[asterisk-users] asterisk sip trunk configure
Hi, I am trying to make external sip calls by using asterisk. Please provide information regarding sip trunk configuration in conf files. Setup is as below, * * *Case A:* Register two soft phones [X-lite] with 1000 and 10001 numbers to asterisk PBX [running in 192.168.1.11] and able to make calls in between. Sip.conf == [general] context=default bindport=5060 bindaddr=192.168.1.11 srvlookup=yes [1000] type=friend nat=yes host=dynamic canreinvite=no context=default allow=ulaw [1001] type=friend nat=yes host=dynamic canreinvite=no context=default allow=ulaw extensions.conf [default] exten = 1000,1,Dial(SIP/1000) exten = 1001,1,Dial(SIP/1001) *Case B:* I have register other phone with ondo sip server running on other PC [192.168.1.12] with number as 6001. Now, Want to make calls between this two [asterisk PBX [1000/1001] on 192.168.1.11 and ondo [6001] on 192.168.1.12]. Please suggest how to configure sip trunk in conf files. Thanks in advance. Regards, Garge. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small PC to build and run Asterisk
On Wed, Jun 16, 2010 at 3:46 AM, Michael Graves mgra...@mstvp.com wrote: Some distro's, like Askozia and Astlinux, have been specifically engineered around running from flash media. This basic form of operation has been well proven in projects like monowall and pfsense. I think you hit the essence of the argument for using these embedded systems, Michael. And we both know from experience and through knowing the people involved that they're both excellent choices! I am now considering using an about-to-be-retired Mac Mini. I'm pretty sure it can be done. How well it might work is another story. I'm pretty much giving up on Skype for Asterisk (and Skype for SIP) now that I realize that they'll be charging a monthly fee that is disproportionately high compared to my need to let Skype users call us. We'll know the pricing in Q4 of 2010, but it looks to be about $15/month for one user. $5 for the channel and $10 for Skype Manager. Maybe something for each name, too? /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail vm-intro played even when temp greetingis setup
Thanks Danny,
I made it working adapting your dialplan
exten = 12345,1,Answer
exten = 12345,n,System(/bin/ls
/var/spool/asterisk/voicemail/default/${EXTENSION}/temp.wav)
exten = 12345,n,verbose(returned ${SYSTEMSTATUS}
exten = 12345,n,Gotoif($[${SYSTEMSTATUS} = SUCCESS]?play1)
exten = 12345,n,Voicemail(${extensi...@test)
exten = 12345,n,hangup
exten = 12345,n(play1),Voicemail(s${extensi...@test)
exten = 12345,n,hangup
Thanks so much,
Jonathan
On Tue, Jun 15, 2010 at 9:58 PM, Danny Nicholas da...@debsinc.com wrote:
This should do the trick – might have to change greet.WAV to some other
value
exten = 930,1,Answer
exten = 930,n,System(/bin/ls
/var/spool/asterisk/voicemail/default/${NUMBER}/greet.WAV)
exten = 930,n,verbose(returned ${SYSTEMSTATUS}
exten = 930,n,Gotoif($[${SYSTEMSTATUS} = SUCCESS]?play1)
exten = 930,n,Voicemail(s${numb...@test)
exten = 930,n,hangup
exten = 930(play1),n,Voicemail(${numb...@test)
exten = 930,n,hangup
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonathan González
*Sent:* Tuesday, June 15, 2010 3:37 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Voicemail vm-intro played even when temp
greetingis setup
Hi there,
I am configuring a small voicemail server and I am facing the following
problem.
Executing this command: exten = 1234,1,VoiceMail(${numb...@test)
When a user does not have a customized temporary greeting vm-intro message
is played asking for the message to the user but when the user has already a
temporary greeting both the temporary greeting and vm-intro are played.
Basically what I would like to do is to avoid this second scenario so when a
user has a customized temporary greeting just that is played and not
vm-intro is played.
I have seen that to avoid the reproduction of vm-intro I can use the s
flag, doing something like this:
exten = 1234,1,VoiceMail(s${numb...@test)
But the problem is that if I do that nothing is played for the users that
don't have personalized greeting.
Any help would be appreciated.
Thanks in advance,
Jonathan
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[asterisk-users] Asterisk +Dahdi does not work with BRI NT
Hi all, As i tested with Asterisk+dahdi+libpri with openvox BRI with NT mode, the ISDN phone does not work. There is the setting and error. ***Enviroment* Asterisk-1.6.1.18 dahdi-linux-2.3.0.1 dahdi-tool-2.30. libpri-1.4.11.2 CentOS-5.5 OpenVox B400P ** In my case , I set prot 1 and port 2 as NT mode ,port 3 and port 4 as TE mode. system.conf*** # Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 16 22:49:10 2010 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,ami # termtype: te bchan=1-2 hardhdlc=3 echocanceller=mg2,1-2 # Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 RED span=2,2,0,ccs,ami # termtype: te bchan=4-5 hardhdlc=6 echocanceller=mg2,4-5 # Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 RED span=3,3,0,ccs,ami # termtype: te bchan=7-8 hardhdlc=9 echocanceller=mg2,7-8 # Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 span=4,4,0,ccs,ami # termtype: te bchan=10-11 hardhdlc=12 echocanceller=mg2,10-11 # Global data loadzone= us defaultzone = us ***dahdi-channles.conf** ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 16 22:49:10 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) group=0,11 context=from-internal switchtype = euroisdn signalling = bri_net channel = 1-2 context = default group = 63 ; Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 RED group=0,12 context=from-internal switchtype = euroisdn signalling = bri_net channel = 4-5 context = default group = 63 ; Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 RED group=0,13 context=from-pstn switchtype = euroisdn signalling = bri_cpe channel = 7-8 context = default group = 63 ; Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 group=0,14 context=from-pstn switchtype = euroisdn signalling = bri_cpe channel = 10-11 context = default group = 63 **dmesg* wcb4xxp :01:03.0: Port 1: NT mode wcb4xxp :01:03.0: Port 2: NT mode wcb4xxp :01:03.0: Port 3: TE mode wcb4xxp :01:03.0: Port 4: TE mode wcb4xxp :01:03.0: Did not do the highestorder stuff wcb4xxp :01:03.0: Configuring span 1 wcb4xxp :01:03.0: new card sync source: port 3 wcb4xxp :01:03.0: Configuring span 1 wcb4xxp :01:03.0: Configuring span 2 wcb4xxp :01:03.0: Configuring span 3 wcb4xxp :01:03.0: Configuring span 4 dahdi_echocan_mg2: Registered echo canceler 'MG2' dahdi: Registered tone zone 0 (United States / North America) wcb4xxp :01:03.0: new card sync source: port 1 ** *CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefault default In Service 1from-internal default In Service 2from-internal default In Service 4from-internal default In Service 5from-internal default In Service 7from-pstn default In Service 8from-pstn default In Service 10from-pstn default In Service 11from-pstn default In Service ** *CLI pri show spans *PRI span 1/0: Provisioned, Down, Active* PRI span 2/0: Provisioned, In Alarm, Down, Active PRI span 3/0: Provisioned, In Alarm, Down, Active PRI span 4/0: Provisioned, Up, Active * TE mode can works well. but the NT mode can not work. I do not know why the status always show Down . Any idea for that? I am looking forward to your help. Thank you! liu xin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with dahdi and with freepbx
