Re: [Asterisk-Users] [Fwd: Call Transfer between phones]
Michael Nolan wrote: Receive call, press flash, call other party, wait for answer, press transfer, hangup. Yes, assuming fairly recent firmware (1.0.5.16+). That does an attended transfer. To blind transfer you can just press transfer exten send and hangup. Question: I saw on the Wiki a report that 1.0.5.16 message button does not play well with Asterisk and sends a broken NOTIFY. Is that true? Anyone here have been using 1.0.5.16 (or later?) and care to comment on problems he or she may be having? The reason is that I have a working setup here, except the attended transfer and I'm willing to upgrade the firmware to get that extra feature but not if it breaks something that already works... :-) Thanks! 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 [EMAIL PROTECTED] 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] OS Choice ?
Alex Brecher wrote: Which Distro is the most commonly used distro with Asterisk please ? I don't know which is most commonly used, but I can tell you which is the easiest to install if you're going to install the OS from scratch anyway and plan to use it with Asterisk: Xorcom Rapid is a Debian/Asterisk distribution program that includes an auto-install and special auto-configuration features. It quickly and effortlessly converts any PC to a functioning Asterisk PBX... http://www.voip-info.org/wiki-Xorcom+Rapid -- 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Reboot Script PRI errors!!
Kevin wrote:
There is a reboot script posted on the wiki to reboot Polycom
telephones. When I execute this script, I get the following messages.
I am concerned as this is causing issues with asterisk and the PRI.
Does anyone have any ideas why this would be happening?
asterisk console:
-- Remote UNIX connection
-- Remote UNIX connection disconnected
and in the Asterisk Log:
Nov 28 22:30:42 NOTICE[1099909936]: PRI got event: 6 on Primary
D-channel of span 1 Nov 28 22:43:08 NOTICE[1099909936]: PRI got event:
6 on Primary D-channel of span 1
Script:
#!/usr/bin/perl -w
use Net::Ping;
use Socket;
$polycompath = '/home//';# Where you keep your config files
$arp = '/sbin/arp'; # Location of arp command
$sipserver = '192.168.XXX.XXX'; # IP of asterisk server
$phone = shift;
checkphone($phone);
touch( arp2config($phone) );
reboot_sip_phone( $phone, $sipserver, Reboot );
sub checkphone { # Checks for existence of phone, makes sure
# it's in arp table
$activephone = shift;
# Populate ARP table
print Checking ARP table.\n;
$p = Net::Ping-new(icmp);
if ( $p-ping( $activephone, 2 ) ) {
print $activephone is ;
print reachable.\n;
}
else { die Polycom at , $activephone, is not reachable!; }
sleep(1);
$p-close();
}
sub arp2config {# Gets mac address from arp table, converts
# to a polycom config filename, makes sure
# the config file exists
$arpip = shift;
open( ARP, $arp -an| ) || die Couldn't open arp table: $!\n;
print checking for polycom config name..., \n;
while (ARP) {
chomp;
$addr = $_;
$ip = $_;
$addr =~ s/.*
([\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+).*/$1/;
$addr =~ s/://g;
$addr = lc($addr) . '.cfg';
$ip =~ s/.*?(\d+\.\d+\.\d+\.\d+).*/$1/;
if ( $ip eq $arpip ) {
last;
}
}
$polycomconfig = $polycompath . $addr;
unless ( -e $polycomconfig ) {
print sorry, polycom config file , $polycomconfig,
is not found.\n\n;
exit;
}
return $polycomconfig;
}
sub touch {# We need to touch the config files or the phone
# won't reboot - it depends on time synchronization
print touching config file , $polycomconfig, \n;
my $now = time;
local (*TMP);
foreach my $file (@_) {
utime( $now, $now, $file )
|| open( TMP, $file )
|| die ($0: Couldn't touch file: $!\n);
}
}
sub reboot_sip_phone {# Send the phone a check-sync to reboot it
$phone_ip = shift;
$local_ip = shift;
$sip_to = shift;
$sip_from = 0;
$tm = time();
$call_id = $tm . msgto$sip_to;
$httptime = `date -R`;
$MESG = NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP $local_ip
From: sip:[EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED]
Event: check-sync
Date: $httptime
Call-ID: [EMAIL PROTECTED]
CSeq: 1300 NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
;
$proto = getprotobyname('udp');
socket( SOCKET, PF_INET, SOCK_DGRAM, $proto );
$iaddr = inet_aton($phone_ip);
$paddr = sockaddr_in( 5060, $iaddr );
bind( SOCKET, $paddr );
$port = 5060;
$hisiaddr = inet_aton($phone_ip);
$hispaddr = sockaddr_in( $port, $hisiaddr );
if ( send( SOCKET, $MESG, 0, $hispaddr ) ) {
print reboot of phone , $phone_ip, was successful, \n;
}
else { print reboot of phone , $phone_ip, failed, \n; }
}
exit;
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Kevin -
I rewrote this some time ago because of some issues with Polycom's latest
bootroom/sip update. Try this:
Also, serctl, part of the ser package, has a cisco_restart parameter that
works on Polycoms as well.
John
#!/usr/bin/perl -w
use Net::Ping;
use Socket;
$polycompath = '/home/PlcmSpIp/';# Where you keep your polycom files
$arp = '/sbin/arp'; # Location of arp command
$sipserver = '192.168.XXX.XXX'; # IP of asterisk server
$phone = shift;
checkphone($phone);
touch( arp2config($phone) );
reboot_sip_phone( $phone, $sipserver, get_extension($phone) );
sub checkphone { # Checks for existence of phone,
makes sure
# it's in arp table
$activephone = shift;
# Populate ARP table
print Checking ARP table.\n;
$p = Net::Ping-new(icmp);
if ( $p-ping( $activephone, 2 ) ) {
print $activephone is ;
print reachable.\n;
}
else { die Polycom at , $activephone, is not reachable!; }
Re: [Asterisk-Users] IP to IP call without server?
What's the diff. between FWD and Skype other than Skype uses proprietary solution? I'm getting a little confused with these softphones and provider companies. I cant tell what's so great of one from the other. Is there something special about FWD I'm missing that make it better than other solution like Skype? Thanks! Skype can only work with their software AFAIK. FWD works with many, many clients. The most important issue with Skype is that is works with no adjustments to your computer and it doesn't care about NAT routers. It also is supposedly encrypted end to end and has built in free conferencing for some small limited number of users. Since it's free and one wouldn't be likely to build a huge commercial app around it, it makes no difference that it is proprietory, you'd just drop it if it no longer works for you. FWD and many other free services will also work and they use a standard protocol (SIP) to set your session up. They will all require some screwing around to get them to work and I have never tried one that worked anywhere near as reliably or with as good quality as Skype. Skype is the Microsoft of voIP in some ways ;) However, discussing skype and ip to ip calling is not the subject of this list, so I'll now shut up. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Billing of outoging calls via CAPI
Hi Patrick, exactly, i am looking for something like AOC. The CDR offers just billing seconds and this is not enough because additionaly i would need rate-tables to calculate accurate price of the outgoing call. hmm...and is not what i want. Maybe there is a way to catch the billing information from D-channel. Is there any standalone application for linux, which is able to filter these charging informations when the Asterisk can't do that? Regards Rastislav On Fri, 2004-11-26 at 10:35 +0100, Rastislav Lukac wrote: Hello all, I would like to get billing/charging informations of all outgoing calls of any PSTN numbers made with my IP-Phone via asterisk. Asterisk automatically generates CDR's (Call Detail Records). They are stored in cdr-csv (or a database if you want it there). Can I obtain in * an accurate charge information of outgoing call via CAPI which destination is any PSTN number? I think you are referring to AOC (Advice of Charge) and afaik chan_capi does not support that nor does the CDR generation code. Does the ISDN signal contain charging informations ? Depends on your telco but if Asterisk does not support AOC, it will not matter... Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial plan for TDM22B
I finally got the TDM22B to work. Just for the rcord. The problem was, that I did not see the zaptel.conf and created a new one in /etc/asterisk, however /etc/zaptel.conf existed, Now I am trying to setup the dial plan for these 4 ports of the TDM22B. I face with my extension.conf following problems: 601 (on an ATA-186) can reach 603 / 604 (on the TDM22B) 603 or 604 cannot dial anything !!! dailing in from the PSTN results in You have configured Asterisk ... - You can stop it by dialing the extension number you want. Unfortunately you also can dial to a remote line (UK, USA) ... [incoming_88097680] exten = _**7680,1,Ringing exten = _**7680,2,Answer exten = _**7680,3,Dial(SIP/601,20,r) exten = _**7680,4,Voicemail,u601 exten = _**7680,105,Voicemail,b601 [incoming_88097074] exten = _**7074,1,Ringing exten = _**7074,2,NoOp exten = _**7074,3,Dial(SIP/601,20,r) exten = _**7074,4,Voicemail,u601 exten = _**7074,105,Voicemail,b601 [TDM22B] exten = 603,1,Dial(ZAP/1,60,tr) exten = 603,2,Voicemail,u603 exten = 603,102,Voicemail,b603 exten = 604,1,Dial(ZAP/2,60,tr) exten = 604,2,Voicemail,u604 exten = 604,102,Voicemail,b604 I also did not find a document about possible options (like t, T, m, r ) . What does the ** mean ??? (I guess it means anything ending with the numbers) bye Ronald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio Drops out at Random - one way
Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesnt happen on every call, about one in 5. Hardware is good, 2mb Connection, QOS enabled. If it was only one Asterisk server I would be ok, but it happens on two completely different places. I cannot work out what is causing it, can anyone offer anyone offer a solution or a method to track this down. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 when I call someone who is busy
Hi, My setup is quite complicated. I have to Asterisk server linked via IAX. My Sip phones are connected to one and go out (PSTN) via the IAX trunk and the other server is connected to a Quintum CMS via H323. Phone---(SIP)---Asterisk1---(IAX)---Asterisk2--(H323)---CMS-- PSTN All work fine but when a call someone who is busy I didn't hear the corresponding tone and asterisk2 go to timeout and after asterisk1. In my log the call is well routed through the two asterisk server. I don't know what is the problem so if you have any idea ? Thanks in advance ! Anthony Vous manquez despace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis. A télécharger gratuitement sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] low quality sound samples
Hi, I'm testing Asterisk installation on my Slackware Linux Box, I am making calls from kphone. I've noticed that standard sample voice files attached to * distro (*.gdm files) havn't good quality. What can be done to improve it? I'm using alsa drivers. BR, Corvin ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] low quality sound samples
voice files attached to * distro (*.gdm files) havn't good quality. sorry I meant *gsm files ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Audio Drops out at Random - one way
Sorry forgot to mention this is with IAX2 only, SIP works fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Sent: 29 November 2004 10:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Audio Drops out at Random - one way Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesnt happen on every call, about one in 5. Hardware is good, 2mb Connection, QOS enabled. If it was only one Asterisk server I would be ok, but it happens on two completely different places. I cannot work out what is causing it, can anyone offer anyone offer a solution or a method to track this down. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Billing of outoging calls via CAPI
On Mon, 2004-11-29 at 10:58 +0100, Rastislav Lukac wrote: [snip] Maybe there is a way to catch the billing information from D-channel. Is there any standalone application for linux, which is able to filter these charging informations when the Asterisk can't do that? I don't know. Seems difficult to have a standalone app catch AOC info while Asterisk needs the ISDN link also. You can ask junghanns.net of they are willing to add AOC support to chan_capi (Possibly at a fee). Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unable to compile testcpuid.c in spandsp in x86_64
Steven Hi, I'm unable to compile testcpuid.c with the __x86_64__ architecture (Athlon 64 processor). The messages are: /tmp/ccONleRV.s: Assembly messages: /tmp/ccONleRV.s: Error: suffix or operands invalid for 'pushf' 'pop' 'push' 'popf' Is it safe to ignore this module? When I attempt to start asterisk, libspandsp.so.0 fails to load because 'top_bit' is undefined. Is this related to the compile problem? Thanks for your help, -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] [Fwd: Call Transfer between phones]
