[asterisk-users] Extensions routing
Greetings! I've been playing around with clustering some Asterisk servers for sake of fail-over and load balancing with DNS round-robin, and came to one problem. If I have, say, 2 servers, and clients register either on 1 or 2, how can I route extensions between them? I mean, if today user with extension 101 is registered on server1, and tomorrow he will register with server2 - how would any of servers know where to route it? As some examples, if I have only 2 servers, things are not so bad. I can use Dial(SIP/101SIP/server2/101) on server1 and vice versa. OR, I can check the hungup code, and if it's 34 (or whatever I get when I try to dial unavailable peer) - try it on another server. But I guess things get tricky when you have 3 or more servers, and besides maybe this solution is not the best one. Could you share some knowledge on this, please? -- With Best Regards Mikhail Lischuk -- _ -- 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] Extensions routing
The way I accomplish this is by having an active/passive cluster. The two or more servers have individual IP addresses and running heartbeat creates a clustered IP address. The active server uses the cluster IP address. If the active server should fail then the cluster IP address moves to another server. Each handset and peer uses the clustered IP address to communicate to the server. This way all devices only communicate to a single server and you don’t have the problem of having different devices connected to different servers. I have created a Wiki page based on this which may help you. http://www.klaverstyn.com.au/david/wiki/index.php?title=Cluster The wiki mentions a script file to copy files between servers to keep the data consistent. To do this more efficiently DRDB should be used but the scripts works well in my situation. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mikhail Lischuk Sent: Saturday, 19 May 2012 5:48 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Extensions routing Greetings! I've been playing around with clustering some Asterisk servers for sake of fail-over and load balancing with DNS round-robin, and came to one problem. If I have, say, 2 servers, and clients register either on 1 or 2, how can I route extensions between them? I mean, if today user with extension 101 is registered on server1, and tomorrow he will register with server2 - how would any of servers know where to route it? As some examples, if I have only 2 servers, things are not so bad. I can use Dial(SIP/101SIP/server2/101) on server1 and vice versa. OR, I can check the hungup code, and if it's 34 (or whatever I get when I try to dial unavailable peer) - try it on another server. But I guess things get tricky when you have 3 or more servers, and besides maybe this solution is not the best one. Could you share some knowledge on this, please? -- With Best Regards Mikhail Lischukmailto:mlisc...@itx.com.ua -- _ -- 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] Slow AMI Originate
Hello, We use AMI to originate calls. Sometimes, lately every morning, the AMI Originate process operates extremely slowly. I cannot see the calls in core show channels verbose, I don't know where they are, what state they are in, after 2-3 minutes the calls go through one after the other. As mentioned, it usually happens in the morning as soon as people start their workday, where there are a lot of logins and calls being made, but no where close to a peak in terms of simultaneous channels, etc. In some cases restarting asterisk, in others just taking the storm and waiting it out solves the problem. Having a hard time coming up with something to troubleshoot this. Any ideas would be appreciated. -- Mehmet Avcioglu meh...@activecom.net -- _ -- 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: RTP stats explaination
Hi Dave, On Fri, May 18, 2012 at 11:27 PM, Dave Platt dpl...@radagast.org wrote: In our app we do not forward packet immediately. After enough packet received to increase rtp packetization time (ptime) the we forward the message over raw socket and set dscp to be 10 so that this time packets can escape iptable rules. From client side the RTP stream analysis shows nearly every stream as problematic. summery for some streams are given below : Stream 1: Max delta = 1758.72 ms at packet no. 40506 Max jitter = 231.07 ms. Mean jitter = 9.27 ms. Max skew = -2066.18 ms. Total RTP packets = 468 ? (expected 468) ? Lost RTP packets = 0 (0.00%) ? Sequence errors = 0 Duration 23.45 s (-22628 ms clock drift, corresponding to 281 Hz (-96.49%) Stream 2: Max delta = 1750.96 ms at packet no. 45453 Max jitter = 230.90 ms. Mean jitter = 7.50 ms. Max skew = -2076.96 ms. Total RTP packets = 468 ? (expected 468) ? Lost RTP packets = 0 (0.00%) ? Sequence errors = 0 Duration 23.46 s (-22715 ms clock drift, corresponding to 253 Hz (-96.84%) Stream 3: Max delta = 71.47 ms at packet no. 25009 Max jitter = 6.05 ms. Mean jitter = 2.33 ms. Max skew = -29.09 ms. Total RTP packets = 258 ? (expected 258) ? Lost RTP packets = 0 (0.00%) ? Sequence