Hi to all, I use FreePBX version 2.7.0.2 with dahdi. The first problem is with dahdi: At the system startup i can't find a way to start correctly Asterisk with Dahdi. My boot configuration is the following: /etc/rc.d/after.local /usr/sbin/rcdahdi start sleep 15 /usr/local/sbin/amportal start /sbin/route add -host 85.38.234.9 gw 192.168.2.1 /usr/bin/python /usr/local/bin/alice-client.py --enable-all But, in this way, dahdi don't work with asterisk... I have to manually stop amportal, stop dahdi and then restart dahdi and amportal to have it to work. I don't figure why, seems a timing problem, but with the sleep 15 no resolution at all... Any Hint? My OS is OpenSuSE 11.2. The second problem is with the web interface. I have a CallerID Lookup Source it uses a mysql query to give the name from the callerID, and this works perfectly. The only thing, most annoying, is the cache results box. If i flag it, hit Submit changes and then apply changes if is necessary, i return to that page and the chace result box is not checked... no conf saved! I have looked at the webserver logs, activated php DisplayErrors, but no hit of what can be. The only method i have found is to connect to the mysql database where are stored the configurations, ad make manually an update of the field cache with an entry of 1. But, in any case, this value returns to 0, i don't know why... Any hint also for this? Thank you to all, have a nice day! Cordially, Claudio. -- Claudio Prono OPST System Developer Gsm: +39-349-54.33.258 @PSS Srl Tel: +39-011-32.72.100 Via San Bernardino, 17Fax: +39-011-32.46.497 10141 Torino - ITALY http://atpss.net/disclaimer PGP Key - http://keys.atpss.net/c_prono.asc -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with dahdi and with freepbx
On Wed, Jun 16, 2010 at 10:28:48AM +0200, Claudio Prono wrote: Hi to all, I use FreePBX version 2.7.0.2 with dahdi. The first problem is with dahdi: At the system startup i can't find a way to start correctly Asterisk with Dahdi. My boot configuration is the following: /etc/rc.d/after.local Running it from there is the first sign of problems. You should have used proper init scripts. E.g. the asterisk and dahdi ones. This would have allowed you restarting some stuff without fully rebooting the system. /usr/sbin/rcdahdi start sleep 15 This is a great indication to the fact that 'sleep' is normally a bad cure for races. Why not just: /etc/init.d/dahdi start (not in the background) Please check the OpenSUSE packaging of Asterisk and DAHDI. /usr/local/sbin/amportal start /sbin/route add -host 85.38.234.9 gw 192.168.2.1 Isn't there a better place for the network settings? /usr/bin/python /usr/local/bin/alice-client.py --enable-all But, in this way, dahdi don't work with asterisk... I have to manually stop amportal, stop dahdi and then restart dahdi and amportal to have it to work. I don't figure why, seems a timing problem, but with the sleep 15 no resolution at all... Any Hint? My OS is OpenSuSE 11.2. Please provide some useful output. To quote one IRC bot: Look buddy, doesn't work is a vague statement. Does it sit on the couch all day long? Does it procrastinate doing the dishes? Does it beg on the street for change? Please be specific! Define 'it' and what it isn't doing. Give us more details so we can help you without needing to ask basic questions like what's the error message?. Ask me about smart questions and errors. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports
Tried this... it got connected, but I can't hear any audio now whereas codec was allowed and both negotiated on alaw. Deepika -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faisal Hanif Sent: 15 June 2010 18:30 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports Try setting insecure=port,invite in sip peer config. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades Sent: Tuesday, June 15, 2010 9:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports Deepika Nijhawan wrote: It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' So the question is why didnt it match anything. If the phones are registering then they should reregister before choosing a different port. Are they going through a firewall by any chance? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports
Sounds like you have a firewall or NAT issue Deepika Nijhawan wrote: Tried this... it got connected, but I can't hear any audio now whereas codec was allowed and both negotiated on alaw. Deepika -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faisal Hanif Sent: 15 June 2010 18:30 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports Try setting insecure=port,invite in sip peer config. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades Sent: Tuesday, June 15, 2010 9:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk reject SIP INTITE from different source ports Deepika Nijhawan wrote: It just gives no matching peer error and doesn't pick their sip configuration, so do not go to any context in extentions.conf. VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from IP:4604' So the question is why didnt it match anything. If the phones are registering then they should reregister before choosing a different port. Are they going through a firewall by any chance? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to pickup an extension, tryi
Jonas Kellens wrote:
Rob,
it's not a macro but a sub. In my previous post I posted more info, I
am not going to post the whole output every time.
I read on the wiki that you set the PICKUPMARK equal to the extension
for that channel, but in my case I'm not using extensions but multiple
SIPaccounts in my dial statement.
Since you are ringing multiple extensions, have you tried looking at the
Callgroup and pickupgroup options in sip.conf ?
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
Its not directed pickup, but might be a better fit to what you are
trying to achieve.
Do you have another wiki ? Because even with the search-option I can
not find the word inbound, as you refer to [macro-inbound].
I'm refering to http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup
Scroll approx. one 3rd of the way down the page to where there is a
large bold bit of text saying:
Example using PICKUPMARK for Asterisk 1.4
[macro-inbound]
exten = s,1,Set(_PICKUPMARK=${MACRO_EXTEN})
exten = s,n,Dial(SIP/SomeSipPhone,20,rwt)
Something similar may work in your case perhaps ?