Yes, 1.0.5.16 breaks the message button. 1.0.5.18 fixes the message button but shows a 403 error 'bout once an hour. Craig - Original Message - From: Gilad Ben-Yossef [EMAIL PROTECTED] To: Michael Nolan [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 29, 2004 4:06 PM Subject: Re: [Asterisk-Users] [Fwd: Call Transfer between phones] Michael Nolan wrote: Receive call, press flash, call other party, wait for answer, press transfer, hangup. Yes, assuming fairly recent firmware (1.0.5.16+). That does an attended transfer. To blind transfer you can just press transfer exten send and hangup. Question: I saw on the Wiki a report that 1.0.5.16 message button does not play well with Asterisk and sends a broken NOTIFY. Is that true? Anyone here have been using 1.0.5.16 (or later?) and care to comment on problems he or she may be having? The reason is that I have a working setup here, except the attended transfer and I'm willing to upgrade the firmware to get that extra feature but not if it breaks something that already works... :-) Thanks! 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] multiple EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
Jean-Michel Hiver wrote: Tomasz Chmielewski wrote: Hello, I'm thinking of deploying Asterisk. I already have a handful of EICON Diva 2.01 PCI ISDN cards. I was thinking if it's possible to insert 4 such cards to my PC-Asterisk server (which I yet have to install) and use them as 4 lines in case anyone has to call me in / I have to call out using ISDN line(s)? From what I have been told on this very list you can only use Diva Server cards with asterisk because the 'cheaper' diva cards do not support some stuff called 'capi'. It appears they have been wrong. I just checked on www.capi.org, and Eicon Diva 2.0 PCI cards offers CAPI 2.0... This same page says this card is is not Linux compatibile, though, but it is. Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Audio Drops out at Random - one way
Craig Waddington wrote: Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesnt happen on every call, about one in 5. Hardware is good, 2mb Connection, QOS enabled. If it was only one Asterisk server I would be ok, but it happens on two completely different places. I cannot work out what is causing it, can anyone offer anyone offer a solution or a method to track this down. Thanks. I have a similar problem with my IAX connection to my termination provider.. No one seems to be able to help and I have replaced or reinstalled just about every component in the chain except the internet itself and the termination provider.. Have updated Asterisk to 1.0.2, have added a switch to my network (was using a hub), have changes to a different firewall, have setup port mapping through the NAT, have tried different DSL routers and put in a high quality microfilter.. So the only things I think it can be are a) my termination provider (but they service many people and I am sure others would have brought it up if it was a problem), b) Asterisk itself or c) my DSL line or ISP.. Unfortunately these are all hard to check and the debug logging on Asterisk didn't help much when I tried looking at it.. I know this doesn't help much but if you come up with anything please let me know.. Its driving us crazy having calls drop on us especially when talking to customers.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Dial plan for TDM22B
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Monday, 29 November 2004 6:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dial plan for TDM22B I finally got the TDM22B to work. Just for the rcord. The problem was, that I did not see the zaptel.conf and created a new one in /etc/asterisk, however /etc/zaptel.conf existed, Now I am trying to setup the dial plan for these 4 ports of the TDM22B. I face with my extension.conf following problems: 601 (on an ATA-186) can reach 603 / 604 (on the TDM22B) 603 or 604 cannot dial anything !!! dailing in from the PSTN results in You have configured Asterisk ... - You can stop it by dialing the extension number you want. Unfortunately you also can dial to a remote line (UK, USA) ... [incoming_88097680] exten = _**7680,1,Ringing exten = _**7680,2,Answer exten = _**7680,3,Dial(SIP/601,20,r) exten = _**7680,4,Voicemail,u601 exten = _**7680,105,Voicemail,b601 [incoming_88097074] exten = _**7074,1,Ringing exten = _**7074,2,NoOp exten = _**7074,3,Dial(SIP/601,20,r) exten = _**7074,4,Voicemail,u601 exten = _**7074,105,Voicemail,b601 [TDM22B] exten = 603,1,Dial(ZAP/1,60,tr) exten = 603,2,Voicemail,u603 exten = 603,102,Voicemail,b603 exten = 604,1,Dial(ZAP/2,60,tr) exten = 604,2,Voicemail,u604 exten = 604,102,Voicemail,b604 I also did not find a document about possible options (like t, T, m, r ) . What does the ** mean ??? (I guess it means anything ending with the numbers) Actually, * is not a wildcard, . is the asterisk wildcard. The problem that you have here is that you are assuming that the PSTN passes calling ID and it does not. Calling ID can only be passed on a PRI or BRI (ISDN) interface. So all you receive on a ZAP interface is caller ID. Since there is no calling ID (as you expect), incoming calls must go the the s extention. Here is what I use on a TDM22B - ; incoming calls from the PSTN should come here [incoming] exten = s,1,Wait(1) ;wait to get caller ID in. exten = s,2,Dial(SIP/102,20) exten = s,3,Voicemail(u102) exten = s,102,Voicemail(b102) exten = s,103,Hangup Not sure what you mean by 603 and 604 cannot dial anything, they should be able to dial each other based on the config you have here. Maybe you have errors in other parts of your config? bye Ronald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Audio Drops out at Random - one way
I found this: http://lists.digium.com/pipermail/asterisk-dev/2003-May/000764.html But it is old, and I am sure lots of changes have been made to the source, since then. Where and how do you set absolutetimeout=0, would this help? A test I want to perform is, we make a call, and say nothing for 20 seconds, and see if that's why the audio stream is being dropped. ??? What I am doing currently is running debug IAX2 when users make a call, to try pinpoint the issue, but I don't know what I am looking for in the output. Are you using Cisco Phones? If so, what firmware, that is the only common thing at my end. This install worked fine for months, the audio issue has just started occurring. The quality is perfect, except this loss of Audio for a few seconds. Is your problem purely outgoing? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: 29 November 2004 13:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Audio Drops out at Random - one way Craig Waddington wrote: Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesn't happen on every call, about one in 5. Hardware is good, 2mb Connection, QOS enabled. If it was only one Asterisk server I would be ok, but it happens on two completely different places. I cannot work out what is causing it, can anyone offer anyone offer a solution or a method to track this down. Thanks. --- - I have a similar problem with my IAX connection to my termination provider.. No one seems to be able to help and I have replaced or reinstalled just about every component in the chain except the internet itself and the termination provider.. Have updated Asterisk to 1.0.2, have added a switch to my network (was using a hub), have changes to a different firewall, have setup port mapping through the NAT, have tried different DSL routers and put in a high quality microfilter.. So the only things I think it can be are a) my termination provider (but they service many people and I am sure others would have brought it up if it was a problem), b) Asterisk itself or c) my DSL line or ISP.. Unfortunately these are all hard to check and the debug logging on Asterisk didn't help much when I tried looking at it.. I know this doesn't help much but if you come up with anything please let me know.. Its driving us crazy having calls drop on us especially when talking to customers.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] D-LINK PoE switch, does it work with cisco or do I need to do the cable trick?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Chester Sent: Monday, 29 November 2004 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] D-LINK PoE switch, does it work with cisco or do I need to do the cable trick? As an alternative to the expensive PoE switches out there, I found the D-Link Web Smart DES-1316 switch for just around $400. Now, the issue is, it does 802.3af power and, as I've found out through previous discussions, this original 76?0 series phones (the non G variety) do not use 802.3af. So I'm assuming I just need to use the special cable thing as per http://www.voip-info.org/wiki-Cisco+POE and it will work. Is it this simple to save myself $1000 over the next lowest price PoE switch? It depends on whether the D-LINK injects the power into the unused wires or whether it injects the power into the TX and RX pairs? Look at the UPDATE at the bottom of the page you mentioned. 3com inject the power into the TX and RX pairs, so if the end device is not fully 802.3af compliant in anyway (like these Cisco phones), then to bad, or buy the 3com covertors first. - http://www.3com.com/products/en_US/detail.jsp?tab=featurespathtype=purchase sku=3CNJVOIP-CPOD ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] low quality sound samples
On Mon, 2004-11-29 at 12:10 +, Corvin wrote: Hi, I'm testing Asterisk installation on my Slackware Linux Box, I am making calls from kphone. I've noticed that standard sample voice files attached to * distro (*.gdm files) havn't good quality. What can be done to improve it? I'm using alsa drivers. Adjust your expectations to telephone quality. Everything is based around 8khz samples and at best around 14 bit quality. The GSM files should be around cell phone quality due to the codec. You are welcome to rerecord any sound prompts you wish and use them at a slightly higher quality but 8khz 16bit PCM is the highest quality you can use. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to call s extension from SIP phone?
BR C. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Fwd: Re: [Asterisk-Users] Adit 600 channel bank in UK setting]
On Thursday 18 November 2004 22:16, Tim Robinson wrote: Channel banks are a peculiar US thing. Be careful! You will almost certainly be better off using voip handsets (SNOMs are cool, avoid Grandstream for anything other than domestic environment) and a few Sipura-type ATA's for the analogue fax machines etc. or some Digium analogue cards. So what would you advise using in the UK to interface with standard 2 wire phones - I'm trying to avoid having to use ata type adapters. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 based bbs
In data Mon, 29 Nov 2004 08:51:08 +, Jean-Michel Hiver [EMAIL PROTECTED] ha scritto: It's a very interesting idea. The more I think about it, the more I wonder . . . Sounds like something that would end up being used as a dating phone service to me :-) yes, I think so. that could be a viable usage. also for small groups of people willing to share a message with each other. l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Best SIP phone for high quality telemarketing
On Friday 19 November 2004 04:42, Luke Connolly wrote: I'm really happy with my Polycom IP 600 http://www.polycom.com About $200 cheaper than cisco and no difference in qual or features. If only :). In the UK, I found 7960G's for £200 (ish) if I could lose $200 (about £105) I'd definitely be tempted with the polycoms. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 compile testcpuid.c in spandsp in x86_64
Hello, I'm unable to compile testcpuid.c with the __x86_64__ architecture (Athlon 64 processor). The messages are: /tmp/ccONleRV.s: Assembly messages: /tmp/ccONleRV.s: Error: suffix or operands invalid for 'pushf' 'pop' 'push' 'popf' Is it safe to ignore this module? I have similar problems compiling under PPC. I just removed that module together with other troubling assembler parts (MMX detection routines). After that it compiled fine and worked well (but the platform is 32 bits, I dunno if it will work under a 64bit platform). Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] low quality sound samples
Adjust your expectations to telephone quality. Everything is based around 8khz samples and at best around 14 bit quality. The GSM files should be around cell phone quality due to the codec. You are welcome to rerecord any sound prompts you wish and use them at a slightly higher quality but 8khz 16bit PCM is the highest quality you can use. Heh sorry but I can't agree with you I am very familiar with GSM technology and I use every day cellphone with EFR codec. And voice quality of what I can hear from kphone is much worse :(. C. --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] UK available SIP phone?
