration adjustment, etc. The voicemail playback volume is significantly worse for those asterisk systems that are further away from the Central Office. That is due to the pstn plant loss (unavoidable) on top of the 10db loss. So, if your asterisk box were 18,000 feet from the CO (as an example only), one would incure about a 7db plant loss in recording the voicemail, the 10db loss noted in bug 2023, plus an additional 7db loss listening to that voicemail via the pstn. That's a total of 24db loss, which makes listening extremely difficult. (If the asterisk machine were colocated in the Central Office, listening to voicemails would only incure the 10db bug, which does not have the same user impact as the 24db loss. So, those folks that are closer to the CO don't complain about this particular bug.) The 10db loss has been noted by users of the TDM card, T1 cards, etc, from digium; it is never seen when using other vendor's pstn gateways. The x100p (and clones) tend to be just okay for low volume soho use, but as you've already experienced, the majority of those cards were designed/targeted to the US market back when WinModems were popular. Some of the clone cards do have chip sets for the non- US market, but the people selling them won't tell you which chip set is on their for-sale cards. Therefore, implementing a clone in non-US (eg, non-600-ohm pstn networks) is a crap-shoot at best.
Re: [Asterisk-Users] Music on hold distorted
Mark, I have heard this problem. I'm not exactly sure what the cause is but check for any duplex mismatches between the phone and the * box. Hope this helps. Scott H Mark Benson wrote: Yesterday I setup music on hold by downloading and installing mpg123 r Now I have music on hold but it sounds terrible - clipping, buzzing, digital distortion, and its too loud (which probably isn't helping) and I'm just running it thru the 'default' line in music onhold.conf line default = quietmp3:/var/lib/asterisk/mohmp3, with the default mp3s. This is a 1.0.3 box, running headless (no x desktop) on FC2. On a P4 2.4GHz I have listened to the music on hold from both xlite and a budgetone 102 and it sounds the same from both. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Payload Type Compatability
We are having problems with DTMF generation with our supplier of IP to PSTN call termination. Their (Entice) soft switch is looking for RFC2833 payload type of 99 but Asterisk is using RFC2833 payload type 101. We are specifically having problems being able to access IVR menus and voice-mail. Does anyone have any ideas on how we can change the RFC2833 payload type to 99 so we can work with their soft switch? The Entice soft switch uses SIP Regards Norman Howlett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] what should SDP show in c=
Hi I am running mediaproxy infront of asterisk, with SER xlite SER ===asterisk (voicemail) || || || Mediaproxy If xlite is behind NAT or not, the mediaproxy replaces the c= header in the SDP part with th IP address of the mediaproxy (tks to Java for help with that) Now at the asterisk debug side, I guess the c= should also have the mediaproxy IP address in it, but it has that of the SER server instead, which is the reason (I guess) that I can leave a message on asterisk, but not here a pre-recorded prompt, even though the debug shows it as playing, any pointers as to how to change this. tks Iqbal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fastagi question
On Tue, 8 Feb 2005 22:09:59 -0800 (PST), Paul Chan [EMAIL PROTECTED] wrote: Hi All, I have a question about Fastagi because I can't get it to work for some reason. Everytime I execute the fastagi command, i get an error: my extensions.conf: .. exten = 1000,1,agi(agi://some_ip_address) .. try this exten = 1000,1,agi(agi://some_ip_address:some_port) Here is the exact line from my extensions.conf I am running on 1.0.5 against a JAGI server as well. exten = 5282,2,agi(agi://10.10.2.250:4573) Hope this helps, -Chuji ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX = FWD down again?
Can anybody confirm if IAX on FWD is down again? I can not register IAX with FWD. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX = FWD down again?
Joseph wrote: Can anybody confirm if IAX on FWD is down again? I can not register IAX with FWD. I got fed up with the yo-yo, which then led me to dump fwd and install asterisk and start playing with inter-asterisk routing via e164.org... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on hold distorted
I installed mpg123.0.59s and that was nasty so installed 0.59r but it was still distorted, eventually deleted s and reinstalled r and after a few mins the music on hold sorted itself out - it just happend as I was testing it after reinstalling - weird - I had looked at the phone/asterisk settings but found nothing odd - anyhoo all sorted now. Cheers, Mark Scott Herrick wrote: Mark, I have heard this problem. I'm not exactly sure what the cause is but check for any duplex mismatches between the phone and the * box. Hope this helps. Scott H Mark Benson wrote: Yesterday I setup music on hold by downloading and installing mpg123 r Now I have music on hold but it sounds terrible - clipping, buzzing, digital distortion, and its too loud (which probably isn't helping) and I'm just running it thru the 'default' line in music onhold.conf line default = quietmp3:/var/lib/asterisk/mohmp3, with the default mp3s. This is a 1.0.3 box, running headless (no x desktop) on FC2. On a P4 2.4GHz I have listened to the music on hold from both xlite and a budgetone 102 and it sounds the same from both. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX = FWD down again?
No problems here - works fine. - Original Message - From: Joseph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 09, 2005 8:17 AM Subject: [Asterisk-Users] IAX = FWD down again? Can anybody confirm if IAX on FWD is down again? I can not register IAX with FWD. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0506-0, 02/08/2005 Tested on: 2/9/2005 8:25:51 AM avast! - copyright (c) 1988-2004 ALWIL Software. http://www.avast.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX = FWD down again?