Rob
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[asterisk-users] Fwd: can't seem to register, status unmonitored
-- Forwarded message --
From: nikhil singhania niksingha...@gmail.com
Date: 16 June 2010 12:15
Subject: Re: [asterisk-users] can't seem to register, status unmonitored
To: Zeeshan Zakaria zisha...@gmail.com
Here is my extensions.conf:
[general]
static=yes ; default values for changes to this file
writeprotect=no ; by the Asterisk CLI
[globals]
; variables go here
[default]
; default context
[phones]
; context for our phones
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 500,1,Answer()
exten = 500,2,Playback(demo-echotest)
; Let them know what's going on
exten = 500,3,Echo
; Do the echo test
exten = 500,4,Playback(demo-echodone)
; Let them know it's over
exten = 500,5,Hangup
exten = _.,1,Dial(SIP/${ext...@wlg-gateway); match anything and
send to wlg-gateway
exten = _.,2,Hangup
[from-wlg-gateway]
; context for calls coming from wlg-gateway
exten = 4980007,1,Dial(SIP/2001SIP/2002)
exten = _.,1,Congestion()
; everyone else gets congestion
..
sip.conf
[general]
context=default ; Default context for incoming calls
port=5060; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes; Enable DNS SRV lookups on outbound calls
[2001]
type=friend ; both send and receive calls from this peer
host=dynamic ; this peer will register with us
username=2001
secret=j0nny
canreinvite=no ; don't send SIP re-invites (ie. terminate rtp stream)
nat=yes ; always assume peer is behind a NAT
context=phones ; send calls to 'phones' context
dtmfmode=rfc2833 ; set dtmf relay mode
allow=all; allow all codecs
[2002]
type=friend
host=dynamic
username=2002
secret=whyfry
canreinvite=no
nat=yes
context=phones
dtmfmode=rfc2833
allow=all
[wlg-gateway]
type=friend
disallow=all
allow=ulaw
context=from-wlg-gateway
host=202.7.4.40
canreinvite=no
dtmfmode=rfc2833
allow=all
.
inbound.php
..
#!/usr/bin/php
?php
ob_implicit_flush(true);
set_time_limit(0);
echo(Hello, world!);
require_once phpagi.php;
error_reporting(E_ALL);
echo(Hello, world!);
$dir_base = /var/www/wizoz/;
echo $dir_base;
$dir_prompt = $dir_base.prompts;
$dir_wav = $dir_base.wav;
$rel_dir_mp3 = mp3;
$dir_mp3 = $dir_base.$rel_dir_mp3;
$agi = new AGI();
echo(created);
$agi-answer();
$agi-exec_dial(SIP,2002);
$agi-stream_file($dir_prompt.'/welcome','123'); fflush($agi-out);
# welcome to yumphone.com
$agi-stream_file($dir_prompt.'/welcome','123'); fflush($agi-out);
echo(Hello, world!);
$result = $agi-get_variable(CALLERID(num));
echo $result;
$phonenum = $result['data'];
if (strlen($phonenum) != '10')
{
$phonenum = substr($phonenum,-10);
}
$uid = $phonenum.time();
$agi-stream_file($dir_prompt.'/record','123'); fflush($agi-out);
# please record your message after the beep. press 0 at the end of the
message
$agi-record_file($dir_wav./.$uid,'wav','0','6',NULL,true,5);
# fname, format, escape, timeout, offset, beep, silence
$agi-stream_file($dir_prompt.'/messagesent','123'); fflush($agi-out);
# your message has been sent
$agi-stream_file($dir_prompt.'/thankyou','123'); fflush($agi-out);
# thank you
?
..
Though I am new, but i am somewhat familiar, and am devoting a great deal of
time. Now you have all the files. I highlited the exec_dial function. This
inbound.php is the file i am executing on the command line on the server.
But I am not gettting the call at my end. May be the way i am doing it is
wrong. Please suggest me. Rest of the code works fine.
On 15 June 2010 18:15, Zeeshan Zakaria zisha...@gmail.com wrote:
The reason I said it'll take you one week, is because you seem new to
asterisk. It may take even more.
Pasting a part of the code is not enough for anybody to be able to help
you. You should paste the relevant parts of your sip.conf, extensions.conf
and the agi script. To me it seems you are new to dial plans, and if this is
true, first you need to focus on understanding dial plans, and then jump to
agi.
Did the other two issue get resolved?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-15 7:49 AM, nikhil singhania niksingha...@gmail.com wrote:
Hi Zeeshan,
Thanx for ur reply!!
The reason for this question was that i am actually doing the 3rd part,
which you said will take me 1 week to learn.
I have modified a
[asterisk-users] Asterisk reject SIP INTITE from different source ports
It's working now after giving nat=yes, thanks. Deepika -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + E1 card
Dear all, I have to install an E1 card in my Asterisk 1.4.23 server and here is my short question: Is it necessary to install or update any Asterisk/Zaptel/Any extra module or the default installation is good enough to just plug and run the E1 card Thanks a lot Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ring no answer / RONA versus HangUp
Hello List: I'm working on a funny scenario, where I'm bouncing calls from a Cisco call center into asterisk. Cisco call center has some logic that if a customer calls in, an agent is logged into a given extension... if Cisco sends a customer call to that extension, and there is a ring with no answer after a preset amount of time, Cisco concludes the agent is unavailable, kicks the agent off the queue, and pulls the customer call back to the front of the queue for the next available agent. This is all good. My problem is that I'm trying to figure out how best to tell asterisk 'let this ring, and don't pick it up,' that is, I want to exercise that Cisco behaviour that I've described. I know if I do not do an Answer() that the call is not yet picked up. However, if I do a HangUp(), is that functionally equivalent? Can you Hangup() a channel you never Answer() ed? I know these are pretty basic questions, but I've never thought about this problem like this before. I'm going to go ahead and play, but wanted to brainstorm this on the list. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + E1 card
Alejandro Cabrera Obed wrote: Is it necessary to install or update any Asterisk/Zaptel/Any extra module or the default installation is good enough to just plug and run the E1 card You'll need to make sure that libpri and dahdi are installed and configured. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + E1 card
Your installation should work, you must configure the card channels and load the card module on your OS. Regards. 2010/6/16 Alejandro Cabrera Obed aco1...@gmail.com Dear all, I have to install an E1 card in my Asterisk 1.4.23 server and here is my short question: Is it necessary to install or update any Asterisk/Zaptel/Any extra module or the default installation is good enough to just plug and run the E1 card Thanks a lot Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan. Linux User #441131 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: can't seem to register, status unmonitored
On Wed, 16 Jun 2010, nikhil singhania wrote: echo(Hello, world!); echo(Hello, world!); echo(created); echo(Hello, world!); echo $result; Each time you echo you are violating the AGI protocol. Remember, your script's STDOUT is feeding requests back to Asterisk. Try enabling AGI debugging and watch the console output for clues. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small PC to build and run Asterisk
On Wed, 16 Jun 2010, Randy R wrote: On Wed, Jun 16, 2010 at 3:46 AM, Michael Graves mgra...@mstvp.com wrote: Some distro's, like Askozia and Astlinux, have been specifically engineered around running from flash media. This basic form of operation has been well proven in projects like monowall and pfsense. I think you hit the essence of the argument for using these embedded systems, Michael. And we both know from experience and through knowing the people involved that they're both excellent choices! I am now considering using an about-to-be-retired Mac Mini. I'm pretty sure it can be done. How well it might work is another story. I'm pretty much giving up on Skype for Asterisk (and Skype for SIP) now that I realize that they'll be charging a monthly fee that is disproportionately high compared to my need to let Skype users call us. We'll know the pricing in Q4 of 2010, but it looks to be about $15/month for one user. $5 for the channel and $10 for Skype Manager. Maybe something for each name, too? I may have missed this part of the thread, but why giving up on SfA? I was just getting ready to start playing with that myself. Thanks, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Blind transfer feature