On Sunday 21 November 2004 12:03, Clive Carter wrote: Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike I use Grandstream Budge Tones. They are cheap, and some people say they look it, but they work ! I have also got ipDialogs SipTone II. They are twice the price, and although I have got the basic functions working, for some reason they just will not connect to VoiceMail I have used both sipura 2k's and ata286 - both worked perfectly with my dect phones. Currently 2K is connected to DECT, ata286 to fax and a 7960G for main use. I have in the past had budgetones and yes, they do look cheap - but so what if it's only for home use. I work from home, hence the 7960G (I simply needed more lines). But imho you'll struggle to beat a sipura 2K with a good quality DECT phone - although that works out a similar price as my 7960G. HTH Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] low quality sound samples
Corvin wrote: Adjust your expectations to telephone quality. Everything is based around 8khz samples and at best around 14 bit quality. The GSM files should be around cell phone quality due to the codec. You are welcome to rerecord any sound prompts you wish and use them at a slightly higher quality but 8khz 16bit PCM is the highest quality you can use. Heh sorry but I can't agree with you I am very familiar with GSM technology and I use every day cellphone with EFR codec. And voice quality of what I can hear from kphone is much worse :(. Not that it is particularly relevant to your issue, but * doesn't use EFR. It uses 06.10, the original GSM codec. EFR is encumbered by Nokia and other patents (and isn't that interesting anyway). Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Audio Drops out at Random - one way
Craig Waddington wrote: I found this: http://lists.digium.com/pipermail/asterisk-dev/2003-May/000764.html But it is old, and I am sure lots of changes have been made to the source, since then. Where and how do you set absolutetimeout=0, would this help? A test I want to perform is, we make a call, and say nothing for 20 seconds, and see if that's why the audio stream is being dropped. ??? What I am doing currently is running debug IAX2 when users make a call, to try pinpoint the issue, but I don't know what I am looking for in the output. Are you using Cisco Phones? If so, what firmware, that is the only common thing at my end. This install worked fine for months, the audio issue has just started occurring. The quality is perfect, except this loss of Audio for a few seconds. Is your problem purely outgoing? If you haven't set absolutetimeout in your dialplan then there will be no reason to have to reset it back to 0.. I tried the debug but didn't see much useful in there.. probably because I don't know what its telling me.. I am using Snom phones but I am almost sure its not the phone to Asterisk leg because I have an analog phone connected directly to the Asterisk box and it also drops calls.. Its almost certainly the IAX leg.. My problem appears to be purely on outgoing but we don't get a huge amount of incoming calls over the IAX and as far as I know we have never dropped an inbound call, but because of the small inbound call volume I wouldn't take it as fact.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Record() and problems converting with sox.
Hi, I'm trying to convert a high(er) quality wav/ulaw/alaw file (captured with Record()) to a gsm file and can't get the bugger to work. The example on the page http://www.voip-info.org/wiki-Asterisk+sound+files says that: $ sox inputfile.wav -r 8000 -c 1 outputfile.gsm resample -ql Should work, but I get the error message: sox: Input and Output rates must be different to use resample effect I get the same error message with $ sox -t la inputfile.alaw -r 8000 outputfile.gsm resample -ql And also with WAV, ulaw and everything else. I've trawled through the man page and am still completely bamboozled - it doesn't explain exactly what is necessary for it to work properly. I'm currently running sox version 12.17.6 (the latest) with gsm support enabled. Any ideas? Cheers, Nick. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] UK available SIP phone?
I should reply back now and say that I managed to get the Tecom IP2005 from solwise.co.uk working with Asterisk. I did however buy a Grandstream Budge Tone when I had almost given up on the Tecom one, so now I have both! :) The external quality of rhe grandstream is not as nice as the Tecom but they it is a bit cheaper. I'm still in the very early stages of Asterisk so I'm not sure what features the Tecom one does/not support yet. Thanks to those who replied. Regs Mike (now if only I could get IAX/FWD/SIP or something to work I'd be happy) On Mon, 29 Nov 2004 13:48:12 +, Jon Lawrence [EMAIL PROTECTED] wrote: On Sunday 21 November 2004 12:03, Clive Carter wrote: Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike I use Grandstream Budge Tones. They are cheap, and some people say they look it, but they work ! I have also got ipDialogs SipTone II. They are twice the price, and although I have got the basic functions working, for some reason they just will not connect to VoiceMail I have used both sipura 2k's and ata286 - both worked perfectly with my dect phones. Currently 2K is connected to DECT, ata286 to fax and a 7960G for main use. I have in the past had budgetones and yes, they do look cheap - but so what if it's only for home use. I work from home, hence the 7960G (I simply needed more lines). But imho you'll struggle to beat a sipura 2K with a good quality DECT phone - although that works out a similar price as my 7960G. HTH Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New T100P Pri install suggestions?
In the next week or so, we'll be turning up a new T1 Pri from Cox Cable in Omaha using a T100P (installed but not yet configured). While discussing the interface parameters with a very knowledgable Cox engineer, we decided on some of the basics including b8zs, esf, dchan=24, callerid, 5 digits of called number forwarded to *, and switchtype=national (NI2). They indicated their CO switch is a DMS500 that is compatible with several different switchtypes including 5ess, dms100, etc. * will clock sync from this pri. This system also has a tdm04b (4-port fxo) installed and working, and is expected to remain after the t100p is implemented. Zttool can see both cards at the moment. Questions: 1. Is a switchtype=national (NI2) a reasonable choice, or are there other types that others have found to be more usable/stable with * in the US? 2. Since the current system has a working tdm04b defined as fxsks=1-4 in /etc/zaptel.conf, how will I know when implementing the t100p whether those b-channels should be 5-28 or 1-23? (eg, which card has channel 1? It almost appears that defining the span= first with bchan=1-23 and follow it with fxsks=25-28 (for the tdm04b), is about the only way to do that without creating ambiguity for the reader.) Is that a reasonable approach or assumption? 3. Other then zttool, what other methods have others found useful for diagnosing layer 1 2 type issues remotely during an initial pri install? (After ensuring layer 1/2 functionality, I'm assuming 'pri debug', etc, is most appropriate for diagnosing higher layer and dialplan issues.) 4. In very general terms, when passing incoming pri calls to a specific extensions.conf context, is using 'exten = s,1...' a reasonable way to handle inbound calls initially? (Is the 'called number' passed in EXTEN like it is with sip calls?) Any other tips/tricks for handling an initial pri installation? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] UK available SIP phone?
I picked up a Sipura SPA-2000 (new) on e-Bay for ~£70. The voice quality is excellent. Peter -Original Message- From: Mike Dent [mailto:[EMAIL PROTECTED] Sent: 29 November 2004 14:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UK available SIP phone? I should reply back now and say that I managed to get the Tecom IP2005 from solwise.co.uk working with Asterisk. I did however buy a Grandstream Budge Tone when I had almost given up on the Tecom one, so now I have both! :) The external quality of rhe grandstream is not as nice as the Tecom but they it is a bit cheaper. I'm still in the very early stages of Asterisk so I'm not sure what features the Tecom one does/not support yet. Thanks to those who replied. Regs Mike (now if only I could get IAX/FWD/SIP or something to work I'd be happy) On Mon, 29 Nov 2004 13:48:12 +, Jon Lawrence [EMAIL PROTECTED] wrote: On Sunday 21 November 2004 12:03, Clive Carter wrote: Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike I use Grandstream Budge Tones. They are cheap, and some people say they look it, but they work ! I have also got ipDialogs SipTone II. They are twice the price, and although I have got the basic functions working, for some reason they just will not connect to VoiceMail I have used both sipura 2k's and ata286 - both worked perfectly with my dect phones. Currently 2K is connected to DECT, ata286 to fax and a 7960G for main use. I have in the past had budgetones and yes, they do look cheap - but so what if it's only for home use. I work from home, hence the 7960G (I simply needed more lines). But imho you'll struggle to beat a sipura 2K with a good quality DECT phone - although that works out a similar price as my 7960G. HTH Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 compile testcpuid.c in spandsp in x86_64
Nicolás Gudiño wrote: Hello, I'm unable to compile testcpuid.c with the __x86_64__ architecture (Athlon 64 processor). The messages are: /tmp/ccONleRV.s: Assembly messages: /tmp/ccONleRV.s: Error: suffix or operands invalid for 'pushf' 'pop' 'push' 'popf' Is it safe to ignore this module? I have similar problems compiling under PPC. I just removed that module together with other troubling assembler parts (MMX detection routines). After that it compiled fine and worked well (but the platform is 32 bits, I dunno if it will work under a 64bit platform). Regards, It would be nice if we customised the code for PPC, and maybe ARM. The only thing I really do in assembly right now is where is the top bit. There is usually an instruction for this on most modern CPUs. It can really speed up a couple of things. It think it should really be a standard C intrinsic. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 based bbs
On November 29, 2004 08:37 am, lenz wrote: yes, I think so. that could be a viable usage. also for small groups of people willing to share a message with each other. Meh -- My time is more valuable than that -- why listen to people talk when I can read a dozen messages in the same amount of time? Sure it's a little more personal but it's the same reason I prefer to listen to radio than watch TV -- there are less senses involved, so I can essentially multitask. Now if I were trying to do some kind of dating service, it's better to involve more senses but when you're talking about straight information exchange and indeed with anything technical, it seems that text and diagrams are best unless there is a specific need to communicate something audibly or that involves change with time (i.e. motion). Just my $0.02, having been in this kind of situation many times over the last two decades or so. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] multiple EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
On Mon, 2004-11-29 at 13:51 +0100, Tomasz Chmielewski wrote: [snip] It appears they have been wrong. I just checked on www.capi.org, and Eicon Diva 2.0 PCI cards offers CAPI 2.0... This same page says this card is is not Linux compatibile, though, but it is. Capi.org is not the same as chan_capi or Asterisk. I have never heard of a cheap (non Server or active) Eicon Diva card working with chan_capi Asterisk. If you want an ISDN based chan_capi/Asterisk solution either buy an Eicon Diva Server, any of the active AVM cards (B1 or C4 iirc) or the cheapest solution: an AVM Fritz! card. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] multiple EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
Patrick wrote: On Mon, 2004-11-29 at 13:51 +0100, Tomasz Chmielewski wrote: [snip] It appears they have been wrong. I just checked on www.capi.org, and Eicon Diva 2.0 PCI cards offers CAPI 2.0... This same page says this card is is not Linux compatibile, though, but it is. Capi.org is not the same as chan_capi or Asterisk. I have never heard of a cheap (non Server or active) Eicon Diva card working with chan_capi Asterisk. If you want an ISDN based chan_capi/Asterisk solution either buy an Eicon Diva Server, any of the active AVM cards (B1 or C4 iirc) or the cheapest solution: an AVM Fritz! card. And I was so happy today because I thought I won't have to buy anything :) So because I have these cheap Eicon Diva cards - does this mean they won't work at all? Or rather that some features will be missing only? I need only incoming and outgoing calls through these cards. Tomek ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] multiple EICON Diva 2.01 PCI ISDN cards with Asterisk - possible?
On Mon, 2004-11-29 at 15:51 +0100, Tomasz Chmielewski wrote: [snip] So because I have these cheap Eicon Diva cards - does this mean they won't work at all? Or rather that some features will be missing only? It just won't work at all (again afaik). I need only incoming and outgoing calls through these cards. That's pretty much everything :) If you want to explore support for plain Eicon Diva cards in chan_capi talk to junghanns.net. If you don't want to do that get an AVM Fritz! card or go the other way and get an HFC-S based ISDN card and use bristuff (also from junghanns.net). For more info about HFC-S bristuff search voip-info.org or google. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] OS Choice ?