errors = 0 Duration 10.28 s (-10181 ms clock drift, corresponding to 76 Hz (-99.05%) Any idea where should we look for the problem? A maximum jitter of 230 milliseconds looks pretty horrendous to me. This is going to cause really serious audio stuttering on the receiving side, and/or will force the use of such a long jitter buffer by the receiver that the audio will suffer from an infuriating amount of delay. Even a local call would sound as if it's coming from overseas via a satellite-radio link. I suspect it's likely due to a combination of two things: (1) The fact that you are deliberately delaying the forwarding of the packets. This adds latency, and if you're forwarding packets in batches it will also add jitter. There is no other ways other than doing this. Because we need enough packets to be queued before doing a repacketization feature. Asterisk also does this by allow:g729:120 in sip.conf. But we have seen that asterisk fails to do that in different circumstances. Because of this we are trying to do it before it goes into asterisk. We can also try learning asterisk development and then try to modify asterisk to meet our needs. But writing a new application seems logical to me, because then we will be bound in one platform. It will be better if this repacketization is telephony platform agnostic. We also have some freeswitch boxes, and some propitiatory platform for which we do not have codes. There is a severe limitation in freeswitch regarding this feature because of its dependence on L16 format when communication with transcoder card. Because of this they only support up to 50ms of ptime for g729. and the other platform vendor does not intend to support it either. But this feature is very critical to our operation. If we want to move this application in kernel space by writing kernel module would it help? What are the constraints we need to be aware if we start writing a kernel module to provide this functionality? (2) Scheduling delays. If your forwarding app fails to run its code on a very regular schedule - if, for example, it's delayed or preempted by a higher-priority task, or if some of its code is paged/swapped out due to memory pressure and has to be paged back in - this will also add latency and jitter. Pushing real-time IP traffic up through the application layer like this is going to be tricky. You may be able to deal with issue (2) by locking your app into memory with mlock() and setting it to run at a real-time scheduling priority. We will test it and post further results. Issue (1) - well, I really think you need to avoid doing this. Push the packets down into the kernel for retransmission as quickly as you can. If you need to rate-limit or rate-pace their sending, use something like the Linux kernel's traffic-shaping features. Is there other network traffic flowing to/from this particular machine? It's possible that other outbound traffic is saturating network-transmit buffers somewhere - either in the kernel, or in an upstream communication node such as a router or DSL modem. If this happens, there's no guarantee that high priority or expedited delivery packets would be given priority over (e.g.) FTP uploads... many routers/switches/modems don't pay attention to the class-of-service on IP packets. To prevent this, you'd need to use traffic shaping features on your system, to pace the transmission of *all* packets so that the total transmission rate is slightly below the lowest-bandwidth segment of your uplink. You'd also want to use multiple queues to give expedited-deliver packets priority over bulk-data packets. The Ultimate Linux traffic-shaper page
Re: [asterisk-users] Extensions routing
On Saturday 19 May 2012, Mikhail Lischuk wrote: I've been playing around with clustering some Asterisk servers for sake of fail-over and load balancing with DNS round-robin, and came to one problem. If I have, say, 2 servers, and clients register either on 1 or 2, how can I route extensions between them? I mean, if today user with extension 101 is registered on server1, and tomorrow he will register with server2 - how would any of servers know where to route it? Won't Dundi serve your purpose? From http://www.dundi.com/ : DUNDi™ is a peer-to-peer system for locating Internet gateways to telephony services. Unlike traditional centralized services (such as the remarkably simple and concise ENUM standard), DUNDi is fully-distributed with no centralized authority whatsoever. DUNDi is not itself a Voice-over IP signaling or media protocol. Instead, it publishes routes which are in turn accessed via industry standard protocols such as IAX™, SIP and H.323. DUNDi can be used within an enterprise to create a fully-federated PBX with no central point of failure, and the ability to arbitrarily add new extensions, gateways and other resources to a trusted web of communication servers, where any adds, moves, changes, failures or new routes are automatically absorbed within the cloud with no additional configuration. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Testing for media?