Hi, Am running 1.4.18 at the moment, and am trying to implement inline blind transfer. I have : [featuremap] blindxfer = *6 ; Blind transfer in features.conf And in extensions .conf under [globals] : DYNAMIC_FEATURES=automon#blindxfr So what am I missing ?? Have read through http://www.voip-info.org/wiki/view/Asterisk+config+features.conf Thanks, Adrian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDD/TTY Support
On voip-info I found a few dated references to TDD support being in the alpha stage and buggy. Can anyone direct me to any newer information on this option? Thanks -- Karl Harris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small PC to build and run Asterisk
On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere j...@sunfone.com wrote: pretty much giving up on Skype for Asterisk (and Skype for SIP) now that I realize that they'll be charging a monthly fee that is disproportionately high compared to my need to let Skype users call us. We'll know the pricing in Q4 of 2010, but it looks to be about $15/month for one user. $5 for the channel and $10 for Skype Manager. Maybe something for each name, too? I may have missed this part of the thread, but why giving up on SfA? I was just getting ready to start playing with that myself. Monthly fees as I mentioned above. In addition to the binary, youneed to pay for Skype Manager and each seat on that (name) - at least that is my understand of their page. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 Trunk Problem calling from Asterisk to Avaya PBX
On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi shinacaly...@gmail.comwrote: Hi! I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide a conference bridge for an existing Avaya PBX. I have no control over the Avaya system, but I am able to speak with the admin in charge when I need stuff done. I am running all this in a VirtualBox virtual instance, with CentOS 5.4 as the asterisk's host operating system. I configured a h323 trunk asterisk based on a few guides I discovered online, and I created a single sip extension (to test), and I am able to make a call from the Avaya PBX extensions successfully to my asterisk-freepbx virtual machine. The problem is when I try to make calls from Asterisk to Avaya, I get no sound whatsover and the call just keeps trying indefinitely until I end it. (I've used Twinkle and Ekiga softphones). This is what I find in the logs: [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Macro [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:12] ExecIf(SIP/16000-0002, 0|AGI|fixlocalprefix) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: ExecIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:13] Set(SIP/16000-0002, OUTNUM=18151) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:14] Set(SIP/16000-0002, custom=AMP) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:15] ExecIf(SIP/16000-0002, 0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: ExecIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:16] Macro(SIP/16000-0002, dialout-trunk-predial-hook|) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk-predial-hook:1] MacroExit(SIP/16000-0002, ) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Macro [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:17] GotoIf(SIP/16000-0002, 0?bypass|1) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: GotoIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:18] GotoIf(SIP/16000-0002, 1?customtrunk) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Goto (macro-dialout-trunk,s,21) [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: GotoIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:21] Set(SIP/16000-0002, pre_num=AMP:h323/Avaya/) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:22] Set(SIP/16000-0002, the_num=OUTNUM) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:23] Set(SIP/16000-0002, post_num=@ 10.100.7.15:1720) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:24] GotoIf(SIP/16000-0002, 1?outnum:skipoutnum) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Goto (macro-dialout-trunk,s,25) [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: GotoIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:25] Set(SIP/16000-0002, the_num=18151) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:26] Dial(SIP/16000-0002, h323/Avaya/18...@10.100.7.15:1720|300|) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Requested transfer capability: 0x00 - SPEECH my h323.conf file is below: [general] port = 1720 bindaddr = 10.101.4.224 amaflags = AVAYA progress_setup = 8 progress_alert = 8 faststart = yes h245tunneling = yes gatekeeper = DISABLE disallow=all allow=g729 allow=g723 dtmfmode=rfc2833 context=from-internal h323id=ObjSysAsterisk callerid=testbridge logfile=/var/log/asterisk/h323_log [Avaya] type=friend context=from-internal host=10.100.7.15 port=1720 disallow=all allow=g729 allow=g723 canreinvite=no dtmfmode=rfc2833 Please help me find out why the call isn't going through. -- best regards, Sina Owolabi 2348034022578 23417203257 23417420690 -- best regards, Sina Owolabi 2348034022578 23417203257 23417420690 --
Re: [asterisk-users] TDD/TTY Support
I'm supposing that it is 1. no better or worse than SMS support 2. dependent on the version you are on _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Harris Sent: Wednesday, June 16, 2010 10:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] TDD/TTY Support On voip-info I found a few dated references to TDD support being in the alpha stage and buggy. Can anyone direct me to any newer information on this option? Thanks -- Karl Harris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring no answer / RONA versus HangUp
On Wednesday 16 June 2010 08:21:17 David Backeberg wrote: I know if I do not do an Answer() that the call is not yet picked up. However, if I do a HangUp(), is that functionally equivalent? Can you Hangup() a channel you never Answer() ed? A Hangup just returns -1, which causes the dialplan to terminate. So yes, you can Hangup() a call you never answered. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDD/TTY Support
On 06/16/2010 11:31 PM, Karl Harris wrote: On voip-info I found a few dated references to TDD support being in the alpha stage and buggy. Can anyone direct me to any newer information on this option? There are installations where the TDD support in spandsp has been integrated with Asterisk, but I don't know if anyone has publicly released the code they use to integrate them. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring no answer / RONA versus HangUp
On Wed, Jun 16, 2010 at 11:50 AM, Tilghman Lesher tles...@digium.com wrote: On Wednesday 16 June 2010 08:21:17 David Backeberg wrote: I know if I do not do an Answer() that the call is not yet picked up. However, if I do a HangUp(), is that functionally equivalent? Can you Hangup() a channel you never Answer() ed? A Hangup just returns -1, which causes the dialplan to terminate. So yes, you can Hangup() a call you never answered. What I was really trying to determine was whether the calling side would get the same behavior (rings, but no pickup) as if the dialplan was simulating a phone where nobody was picking it up. I ended up doing a: exten = s,1,Wait(10) exten = s,n,HangUp essentially, which was good enough for me to simulate 5 seconds of no-pickup, as perceived by the caller. Thanks much. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] read data from file system and put in a variable
I am looking at http://www.voip-info.org/wiki/view/Asterisk+cmd+System I dont see how to execute a system command and set an asterisk variable to the string that is in my file /tmp/my_data. How is that done? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Qwest PRIs