On Mon, 29 Nov 2004 10:09:26 +0200, Gilad Ben-Yossef wrote: Alex Brecher wrote: Which Distro is the most commonly used distro with Asterisk please ? I don't know which is most commonly used, but I can tell you which is the easiest to install if you're going to install the OS from scratch anyway and plan to use it with Asterisk: Xorcom Rapid is a Debian/Asterisk distribution program that includes an auto-install and special auto-configuration features. It quickly and effortlessly converts any PC to a functioning Asterisk PBX... Since I had to rebuild my * server over the weekend I had a go with this Xorcom thingy. It pretty much did as it promised, with minimal user interaction it created a working * installation with a handy text mode shell. However, being a Linux newbie I found that it lacked a few basic things that I needed to make it work for me...most significantly the ability to use SSH to connect from my desktop transfer config files and otherwise and administer *. Had I been able to do that I would probably have tried it out for a while. Oh, also the version of * it installed was quite old...CVS 5/11/04 if I recall. That was also a major concern. If I have to build a new server for my home office some time in the future I'll try the AstLinux ISO which is an embedded version of Gentoo with Asterisk 1.0. Runs on PC Engines WRAP boards. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] OS Choice ?
Michael, Xorcom can allow ssh. You didn't read the instructions properly (lord knows I didn't the first few dozen times). When you insert the disk for the first time instead of typing linux or pressing enter to start the install type expert This will halt the installation at each section to ask you various questions (most of which you can ignore) but it will allow you to install ssh and then you can continue the rest of the installation remotely. Cheers, Dean (yeh one question I was able to answer). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Monday, November 29, 2004 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OS Choice ? On Mon, 29 Nov 2004 10:09:26 +0200, Gilad Ben-Yossef wrote: Alex Brecher wrote: Which Distro is the most commonly used distro with Asterisk please ? I don't know which is most commonly used, but I can tell you which is the easiest to install if you're going to install the OS from scratch anyway and plan to use it with Asterisk: Xorcom Rapid is a Debian/Asterisk distribution program that includes an auto-install and special auto-configuration features. It quickly and effortlessly converts any PC to a functioning Asterisk PBX... Since I had to rebuild my * server over the weekend I had a go with this Xorcom thingy. It pretty much did as it promised, with minimal user interaction it created a working * installation with a handy text mode shell. However, being a Linux newbie I found that it lacked a few basic things that I needed to make it work for me...most significantly the ability to use SSH to connect from my desktop transfer config files and otherwise and administer *. Had I been able to do that I would probably have tried it out for a while. Oh, also the version of * it installed was quite old...CVS 5/11/04 if I recall. That was also a major concern. If I have to build a new server for my home office some time in the future I'll try the AstLinux ISO which is an embedded version of Gentoo with Asterisk 1.0. Runs on PC Engines WRAP boards. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] app_sms: problems sending a sms
Thanks Steffen. Please update me if this ever works. Seshu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steffen Koepf Sent: Friday, November 26, 2004 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] app_sms: problems sending a sms Hello Seshu, no it still does not work. I started to debug this thing, and as far as i can say, the problem is that app_sms does not recognize the initial connection established packet, that the SM-SC sends when answering the call. I don't know now what exactly the problem is at the moment, possible that it is the level of the signal. I will examine this in the next days. cu, Steffen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cannot get two TE410Ps to operate correctly in the same machine
Your description makes perfect sense. My system is still getting HDLC overruns, which are certainly a consequence of frame slips because the second card is not getting clocked from the external source. I come back to my basic question: How do you configure an asterisk system so that a second TE410P card recognizes an external clock source? No matter what I've done the second card is always reported as "internally clocked". Configuring span 5 (the first on the second board) with exactly the same timing parameters as for span 1 (which works great) is not doing it. What is the trick? Can anybody share the config files for a succesful installation with more than one TE410P boards? Fernando Rich Adamson wrote: It seems to me that if not all cards are clocked from the same source, then each one should be able to get its own external clock. However, card 0 has an external clock, but card 1 does not. I've read the early posts relating to this and there still seems to be a misunderstanding on this clock sync issue. This stuff has been around for a long time in the telephony business, but it seems like not many people understand it on this list. Syncing a digium card's clock has nothing to do with universal time, expensive add-on clocking hardware, or my clock is more accurate then your clock. Every single T1/E1 data stream has clocking information embedded in the data stream. There is no such thing as turning _off_ the clocking. The telco can't do it and you can't do it. Also, there is absolutely no need for a clock on one digium card to be in sync with the clock on another digium card (or any other manufacturer of T1/E1 cards). That discussion is totally irrelevant to using those ports with one exception, and I'll mention that further in this text. The clock syncing issue is really very simple. For asterisk cards, do you connect to some system/device via a T1/E1 that you think interconnects with _other_ systems/telcos/ITSP's? If so, your digium card should accept clock syncing from that source (and only that source). You only need to choose one, but on the four-port digium card you're also given the option of selecting a secondary (etc) source should the first one fail. What's the issue? The telco (or whatever your connecting to) is doing exactly the same thing; they accept clock sync from another telco or long distance provider that provides T1/E1 connections to them. Assume purely for discussion purposes, the telco's clock is supposed to run at exactly 1.544 mhz/sec. What happens if their clock is actually running at 1,543,900 hz instead (clocks do drift)? You could have the most accurate clock in the world that you're syncing to, and in this case you are going to have a problem because the two clocks are _different_. There is going to be frame slips that occur, and the rate of slips is directly related to how far apart the two clocks really are. How do you get around that? By simply telling your T1/E1 card that you are syncing to that span. No more slips, period. The digium T1/E1 cards only have a single clock chip on the card. There is no reason to have four clocks on a four port card. By telling your card to accept clock sync on port 1 (as an example), the digium card has the smarts to adjust that single clock chip to be in sync with that span. Since practically every telephone company has T1/E1 feeds from other telco's, they follow a hierarchical engineering approach that basically says all telco's are in sync. That engineering approach has been around since T1's were invented, and the telco engineers know that very well. So, what happens if port #2 goes to yet another telco? No problem, because the telco's are already in sync. Choose one as your primary sync and the other for backup (secondary). What happens if port #3 goes to one of your channel banks? No problem, your digium card has embedded the clocking information in the T1/E1 data stream (you don't have a choice either), and you configure your channel bank to _accept_ clock sync from that T1/E1. The only exception (as mentioned above) to this is "if" you happen to engineer a combination of three or more asterisk systems in a circular fashion, and you've told all three to sync from its neighbor. In other words, system A connects to system B, then C, which connects back to system A, and you have B accepting clock sync from A, C from B, and then on the last leg from C back to A you tell A to use clocking from C. Any clock drifting that might occur will be passed around the closed loop and _could_ present a problem if the clock drifted far enough off frequency. E.g., one bad card could impact all three spans under the right conditions. What happens if you have a second T1/E1 card in the system? It doesn't make any difference. If a port on that card _is_ connected to a telco, then accept clock sync from it. If none of the ports have such a connection, then simply configure the devices at the far
Re: [Asterisk-Users] how to call s extension from SIP phone?
*Which* SIP phone? Some of them, like the Gradstream Budgettone, have a dial this number when user picks up configuration option. If you have such a phone you can create an alias for the s extention like so: exten = 666,1,Goto(s,1) And instruct the phone to dial '666' when the phone is off the hook. Hope this helps, 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: how to call s extension from SIP phone?
- Original Message - snip build another extension that uses the goto() application. exten=xxx,goto(context,s,1) will send the call to that context. think about restating your question in the body of your email when you post. Jason Kawakami www.optellabs.com Salt Lake City, UT ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] overriding DTMF and codec from dialplan?
OH But it is just that simple. You also have: -= Info about application 'ImportVar' =- [Synopsis]: Set variable to value [Description]: ImportVar(#n=channel|variable): Sets variable n to variable as evaluated on the specified channel (instead of current). If prefixed with _, single inheritance assumed. If prefixed with __, infinite inheritance is assumed. bkw It's not so simple. Check http://bugs.digium.com/bug_view_advanced_page.php?bug_id=928 for the details. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Registering on Gatekeeper
Thanks in advance kido. Please someone could help me. I am so closed to the solution :) My provider sent me this log from his gnuGK: RCF|200.68.216.107:1720|root:h323_ID|terminal|7000_endp; URQ|200.68.216.107:32877|7000_endp|maintenance; SoftPBX: Endpoint 200.68.216.107:1720 unregistered! My Asterisk keep sending root as h323_id. My h323.conf is this: ; h323.conf [general] listenAddress=0.0.0.0 listenPort=1720 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=yes h245tunnelling=yes h245inSetup=yes gatekeeper=###.###.###.### gatekeeperTTL=600 AllowGKRouted = yes userInputMode=Q931 amaFlags=default accountCode=4571025814 disallow=all allow=all wrapLibTraceLevel=1 libTraceLevel=9 libTraceFile=/root/pruebalogasterisk [##] ; Mi h323_id type=h323 prefix=99,0343 context=casanahuel I have already done a log watch, a h.323 debug and a h.323 trace 10, and Asterisk send root as h323_id. The h.323 trace 1 give me millons of this lines: 0:23.901 Transactor:81395c8h323trans.cxx(709) Trans registrationRequest rejected: fullRegistrationRequired 0:24.922 Transactor:81395c8h323trans.cxx(709) Trans registrationRequest rejected: fullRegistrationRequired 0:25.359 Transactor:81395c8h323trans.cxx(709) Trans registrationRequest rejected: fullRegistrationRequired 0:26.003 Transactor:81395c8h323trans.cxx(709) Trans registrationRequest rejected: fullRegistrationRequired On Sun, 28 Nov 2004 20:41:17 -0300, Nahuel Alejandro Ramos [EMAIL PROTECTED] wrote: I have already tryed this but asterisk always send root as h323_id On Sun, 28 Nov 2004 21:31:52 -, kido noagbodji [EMAIL PROTECTED] wrote: If you are using GnuGK, i think this should do, in your h323.conf file, configure an asterisk endpoint as follow for instance [time] Username type=h323 e164=99 context=test K. - Original Message - From: Nahuel Alejandro Ramos [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 28, 2004 5:52 PM Subject: [Asterisk-Users] Registering on Gatekeeper Hi, Anyone know how can I send a username or account id (h.323) and a password to register on a remote Gatekeeper. I am using the Nuphone channel with the h323.conf. I tryed everything but Asterisk always send root as account id and the Gatekeeper rejected me. Thank you very much... Nahuel Ramos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Warnings - chan_iax2.c:1464 attempt_transmit
I am getting quite a few of these warnings lately, and audio is sometimes dropping to one way. Is this some way related? Latency to my IAX provider is minimal, and no major packet loss. Nov 29 15:04:00 WARNING[1095035200]: chan_iax2.c:1464 attempt_transmit: Max retries exceeded to host X.X.X.X on IAX2/marg/2 (type = 6, subclass = 2, ts=120009, seqno=36) Nov 29 15:04:00 WARNING[1095035200]: chan_iax2.c:1464 attempt_transmit: Max retries exceeded to host X.X.X.X on IAX2/marg/2 (type = 6, subclass = 11, ts=120012, seqno=37) Nov 29 15:10:42 WARNING[1089370688]: chan_sip.c:675 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Critical Request) Nov 29 15:10:46 WARNING[1089370688]: chan_sip.c:675 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Can anyone bring some light to this? Using cisco phones. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Subject: IAXy and ADPCM codec problem.
Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only in one way, can anyboddy help me with this isue??? I don't thing the ADPCM codec works like this normally. Thanks for all -- Carlos Clemares Director 58 (0) 212 740-53-12/17 [EMAIL PROTECTED] www.radiumtec.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 phones cutting out with Asterisk??