I'm in the process of setting up an asterisk box that will stand between PBX's and the SIP providers. So a trunking server. How can I 'test' to see if this trunking server is stepping out of the media path during calls? Thanks David -- -- www.ringfree.biz 828-575-0030 -- _ -- 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] Testing for media?
tcpdump and wireshark would help I guess. Just sniff for sip traffic and look out for what's happening there. My 2 cents Sent from my iPhone On May 19, 2012, at 8:33 PM, David Wessell da...@ringfree.biz wrote: I'm in the process of setting up an asterisk box that will stand between PBX's and the SIP providers. So a trunking server. How can I 'test' to see if this trunking server is stepping out of the media path during calls? Thanks David -- -- www.ringfree.biz 828-575-0030 -- _ -- 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] Best practices to route calls according holidays
Olivier wrote:
Hi,
At the moment, I'm mostly using a Day/Night toggle button to let
users deal with week-ends, holidays and opening hours.
As Asterisk 1.8 introduces Calendar capabilities, I'm wondering if
better alternatives now exist.
Is it possible, safe, reliable and easy to refer from Asterisk to a
public calendar resource listing holidays, for a given country ?
Should you instead refer to a private resource, to avoid depending on
an externaly managed resource ? If you go this way, which tools would
you recommend to build and update a private calendar ?
Suggestions ?
Regards
--
The database approach that others have suggested sounds pretty good. What
I did was to write a simple agi script that dispatches a subroutine based
on the holiday. I hard coded the holidays in the script, but they could
just as easily be stored in a db.
Here's the key portion of the script. (The formatting may get goofed up
in the email).
#!/usr/bin/perl
use strict;
use warnings;
use Asterisk::AGI;
use Date::Calendar;
$|++;
my ($min, $hr, $day, $mo, $yr, $dow) = (localtime)[1..6];
$mo++;
$yr += 1900;
my $today = sprintf(%d%02d%02d, $yr,$mo,$day);
my $holidays = {
New Year's Day = #Jan/1,
Easter = +0,
Memorial Day = 5/Mon/May,
Independence Day = #Jul/4,
Labor Day = 1/Mon/Sep,
Thanksgiving = 4/Thu/Nov,
Black Friday = 4/Fri/Nov,
Christmas Eve = #Dec/24,
Christmas Day = #Dec/25,
Christmas Dayafter = #Dec/26,
New Year's Eve = #Dec/31
};
my %dispatch = (
New Year's Day = \new_years_day,
Easter = \easter,
Memorial Day = \memorial_day,
Independence Day = \july4,
Labor Day = \labor_day,
Thanksgiving = \thanksgiving,
Black Friday = \black_friday,
Blackout Period= \blackout_hrs,
Christmas Eve = \christmas_eve,
Christmas Day = \christmas_day,
Christmas Dayafter = \christmas_dayafter,
New Year's Eve = \new_years_eve,
);
my $agi= Asterisk::AGI-new;
my $calendar = Date::Calendar-new( $holidays );
$calendar-year( $yr );
foreach my $holiday ( keys %$holidays ) {
my @holiday = $calendar-search( $holiday );
my $holidaydate = sprintf(%d%02d%02d, $holiday[0]-year,
$holiday[0]-month,
$holiday[0]-day
);
if ( $today == $holidaydate ) {
$dispatch{ $holiday }-($agi);
exit;
}
}
if ( in_blkout_period( $today ) ) {
$dispatch{Blackout Period}-( $agi, $dow, $hr );
exit;
}
##
sub playback {
my ($agi, $holiday, $hrs) = @_;
$agi-stream_file([
'frys/thank_you_for_calling',
frys/$holiday,
frys/$hrs,
'frys/enjoy',
'frys/frys_goodbye'
]
);
}
sub new_years_day {
my $agi = shift;
$agi-exec('noop', Incoming call on New Year's Day);
if ($hr 10 or $hr = 19) {
playback($agi, 'new_years_day', '10to7');
$agi-hangup();
}
else {
$agi-exec('Goto', 'welcome');
}
}
--
Ron Bergin
Network Operations Administrator
Fry's Electronics, Inc.