Ok got it up and running. In the case for Qwest with NFAS they reserve what they call Interface ID 1 for the circuit with the backup d channel. In our case we only have two circuits with a single d channel. The real key was realizing the logical span number in the spanmap translated into interface ID so here are the spanmaps that worked for us: [trunkgroups] trunkgroup = 1,24 spanmap = 1,1,0 spanmap = 2,1,2 group=1 switchtype=dms100 echocancel=yes signalling=pri_cpe channel =1-23,25-48 Notice the switchtype. While they told me that their switchtype was NI2 (National ISDN 2) they did say they were using a dms100, so I would always ask your carrier what switch they have. Anyway thanks for your help all. On Mon, Jun 14, 2010 at 4:47 PM, Steve Edwards asterisk@sedwards.comwrote: On Mon, 14 Jun 2010, C F wrote: One more thing, read the comments here: http://www.voip-info.org/wiki/index.php?page_id=573tk=2ff846f8169b7694aed5comments_page=1 Don't forget to have a beer ready :P Now that's really funny. I read along with this and was thinking this was exactly my experience with some Qwest PRIs a couple of years ago. Then I noticed -- it was me :) I guess I had too many beers. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] read data from file system and put in a variable
On Wed, 16 Jun 2010, Jerry Geis wrote: I am looking at http://www.voip-info.org/wiki/view/Asterisk+cmd+System I dont see how to execute a system command and set an asterisk variable to the string that is in my file /tmp/my_data. Read up on the application readfile(). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small PC to build and run Asterisk
On Wed, 16 Jun 2010, Randy R wrote: On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere j...@sunfone.com wrote: pretty much giving up on Skype for Asterisk (and Skype for SIP) now that I realize that they'll be charging a monthly fee that is disproportionately high compared to my need to let Skype users call us. We'll know the pricing in Q4 of 2010, but it looks to be about $15/month for one user. $5 for the channel and $10 for Skype Manager. Maybe something for each name, too? I may have missed this part of the thread, but why giving up on SfA? I was just getting ready to start playing with that myself. Monthly fees as I mentioned above. In addition to the binary, youneed to pay for Skype Manager and each seat on that (name) - at least that is my understand of their page. Ack! I thought SfA was a one time charge, like their G.729 license. j-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDD/TTY Support
On 06/16/2010 11:44 PM, Danny Nicholas wrote: I’m supposing that it is 1. no better or worse than SMS support What relevance does SMS support have to TDD/TTY support? 1. dependent on the version you are on I don't think the TDD support has been touched for years, so I doubt the version makes much difference. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Karl Harris *Sent:* Wednesday, June 16, 2010 10:31 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] TDD/TTY Support On voip-info I found a few dated references to TDD support being in the alpha stage and buggy. Can anyone direct me to any newer information on this option? Thanks -- Karl Harris Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] read data from file system and put in a variable
Read up on the application readfile(). Steve, on my 1.4 system help readfile says no such command. searching a little more shows readfile as an AGI command. Is this what your refering to? http://www.voip-info.org/wiki/view/Long+Distance+or+Local+Python+AGI jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] read data from file system and put in a variable
Do a 'show application ReadFile' Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-16 12:49 PM, Jerry Geis ge...@pagestation.com wrote: Read up on the application readfile(). Steve, on my 1.4 system help readfile says no such command. searching a little more shows readfile as an AGI command. Is this what your refering to? http://www.voip-info.org/wiki/view/Long+Distance+or+Local+Python+AGI jerry -- _ -- Bandwidth and C... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] read data from file system and put in avariable
core show application readfile is the new command to not get the deprecated message. To answer the OP's query, here's the dialplan line Exten = 1234,1,readfile(foo,/tmp/my_data) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Wednesday, June 16, 2010 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] read data from file system and put in avariable Do a 'show application ReadFile' Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-16 12:49 PM, Jerry Geis ge...@pagestation.com wrote: Read up on the application readfile(). Steve, on my 1.4 system help readfile says no such command. searching a little more shows readfile as an AGI command. Is this what your refering to? http://www.voip-info.org/wiki/view/Long+Distance+or+Local+Python+AGI jerry -- _ -- Bandwidth and C... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small PC to build and run Asterisk
On Tue, Jun 15, 2010 at 08:46:17PM -0500, Michael Graves wrote: On Tue, 15 Jun 2010 07:58:34 -0400, SIP wrote: Danny Nicholas wrote: Also cheaper to replace flash card than hard drive. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Monday, June 14, 2010 4:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Small PC to build and run Asterisk Why no flash? * Small pre-built PC (not buying board, case, all parts separately) * Low power consumption * No fan or very small fan * Hard drive (not flash memory) An ssd uses less power, so generates less warmth, hence less need for fan in the drive area. Also less noise.. I like this one, or its smaller brother: http://www.fit-pc.com/web/fit-pc2/fit-pc2i-specifications/ But a flash card needs replacing more often than a hard drive. It's just not designed for the same sort of lifecycle of writes that a hard drive is. Sure, the number is always increasing as they increase the capacity, but it WILL NOT LAST. Dependent on the type of filesystem access you need, SSD could be a great choice. But if you're heavy on logging and writing small data bits here and there (which isn't always something you can control if you don't write all the software), then a hard drive is just going to be the better choice to hold up for a long period of time. This need not be the case. It depends upon what Asterisk distro you're using. I ran Astlinux from a vintage 256 MB CF card for several years without a problem. If you simply build up a server and use flash media in place of a disk then you will likely kill the media in a short period. The behaviour of the system needs to be tailored to running from Flash. Some distro's, like Askozia and Astlinux, have been specifically engineered around running from flash media. This basic form of operation has been well proven in projects like monowall and pfsense. For very large installations with a lot of I/O intensive extra activities running on the server running from flash may never be appropriate. I'm not sure how much this is an issue. We've had our share of embedded (read: small, headless) systems where I work. We originally used our own specifically-engineered distro. It was basically Debian, where the root file system was read-only, some changed parts were on a ram-disk, with an option to actively sync them. We basically used http://packages.debian.org/flashybrid for that. Later on we decided to check what would happen if you take a standard distro (Elastix, eventually. Ypu, FreePBX-based, with MySQL and the lot) and put it on our embedded systems. Well, it happens to work. Now, you have to be careful about the flash you use. We did test it. Our own office's PBX also ran on a box with such a flash for quite some time (and we employ active very active logging, and occasional call recording). It's still not as cheap (per MB) as magnetic disks. But it may be cheap enough for you. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfer feature
Am running 1.4.18 at the moment, and am trying to implement inline blind transfer. I have : [featuremap] blindxfer = *6 ; Blind transfer Do remember that asterisk needs to be in the media stream for this to work, so you'll want to make sure (in the case of SIP devices) you've set canreinvite=no You might also want to increase the feature code timeout (both activation and interdigit) - I think the default is something like 500ms, which most users find far too short to use reliably. Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to pickup an extension
Hello.