Hi The problem is not normally the phones, check that you don't have busy detect on this sometimes can cause the phone to cut out. What card are you using? I have had the same problem with Digium FXO cards, we changed to Voictronix and the problem went away. If you are using an ISDN card it is most probably the busy detect. Doug -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Tim JacksonSent: Friday, November 26, 2004 9:44 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] SIP phones cutting out with Asterisk?? Ive had the same problem. I posted to the list earlier about the problem, and from what I can tell, its a Polycom issue (not a network issue as stated in the other post). It happens after the phones have been on for about 2-3 days from what I can tell. My solution to this was to use a script to reboot the phones every night at like 3am, and the problem has almost disappeared. If you find any other solutions, please post them to the list. -Tim From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave HendersonSent: Friday, November 26, 2004 12:11 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] SIP phones cutting out with Asterisk??Importance: High Hi folks, I've got a very bizarre problem recurring when making calls with Polycom SoundPoint IP500 SIP phones and Asterisk. Sometimes when a call comes in to an IP500, one of the sides of the conversation is cut off (i.e. the caller can't hear the callee, or vice-versa). This isn't easily repeated, and rebooting the phone, or restarting Asterisk, doesn't seem to have an effect. Has anybody else experienced this sort of thing happening? I've seen this with both CVS-v1-0-10/27/04-21:54:17 and CVS-HEAD-09/02/04-22:57:21. Thanks for any insight, Dave HendersonCustomer Service ManagerThe IT Department, Inc.ph: 613-523-2322x321fx: 613-526-3949 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXy and ADPCM codec problem
Hi everyone, I'm using the IAXy boxes and i'm having some trouble when I use it with the ADPCM codec. When I use the ADPCM codec only one person (out of the two of the conversation) is able to hear the other, but when I switch to ULAW codec everybody can hear the other. The ULAW codec is too heavy for my bandwidth (64Kbits/s) and its sounds choppy, the ADPCM codec sounds good but only in one way, can anyboddy help me with this isue??? I don't thing the ADPCM codec works like this normally. Thanks for all -- Carlos Clemares ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cannot get two TE410Ps to operate correctly in the same machine
Your description makes perfect sense. My system is still getting HDLC overruns, which are certainly a consequence of frame slips because the second card is not getting clocked from the external source. I come back to my basic question: How do you configure an asterisk system so that a second TE410P card recognizes an external clock source? No matter what I've done the second card is always reported as internally clocked. Configuring span 5 (the first on the second board) with exactly the same timing parameters as for span 1 (which works great) is not doing it. What is the trick? Can anybody share the config files for a succesful installation with more than one TE410P boards? I don't recall from your previous postings on this topic, so I'll ask. After config'ing span 5 in /etc/zaptel.conf, are you reloading/restarting the drivers? (That _is_ necessary.) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax pass-throught.
I've already search in the mailing list and voip-wiki site but I can not find any examples in how to send a FAX through a IAX channel. I've found the fax extention setting, but this is not what I want to do. I'd like to dial from the line on the other side of the IAX channel to a fax, to cut long distance costs, and send a FAX from the source IAX channel. Like bellow: source destination FAX --- Asterisk --- internet --- Asterisk --- external line - PSTN -- FAX I've also tried setting the codec to G.711 but it has not worked either. Can anyone shed a light on this matter? I am using Asterisk 1.0.2. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Fax pass-throught.
Alessandro Ren wrote: I've found the fax extention setting, but this is not what I want to do. I'd like to dial from the line on the other side of the IAX channel to a fax, to cut long distance costs, and send a FAX from the source IAX channel. Like bellow: source destination FAX --- Asterisk --- internet --- Asterisk --- external line - PSTN -- FAX I haven't done this, but I've heard that faxing through a voip connection is problematic. Have you considered the possibility (if you control both Asterisk installations in your diagram) that you could fax to a virtual fax on the source Asterisk system which would capture to a file, email or file transfer the image to the other Asterisk box which would then dial out and send to the final destination? This assumes your source and destination actually have to be real fax machines, otherwise you have even more options. Check out: http://scottstuff.net/scott/archives/000152.html and see if it gives you any ideas. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spawn extension
hi, calling from Asterisk to PBX via Eicon Diva 4BRI gives me the following error. -- Executing NoOp(SIP/2004-41dc, call for 998004) in new stack -- Executing Dial(SIP/2004-41dc, CAPI/99:8004|20|r) in new stack == Everyone is busy/congested at this time -- Executing Congestion(SIP/2004-41dc, ) in new stack == Spawn extension (default, 998004, 3) exited non-zero on 'SIP/2004-41dc' What is the meaning of the exited non-zero line? thx for your feedback __ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: how to call s extension from SIP phone?
- Original Message - snip build another extension that uses the goto() application. exten=xxx,goto(context,s,1) will send the call to that context. think about restating your question in the body of your email when you post. Jason Kawakami www.optellabs.com Salt Lake City, UT ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] OS Choice ?
Michael Graves wrote: On Mon, 29 Nov 2004 10:09:26 +0200, Gilad Ben-Yossef wrote: Alex Brecher wrote: Which Distro is the most commonly used distro with Asterisk please ? I don't know which is most commonly used, but I can tell you which is the easiest to install if you're going to install the OS from scratch anyway and plan to use it with Asterisk: Xorcom Rapid is a Debian/Asterisk distribution program that includes an auto-install and special auto-configuration features. It quickly and effortlessly converts any PC to a functioning Asterisk PBX... Since I had to rebuild my * server over the weekend I had a go with this Xorcom thingy. It pretty much did as it promised, with minimal user interaction it created a working * installation with a handy text mode shell. However, being a Linux newbie I found that it lacked a few basic things that I needed to make it work for me...most significantly the ability to use SSH to connect from my desktop transfer config files and otherwise and administer *. Had I been able to do that I would probably have tried it out for a while. Oh, also the version of * it installed was quite old...CVS 5/11/04 if I recall. That was also a major concern. If I have to build a new server for my home office some time in the future I'll try the AstLinux ISO which is an embedded version of Gentoo with Asterisk 1.0. Runs on PC Engines WRAP boards. Michael Micheal, Actually, AstLinux is not available as an ISO just yet. I am waiting until I hear more feedback from current users of AstLinux before I create an ISO. When I do create the ISO I will let you know. -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fxo connection in the UK
I located in the UK and am looking into connecting three analog BT lines to an astersik system which is replacing our current pbx. I could use three Digium wildcard x100p cards for that but I rather use a unit which is external to the computer to have a better separation of analogue/digital side. I would not like to go ISDN because the analog lines have so far sufficed in every repect and I tend not to fix what isn't broken. Today I found a unit on a supplier's website http://www.peripheralcorner.co.uk/product_info.php/cPath/113/products_id/544 which is a Micronet SP5054 VoIP Gateway 4 FXO Ports The website for this product is http://www.micronet.info/Products/voip/SP5054.asp and it appears to me that this unit would (similar to a channel bank) multiplex our three BT lines into one LAN port. If so, I could simply connect such a box to a LAN port in my asterisk server. I suspect the unit would appear to the asterisk box like three SIP-to-fxo converters (sorry for the horrible beginner-jargon). I basically would like to know whether I could use this unit instead of three x100P cards and it would functionally replace them. Questions * Would it be legal in the UK to connect such a unit to the PSTN ? In the specifications http://www.micronet.info/Products/voip/SP5054.asp#Specif there seems to be CE regulatory approval (see bottom row of table, Emission), but I don't know whether that is sufficient for use with a PSTN in the UK. * Could I operate the unit in a transparent fashion, i.e. it would look to the asterisk machine as if I had connected three SIP-to-fxo converters which I can control independently of each other from the asterisk machine? For example, could I initiate / receive a phone call while another phone conversation is running? For outgoing calls, could I specify which fxo port to use / for incoming calls, could I find out which port answered it? * Would the fxo ports match the UK PSTN specifications (impedance)? I am asking in this list just in case that someone has used / is using such voip gateways. I am still very much in the enquiry phase. Thank you very much for your consideration. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Spawn extension
== Spawn extension (default, 998004, 3) exited non-zero on 'SIP/2004-41dc' What is the meaning of the exited non-zero line? afaik, after something executes, zero is returned: Everything went OK or -1 Something bad happened, you can branch conditionally in the dialplan based on that. Unfortunately, there doesn't seem to be any kind of granularity there so you can't branch based on what went wrong, just that something went wrong. hth. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 on a notebook: Modem = FXO?
Hi, I've got a (maybe stupid) question: I'd like to install Asterisk on my notebook (just to play around with it). Is my internal phone modem equivalent to an FXO port? Hence, could I make phonecalls to my pots-line with it? thanks philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 on a notebook: Modem = FXO?
Philipp Ebneter wrote: Hi, I've got a (maybe stupid) question: I'd like to install Asterisk on my notebook (just to play around with it). Is my internal phone modem equivalent to an FXO port? Hence, could I make phonecalls to my pots-line with it? thanks philipp If the modem has an Intel 537 chipset ( or 517? ), I don't see why not. But that's just a guess. :) Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 on a notebook: Modem = FXO?
On Mon, 2004-11-29 at 17:46 +0100, Philipp Ebneter wrote: Hi, I've got a (maybe stupid) question: I'd like to install Asterisk on my notebook (just to play around with it). Is my internal phone modem equivalent to an FXO port? Hence, could I make phonecalls to my pots-line with it? While yes your modem is an FXO port, it is highly unlikely you will get a driver for that modem working with asterisk. Your best option for a laptop would be to consider a VoIP appliance that would provide FXO or FXS ports for the laptop. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cannot get two TE410Ps to operate correctly in the same machine
Yes. I've tried: - ztcfg - rmmod the wct4xxp module and zaptel and modprob'ing again - rebooting the server. No matter what I do, the second card is always "internally clocked". The E1 I have plugged into that board is good. It can clock the first card just fine. This is my complete zaptel.conf, just for reference: # E1 ISDN PRI. span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 # E1 R2 span=2,2,0,cas,hdb3 cas=32-46:1101 cas=48-62:1101 # E1 ISDN PRI span=3,0,0,ccs,hdb3 bchan=63-77 dchan=78 bchan=79-93 # T1 Channel bank. span=4,0,0,esf,b8zs fxsks=94-101 fxoks=102-117 # E1 ISDN PRI. Same carrier as span 1. Line can be exchanged with span 1 and all is OK. span=5,1,0,ccs,hdb3 bchan=118-132 dchan=133 bchan=134-148 # E1 Unused. span=6,0,0,ccs,hdb3 unused=149-163 unused=164 unused=165-179 # E1 Unused span=7,0,0,ccs,hdb3 unused=180-194 unused=195 unused=196-210 # T1 Channel bank. span=8,0,0,esf,b8zs fxoks=211-234 Fernando Rich Adamson wrote: Your description makes perfect sense. My system is still getting HDLC overruns, which are certainly a consequence of frame slips because the second card is not getting clocked from the external source. I come back to my basic question: How do you configure an asterisk system so that a second TE410P card recognizes an external clock source? No matter what I've done the second card is always reported as "internally clocked". Configuring span 5 (the first on the second board) with exactly the same timing parameters as for span 1 (which works great) is not doing it. What is the trick? Can anybody share the config files for a succesful installation with more than one TE410P boards? I don't recall from your previous postings on this topic, so I'll ask. After config'ing span 5 in /etc/zaptel.conf, are you reloading/restarting the drivers? (That _is_ necessary.) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sending triggers through SIP
Hi! I have a Asterisk implementation where I would like to start an update of the PBX (config files, etc) by sending some kind of trigger from a remote machine through SIP and catching that in a (AGI) script in the Asterisk PBX. What is the simplest way to archive this? I don't want to set up other services than SIP on the machine. I am not a SIP expert and I am planning to use one of the available SIP libraries. Thanks, Håkan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Regular Phones - ISDN NT - FXS Adapters
Hello! I'd like to connect regular analog phones (Cordless Panasonic, for instance) to Asterisk. What's the cheapest FXS device on the market that can be used with Asterisk? Where can I find it? It has to be the cheapest to avoid huge import taxes, since I'm planning to buy in bulk. I'm also looking for BRI ISDN cards (cheap and used, preferable) that can be used in NT mode so I can connect my ISDN phones to Asterisk but it seems hard to find on E-bay or any web site Can anybody point me to some direction? Any tips on brand and model? In case there's any retail/vendor reading this, please contactme in private. Thanks! -Dhennys ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap'
Hi Asterisk-ians! Need all of your help. I am stuck with this issue for last few days. I have one X100P installed in my system. My Asterisk is registered with another Asterisk Server/SIP provider as client and the call is successfully received by my Asterisk server (named as starwars). Now, the extentions.conf file states, the incoming INTO * should go out to fxo as below: exten = s,1,Dial(Zap/1/403142142) exten = s,2,Dial(Zap/1/403132663) exten = s,3,hangup whereas other file config is as below: zapata.conf [channels] relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=yes context=bell signalling=fxs_ks callerid=asreceived channel = 1 zaptel fxsks=1 loadzone=us defaultzone=us sip.conf register = 7062210:9211:[EMAIL PROTECTED] [MyService] type=peer username=7062210 fromuser=7062210 secret=9211 host=192.168.7.16 context=incoming fromdomain=sipdom.inf nat=no canreinvite=no dtmfmode=inband so whenever the call comes in from service provider's asterisk to my starwars asterisk, I get the error messages captured below: starwars*CLI sip show registry HostUsername Refresh State 192.168.7.16:5060 7062210105 Registered -- Executing Dial(SIP/192.168.7.14-085a4790, Zap/1/67742142) in new stack Nov 30 01:41:52 NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Dial(SIP/192.168.7.14-085a4790, Zap/1/61002663) in new stack Nov 30 01:41:52 NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Hangup(SIP/192.168.7.14-085a4790, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'SIP/192.168.7.14-085a4790' -- Executing Dial(SIP/192.168.7.14-085a4790, Zap/1/67742142) in new stack Nov 30 01:41:52 NOTICE[524305]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Dial(SIP/192.168.7.14-085a4790, Zap/1/61002663) in new stack Nov 30 01:41:52 NOTICE[524305]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Hangup(SIP/192.168.7.14-085a4790, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'SIP/192.168.7.14-085a4790' please note the output of the following commands: starwars*CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefaultdefault starwars*CLI zap show channel 1 Unable to find given channel 1 starwars*CLI sip show registry HostUsername Refresh State 192.168.7.16:5060 7062210105 Registered starwars*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status MyService/7062210 192.168.7.16255.255.255.255 5060 Unmonitored Thanks Thanks Abdullah. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap gives no ring to the caller...