--
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[asterisk-users] make and receive call from dial-up modem
is it possible to use data voice dial-up modem to make and receive call for ivr system.. -- _ -- 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] Realtime peers and trunks coming from the same IP
I use an SBC to protect my pool of asterisk servers and as trunking endpoint with SIP Telcos. Now I'm trying to implement SIP phone registration to be delegated through the SBC, as a proxy. It doesn't work. It just works when I don't use realtime peers at the asterisk servers. Using realtime SIP peers, since there is one SIP phone that gets his registration delegated through the SBC, any inbound call that tries to reach any asterisk server, coming from any SIP Telco trunk ended at my SBC, gets refused in asterisk. As asterisk records the IP of the SBC as the IP of the phone that has been registered, it thinks that those calls coming from the SBC are calls coming from that phone, and it refuses them with 401 Unauthorized replies. I'm using asterisk 1.8.11. How can I surpass this problem? Is there any configuration that I'm lacking on, or is this a limitation of asterisk? Thanks, Ricardo. -- _ -- 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] make and receive call from dial-up modem
Mahendra Dobariya wrote: is it possible to use data voice dial-up modem to make and receive call for ivr system.. This should help: http://www.voip-info.org/wiki/view/X100P+clone 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] make and receive call from dial-up modem
Hey Mahendra, We happen to still stock the same X100P cards in India. If you require one, do connect with me off the list. Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Sat, May 19, 2012 at 9:59 PM, Doug Lytle supp...@drdos.info wrote: Mahendra Dobariya wrote: is it possible to use data voice dial-up modem to make and receive call for ivr system.. This should help: http://www.voip-info.org/wiki/**view/X100P+clonehttp://www.voip-info.org/wiki/view/X100P+clone 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-**usershttp://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
[asterisk-users] SET SIP_CODEC and Video issues
Greetings List.
I Have a small test server and i'm facing a small issue.
i have setup two SIP PEERS and they are able to do Video calls.
now I'm testing SET SIP_CODEC in a dial plan and when ever i'm setting the
codec .. the inbound (=first) leg stops receiving or sending video and SIP SHOW
CHANNELS shows only the Codec i set in the dialplan.
is it possible to avoid this problem?
Asterisk version
1.8.11.0
SIP.CONF
===
[TK1000]
type=friend
secret=0jCiOdT81P
videosupport=yes
qualify=yes
host=dynamic
dtmfmode=rfc2833
context=DER-TEST
canreinvite=yes
disallow=all
allow=ulaw,alaw,gsm,h263,h263p
[TK1000]
type=friend
secret=0jCiOdT81P
videosupport=yes
qualify=yes
host=dynamic
dtmfmode=rfc2833
context=DER-TEST
canreinvite=yes
disallow=all
allow=ulaw,alaw,gsm,h263,h263p
EXTENSIONS.CONF
[DER-TEST]
;exten = _.,1,NoCDR()
exten = _.,1,Set(SIP_CODEC=alaw)
exten = _.,2,Set(SIP_CODEC_OUTBOUND=gsm)
;exten = _.,2,Set(SIP_CODEC_INBOUND=gsm)
exten = _.,n,DIAL(SIP/TK${EXTEN})
exten = h,1,Hangup()
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
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[asterisk-users] IAX2 passing back and forth variables
Hi all,
I have two asterisk servers A and B.
And I would like from A, dial to B passing some IAX variables.
Then B handles the calls, setup some other variables that become available
to A which can continue.
So far, I have used IAXVAR function.
It works when sending call from A to B
But variables setup on B are not available on A.
Any idea how I can do it ?
Here are my dialplans.
+++
SERVER A
+++
[contextA]
exten = s,1,Set(IAXVAR(TESTVAR1)=abcd)
exten = s,n,Dial(IAX2/serverb/s,30,g)
exten = s,n,Noop( The out variable is : ${IAXVAR(TESTVAR2)} ) ;
Does not work
+++
SERVER B
+++
[contextB]
exten = s,1,Noop( ${IAXVAR(TESTVAR1)} ) - Does work
exten = s,n,Set(IAXVAR(TESTVAR2))
exten = s,n,Hangup
--
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Re: [asterisk-users] IAX2 passing back and forth variables
Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2 to
anything. You would need some sort of Set(IAXVAR(TESTVAR2)=...)
Noah
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 passing back and forth variables
Hi all,
I have two asterisk servers A and B.
And I would like from A, dial to B passing some IAX variables.
Then B handles the calls, setup some other variables that become available to A
which can continue.
So far, I have used IAXVAR function.
It works when sending call from A to B
But variables setup on B are not available on A.
Any idea how I can do it ?
Here are my dialplans.
+++
SERVER A
+++
[contextA]
exten = s,1,Set(IAXVAR(TESTVAR1)=abcd)
exten = s,n,Dial(IAX2/serverb/s,30,g)
exten = s,n,Noop( The out variable is : ${IAXVAR(TESTVAR2)} ) ; Does
not work
+++
SERVER B
+++
[contextB]
exten = s,1,Noop( ${IAXVAR(TESTVAR1)} ) - Does work
exten = s,n,Set(IAXVAR(TESTVAR2))
exten = s,n,Hangup
--
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Re: [asterisk-users] IAX2 passing back and forth variables
Sorry, the dialplan is really on server B
exten = s,n,Set(IAXVAR(TESTVAR2)=efgh)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah
Engelberth
Sent: 2012-05-19 14:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 passing back and forth variables
Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2
to anything. You would need some sort of Set(IAXVAR(TESTVAR2)=.)