This is what I have :
suppose ${SIPaccounts}=SIP/testcorp5SIP/testcorp6
exten = group,1,Set(_PICKUPMARK=${SIPaccounts})
exten = group,n,Dial(${SIPaccounts})
This is what happens when I try to pickup an extension :
[Jun 16 20:39:33] -- Executing [...@sub-routing:13]
Set(SIP/testcorp4-0005,
*_PICKUPMARK=SIP/testcorp5SIP/testcorp6*) in new stack
[Jun 16 20:39:38] -- Executing [...@sub-routing:14]
Dial(SIP/testcorp4-0005, SIP/testcorp5SIP/testcorp6|30) in new
stack
[Jun 16 20:39:38] -- Called testcorp5
[Jun 16 20:39:38] -- Called testcorp6
[Jun 16 20:39:38] -- SIP/testcorp6-0008 is ringing
[Jun 16 20:39:38] -- SIP/testcorp5-0007 is ringing
...
[Jun 16 20:39:40] -- Executing [*...@from-testcorp:8]
Pickup(SIP/testcorp1-0009, *testco...@pickupmark*) in new stack
[Jun 16 20:39:40] NOTICE[2936]: app_directed_pickup.c:159 pickup_exec:
*No target channel* found for testcorp6.
[Jun 16 20:39:40] -- Executing [*...@from-testcorp:9]
NoOp(SIP/testcorp1-0009, ) in new stack
[Jun 16 20:39:40] -- Executing [*...@from-testcorp:10]
Pickup(SIP/testcorp1-0009, *SIP/testco...@pickupmark*) in new stack
[Jun 16 20:39:40] NOTICE[2936]: app_directed_pickup.c:159 pickup_exec:
*No target channel* found for SIP/testcorp6.
So again my question: how to use Pickup() when I have multiple
SIPaccounts in the Dial()-statement ??
The question remains.
Jonas
On 06/16/2010 12:25 PM, Rob Coward wrote:
Scroll approx. one 3rd of the way down the page to where there is a
large bold bit of text saying:
Example using PICKUPMARK for Asterisk 1.4
[macro-inbound]
exten = s,1,Set(_PICKUPMARK=${MACRO_EXTEN})
exten = s,n,Dial(SIP/SomeSipPhone,20,rwt)
Something similar may work in your case perhaps ?
Rob
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Re: [asterisk-users] a2billing for residential voip usage
I'm unable to place any calls through a2billing. I followed instructions here: http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q to DISABLE PIN number request Prompt for some users but, I'm not able to place any calls. I created a trunk with the same name as in my sip.conf and I'm not able to make any calls. I don't know what I'm missing. This is the output when trying to call: == Using SIP RTP CoS mark 5 -- Executing [812022418...@a2billing:1] Answer(SIP/1433631307-0015, ) in new stack -- Executing [812022418...@a2billing:2] Wait(SIP/1433631307-0015, 2) in new stack -- Executing [812022418...@a2billing:3] AGI(SIP/1433631307-0015, a2billing.php,3) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- SIP/1433631307-0015AGI Script a2billing.php completed, returning -1 I can't debug it or anything I'm stuck please help. --- On Tue, 6/15/10, Faisal Hanif fai...@vopium.com wrote: From: Faisal Hanif fai...@vopium.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:26 PM You need to copy or soft link a2billing.conf to /etc/ folder as by default latest version search for it in /etc/ Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Tuesday, June 15, 2010 9:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] a2billing for residential voip usage I copied the config to the a2billing.conf in /etc/asterisk folder. I'm still not able to place any calls yet. Looks like I have to read more on how to configure trunks and providers whick got me confused. I'll learn though. --- On Tue, 6/15/10, Vardan Harutyunyan hvarda...@gmail.com wrote: From: Vardan Harutyunyan hvarda...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 8:03 AM look manual, but in any case the a2billing.conf is in /etc/asterisk/ on can say, where you have place your asterisk configuration files -- Vardan Harutyunyan, Senior System Administrator Enterprise Incubator Foundation 123 Hovsep Emin Street, Yerevan 0051, Republic of Armenia Tel: + 374 10 219735 Fax: + 374 10 219777 E-mail: i...@eif.am www.eif-it.com Jimmy Godbout wrote: Hi, Maybe you can just use a reporting tool that will look at the CDR and tell you who's using the phone the most. Some of them will use a DB to store the CDR. If you want, you can even use Excel to look at the csv file created by default and make your own report. http://www.voip-info.org/wiki/view/Asterisk+billing http://www.voip-info.org/wiki/view/Asterisk+GUI (in Billing Call Detail Reporting) http://www.voip-info.org/wiki/view/Asterisk+CDR+Areski+GUI Jimmy -Original Message- From: landysacco...@yahoo.com Sent: Tue, 15 Jun 2010 00:11:51 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] a2billing for residential voip usage Ram. Thanks for replying. I have searched / googled about it but can't find a solution to monitor the 4 extensions I have at home. A2billing asks for the number I want to dial but, I don't need that. I would like the extensions to dial out normally and a2billing just record the time and talked time for later review. Thanks. --- On Tue, 6/15/10, ramtalk2...@gmail.com wrote: From: ramtalk2...@gmail.com Subject: Re: [asterisk-users] a2billing for residential voip usage To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, June 15, 2010, 1:05 AM you see lot of documentation on wiki Google them many success case you see Ram On Tue, Jun 15, 2010 at 7:01 AM, Landy Landylandysacco...@yahoo.com wrote: Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for •VoIP residential services? if yes, how? if no, please guide me to another application I can use along side asterisk. Thanks in advanced for your time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Unable to pickup an extension
Hi!