I have a E1 conected to asterisk all zap channels are ok, but when calls come into Asterisk caller don't hear none ring, the call goes straight into the menu, how can i simulate 2 or 3 rings?
here it is my conf.
exten => s,1,Answer
exten => s,2,Wait,2
exten => s,3,NoOp(${CALLERID})
exten => s,4,ResponseTimeout,45
exten => s,5,DigitTimeout,3
exten => s,6,Background(wellcome)
Adrià Vidal
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Re: [Asterisk-Users] Zap gives no ring to the caller...
On November 29, 2004 12:35 pm, adria vidal wrote:
I have a E1 conected to asterisk all zap channels are ok, but when
calls come into Asterisk caller don't hear none ring, the call goes
straight into the menu, how can i simulate 2 or 3 rings?
exten = s,1,Answer
exten = s,2,Wait,2
exten = s,3,NoOp(${CALLERID})
Wait() before Answer() if you want to have the caller hear a few rings. Why I
don't know, but hey, it's your setup. :-)
-A.
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Re: [Asterisk-Users] Zap gives no ring to the caller...
Have you tried exten = s,2,Ringing
phoenix*CLI show application ringing
phoenix*CLI
-= Info about application 'Ringing' =-
[Synopsis]:
Indicate ringing tone
[Description]:
Ringing(): Request that the channel indicate ringing tone to the user.
Always returns 0.
From: adria vidal [EMAIL PROTECTED]
Subject: [Asterisk-Users] Zap gives no ring to the caller...
Date: Mon, 29 Nov 2004 18:35:36 +0100
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
I have a E1 conected to asterisk all zap channels are ok, but when
calls come into Asterisk caller don't hear none ring, the call goes
straight into the menu, how can i simulate 2 or 3 rings?
here it is my conf.
exten = s,1,Answer
exten = s,2,Wait,2
exten = s,3,NoOp(${CALLERID})
exten = s,4,ResponseTimeout,45
exten = s,5,DigitTimeout,3
exten = s,6,Background(wellcome)
Adrià Vidal
---End of Original Message-
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Re: [Asterisk-Users] Cannot get two TE410Ps to operate correctly in the same machine
Guess at this point I'd call digium support and open a ticket. Yes. I've tried: - ztcfg - rmmod the wct4xxp module and zaptel and modprob'ing again - rebooting the server. No matter what I do, the second card is always internally clocked. The E1 I have plugged into that board is good. It can clock the first card just fine. This is my complete zaptel.conf, just for reference: # E1 ISDN PRI. span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 # E1 R2 span=2,2,0,cas,hdb3 cas=32-46:1101 cas=48-62:1101 # E1 ISDN PRI span=3,0,0,ccs,hdb3 bchan=63-77 dchan=78 bchan=79-93 # T1 Channel bank. span=4,0,0,esf,b8zs fxsks=94-101 fxoks=102-117 # E1 ISDN PRI. Same carrier as span 1. Line can be exchanged with span 1 and all is OK. span=5,1,0,ccs,hdb3 bchan=118-132 dchan=133 bchan=134-148 # E1 Unused. span=6,0,0,ccs,hdb3 unused=149-163 unused=164 unused=165-179 # E1 Unused span=7,0,0,ccs,hdb3 unused=180-194 unused=195 unused=196-210 # T1 Channel bank. span=8,0,0,esf,b8zs fxoks=211-234 Fernando Rich Adamson wrote: Your description makes perfect sense. My system is still getting HDLC overruns, which are certainly a consequence of frame slips because the second card is not getting clocked from the external source. I come back to my basic question: How do you configure an asterisk system so that a second TE410P card recognizes an external clock source? No matter what I've done the second card is always reported as internally clocked. Configuring span 5 (the first on the second board) with exactly the same timing parameters as for span 1 (which works great) is not doing it. What is the trick? Can anybody share the config files for a succesful installation with more than one TE410P boards? I don't recall from your previous postings on this topic, so I'll ask. After config'ing span 5 in /etc/zaptel.conf, are you reloading/restarting the drivers? (That _is_ necessary.) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] IP to IP call without server?
On Mon, 29 Nov 2004 16:38:00 +0900, nkb wrote: I'm not a big fan of supporting proprietary soltuions so I'd avoid Skype. However, what about Free World Dialup? Uses common sip clients, they have the new Pulver communicator which supports video, voice, and text chat. Seems like a good solution. What's the diff. between FWD and Skype other than Skype uses proprietary solution? I'm getting a little confused with these softphones and provider companies. I cant tell what's so great of one from the other. Is there something special about FWD I'm missing that make it better than other solution like Skype? Thanks! From my standpoint as the user of * for my home office I prefer FWD. They have an experimental program that lets me use IAX2 to connect my server to theirs. While it's not a production program yet (Ed Guy assures us that will happen soon) it has been useful. The fact that they support the * community makes me inclibed to use/support their effort over Skype. FWIW, Skype isn't anything spectacularily innovative anyway. There's been much coverage of this fact in the bloggosphere. It works, connects Skype users, PSTN if you pay. How is that any different of better than anything else? They handle NAT well, but so do IAX2 clients. They generate a lot of hype, which many SIP providers don't. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Parking from call group problems
Trouble in Parking Paradise! Good Day all, I have a situation that I have tracked as far as I can take it and am looking for assistance into the matter. My setup. Asterisk 1.0.1 with the AMP config environment. When I have auto attendant answer the phone and I dial my extension 2204 The call come through and If I park it it gets place into extension 701 (the first parking spot). I am able to successfully pick the call up and all works fine. Then I repeat step one, however If I leave the call parked, and upon the timeout the call rings back to my phone. Excellent this is what I expected. NOW The problem A call comes in and goes to a call group (700) in this case. This call group then rings 3 extensions (2203, 2204, 2201) I can pick up the call and place it into parking . I am then able to grab the call out of parking slot with no problems. BUT. If I let the call in the parking slot, upon it timing out, the person who was parked gets a loud screeching noise and the call is eventually dropped. I have it narrowed down to the following. When the call is parked from the auto attendant the ring back is set to snip from pastebin.ca/2452 *** Works Correctly and send call back to the extension that parked the call ** 069 == Parked Zap/3-1 on 701. Will timeout back to aa_1,2204,1 in 45 seconds 070 -- Playing 'digits/7' (language 'en') 071 -- Playing 'digits/0' (language 'en') 072 -- Playing 'digits/1' (language 'en') 073 -- Stopped music on hold on Zap/3-1 074 -- Started music on hold, class 'default', on Zap/3-1 075 -- Added extension '701' priority 1 to parkedcalls 076 == Spawn extension (macro-dial, s, 1) exited KEEPALIVE in macro 'dial' on 'Zap/3-1' 077 == Spawn extension (macro-exten-vm, s, 4) exited KEEPALIVE in macro 'exten-vm' on 'Zap/3-1' 078 == Spawn extension (aa_1, 2204, 1) exited KEEPALIVE on 'Zap/3-1' 079 == Timeout for Zap/3-1 parked on 701. Returning to aa_1,2204,1 snip from pastebin.ca/2453 *** DOES NOT Work Correctly ** 078 == Parked Zap/3-1 on 701. Will timeout back to ext-group,700,3 in 45 seconds 079 -- Playing 'digits/7' (language 'en') 080 -- Playing 'digits/0' (language 'en') 081 -- Playing 'digits/1' (language 'en') 082 -- Stopped music on hold on Zap/3-1 083 -- Started music on hold, class 'default', on Zap/3-1 084 -- Added extension '701' priority 1 to parkedcalls 085 == Spawn extension (macro-dial, s, 1) exited KEEPALIVE in macro 'dial' on 'Zap/3-1' 086 == Spawn extension (macro-rg-group, s, 4) exited KEEPALIVE in macro 'rg-group' on 'Zap/3-1' 087 == Spawn extension (ext-group, 700, 3) exited KEEPALIVE on 'Zap/3-1' 088 == Timeout for Zap/3-1 parked on 701. Returning to ext-group,700,3 This is as far as my skill level allows me to go with this problem. Anyone willing to assist in determining how to make the call group actually set the ringback extension to the actual extension that did the parking would be most appreciated. Thanks for your time list. Files which may be helpful. http://pastebin.ca/2452 Manager debug messages of working parking situation http://pastebin.ca/2453 Manager Debug messages of NON_Functional Parking http://pastebin.ca/2454 Extensions.conf http://pastebin.ca/2455 Extensions_additional.conf ~ron ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap'
Looks like asterisk is trying to send the call out Zap/1, but is having an issue that appears almost like there is no telephone line attached to your x100p card. Is this machine located in the US and are you sure the pstn line is properly plugged to the card? Another remote possibility is that asterisk is detecting a busy signal on the pstn line. If you are in the US, what is 403142142? That isn't a standard US telephone number. (Nine digits?) Again, if this is in the US, best guess is that sending those digits out the pstn line is resulting in some sort of busy/congestion tone coming back from your telco. Hi Asterisk-ians! Need all of your help. I am stuck with this issue for last few days. I have one X100P installed in my system. My Asterisk is registered with another Asterisk Server/SIP provider as client and the call is successfully received by my Asterisk server (named as starwars). Now, the extentions.conf file states, the incoming INTO * should go out to fxo as below: exten = s,1,Dial(Zap/1/403142142) exten = s,2,Dial(Zap/1/403132663) exten = s,3,hangup whereas other file config is as below: zapata.conf [channels] relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=yes context=bell signalling=fxs_ks callerid=asreceived channel = 1 zaptel fxsks=1 loadzone=us defaultzone=us sip.conf register = 7062210:9211:[EMAIL PROTECTED] [MyService] type=peer username=7062210 fromuser=7062210 secret=9211 host=192.168.7.16 context=incoming fromdomain=sipdom.inf nat=no canreinvite=no dtmfmode=inband so whenever the call comes in from service provider's asterisk to my starwars asterisk, I get the error messages captured below: starwars*CLI sip show registry HostUsername Refresh State 192.168.7.16:5060 7062210105 Registered -- Executing Dial(SIP/192.168.7.14-085a4790, Zap/1/67742142) in new stack Nov 30 01:41:52 NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Dial(SIP/192.168.7.14-085a4790, Zap/1/61002663) in new stack Nov 30 01:41:52 NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Hangup(SIP/192.168.7.14-085a4790, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'SIP/192.168.7.14-085a4790' -- Executing Dial(SIP/192.168.7.14-085a4790, Zap/1/67742142) in new stack Nov 30 01:41:52 NOTICE[524305]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Dial(SIP/192.168.7.14-085a4790, Zap/1/61002663) in new stack Nov 30 01:41:52 NOTICE[524305]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Hangup(SIP/192.168.7.14-085a4790, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'SIP/192.168.7.14-085a4790' please note the output of the following commands: starwars*CLI zap show channels Chan Extension Context Language MusicOnHold pseudodefaultdefault starwars*CLI zap show channel 1 Unable to find given channel 1 starwars*CLI sip show registry HostUsername Refresh State 192.168.7.16:5060 7062210105 Registered starwars*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status MyService/7062210 192.168.7.16255.255.255.255 5060 Unmonitored ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Directed call pickup
Is anyone successfully using directed call pickup with asterisk? *8exten to only pick up that persons extension if the phone is ringing.. It says in the wiki asterisk supports it but I can not get it to work.. Thanks -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vonage integration... Hardware or Softphone type acct.