Can anybody confirm if IAX on FWD is down again? I can not register IAX with FWD. Works fine for me. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk Compile Problem on Red Hat 9
I get the following error when trying to compile asterisk 1.05 on red hat 9. Is this the tarball available for download from the asterisk website? You might try CVS instead - try the CVS HEAD release: # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout zaptel libpri asterisk Or, if that doesn't do it, you can try CVS Stable # cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds When you compile, make sure you do a make clean, first, then make install If neither of these works, I might suggest trying a different OS - Red Hat 9 is no longer updated. If you're attached to red hat, you might try Tao Linux or Whitebox Linux - both are essentially Red Hat Enterprise, but they are free and both provide updates (security and otherwise). -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN `ls *.c` : invalid option Usage: /bin/sh [GNU long option] [option] ... *** You don't have mpg123 installed. You're going to need *** *** it if you want MusicOnHold *** BTW: You can get mpg123 here: http://www-ti.informatik.uni-tuebingen.de/~hippm/mpg123.html Download the 0.59r version. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web based Asterisk management tool
Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download at sourceforge and does exactly what you are looking for. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based Asterisk management tool Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem using TDM400P FXS card
After having spent several time to look for any solution for my problem, I decided to write here. Here is the problem got a Digium X101 FXO card and Digium TDM400P alias freshmaker (1 fxs module on it on first port ) in my asterisk box. The X101 works perfectly. The problem comes from the TDM400, when i plug a tested working phone on it , i get no dialtone; the goal is to connect a few old non-ip phones to the PBX. The modules zaptel and wctdm compiled and are loaded successfully. Here are some infos : debian linux asterisk 1.0.5 cvs zaptel zaptel.conf fxoks=2 ; the x101 card uses channel 1 with fxsks=1 loadzone=fr default zone = fr Your /etc/zaptel.conf file should have fxoks=1 ; for the tdm card with a fxs module for telephone use fxsks=2 ; for the x100p card connected to a pstn line You might have to reverse the 1 and 2 in the above depending upon whether the x100p card or the tdm card is discovered/listed first. Then do a ztcfg - to see what's registered, etc. when i do a ztcfg it succesfully says me registered zone 2 But i cant see the phone in the zttool /etc/asterisk/zapata.conf [channels] language=fr context=default signalling=fxo_ks echocancel=yes group= 1 busydetect=yes channel = 2 Asterisk loads succesfully; and a zap show channels tells me : pseudocartezapfr 2cartezapfr But the problem is : i still dont have any dialtone on the phone. Anyone has an idea ? I decided to take another approach to se if it s not my card that is faulty ; removed the X101 card , only let the TDM , I put the FXS module on another port (maybe the first is bad). Here are the infos : /etc/zaptel.conf fxoks=2 loadzone = fr defaultzone=fr /etc/asterisk/zapata.conf [channels] language=fr context=cartezap ; context with a few extensions signalling=fxo_ks echocancel=yes busydetect=yes channel = 2 ztcfg -vvv gives me : Channel 02: FXO Kewlstart (Default) (Slaves: 02) 1 channels configured. in CLI of asterisk : zap show channels : Chan Extension Context Language MusicOnHold pseudocartezapfr 2cartezapfr The problem is i have still no diatone on the phone ( i m sure it works tested different ways) Any other Idea ? Thx in advance Nicolas Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk Compile Problem on Red Hat 9
I have both Stable version (asterisk-1.0-RC2) and the CVS version (asterisk v1-0-5) running on different Red Hat 9 boxes and there is no problem. I have only problem when I installed the oh323 driver (asterisk-oh323). Make sure you install Red Hat with required Package to run Asterisk. Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller Sent: mercredi 9 février 2005 15:37 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Asterisk Compile Problem on Red Hat 9 I get the following error when trying to compile asterisk 1.05 on red hat 9. Is this the tarball available for download from the asterisk website? You might try CVS instead - try the CVS HEAD release: # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login - the password is anoncvs. # cvs checkout zaptel libpri asterisk Or, if that doesn't do it, you can try CVS Stable # cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds When you compile, make sure you do a make clean, first, then make install If neither of these works, I might suggest trying a different OS - Red Hat 9 is no longer updated. If you're attached to red hat, you might try Tao Linux or Whitebox Linux - both are essentially Red Hat Enterprise, but they are free and both provide updates (security and otherwise). -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN `ls *.c` : invalid option Usage: /bin/sh [GNU long option] [option] ... *** You don't have mpg123 installed. You're going to need *** *** it if you want MusicOnHold *** BTW: You can get mpg123 here: http://www-ti.informatik.uni-tuebingen.de/~hippm/mpg123.html Download the 0.59r version. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web based Asterisk management tool
How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: mercredi 9 février 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download at sourceforge and does exactly what you are looking for. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based Asterisk management tool Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie help/pointers required - configure xlite with asterisk
I just want one of my incoming numbers to go to an IVR service that will allow me to select what I want. For example Press 1 for Mike, 2 for Karen, 3 for other, 9 for voicemail etc Just need to learn how to configure services now so that I can put a menu on one of my numbers! Elaborate please, I'm not clear on put a menu on one of my numbers. Give an example of what you want to accomplish and I'm sure many people here will help you. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.6 - Release Date: 07/02/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web based Asterisk management tool
That would be the AMP database, I don't know. Ping the amp list and find out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa Sent: Wednesday, February 09, 2005 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: mercredi 9 février 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download at sourceforge and does exactly what you are looking for. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based Asterisk management tool Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 codec for X-lite soft phone
Hello all, Is X-lite soft phone support G.729? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with meetMe
I try to use meetme app after reading manual i compile and install zaptel with ztdummy when i make lsmod i have ztdummy 2532 0 (unused) wcusb 20064 0 (unused) zaptel179168 4 [ztdummy wcusb] usb-uhci 26348 0 [ztdummy] usbcore51616 0 [wcusb usb-uhci] after it i recompile asterisk and after it i have app_meetme in meetme.conf i put ; Configuration file for MeetMe simple conference rooms ; for Asterisk of course. [rooms] ; Usage is conf = confno[,pin] conf = 4000 in extentions.conf [outgoing] exten = _4000,1,Answer exten = _4000,2,Wait(1) exten = _4000,3,MeetMe(4000,Mp) exten = _4000,4,Hungup when i try call from one ATA186 (using oh323) i hear message that i am alone in this conference and start play music on holdOn when i call from another ATA186 i hear in first telephone signal that somebody conect to conference and after it i don't hear anything i try to change codec G711u,G711a,G29 the situation is the same there is the logs from console -- Executing Answer(OH323/R14186, ) in new stack -- Executing Wait(OH323/R14186, 1) in new stack -- Executing MeetMe(OH323/R14186, 4000|Mp) in new stack == Parsing '/usr/local/asterisk_2/etc/asterisk/meetme.conf': Found -- Created MeetMe conference 1023 for conference '4000' -- Playing 'conf-onlyperson' (language 'en') -- Started music on hold, class 'default', on OH323/R14186 -- Executing Answer(OH323/R19096, ) in new stack -- Executing Wait(OH323/R19096, 1) in new stack -- Executing MeetMe(OH323/R19096, 4000|Mp) in new stack -- Stopped music on hold on OH323/R14186 Can anybody help me? thanks a lot ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] polycom soundpoint ip 300
hello,
I try to set up two lines per ip 300 phone,
registration is ok but i get Failure to authenticate
407 for subscribe.
Anybody could help me to configure Asterisk in order
to set instant message and presence ?
I've tried with Ondo sip server it's ok !
Regards
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; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
;
; The General category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without the ';')
; Note that this is different from the include command that includes contexts
within
; other contexts. The #include command works in all asterisk configuration
files.