suppose ${SIPaccounts}=SIP/testcorp5SIP/testcorp6
exten = group,1,Set(_PICKUPMARK=${SIPaccounts})
If I was doing this I'd rather do
Set(_PICKUPMARK=group)
or
Set(_PICKUPMARK=${EXTEN})
but that is probably just me. But let's look at two of your lines:
Set(SIP/testcorp4- 0005,_PICKUPMARK=SIP/testcorp5SIP/testcorp6)
Pickup(SIP/testcorp1-0009, testco...@pickupmark)
Can you see the difference? That's what you need to change so that it
matches. SIP/testcorp5SIP/testcorp6 is not the same as testcorp6.
Philipp
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[asterisk-users] Call hangs up after exactly 1 minute
Hello list, using Asterisk 1.4.30. [Jun 16 21:35:12] -- Executing [...@sub-routing:12] Dial(SIP/user110-005a, SIP/user2|999) in new stack [Jun 16 21:35:12] -- Called user2 [Jun 16 21:35:12] -- SIP/user2-005c is ringing [Jun 16 21:36:12] WARNING[1991]: chan_sip.c:13073 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '0ae668e73053d17f33c852253f965...@192.168.1.150'. Giving up. [Jun 16 21:36:12] -- SIP/user2-005c is circuit-busy [Jun 16 21:36:12] == Everyone is busy/congested at this time (1:0/1/0) After exactly 60 seconds, the call is terminated, although I have given a timeout-value of 999... How come ?? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
On Wed, 16 Jun 2010, Landy Landy wrote: I'm unable to place any calls through a2billing. I followed instructions here: http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q to DISABLE PIN number request Prompt for some users but, I'm not able to place any calls. I created a trunk with the same name as in my sip.conf and I'm not able to make any calls. I don't know what I'm missing. This is the output when trying to call: == Using SIP RTP CoS mark 5 -- Executing [812022418...@a2billing:1] Answer(SIP/1433631307-0015, ) in new stack -- Executing [812022418...@a2billing:2] Wait(SIP/1433631307-0015, 2) in new stack -- Executing [812022418...@a2billing:3] AGI(SIP/1433631307-0015, a2billing.php,3) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php -- SIP/1433631307-0015AGI Script a2billing.php completed, returning -1 I can't debug it or anything I'm stuck please help. Try enabling AGI debugging. If that does not yield a clue, maybe an a2billing mailing list would be a more appropriate forum. [snipping remaining 250+ lines that are probably irrelevant at this point in time.] -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call ended after 31 seconds
Yes, I'm using XLite... Anahi Ludueña From: l...@virtutel.ca To: asterisk-users@lists.digium.com Date: Fri, 11 Jun 2010 20:05:39 -0400 Subject: Re: [asterisk-users] Call ended after 31 seconds You`re using Xlite/eyeBeam by any chance? Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anahi Ludueña Sent: Friday, June 11, 2010 16:12 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call ended after 31 seconds Hi people, I have a problem with some extensions. The calls are ended after 31/35 seconds, also, it depends on the number which I call. This is the log, but I've not been able to find something wrong... Any ideas? [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: ExecIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:16] Macro(SIP/3000-6d07, dialout-trunk-predial-hook|) in new stack [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk-predial-hook:1] MacroExit(SIP/3000-6d07, ) in new stack [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: Macro [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:17] GotoIf(SIP/3000-6d07, 0?bypass|1) in new stack [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:18] GotoIf(SIP/3000-6d07, 0?customtrunk) in new stack [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:19] Dial(SIP/3000-6d07, SIP/GAF/|300|) in new stack [Jun 11 15:50:46] NOTICE[26071] app_dial.c: Hey! chan SIP/3000-6d07's context='macro-dialout-trunk', and exten='s' [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Called SIP/GAF/ [Jun 11 15:50:49] VERBOSE[26071] logger.c: -- SIP/GAF-6 is ringing [Jun 11 15:50:49] VERBOSE[26071] logger.c: -- SIP/GAF-6 is making progress passing it to SIP/3000-6d07 [Jun 11 15:50:56] VERBOSE[26071] logger.c: -- SIP/GAF-6 answered SIP/3000-6d07 [Jun 11 15:50:56] VERBOSE[26071] logger.c: -- Packet2Packet bridging SIP/3000-6d07 and SIP/GAF-6 [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-dialout-trunk:1] Macro(SIP/3000-6d07, hangupcall|) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:1] GotoIf(SIP/3000-6d07, 1?skiprg) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Goto (macro-hangupcall,s,4) [Jun 11 15:51:27] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:4] GotoIf(SIP/3000-6d07, 1?skipblkvm) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Goto (macro-hangupcall,s,7) [Jun 11 15:51:27] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:7] GotoIf(SIP/3000-6d07, 1?theend) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Goto (macro-hangupcall,s,9) [Jun 11 15:51:27] DEBUG[26071] app_macro.c: Executed application: GotoIf [Jun 11 15:51:27] VERBOSE[26071] logger.c: -- Executing [...@macro-hangupcall:9] Hangup(SIP/3000-6d07, ) in new stack [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3000-6d07' in macro 'hangupcall' [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn h extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/3000-6d07' [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/3000-6d07' in macro 'dialout-trunk' [Jun 11 15:51:27] VERBOSE[26071] logger.c: == Spawn extension (from-internal, xxx, 5) exited non-zero on 'SIP/3000-6d07' Thanks, Anahi Ludueña Noticias, servicios, tendencias. Haz de MSN.ES tu pág. de inicio _ Sé el protagonista de GQ con Messenger y Vodafone Blackberry. ¡Y gana premios! http://serviciosmoviles.es.msn.com/messenger/vodafone.aspx-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to pickup an extension
Yes, so I've noticed that I can name _PICKUPMARK anything I want... OK so the name does not mather and has nothing to do with the different SIPaccount that it holds... Another problem is that when another call come in, the _PICKUPMARK variable is overwritten and I can no longer pick up the first incoming call. How to overcome this ?? Jonas. On 06/16/2010 09:26 PM, Philipp von Klitzing wrote: Set(SIP/testcorp4- 0005,_PICKUPMARK=SIP/testcorp5SIP/testcorp6) Pickup(SIP/testcorp1-0009, testco...@pickupmark) Can you see the difference? That's what you need to change so that it matches. SIP/testcorp5SIP/testcorp6 is not the same as testcorp6. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI PRI error message