Hi All, I've got an * PBX up with couple of stations and now I'd like to integrate my Vonage service for outgoing PSTN calls. Is this possible if I have an account with them that uses their hardware box (ATA186) or do I need a 'softphone' account? thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Directed call pickup
On Mon, 29 Nov 2004, Matthew Marlowe wrote: Is anyone successfully using directed call pickup with asterisk? *8exten to only pick up that persons extension if the phone is ringing.. It says in the wiki asterisk supports it but I can not get it to work.. You could use app_intercept from http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002692 Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] overriding DTMF and codec from dialplan?
Brian West wrote: OH But it is just that simple. You also have: -= Info about application 'ImportVar' =- [Synopsis]: Set variable to value [Description]: ImportVar(#n=channel|variable): Sets variable n to variable as evaluated on the specified channel (instead of current). If prefixed with _, single inheritance assumed. If prefixed with __, infinite inheritance is assumed. I give up, my mistake. bkw It's not so simple. Check http://bugs.digium.com/bug_view_advanced_page.php?bug_id=928 for the details. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: IAX2 and FWD problems?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: November 29, 2004 1:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: IAX2 and FWD problems? If there are problems with FWD's IAX gateway, please fill out a trouble report at FWD. If enough of us do it, the guys on the desk will realize there really is a problem and let someone know! Hi Ed, You can't not know that this has been talked about for weeks on the Pulver board, including people saying they sent trouble tickets with no reply. I'm sure he does know. The conclusion most people have drawn is basically what you state, that FWD is experimental. Keep in mind that Ed does not own Pulver, and is almost certainly not the sole decision-maker with repect to their policies. What I hear him saying is that the more of us that make our voices heard with respect to FWD and IAX, the easier it will be for him to make IAX support a priority for rest of the executive team at Pulver. If you'll recall, Ed represented Pulver at Astricon (and gave a VERY enjoyable and informative presentation; thanks Ed) so be assured that Ed is someone who champions Asterisk and IAX. IAX will be fully supported in the very near future! Can't be any plainer than that. We can help speed up the process by doing some cheering for IAX. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to rid yourself of Broadvoice
If you're having problems cancelling your Broadvoice service like I was simply do the following: 1) Dial 1-978-418-7300 (don't block your caller ID/CPN). 2) Announce your displeasure with the service to their voicemail (it should tell you something that you have to leave voicemail to get support). Expletives will reduce the waiting in step 3 from 10 minutes to 5 minutes. 3) Wait 10 minutes and your phone will ring with a Broadvoice representative ready to receive your appreciation for being a great company. After two months of no service, dozens of e-mails and phone calls, and canned we don't support Asterisk responses this finally got the job done. American Express is doing the rest of the legwork in getting my $30+ back for the entire month the service didn't work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Fwd: Re: [Asterisk-Users] Adit 600 channel bank in UK setting]
Jon, I actually had some more discussions with Tim on this issue, and it seems that the channel bank would still be a good option to choose for internal purposes. I would not see any other solution than a channel bank to connect many 2wire phones into one asterisk box. I had a talk today with Carrier Access, and it seems that the adit would do us fine. The fxs cards of the adit 600 are actually reprogrammable for uk phones (dip switches). We have requested a test platform from CAC which I hope would arrive here shortly, and we would test how the fxs ports work with different uk phones. I would really be unhappy to scratch our existing phone cable network and to lay an entire new LAN and to buy many IP phones. First of all - new installations always have teething problems. Then the admin headache with the many IP phones. Also - the solution doesn't scale very easily. For each new phones we need a new network socket... or a hub. Then one mains connection per phone (with power supply - more fire risk). And on and on... Using 2wire phones eliminates all that - cables are there already, users can buy any phone they like, we can put in additional sockets without admin effort and so on. Really - 2wire rocks! The pstn connectivity is an entirely different matter. The fxo cards of the adit600 seem not to be EU approved for public ( = pstn) connection (mainly) because of emission regulations. Also, I would not be sure how well the impedances of the fxo ports and the BT lines match (Echo problem, noise etc.). I am looking into using a voip gateway with fxo ports for our BT line connectivity. I found one on sale on a UK website, see http://lists.digium.com/pipermail/asterisk-users/2004-November/075118.html However, it looks to me more and more that for our internal phones we use the adit 600. Lets see what the testing will show... Peter On Thursday 18 November 2004 22:16, Tim Robinson wrote: Channel banks are a peculiar US thing. Be careful! You will almost certainly be better off using voip handsets (SNOMs are cool, avoid Grandstream for anything other than domestic environment) and a few Sipura-type ATA's for the analogue fax machines etc. or some Digium analogue cards. So what would you advise using in the UK to interface with standard 2 wire phones - I'm trying to avoid having to use ata type adapters. Jon -- There are 10 kinds of people in the world, those who understand binary, and those who don't. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TOS Settings to DSCP
I am assuming that the TOS values directly map to DSCP values in the ip header. Is this a correct assumption? If so, can someone tell me the correct setting to set call control packets with a DSCP of AF31(011010) and media with EF(101110)? So would the setting for AF be TOS=46?? Is it possible to mark the media and call control separately?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TOS Settings to DSCP
I am assuming that the TOS values directly map to DSCP values in the ip header. Is this a correct assumption? If so, can someone tell me the correct setting to set call control packets with a DSCP of AF31(011010) and media with EF(101110)? So would the setting for AF be TOS=46?? Is it possible to mark the media and call control separately?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Variable substitution - How can I do Dial(${DIALSTRING}) where ${DIALSTRING} is 'SIP/201, 15, tT'?
I've been banging my head against a brick wall for the last hour and I'm
sure this is one of those easy to solve things - just that I can't see the
wood for the trees.
I'm trying to do:
---
[some-context]
Exten = s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass)
[macro-dodial]
Exten = s,1,SetCallerID(${ARG2})
Exten = s,2,SetMusicOnHold(${ARG3})
Exten = s,3,Dial(${ARG1})
---
(there's a lot more to it than that, but the above should give you an idea
of what I'm trying to achieve)
A command starting at some-context,s,1 returns the following error:
-- Executing Macro(SIP/200-b9d9,
dodial|SIP/201,15,tT|123456|MOHClass) in new stack
-- Executing SetCallerID(SIP/200-b9d9, 123456) in new stack
-- Executing SetMusicOnHold(SIP/200-b9d9, MOHClass) in new stack
Nov 29 19:43:09 WARNING[802835]: pbx.c:1280 pbx_extension_helper: No
application 'Dial{${ARG1})' for extension (macro-dodial, s, 5)
== Spawn extension (macro-dodial, s, 5) exited non-zero on 'SIP/200-b9d9'
in macro 'dodial'
== Spawn extension (from-sip, 201, 2) exited non-zero on 'SIP/200-b9d9'
Is there any way I can achieve this without having to pass each of the dial
parameters to the macro individually?
Cheers,
Nick.
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[Asterisk-Users] Prepaid
Is anyone successfully using asterisk-prepaid-0.3.1? I try to configure but doesn't work. It said that you need to do a few step, copy a few files and that is. Please, if someone has any tips about the configuration, answer me. Sebastian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fedora Core 3 TDM400P cards?
Has anybody managed to get Fedora Core 3 and the zapata drivers working with the TDM400P cards? Everything is going fine untilI do modprobe mcfxs Any ideas? It's a straight install of Fedora Core 3. Graeme ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXy power source from Radio Shack
Anyone using this 110-220v 9v 1500ma supply? If so, which DC plug adapter does the IAXy need? A friend brought the wall wart over here for me but she said you get one plug free and she didn't know which one it took. Does anyone here know? http://www.radioshack.com/category.asp?catalog%5Fname=CTLGcategory%5Fname=CTLG%5F009%5F001%5F001%5F003Page=1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Uniden UIP200 -- configured, but not working?
On Mon, 2004-29-11 at 00:21 -0500, Ken D'Ambrosio wrote: no calls actually take place, either in-bound or out-bound. With sip debug going, I get this: The phone's firmware rev. is BS4.59a The unidenmac.txt file is as follows: MyLcdDisplay22 MyDialNumber22 UserNameForRegistrar22 PasswordForRegistrarfoo TimeDisplay Enable Try adding : UserNameForProxy 22 PasswordForProxy foo Lastly, if, in the unidencom.txt file, I put a proxy bit in, I get an honest-to-goodness busy signal, which certainly seems better than nothing. But I'm not using a proxy -- I'm going straight to the Asterisk box. I don't use a proxy either, but always define ProxyServer and OutboundProxy1 in unidencom.txt (in addition to Registrar1/2) Ryan PS: Try using Ethereal to debug problems like this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Fwd: Re: [Asterisk-Users] Adit 600 channel bank in UK setting]
On Monday 29 November 2004 19:12, Peter Hoppe wrote: Jon, I actually had some more discussions with Tim on this issue, and it seems that the channel bank would still be a good option to choose for internal purposes. I would not see any other solution than a channel bank to connect many 2wire phones into one asterisk box. I had a talk today with Carrier Access, and it seems that the adit would do us fine. The fxs cards of the adit 600 are actually reprogrammable for uk phones (dip switches). We have requested a test platform from CAC which I hope would arrive here shortly, and we would test how the fxs ports work with different uk phones. I'd be interested if you could report back to the list with your findings. I would really be unhappy to scratch our existing phone cable network and to lay an entire new LAN and to buy many IP phones. First of all - new installations always have teething problems. Then the admin headache with the many IP phones. Also - the solution doesn't scale very easily. For each new phones we need a new network socket... or a hub. Then one mains connection per phone (with power supply - more fire risk). And on and on... Using 2wire phones eliminates all that - cables are there already, users can buy any phone they like, we can put in additional sockets without admin effort and so on. Really - 2wire rocks! Wiring isn't my problem - all our connections are over cat5e. In our building we rent out most of the offices, the clients provide their own phones and we simply provide the lines. Some of their phones look pretty expensive, so I'd rather not tell them that they can't use them anymore. As with any office scenario power sockets are an issue - people never put enough in when they design the rooms. The pstn connectivity is an entirely different matter. Our connection is via pri so this isn't a great issue. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Variable substitution - How can I do Dial(${DIALSTRING}) where ${DIALSTRING} is 'SIP/201, 15, tT'?