;#include filename.conf
; The Globals category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
[sip]
exten = 100,1,Dial(SIP/100)
exten = 150,1,Dial(SIP/150)
exten = 200,1,Dial(SIP/200)
exten = 200,1,Dial(SIP/250)
;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/[EMAIL PROTECTED] to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED]
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
[general]
context=sip ; Default context for incoming calls
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
realm=home.net ; Realm for digest authentication
; defaults to asterisk
; Realms MUST be globally unique according to
RFC 3261
; Set this to your host name or domain name
port=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=192.168.0.2; IP address to bind to (0.0.0.0 binds to all)
[100]
type=friend
username=100
secret=100
fromuser=100
host=dynamic
context=sip
dtmfmode=rfc2833; Choices are inband, rfc2833, or info
progressinband=no ; Polycom phones don't work properly with
never
[150]
type=friend
username=150
secret=150
fromuser=150
host=dynamic
context=sip
dtmfmode=rfc2833; Choices are inband, rfc2833, or info
[200]
type=friend
username=200
fromuser=200
secret=200
host=dynamic
context=sip
dtmfmode=rfc2833; Choices are inband, rfc2833, or info
progressinband=no ; Polycom phones don't work properly with
never
[250]
type=friend
username=250
fromuser=250
secret=250
host=dynamic
context=sip
dtmfmode=rfc2833; Choices are inband, rfc2833, or info
progressinband=no ; Polycom phones don't work properly with
never
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[Asterisk-Users] VoIP guide for business people
I regularly get asked by business people, What's the point of VoIP?, so I put together a guide: http://integrics.com/tips/voip_for_business/ I'd be interested in hearing your feedback, and ideas for expansion. -- Alistair Cunningham, Integrics Ltd, Telephony, database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to load module iax.conf
Joseph wrote: When I try to load iax.conf I get (*-1.0.5): loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/iax.conf: cannot open shared object file: No such file or directory iax.conf is not something you can load. chan_iax2.so is, though. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip_notify.conf
Altus Snyman wrote: Good day all What is the file sip_notify.conf for Read the Mantis bugnotes about it when it was added. It's very useful. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] limit iax calls
Altus Snyman wrote: Good day all We have 2 asterisk servers,connected with iax2 and the phone via SIP They dont have a very big line so I want to restrict the call limet to 3 iax2 calls at a time,and for instance it the 4th call is made it will say something like all lines are being use try later Try show applications like group. Also see http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20CheckGroup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: polycom soundpoint ip 300
Hi Harry - I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. What version of the SIP firmware are you using? I've had success with 1.3.0, 1.3.1, 1.3.4, and 1.4.1. My sip.conf entries for my Polycom phones look like this: [12] type=friend username=12 secret=12 callerid=12 host=dynamic dtmfmode=inband context=no-callwaiting [EMAIL PROTECTED] disallow=all allow=ulaw Are you configuring directly on the phone, or using an FTP or TFTP server? Anybody could help me to configure Asterisk in order to set instant message and presence ? To the best of my knowledge the Presence feature of the Polycom phones does not work with Asterisk. I believe it only works with other IM clients. Hope this helps! Noah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] loader.c:301 __load_resource: libpt_linux_x86_r.so.1.8.1: cannot open shared object file... [solution found, but quick question]
All, I followed the channels/h323/README to the letter and everything does compile properly. When I start asterisk I get the following error: [chan_h323.so]Feb 9 10:30:51 WARNING[30700]: loader.c:301 __load_resource: libpt_linux_x86_r.so.1.8.1: cannot open shared object file: No such file or directory Running ldd, I see some missing libs: [EMAIL PROTECTED]:/usr/lib# ldd /usr/lib/asterisk/modules/chan_h323.so libdl.so.2 = /lib/libdl.so.2 (0x4003f000) libpt_linux_x86_r.so.1.8.1 = not found libh323_linux_x86_r.so.1.15.1 = not found libcrypto.so.0 = /usr/lib/libcrypto.so.0 (0x41072000) libssl.so.0 = /usr/lib/libssl.so.0 (0x41171000) libexpat.so.0 = /usr/lib/libexpat.so.0 (0x411a2000) libgcc_s.so.1 = /usr/lib/libgcc_s.so.1 (0x411c2000) libc.so.6 = /lib/libc.so.6 (0x411ca000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) libpthread.so.0 = /lib/libpthread.so.0 (0x412f9000) libasound.so.2 = /usr/lib/libasound.so.2 (0x4134a000) libresolv.so.2 = /lib/libresolv.so.2 (0x413fa000) libstdc++.so.5 = /usr/lib/libstdc++.so.5 (0x4140c000) libm.so.6 = /lib/libm.so.6 (0x414c3000) My fix was to create a symbolic link to my source in /usr/lib [EMAIL PROTECTED]:/usr/lib# ln -s /usr/local/src/pwlib/lib/libpt_linux_x86_r.so.1.8.1 [EMAIL PROTECTED]:/usr/lib# ln -s /usr/local/src/openh323/lib/libh323_linux_x86_r.so.1.15.1 [EMAIL PROTECTED]:/usr/lib# ldd /usr/lib/asterisk/modules/chan_h323.so libdl.so.2 = /lib/libdl.so.2 (0x4003f000) libpt_linux_x86_r.so.1.8.1 = /usr/lib/libpt_linux_x86_r.so.1.8.1 (0x40042000) libh323_linux_x86_r.so.1.15.1 = /usr/lib/libh323_linux_x86_r.so.1.15.1 (0x40401000) libcrypto.so.0 = /usr/lib/libcrypto.so.0 (0x41072000) libssl.so.0 = /usr/lib/libssl.so.0 (0x41171000) libexpat.so.0 = /usr/lib/libexpat.so.0 (0x411a2000) libgcc_s.so.1 = /usr/lib/libgcc_s.so.1 (0x411c2000) libc.so.6 = /lib/libc.so.6 (0x411ca000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) libpthread.so.0 = /lib/libpthread.so.0 (0x412f9000) libasound.so.2 = /usr/lib/libasound.so.2 (0x4134a000) libresolv.so.2 = /lib/libresolv.so.2 (0x413fa000) libstdc++.so.5 = /usr/lib/libstdc++.so.5 (0x4140c000) libm.so.6 = /lib/libm.so.6 (0x414c3000) My question is, how come the LD_LIBRARY_PATH defined in /etc/profile did not link the libs properly? PWLIBDIR=/usr/local/src/pwlib export PWLIBDIR OPENH323DIR=/usr/local/src/openh323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH Thanks inadvance --- Paul Belanger (mailto:[EMAIL PROTECTED]) Technical Support Specialist Cisco Certified Network Associate Pronexus Inc. - A Powerful Voice in Communication Solutions --- Tel: 613.271.8989 ext. 516 Fax: 613.271.8388 http://support.pronexus.com --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is there a Caller ID issue in the latest CVS Stable
Greetings, 2 nights ago I upgraded one of my remote servers to the latest CVS Stable, Asterisk CVS-v1-0-02/07/05-16:13:48, and my inbound and outbound caller ID stopped working. At first I thought it was my carrier, so Ive been yelling at them for the past day or so. I then upgraded my core network (about 8 asterisk servers) to the latest stable as of 3am last night, Asterisk CVS-v1-0-02/08/05-23:33:03, and now my inbound and outbound caller ID doesnt work for my entire network. It appears to be a bug with the latest Stable version of Asterisk. All calls, internal phone to phone, or external to phone calls, all the caller ID says is asterisk with no number or name. So the cidname and cidnum are both set to asterisk. When I dial outbound, the carrier wont accept alpha characters in the cidnum field and will set my calls as private, so all my outbound calls are going out as private/blocked. I dont want to downgrade back to my 10-26-04 version, but I need to know if this bug is being worked on? Help? Thanks, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] no caller ID presented from 12SP+
Are you running the latest stable? I think there's a bug with it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Sunday, February 06, 2005 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] no caller ID presented from 12SP+ Hi folks, I have got a Cisco 12SP+ working (thanks Derek!) but I'm having a minor issue with it. When I use it to call another desk I get no CallerID. The receiving phone diplsays asterisk as the CID. Below is the skinny.conf stanza. [2207] device=SEP00308062B006 version=P002L2J2 context=intern callerid=Luke's Room 2207 mailbox=2207 transfer=1 callwaiting=1 threewaycalling=1 line = 2207 Thanks Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] loader.c:301 __load_resource: libpt_linux_x86_r.so.1.8.1: cannot open shared object file... [solution found, but quick question]
Paul Belanger [EMAIL PROTECTED] wrote: [...] My question is, how come the LD_LIBRARY_PATH defined in /etc/profile did not link the libs properly? LD_LIBRARY_PATH is occasionally ignored for security reasons. If you wish to globally add a directory to the library search path, you should put it in /etc/ld.so.conf. You may want to re-run ldconfig afterwards to clean up and correct symlinks. -- The young always have the same problem - how to rebel and conform at the same time. They have now solved this by defying their parents and copying one another. - Quentin Crisp ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 codec for X-lite soft phone
you have to buy pro version for 729, $50. -Matthew - Original Message - From: Daniel Eboa [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 09, 2005 9:00 AM Subject: [Asterisk-Users] G.729 codec for X-lite soft phone Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: How do I match a D? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
-Original Message- From: Gilad Ben-Yossef [mailto:[EMAIL PROTECTED] I'm prbably stupid, but wont this do what you want? exten = 1,1,Goto(bye,s,1) No, because I wanted to match on D, not 1. Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't realize this because I wasn't waiting for message playback to finish. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX = FWD down again?