Hello- After configuring DAHDI and starting asterisk, I get the following message continuously on the Asterisk console: WARNING[2057]: chan_dahdi.c:4158 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! My card is a D410P configured for E1, only the first span is configured, and configuration snippets are as follows: From /etc/dahdi/system.conf: (auto configured, first span only shown:) # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 -- From /etc/asterisk/dahdi-channels.conf (included in chan_dahdi.conf): ; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 context = default group = 63 -- QUESTION: Shouldn't asterisk pick up from dahdi.conf that the signalling channel is 16? Why the error message? Thanks Scott -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + Dahdi does not work with BRI NT mode
Hi all, As i tested with Asterisk+dahdi+libpri with openvox BRI with NT mode, the ISDN phone does not work. There is the setting and error. ***Enviroment** Asterisk-1.6.1.18 dahdi-linux-2.3.0.1 dahdi-tool-2.30. libpri-1.4.11.2 CentOS-5.5 OpenVox B400P In my case , I set prot 1 and port 2 as NT mode ,port 3 and port 4 as TE mode. system.conf* # Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 16 22:49:10 2010 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) span=1,1,0,ccs,ami # termtype: te bchan=1-2 hardhdlc=3 echocanceller=mg2,1-2 # Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 RED span=2,2,0,ccs,ami # termtype: te bchan=4-5 hardhdlc=6 echocanceller=mg2,4-5 # Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 RED span=3,3,0,ccs,ami # termtype: te bchan=7-8 hardhdlc=9 echocanceller=mg2,7-8 # Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 span=4,4,0,ccs,ami # termtype: te bchan=10-11 hardhdlc=12 echocanceller=mg2,10-11 # Global data loadzone= us defaultzone = us ***dahdi-channles.conf** ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Jun 16 22:49:10 2010 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) group=0,11 context=from-internal switchtype = euroisdn signalling = bri_net channel = 1-2 context = default group = 63 ; Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 RED group=0,12 context=from-internal switchtype = euroisdn signalling = bri_net channel = 4-5 context = default group = 63 ; Span 3: B4/0/3 B4XXP (PCI) Card 0 Span 3 RED group=0,13 context=from-pstn switchtype = euroisdn signalling = bri_cpe channel = 7-8 context = default group = 63 ; Span 4: B4/0/4 B4XXP (PCI) Card 0 Span 4 group=0,14 context=from-pstn switchtype = euroisdn signalling = bri_cpe channel = 10-11 context = default group = 63 **dmesg* wcb4xxp :01:03.0: Port 1: NT mode wcb4xxp :01:03.0: Port 2: NT mode wcb4xxp :01:03.0: Port 3: TE mode wcb4xxp :01:03.0: Port 4: TE mode wcb4xxp :01:03.0: Did not do the highestorder stuff wcb4xxp :01:03.0: Configuring span 1 wcb4xxp :01:03.0: new card sync source: port 3 wcb4xxp :01:03.0: Configuring span 1 wcb4xxp :01:03.0: Configuring span 2 wcb4xxp :01:03.0: Configuring span 3 wcb4xxp :01:03.0: Configuring span 4 dahdi_echocan_mg2: Registered echo canceler 'MG2' dahdi: Registered tone zone 0 (United States / North America) wcb4xxp :01:03.0: new card sync source: port 1 *CLI dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefault default In Service 1from-internal default In Service 2from-internal default In Service 4from-internal default In Service 5from-internal default In Service 7from-pstn default In Service 8from-pstn default In Service 10from-pstn default In Service 11from-pstn default In Service *** *CLI pri show spans *PRI span 1/0: Provisioned, Down, Active* PRI span 2/0: Provisioned, In Alarm, Down, Active PRI span 3/0: Provisioned, In Alarm, Down, Active PRI span 4/0: Provisioned, Up, Active *** TE mode can works well. but the NT mode can not work. I do not know why the status always show Down . Any idea for that? I am looking forward to your help. Thank you! liu xin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Dahdi does not work with BRI NT mode
On Thu, Jun 17, 2010 at 09:38:43AM +0800, liuxin wrote: Hi all, As i tested with Asterisk+dahdi+libpri with openvox BRI with NT mode, the ISDN phone does not work. There is the setting and error. ***Enviroment** Asterisk-1.6.1.18 Note that this specific version of Asterisk (well - any version of Asterisk that has currently been released) only supports NT PtP. For ISDN phones you need NT PtMP. That said, I believe that there's a mess right now with detecting NT/TE mode in the wcb4xxp driver. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing for residential voip usage
On 6/17/10 12:49 AM, Steve Edwards wrote: On Wed, 16 Jun 2010, Landy Landy wrote: I'm unable to place any calls through a2billing. I followed instructions here: http://trac.asterisk2billing.org/cgi-bin/trac.cgi/wiki/F.A.Q to DISABLE PIN number request Prompt for some users but, I'm not able to place any calls. I created a trunk with the same name as in my sip.conf and I'm not able to make any calls. I don't know what I'm missing. This is the output when trying to call: == Using SIP RTP CoS mark 5 -- Executing [812022418...@a2billing:1] Answer(SIP/1433631307-0015, ) in new stack -- Executing [812022418...@a2billing:2] Wait(SIP/1433631307-0015, 2) in new stack -- Executing [812022418...@a2billing:3] AGI(SIP/1433631307-0015, a2billing.php,3) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php --SIP/1433631307-0015AGI Script a2billing.php completed, returning -1 I can't debug it or anything I'm stuck please help. If you have CLI version of PHP installed, you can also try running /var/lib/asterisk/agi-bin/a2billing.php directly from the shell, and keep feeding it CR/LF, you'll see step-by-step variable assignment and hopefully the error message that stops it from working. You'll need display_errors on in php.ini for this as well. Most probably you're missing a PHP module or your SQL connection is failing. HTH, Vahan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