Nick Barnes wrote:
snip
Exten = s,3,Dial(${ARG1})
snip
---
Nov 29 19:43:09 WARNING[802835]: pbx.c:1280 pbx_extension_helper: No
application 'Dial{${ARG1})' for extension (macro-dodial, s, 5)
snip
Look very close at this output. Unless you did a copy-and-paste of your
extensions.conf segment, I would say that you typed, in your actual
extensions.conf, 'Dial{${ARG1})' instead of 'Dial(${ARG1})'. Notice the
'{' vs '(' ? That, or did you type out that entire CLI output?
Anyway, my $0.00. (feeling stingy today :)
Chris
--
Christopher L. Wade Unistar-Sparco Computers, Inc.
Senior Systems Administratordba Sparco.com
Email: [EMAIL PROTECTED] 7089 Ryburn Drive
Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053
Fax: (901) 872 8482 USA
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Re: [Asterisk-Users] fxo connection in the UK
On 29 Nov 2004, at 17:30, Peter Hoppe wrote: Micronet SP5054 VoIP Gateway 4 FXO Ports We bought one of these units and had a lot of grief with them. The SIP firmware isn't great at all IMHO. For H.323 they work just fine. Stephan Wik ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Fedora Core 3 TDM400P cards?
Has anybody managed to get Fedora Core 3 and the zapata drivers working with the TDM400P cards? Everything is going fine until I do modprobe mcfxs Any ideas? It's a straight install of Fedora Core 3. The driver for the TDM400P card is wctdm. Not sure what a mcfxs happens to be. The old wcfxs was renamed to wctdm some time ago. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Reboot Script PRI errors!!
Has anyone written an equivalent script to remote reboot Cisco 79XX phones?
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Baker
Sent: Monday, 29 November 2004 17:29
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom Reboot Script PRI errors!!
Kevin wrote:
There is a reboot script posted on the wiki to reboot Polycom
telephones. When I execute this script, I get the following messages.
I am concerned as this is causing issues with asterisk and the PRI.
Does anyone have any ideas why this would be happening?
asterisk console:
-- Remote UNIX connection
-- Remote UNIX connection disconnected
and in the Asterisk Log:
Nov 28 22:30:42 NOTICE[1099909936]: PRI got event: 6 on Primary
D-channel of span 1 Nov 28 22:43:08 NOTICE[1099909936]: PRI got event:
6 on Primary D-channel of span 1
Script:
#!/usr/bin/perl -w
use Net::Ping;
use Socket;
$polycompath = '/home//';# Where you keep your config files
$arp = '/sbin/arp'; # Location of arp command
$sipserver = '192.168.XXX.XXX'; # IP of asterisk server
$phone = shift;
checkphone($phone);
touch( arp2config($phone) );
reboot_sip_phone( $phone, $sipserver, Reboot );
sub checkphone { # Checks for existence of phone, makes sure
# it's in arp table
$activephone = shift;
# Populate ARP table
print Checking ARP table.\n;
$p = Net::Ping-new(icmp);
if ( $p-ping( $activephone, 2 ) ) {
print $activephone is ;
print reachable.\n;
}
else { die Polycom at , $activephone, is not reachable!; }
sleep(1);
$p-close();
}
sub arp2config {# Gets mac address from arp table, converts
# to a polycom config filename, makes sure
# the config file exists
$arpip = shift;
open( ARP, $arp -an| ) || die Couldn't open arp table: $!\n;
print checking for polycom config name..., \n;
while (ARP) {
chomp;
$addr = $_;
$ip = $_;
$addr =~ s/.*
([\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+:[\d\w]+).*/$1/;
$addr =~ s/://g;
$addr = lc($addr) . '.cfg';
$ip =~ s/.*?(\d+\.\d+\.\d+\.\d+).*/$1/;
if ( $ip eq $arpip ) {
last;
}
}
$polycomconfig = $polycompath . $addr;
unless ( -e $polycomconfig ) {
print sorry, polycom config file , $polycomconfig,
is not found.\n\n;
exit;
}
return $polycomconfig;
}
sub touch {# We need to touch the config files or the phone
# won't reboot - it depends on time synchronization
print touching config file , $polycomconfig, \n;
my $now = time;
local (*TMP);
foreach my $file (@_) {
utime( $now, $now, $file )
|| open( TMP, $file )
|| die ($0: Couldn't touch file: $!\n);
}
}
sub reboot_sip_phone {# Send the phone a check-sync to reboot it
$phone_ip = shift;
$local_ip = shift;
$sip_to = shift;
$sip_from = 0;
$tm = time();
$call_id = $tm . msgto$sip_to;
$httptime = `date -R`;
$MESG = NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP $local_ip
From: sip:[EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED]
Event: check-sync
Date: $httptime
Call-ID: [EMAIL PROTECTED]
CSeq: 1300 NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
;
$proto = getprotobyname('udp');
socket( SOCKET, PF_INET, SOCK_DGRAM, $proto );
$iaddr = inet_aton($phone_ip);
$paddr = sockaddr_in( 5060, $iaddr );
bind( SOCKET, $paddr );
$port = 5060;
$hisiaddr = inet_aton($phone_ip);
$hispaddr = sockaddr_in( $port, $hisiaddr );
if ( send( SOCKET, $MESG, 0, $hispaddr ) ) {
print reboot of phone , $phone_ip, was successful, \n;
}
else { print reboot of phone , $phone_ip, failed, \n; }
}
exit;
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Kevin -
I rewrote this some time ago because of some issues with Polycom's latest
bootroom/sip update. Try this:
Also, serctl, part of the ser package, has a cisco_restart parameter that
works on Polycoms as well.
John
#!/usr/bin/perl -w
use Net::Ping;
use Socket;
$polycompath = '/home/PlcmSpIp/';# Where you keep your polycom files
$arp = '/sbin/arp'; # Location of arp command
$sipserver = '192.168.XXX.XXX'; # IP of asterisk server
$phone = shift;
checkphone($phone);
touch( arp2config($phone) );
reboot_sip_phone( $phone, $sipserver, get_extension($phone) );
sub checkphone { # Checks for existence
Re: [Asterisk-Users] OH323 Rocks :) --- H323 guys, use it to solve no answer at this time problem!!!
Thank you very much. I have been trying it but I get in trouble installing it I will try it again. Nahuel Ramos. On Fri, 26 Nov 2004 12:11:20 +0200, Michael Manousos [EMAIL PROTECTED] wrote: Thanks. I appreciate that. Michael. kido noagbodji wrote: i have had some problems with the H323 channel ... Other party not anwsering SIP 2 H323 bridge. the chan_oh323 solves the problem. Use it. (Even though it is quite complicated to install but READ the README file) Nahuel that should solve it!! Kido ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Fedora Core 3 TDM400P cards?
--Original Message Text--- From: Graeme Ogilvie Date: Mon, 29 Nov 2004 19:45:41 - Has anybody managed to get Fedora Core 3 and the zapata drivers working with the TDM400P cards? Everything is going fine until I do modprobe mcfxs Any ideas? It's a straight install of Fedora Core 3. Graeme I tried to use FC3 in rebuilding an * server over the weekend. However, the Anaconda installer failed very early in the install so I ended up going back to FC1. The TDM400P works great though! Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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: IAX2 and FWD problems?
On Mon, 29 Nov 2004 14:03:42 -0500, Jim Van Meggelen wrote: What I hear him saying is that the more of us that make our voices heard with respect to FWD and IAX, the easier it will be for him to make IAX support a priority for rest of the executive team at Pulver. If you'll recall, Ed represented Pulver at Astricon (and gave a VERY enjoyable and informative presentation; thanks Ed) so be assured that Ed is someone who champions Asterisk and IAX. IAX will be fully supported in the very near future! Can't be any plainer than that. We can help speed up the process by doing some cheering for IAX. Whatever the case in the future I see from a simple test that IAX2 to FWD is working a of 2pm CST. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Reboot Script PRI errors!!
Simon Brown wrote: Has anyone written an equivalent script to remote reboot Cisco 79XX phones? Simon Check the wiki... there are two versions for cisco, one that uses the check-sync sip message, the other that simply logs into the phone via telnet and 'reset's the phone. I have made some changes to the latter and use it in production along with some other console commands to remotely setup things like my intercom line, server push, etc. Anyway, the wiki link is http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx If you look about 1/3 way down the page you'll see a link to 'reboot.pl' pointing to http://mklein.bendtel.net/mkreboot.pl That should get you going. Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 USA ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Fedora Core 3 TDM400P cards?
The card must have shipped with some old documentation then. I'll give that a go tomorrow. Thanks. Graeme -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: 29 November 2004 20:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fedora Core 3 TDM400P cards? Has anybody managed to get Fedora Core 3 and the zapata drivers working with the TDM400P cards? Everything is going fine until I do modprobe mcfxs Any ideas? It's a straight install of Fedora Core 3. The driver for the TDM400P card is wctdm. Not sure what a mcfxs happens to be. The old wcfxs was renamed to wctdm some time ago. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Vonage integration... Hardware or Softphone typeacct.
Vonage hard lines only run through an FXO port. There's no feasible way to get the SIP credentials in order to terminate directly into *. You can terminate a soft line directly into asterisk, both for incoming and outgoing calls -- for configuration examples, simply search the list archives (use google and add site:lists.digium.com to your search-term). -Original Message- From: Angus Berry [mailto:[EMAIL PROTECTED] Sent: Monday, November 29, 2004 12:27 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Vonage integration... Hardware or Softphone typeacct. Hi All, I've got an * PBX up with couple of stations and now I'd like to integrate my Vonage service for outgoing PSTN calls. Is this possible if I have an account with them that uses their hardware box (ATA186) or do I need a 'softphone' account? thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Packet8 integration into Asterisk?
Hello everybody Does anybody may tell me how to make the configuration SIP on Asterisk for one Packet8 account? I want to use it with my Asterisk Regards John F --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comparision of IAX2, FWD, iaxtel etc etc.
Hi, I've been setting up * recently and slowly getting to grips with it, however I'm getting rather confused with all the different configs for IAX calls, FWD calls iaxtel etc etc. What I think I need it a basic understanding or even a comparison of these different voip systems (if thats what they are?) I'd like to be able to make calls to other voip users, both in the UK and abroad and also make calls out on to the UK and US networks if possible. I'd also like to receive calls via a voip number? My * box is on my local LAN behind a nat router if that helps. Many thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 newsgrup proposal or phpBB forum
Hi all, I can see huge traffic here over 400 post in 4 days. My proposal is to create asterisk newsgrup proposal or phpBB forum what do think about it ? BR, Corvin btw. I'm admin of phpBB Forum (slackware forum - polish language), nearly 900 users. I think if someone will prepare it good it can be great project. (but I have 7 person team). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