On Thu, 2005-02-10 at 01:22 +1100, Duane wrote: Joseph wrote: Can anybody confirm if IAX on FWD is down again? I can not register IAX with FWD. I got fed up with the yo-yo, which then led me to dump fwd and install asterisk and start playing with inter-asterisk routing via e164.org... I wander what is causing the problem, I was thinking that it was something on my part but I did not change any settings and IAX2 registry is: Host UsernamePerceived Refresh State 65.39.205.121:4569491581 Unregistered 60 Request Sent -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voip as a secure service?
Michael Graves wrote: Hi All, I was just reading through Info Week while on a flight and happened upon an brief piece about a new VOIP security intiative worked up by a handful of the usual suspects; Alcatel, SMU, NIST, Symantec, etc. All of this begs the question of can't we get just do this as a user community? I understand that the Zultys phone, which I own several, support AES encryption of the RTP stream. There's been some preliminary work on encrypted IAX2 streams. We're moving in the right direction. How does effort towards srtp and the like compare/contrast to VPN based connectivity? Forgive the simplistic question, but how compute intensive is a VPN? Could an ITSP like VoipJet or Nufone offer termination over a VPN based connection for a premium rate? Their rates are very low already. Would someone/anyone care enough to pay a premium for the service? Maybe double the usual rate? Is that adequate incentive for ITSPs to offer the service? Are there legal/CALEA implications? I understand that at present secure voip has been fundamentally the domain of larger enterprises that need to secure geographically disperse organisations over owned or hired WAN infrastructure. I work for an SME, yet my primary access to my email is via a PPTP tunnel to an Exchange server some 3000 miles away. If we wanted to deploy our own infrastructure we could use secure voip between locations, but not through ITSPs...at least not yet. It's a very interesting area, one that could be an interesting business in the near future. It would be pretty easy for these service providers to allow users to use a VPN tunnel to connect to them; I think the compute expense of this probably would be less than the compute expense of codec translation.. Setting up OpenVPN for users is pretty easy as well.. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Interaction: Agents and Extensions
Michael Welter wrote: SER newbie here. Why do you need Asterisk for Sip-SIP setup? And if there is a reinvite, is that for the RTP stream only or for the SIP transactions as well? Will you lose the BYE transaction if there is a reinvite? Also, how many SIP registrations do you expect to maintain on each SER box? Good questions. I need asterisk for SIP-SIP setups because it can do interesting things with the call. Like capture digits, play prompts,listen for key codes during a live coversation to transfer and other neat things. As far as I know (and please correct me if I am wrong!) but SER can't do these kinds of things since it's totally disconnected from the RTP Stream. As for the reinvite, I'm *hoping* that it's just the establishment of the RTP stream and not the entire signalling path. However I could be wrong. Looking for feedback from anyone who knows better. I'm expecting SER to maintain thousands of registrations. To quote the SER Admin manual: With a $3000 dual-CPU PC, the SIP Express Router is able to power IP telephony services in an area as large as the Bay Area during peak hours. Even on an IPAQ PDA, the server withstands 150 calls per second (CPS)! That is why you want SER. The two together provide features + scalability. I'm still not sure how to provide services that interact one phone with another phone's RTP stream. Like call pickup. How can I pickup a call on another asterisk server? Hmm Hm -Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Interaction: Agents and Extensions
Why not let asterisk be your PSTN GW? It is in our case, just throwing out my $0.02. Most of the cases I can think of I can get around. The one I can't seem to figure out is 'Agents'. Agents will need to login/logout using 1 number. I can forward that number from SER to asterisk by looking for it, no biggie there. The problem lies in having 1 SER box and many * boxes. If Agent 1 logs in at Asterisk-1, and agent 2 logs in at Asterisk-4, and a call comes into the queue at Asterisk-3, what then? Can Asterisk-3 see that the 2 agents are loggedin? Asterisk boxes need to share states and that seems difficult to accomplish. -Matthew - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 08, 2005 4:33 PM Subject: Re: [Asterisk-Users] SER Interaction: Agents and Extensions Matthew Boehm wrote: With all of these caveats, it seems to me that a SER-Asterisk solution isn't that great. If anyone else out there can show me otherwise... Thanks, Matthew I think you might be missing the point here. SER is a raw SIP processor. So for a second throw everything you know about Asterisk + SIP out the window and go back to vanilla SIP. Getting used to a B2BUA in the call path kinda beats some of the raw power of SIP up. Think of how a SIP URI is formed. That domain portion is kinda like a context, right? furthermore, SER can do stuff with that. I'm doing my own eval with SER for a very large deployment. But I'm just getting started. I had SER running about a year ago, but it's been about that long since I really toyed with it. One of the call flows I'm about to try is: PSTN GW - SER - Asterisk Transfer/re-invite - SER - Phone The idea is that SER manages my PSTN gateway. I can always just stack more Asterisk servers on, SER I'll never really need to expand (there is a redundant SER Server, removing the need for clustering). Then the call gets sent to asterisk for smart call processing, however actual setup of the media gets resent back to SER. I'm not sure if I'll be able to do this, but I may be able to do it with re-invites. Any thoughts? -Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: How do I match a D? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
David Brodbeck wrote: Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't realize this because I wasn't waiting for message playback to finish. Please enter a bug in Mantis for this; it should very likely be corrected, as I don't see any reason to ignore A-D in Background(). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Interaction: Agents and Extensions
[EMAIL PROTECTED] wrote: I think you might be missing the point here. SER is a raw SIP processor. So for a second throw everything you know about Asterisk + SIP out the window and go back to vanilla SIP. Getting used to a B2BUA in the call path kinda beats some of the raw power of SIP up. Think of how a SIP URI is formed. That domain portion is kinda like a context, right? furthermore, SER can do stuff with that. I'm doing my own eval with SER for a very large deployment. But I'm just getting started. I had SER running about a year ago, but it's been about that long since I really toyed with it. One of the call flows I'm about to try is: PSTN GW - SER - Asterisk Transfer/re-invite - SER - Phone The idea is that SER manages my PSTN gateway. I can always just stack more Asterisk servers on, SER I'll never really need to expand (there is a redundant SER Server, removing the need for clustering). Then the call gets sent to asterisk for smart call processing, however actual setup of the media gets resent back to SER. I'm not sure if I'll be able to do this, but I may be able to do it with re-invites. Any thoughts? -Brett SER newbie here. Why do you need Asterisk for Sip-SIP setup? And if there is a reinvite, is that for the RTP stream only or for the SIP transactions as well? Will you lose the BYE transaction if there is a reinvite? Also, how many SIP registrations do you expect to maintain on each SER box? I was hoping to maintain somewhere around 1000 registrations per SER box. I'm pretty sure Asterisk would get sluggish maintaining that many SIP registrations. My biggest concern is this: bandwidth. If we get a customer who has 20 stations (aka UAs) but only wants to be able to have 4 inc/outgoing PSTN conversations, then we only need G729 bandwidth * 4 between our main asterisk server and this customer. If any office-mates want to 4-digit dial eachother, those conversations should not traverse the main bandwidth. It saves my company time, money and effort because we could sign up 14 more customers like the one above and still only need 1 T1 between the asterisk box and the internet. But if all inter-office communications traversed the asterisk box, we would need alot more bandwidth. Our first solution to this was to put asterisk boxes at any customer location that needed more than 10 UAs. But then we run into a billing problem cause those 10 UAs would register to the local asterisk box and not the main server and I would not get account code information from each UA cause the main server's cdr's would show the call comming from same location no matter which UA made the call. (can you say run-on sentence?) If I can get re-invites working great, then I should have no worries about inter-office communication. SER should be able to connect 2 office-mates to eachother even if they are both behind the same NAT, or behind different NATs. -Matthe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web based Asterisk management tool
Thanks for the responses, isn't [EMAIL PROTECTED] an pre-configured distribution for small systems?? I am looking for an open source web management tool to use on any size asterisk server (even ones that are already up and running) the user base could be anything between small and large with many external lines, Ive looked at AMP, is it free ? and are there any alternatives or is AMP the only open source web management tool ? -Original Message- From: dean collins [mailto:[EMAIL PROTECTED] Sent: 09 February 2005 15:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool That would be the AMP database, I don't know. Ping the amp list and find out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa Sent: Wednesday, February 09, 2005 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: mercredi 9 février 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download at sourceforge and does exactly what you are looking for. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based Asterisk management tool Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Interaction: Agents and Extensions
I'm still not sure how to provide services that interact one phone with another phone's RTP stream. Like call pickup. How can I pickup a call on another asterisk server? Hmm Hm -Brett Aw crap. I completly forgot about call pickup. Good point. If you have a call come into one of your many asterisk boxes that rings ext 101, how can you at ext 102 pickup that call if going SER-Asterisk? Yes, you can forward the *8 to asterisk, but who knows which asterisk box will get that *8? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: DTMF Payload Type Compatability
Norman == Norman Howlett [EMAIL PROTECTED] writes: Norman We are having problems with DTMF generation with our supplier Norman of IP to PSTN call termination. Their (Entice) soft switch is Norman looking for RFC2833 payload type of 99 but Asterisk is using Norman RFC2833 payload type 101. From a quick glance at RFC2833, it looks like the payload type is chosen dynamically. How is it announced by the peer? (use sip debug) Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: How do I match a D? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
-Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] David Brodbeck wrote: Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't realize this because I wasn't waiting for message playback to finish. Please enter a bug in Mantis for this; it should very likely be corrected, as I don't see any reason to ignore A-D in Background(). http://bugs.digium.com/bug_view_page.php?bug_id=0003538 A simple fix is included, though I don't have a deep understanding of the Asterisk code, so it's possible it has side effects I'm not aware of. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with running ztcfg
Hi All, I just installed Asterisk 1.0.5, and the installation went fine (I ran modprobe zaptel and modprobe wcfxo). However, when I ran ztcfg I get the following: ioctl(ZT_LOADZONE) failed: Invalid argument Notice: Configuration file is /etc/zaptel.conf line 135: Unable to register tone zone 'us' After that I ran Asterisk and it seem to started ok, except that it won't pick up any calls. Has anyone seen this or know what could be the problem? Thanks for your help in advance! __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Interaction: Agents and Extensions
Matthew Boehm wrote: If I can get re-invites working great, then I should have no worries about inter-office communication. SER should be able to connect 2 office-mates to eachother even if they are both behind the same NAT, or behind different NATs. You can accomplish that with a low-end box running siproxd at the site where the UAs are. The registrations still go into Asterisk (via a proxy), but the media can stay local when appropriate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: How do I match a D? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
On Wed, 2005-02-09 at 09:15 -0700, Kevin P. Fleming wrote: David Brodbeck wrote: Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't realize this because I wasn't waiting for message playback to finish. Please enter a bug in Mantis for this; it should very likely be corrected, as I don't see any reason to ignore A-D in Background(). I would suggest that that be added as a optional switch to background to get the extra digits. While I do not know if A-D are easy to hit with Talk-Off, it is 4 more potential digits to hit. Also it would be less surprise to a user to be required a flag to Background() to get those than it would be to diagnose why you occasionally get dumped in an invalid extension. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Interaction: Agents and Extensions
Matthew Boehm wrote:
Why not let asterisk be your PSTN GW? It is in our case, just throwing out
my $0.02.
Most of the cases I can think of I can get around. The one I can't seem to
figure out is 'Agents'.
Agents will need to login/logout using 1 number. I can forward that number
from SER to asterisk by looking for it, no biggie there. The problem lies in
having 1 SER box and many * boxes. If Agent 1 logs in at Asterisk-1, and
agent 2 logs in at Asterisk-4, and a call comes into the queue at
Asterisk-3, what then? Can Asterisk-3 see that the 2 agents are loggedin?
Asterisk boxes need to share states and that seems difficult to accomplish.
-Matthew
Because my PSTN network is SS7 connected. :) and it's a decently large
interconnection. Today I'd need somewhere on the order of 100-150 T1s
and that's really just initial numbers.
I've looked at the SS7 solution and it's not quite where i need it yet..
but maybe soon.
As for your problem, I think you can still do it. The asterisk box
managing the agent needs to manage the dial in queue as well.. So calls
coming in must hit that box, but I don't think anything says agents need
to be connected to it.. I haven't tried it, but can you register an
agent on a Local channel? If so, there is a lot of stuff you can do
Like AddQueueMember(techsupport|Local/[EMAIL PROTECTED]) I bet
that works.
I think most people probably do something like:
AddQueueMember(techsupport|SIP/${CALLERIDNUM}), but I bet you can put
any valid channel name in there.
-Brett
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Re: [Asterisk-Users] SER Interaction: Agents and Extensions
Matthew Boehm wrote: I'm still not sure how to provide services that interact one phone with another phone's RTP stream. Like call pickup. How can I pickup a call on another asterisk server? Hmm Hm -Brett Aw crap. I completly forgot about call pickup. Good point. If you have a call come into one of your many asterisk boxes that rings ext 101, how can you at ext 102 pickup that call if going SER-Asterisk? Yes, you can forward the *8 to asterisk, but who knows which asterisk box will get that *8? -Matthew Actually, I'm thinking we can do that too.. But I need to test this in a lab.. So *8 gets registered to an extension in a particular context right? So SER sends the call (the *8 call) to the appropriate server (hmm how do we determine this...) and it hits a particular sip registration that maps it to a context that has access to that *8 (and callgroup for that matter). The trick is knowing what server to send that *8 sequence to.. perhaps we can use a DB to indicate where ringing channels are occuring for which callgroups.. look at the one with the latest timestamp and send the call there.. Of course, this would require either a manager interface server or a dialplan app that could access what callgroups a call is a member of.. Problem with that is we don't know until the call is passed to the channel. Perhaps if we populated (manually) the callgroup to the db, it would work.. hmm. This will have to be a test in my lab. I wish there was a way to query the callgroups a channel belongs to. -Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WORKING but How Long! IAX = FWD down again?
[snip] It is working again :-) It appears something broke after recent upgrades (on Gentoo) as I wasn't even able to dial VOIPJET, though I don't know what was broken :-/ I re-emerged asterisk and it is working gain. -- #Joseph I wander what is causing the problem, I was thinking that it was something on my part but I did not change any settings and IAX2 registry is: Host UsernamePerceived Refresh State 65.39.205.121:4569491581 Unregistered 60 Request Sent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Asterisk as sip user agent with more than one device
Here is the situation at hand, awaiting input from the collective minds
of the asterisk-users list:
1. I have Asterisk registered as a SIP user agent to another Asterisk
(which stands for the real provider, to another location).
register = user:[EMAIL PROTECTED]
2. I have a peer section for the provider.
[provider]
type=peer
host=10.0.0.1
fromuser=user
3. I have two phones (FXS) on the Asterisk box. They work fine, and can
call each other, as well as receive calls through SIP from other
agents.
4. When I call from the phones through Asterisk to the (simulated)
provider, I get Failure to authenticate user. The line that places
the call to the provider is:
exten = _4.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
5. The problem seems to be that Asterisk sets the following as the
From: line in the SIP packets:
From: asterisk sip:[EMAIL PROTECTED]
And the (simulated by Asterisk) provider uses asterisk for the
authentication and fails...
I have tried setting the CALLERID before the Dial, but the string
remains the same. That is the following extension plan has no effect on
the From header:
exten = _4.,1,SetGlobalVar(CALLERID=user)
exten = _4.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
And here are my questions:
How can I force asterisk to be a string of my choosing instead of the
original caller id? Is it even possible (without coding it)? Is it
proper behaviour?
OR
Is the behaviour of Asterisk at 10.0.0.1 (simulating the provider's
system) correct? In the 'From: xxx sip:[EMAIL PROTECTED]', is xxx
really to be used over yyy?
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RE: How do I match a D? (Was: RE: [Asterisk-Users] In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
I'll ask a stupid question, how does a user hit an alpha letter from his touchtone? I know that the Cisco 7960's support entering alpha letters, and it could potentially do it (maybe), but how does the average end user enter an a b c or d from their touchtone phone? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, February 09, 2005 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: How do I match a D? (Was: RE: [Asterisk-Users] In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?) On Wed, 2005-02-09 at 09:15 -0700, Kevin P. Fleming wrote: David Brodbeck wrote: Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't realize this because I wasn't waiting for message playback to finish. Please enter a bug in Mantis for this; it should very likely be corrected, as I don't see any reason to ignore A-D in Background(). I would suggest that that be added as a optional switch to background to get the extra digits. While I do not know if A-D are easy to hit with Talk-Off, it is 4 more potential digits to hit. Also it would be less surprise to a user to be required a flag to Background() to get those than it would be to diagnose why you occasionally get dumped in an invalid extension. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: How do I match a D? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
Steven Critchfield wrote: I would suggest that that be added as a optional switch to background to get the extra digits. While I do not know if A-D are easy to hit with Talk-Off, it is 4 more potential digits to hit. Also it would be less surprise to a user to be required a flag to Background() to get those than it would be to diagnose why you occasionally get dumped in an invalid extension. Unless I'm mistaken, if you don't have a D extension defined in the context, Background() is going to ignore the D keypress anyway. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: How do I match a D? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
David Brodbeck wrote: -Original Message- From: Gilad Ben-Yossef [mailto:[EMAIL PROTECTED] I'm prbably stupid, but wont this do what you want? exten = 1,1,Goto(bye,s,1) No, because I wanted to match on D, not 1. I am stupid - I thought you meant the DTMF for the D button (aka 3DEF) :-) Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't realize this because I wasn't waiting for message playback to finish. Cool, I didn't realize you can match on these DTMF signals as Israeli phones usually don't have them :-) Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Interaction: Agents and Extensions
[EMAIL PROTECTED] wrote:
I think most people probably do something like:
AddQueueMember(techsupport|SIP/${CALLERIDNUM}), but I bet you can put
any valid channel name in there.
And you would win that bet :-)
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[Asterisk-Users] SIP ActiveX
I search a ActiveX to develop one softphone SIP with codec G723. Who can help me? Thank´s João Carlos Moura ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web based Asterisk management tool
You need to go back and reread. It is just pretty much an asterisk configuration tool (ok some minor things in the backend but it's the best out there). AMP is available for free download but they make their money by offering support. [EMAIL PROTECTED] is just an automated way of installing AMP and FOP and Web Meetme. If you really have a need to support thousands of extensions as you suggest then you should really go back and learn how to program asterisk with a database yourself from scratch. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Web based Asterisk management tool Thanks for the responses, isn't [EMAIL PROTECTED] an pre-configured distribution for small systems?? I am looking for an open source web management tool to use on any size asterisk server (even ones that are already up and running) the user base could be anything between small and large with many external lines, Ive looked at AMP, is it free ? and are there any alternatives or is AMP the only open source web management tool ? -Original Message- From: dean collins [mailto:[EMAIL PROTECTED] Sent: 09 February 2005 15:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool That would be the AMP database, I don't know. Ping the amp list and find out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa Sent: Wednesday, February 09, 2005 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: mercredi 9 février 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download at sourceforge and does exactly what you are looking for. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based Asterisk management tool Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER Integration together
I know FWD uses a SER/Asterisk combo, and I keep hearing about the massive benefits, however, my initial playing around in SERs configuration indicates its NOTHING like Asterisk at all, and almost 5x as difficult to understand and configure. But thats only after a few hours of playing with it. Im interested in learning SER more, especially the integration with Asterisk. Is there a good how-to guide with lots of examples on how to accomplish this optimal setup? Anybody got any good links or resources or can help me with examples? Right now I have Asterisk doing all the work and its getting frustrating w/ Quality issues left and right and such. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: How do I match a D? (Was: RE: [Asterisk-Users] In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
Paul Rodan wrote: I'll ask a stupid question, how does a user hit an alpha letter from his touchtone? I know that the Cisco 7960's support entering alpha letters, and it could potentially do it (maybe), but how does the average end user enter an a b c or d from their touchtone phone? Some phones actually have 'A', 'B', and 'C' keys - a few even have a 'D' key. Haven't seen any, but I know they do exist. -Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Background() ignoring digits A-D (Was: RE: How do I match a D?)
-Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] On Wed, 2005-02-09 at 09:15 -0700, Kevin P. Fleming wrote: David Brodbeck wrote: Anyway, I figured it out. The extension was working, but Background() ignores the tones A through D by default. I didn't realize this because I wasn't waiting for message playback to finish. Please enter a bug in Mantis for this; it should very likely be corrected, as I don't see any reason to ignore A-D in Background(). I would suggest that that be added as a optional switch to background to get the extra digits. While I do not know if A-D are easy to hit with Talk-Off, it is 4 more potential digits to hit. Also it would be less surprise to a user to be required a flag to Background() to get those than it would be to diagnose why you occasionally get dumped in an invalid extension. At very least it should be documented. It was very confusing to see the digits come up in the debug output without provoking any response from *. Ideally, I suppose, Background() would ignore *any* invalid digits, but that would require it to understand the dialplan, which is probably impractical. It would minimize the chances of talk-off, though. (Why are people talking during the menus, anyhow? ;) ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: How do I match a D? (Was: RE: [Asterisk-Users] In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
Paul Rodan wrote: I know that the Cisco 7960's support entering alpha letters, and it could potentially do it (maybe), but how does the average end user enter an a b c or d from their touchtone phone? They don't. Most phones (99.9%) don't have any way to generate DTMF A through D. There are test sets that do, and of course softphones could easily do so. These tones could also be generated by automated applications, although I don't know why one of them would be talking to Background() on an Asterisk server... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Interaction: Agents and Extensions
On Wed, 9 Feb 2005, Matthew Boehm wrote: I'm still not sure how to provide services that interact one phone with another phone's RTP stream. Like call pickup. How can I pickup a call on another asterisk server? Hmm Hm Aw crap. I completly forgot about call pickup. Good point. If you have a call come into one of your many asterisk boxes that rings ext 101, how can you at ext 102 pickup that call if going SER-Asterisk? Yes, you can forward the *8 to asterisk, but who knows which asterisk box will get that *8? Perhaps the app_intercept patch would work better? It is a lot less of a kludge and more flexible than the *8 that is in Asterisk. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVS Stable
Hello, 2 nights ago I upgraded one of my remote servers to the latest CVS Stable, Asterisk CVS-v1-0-02/07/05-16:13:48, and my inbound and outbound caller ID stopped working. My suggestion would be to downgrade to 1.0.3. It might solve your problem. There were a number of changes in callerid handling in the last couple of weeks. Many manager based applications stopped working because of them. Maybe your setup is affected too. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web based Asterisk management tool
Gary, contact me off-list. I have developed a GUI Windows based tool that will allow management of configuration files if you are running RealTime. It supports sip,iax,extensions,voicemail currently. It will also display CDR's and the various schema's used by the Asterisk box. Tom Chandler [EMAIL PROTECTED] - Original Message - From: dean collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 09, 2005 10:47 AM Subject: RE: [Asterisk-Users] Web based Asterisk management tool You need to go back and reread. It is just pretty much an asterisk configuration tool (ok some minor things in the backend but it's the best out there). AMP is available for free download but they make their money by offering support. [EMAIL PROTECTED] is just an automated way of installing AMP and FOP and Web Meetme. If you really have a need to support thousands of extensions as you suggest then you should really go back and learn how to program asterisk with a database yourself from scratch. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Web based Asterisk management tool Thanks for the responses, isn't [EMAIL PROTECTED] an pre-configured distribution for small systems?? I am looking for an open source web management tool to use on any size asterisk server (even ones that are already up and running) the user base could be anything between small and large with many external lines, Ive looked at AMP, is it free ? and are there any alternatives or is AMP the only open source web management tool ? -Original Message- From: dean collins [mailto:[EMAIL PROTECTED] Sent: 09 February 2005 15:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool That would be the AMP database, I don't know. Ping the amp list and find out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa Sent: Wednesday, February 09, 2005 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: mercredi 9 février 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download at sourceforge and does exactly what you are looking for. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based Asterisk management tool Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
Re: [Asterisk-Users] SER Interaction: Agents and Extensions
Peter Svensson wrote: On Wed, 9 Feb 2005, Matthew Boehm wrote: I'm still not sure how to provide services that interact one phone with another phone's RTP stream. Like call pickup. How can I pickup a call on another asterisk server? Hmm Hm Aw crap. I completly forgot about call pickup. Good point. If you have a call come into one of your many asterisk boxes that rings ext 101, how can you at ext 102 pickup that call if going SER-Asterisk? Yes, you can forward the *8 to asterisk, but who knows which asterisk box will get that *8? Perhaps the app_intercept patch would work better? It is a lot less of a kludge and more flexible than the *8 that is in Asterisk. Oh right.. I remember seeing that.. yeah that looked a whole lot more elegant than *8. Why isn't it in HEAD? http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002692 This app is an external verison of what *8 does for Zap channels. set a var called INTERCEPT_TAG before you dial somewhere and another channel can execute Intercept(xyz) to take the unanswered call. [Synopsis]: Intercept an unaswered Call. [Description]: Intercept([Channel Name|varmatch|auto]) Intercept an unanswered channel: A) who's name begins with Channel Name B) Containing the variable INTERCEPT_TAG matching varmatch C) The first encountered unanswered channel if 'auto' was specified. Nice.. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] announcement: astfax 1.0
On Wednesday 09 February 2005 12:53 am, Peer Oliver Schmidt wrote: Steven Critchfield wrote: astfax allows you to create an email to fax gateway. Are we going to see some integration of astfax with Courier-MTA/IMAP? If you look at the instructions, you only need to make a some form of default matching rule to catch all phone numbers and then pipe the resulting message to the given command. You were right in suggesting to look at the instructions. I did not even look at the package yet. From reading the announcement it is obvious that most any MTA should be able to utilize ast_fax. But, seeing that inter7.com (used) to be the home of the Courier-IMAP and MTA suite, and courier incl. built-in support for utilizing the mgetty+sendfax setup, you might concur that my question does carry at least some interest. Sure, you could use it with the courier-mta. Not really with courier-imap since imap does not handle sending email. There is a way to get courier-imap to handle sending emails a regular smtp server should be fine. We use astfax with qmail but it should work just fine with sendmail, postfix or any other MTA that handles dot forward type files. Ken Jones ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: How do I match a D? (Was: RE: [Asterisk-Users] In-band disc onn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
-Original Message- From: Paul Rodan [mailto:[EMAIL PROTECTED] I'll ask a stupid question, how does a user hit an alpha letter from his touchtone? I know that the Cisco 7960's support entering alpha letters, and it could potentially do it (maybe), but how does the average end user enter an a b c or d from their touchtone phone? They don't. Most phones lack the fourth column that has those keys. Some PBX phones have it, though, and it's common on 2-way radios that include DTMF keypads. (Believe it or not, before the advent of cell phones some businesses provided a phone patch interface to allow making phone calls from their 2-way radios. This is still quite common in the railroad industry, AFAIK.) In my case, I need it because it's how my PBX does disconnect notification to the voice mail system. When the line is hung up, it sends a D. I expect these digits are very rarely used, which is probably why no one thought to document that Background() ignores them. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web based Asterisk management tool
Thanks Dean, you say that [EMAIL PROTECTED] is just an automated way of installing AMP and FOP and Web Meetme. But from the installation instructions I have read, you download an ISO image that installs a linux distro for you (destroying current install) and then configures itself for use I do not want to overwrite any existing systems, here is a quote from the installation page of [EMAIL PROTECTED] 1) Burn [EMAIL PROTECTED] iso to a blank CD 2) Boot your Asterisk PC with the CD and press enter NOTE: This will erase all data on the hard drive of the PC!!! Etc etc All I want is the web config tool ! Apologies if I am misunderstanding you here, as I say I am quite new to this and need to get up to speed fast For an Admin only web based product, is AMP my only option ?? Cheers again -Original Message- From: dean collins [mailto:[EMAIL PROTECTED] Sent: 09 February 2005 16:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool You need to go back and reread. It is just pretty much an asterisk configuration tool (ok some minor things in the backend but it's the best out there). AMP is available for free download but they make their money by offering support. [EMAIL PROTECTED] is just an automated way of installing AMP and FOP and Web Meetme. If you really have a need to support thousands of extensions as you suggest then you should really go back and learn how to program asterisk with a database yourself from scratch. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Web based Asterisk management tool Thanks for the responses, isn't [EMAIL PROTECTED] an pre-configured distribution for small systems?? I am looking for an open source web management tool to use on any size asterisk server (even ones that are already up and running) the user base could be anything between small and large with many external lines, Ive looked at AMP, is it free ? and are there any alternatives or is AMP the only open source web management tool ? -Original Message- From: dean collins [mailto:[EMAIL PROTECTED] Sent: 09 February 2005 15:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool That would be the AMP database, I don't know. Ping the amp list and find out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa Sent: Wednesday, February 09, 2005 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: mercredi 9 février 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download at sourceforge and does exactly what you are looking for. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based Asterisk management tool Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___
[Asterisk-Users] ISDN in Spain
Hi list! Sorry for this slightly off-topic message but does anybody know if the standard for ISDN BRI is the same in Spain as it is in the rest of Europe (or the Netherlands). Will a standard HFC-S card work? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] announcement: astfax 1.0
On Tuesday 08 February 2005 5:42 pm, Remco Barende wrote: Looks really cool :) My company requires that for every fax we send we get a printed status report that includes the number we sent the fax to, the number the other fax reported, time+date, tx time and if the fax was sent ok or not plus (and here's the catch) a smaller image of the pages we faxed printed on the same page as the status report. Is there any chance that something like that will ever be implemented? (can it even be implemented like that??) snip I'm not sure if spandsp's txfax program supports returning status. If it did, there might be a way to implement your requirements. Ken Jones inter7.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX = FWD down again?
Joseph wrote: I wander what is causing the problem, I was thinking that it was something on my part but I did not change any settings and IAX2 registry At the time the only thing I could put it down to was congestion... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER Interaction: Agents and Extensions
On Wed, 9 Feb 2005, [EMAIL PROTECTED] wrote: Oh right.. I remember seeing that.. yeah that looked a whole lot more elegant than *8. Why isn't it in HEAD? I'm not sure. Once it started getting some testing BKW closed it. If someone is interested in testing the patch I'm sure the bug could be reopened. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable
I thought that as long as I stuck to the stable branch, only major bug fixes would be included, no new features or changing of the way things are handled? I mean, isn't the latest CVS Stable better than 1.03? I'm in the asterisk-cvs list and every day I see bug fixes added to the stable branch that fixes segfaults and divide by 0's and typo's here and a mistake there, etc. etc. won't all those bugs be present in the 1.03 version? I don't want my system to seg fault as the cvs list would indicate it could. So there are known issues with the latest CVS Stable? What is the best known version of Asterisk to date? 1.03? 1.05? I'm not interested in new features, just stability and quality. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nicolás Gudiño Sent: Wednesday, February 09, 2005 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable Hello, 2 nights ago I upgraded one of my remote servers to the latest CVS Stable, Asterisk CVS-v1-0-02/07/05-16:13:48, and my inbound and outbound caller ID stopped working. My suggestion would be to downgrade to 1.0.3. It might solve your problem. There were a number of changes in callerid handling in the last couple of weeks. Many manager based applications stopped working because of them. Maybe your setup is affected too. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 codec for X-lite soft phone
Hello all, Is X-lite soft phone support G.729? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. /Snip/ Daniel, You know that X-lite does not support G.729 and you also know where to have it, dont you? if you read your questions a couple of times, you will find answers there. Also, if you ever Visit Xten site and look at the information there, you will know what is, and what is not , supported in X-Lite and X-Pro. Sending questions like these toa busy forum like Asterisk only make it ever more difficult for people who are trying to wade through the thousands of emails posted here, for useful information. Please be considerate Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Voice Quality Issues
I am running * 1.0.5 and have been having lots of problems with outgoing calls and their sound quality. I am using ULAW for the codec and sixtel for termination. Basically the problem is that portions of the call seem to be lost and replaced with silence. Sometimes I can't hear the person talking othertimes they can't hear me. This situation comes and goes throughout the call. Bandwidth isn't an issue as I have a 3MB/1MB connection and there is at most 2 concurrent connections. Also using pingplotter to monitor iax2.sixtel.net shows little or no packetloss. Just as further info, I am using a SPA-2000 to connect to * with G711u as the preferred codec. Anyone else experience the like or have any suggestions on what may be causing this or ideas on how to debug? Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Web based Asterisk management tool
Why not share with the community? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Chandler Sent: Wednesday, February 09, 2005 9:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Web based Asterisk management tool Gary, contact me off-list. I have developed a GUI Windows based tool that will allow management of configuration files if you are running RealTime. It supports sip,iax,extensions,voicemail currently. It will also display CDR's and the various schema's used by the Asterisk box. Tom Chandler [EMAIL PROTECTED] - Original Message - From: dean collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 09, 2005 10:47 AM Subject: RE: [Asterisk-Users] Web based Asterisk management tool You need to go back and reread. It is just pretty much an asterisk configuration tool (ok some minor things in the backend but it's the best out there). AMP is available for free download but they make their money by offering support. [EMAIL PROTECTED] is just an automated way of installing AMP and FOP and Web Meetme. If you really have a need to support thousands of extensions as you suggest then you should really go back and learn how to program asterisk with a database yourself from scratch. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 11:17 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Web based Asterisk management tool Thanks for the responses, isn't [EMAIL PROTECTED] an pre-configured distribution for small systems?? I am looking for an open source web management tool to use on any size asterisk server (even ones that are already up and running) the user base could be anything between small and large with many external lines, Ive looked at AMP, is it free ? and are there any alternatives or is AMP the only open source web management tool ? -Original Message- From: dean collins [mailto:[EMAIL PROTECTED] Sent: 09 February 2005 15:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool That would be the AMP database, I don't know. Ping the amp list and find out. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa Sent: Wednesday, February 09, 2005 9:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dean collins Sent: mercredi 9 février 2005 15:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Web based Asterisk management tool Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download at sourceforge and does exactly what you are looking for. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary Sent: Wednesday, February 09, 2005 8:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Web based Asterisk management tool Hi there I am new to Asterisk and am looking for a web based management tool, for managers to manage hunt groups, extensions etc and for user to have access to there own phone features. I have seen there are a number of commercial tools available for this, but I presume there are some freeware options too I noticed one that I like at http://www.thirdlane.com/screenshots.htm but I am assuming this is just a freeware product that has been re-badged so to speak. If any body can give me some suggestions that would be great Regards Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list