Noah
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 passing back and forth variables
Hi all,
I have two asterisk servers A and B.
And I would like from A, dial to B passing some IAX variables.
Then B handles the calls, setup some other variables that become available
to A which can continue.
So far, I have used IAXVAR function.
It works when sending call from A to B
But variables setup on B are not available on A.
Any idea how I can do it ?
Here are my dialplans.
+++
SERVER A
+++
[contextA]
exten = s,1,Set(IAXVAR(TESTVAR1)=abcd)
exten = s,n,Dial(IAX2/serverb/s,30,g)
exten = s,n,Noop( The out variable is : ${IAXVAR(TESTVAR2)} ) ;
Does not work
+++
SERVER B
+++
[contextB]
exten = s,1,Noop( ${IAXVAR(TESTVAR1)} ) - Does work
exten = s,n,Set(IAXVAR(TESTVAR2))
exten = s,n,Hangup
--
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Re: [asterisk-users] Configuring OpenVOX A400P issues
Thanks Vladimir for the response and apologies for my extremely late one! This week was quite crazy!! I have just upgraded my server to FreePBX to take advantage of an up-to-date system. I migrated the original config over with a few set changes. On 05/14/2012 04:42 AM, Vladimir Mikhelson wrote: Kaya, I do not have a definitive answer for you, but here are several things I noticed. 1. fxo*ls*=1 -- I would definitely try /fxoks/ instead I altered this in /etc/dahdi/system.conf: # cat system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Sat May 19 20:22:49 2012 # 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: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) fxoks=1 echocanceller=mg2,1 fxsks=2 echocanceller=mg2,2 # channel 3, WCTDM/4/2, no module. # channel 4, WCTDM/4/3, no module. # Global data loadzone= uk defaultzone= uk 1. fxs*ls*=2 -- I am not sure about your provider but I would try /fxsks/ instead I think the above took care of that too. 1. [May 13 13:15:31] WARNING[3624] chan_dahdi.c: CallerID feed failed: *No such file or directory*-- It looks like your installation is missing some executable files or they ended up in some unexpected places. New build straight from CD so no issues here anymore as far as I could see in the logs. 1. 2 FXO FXSLS (In use) (SWEC: MG2) *RED *It looks your PSTN line is in red condition Still shows up as red :-( # lsdahdi ### Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 FXSFXOKS (In use) 2 FXOFXSKSRED 3 unknownReserved 4 unknownReserved Channel 1 is not in use though it's free as in no one is using it? -plus when press green button on DECT handset I get dialtone. However, plugging an old non-cordless POTs handset into the 10meter cable I have going from the PSTN line over to the Asterisk server did come up with a lot of noise on the line. I think it's because it's a 10 meter cable and it does cross over some power sockets so probably picking up 50Hz hum and additionally other signal residue from around (Electronics 1-0-1 long cable = antenna). - Can't asterisk clean this up using digital filters, or can anything else be done for it? I run my PBX in a 72 rack sitting in my living room and there isn't enough space to put the thing next to the PSTN socket! 1. It would also help to see the contents of the /etc/asterisk/dahdi-channels.conf and /etc/asterisk/chan_dahdi_additional.conf. # cat dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Sat May 19 20:22:49 2012 ; 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: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) ;;; line=1 WCTDM/4/0 FXOKS (In use) signalling=fxo_ks callerid=Channel 1 4001 mailbox=4001 group=5 context=from-internal channel = 1 callerid= mailbox= group= context=default ;;; line=2 WCTDM/4/1 FXSKS signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 2 callerid= group= context=default # cat chan_dahdi_additional.conf ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; ;;[250] signalling=fxo_ks pickupgroup= mailbox=250@default immediate=no echotraining=800 echocancelwhenbridged=no echocancel=yes context=from-internal callprogress=no callgroup= callerid=Communal Extension 250 busydetect=yes busycount=7 accountcode= channel=1 The following thread can have some relevance to your case. Please run fxstest and post the results here. http://forums.digium.com/viewtopic.php?f=1t=80253start=0 I looked at this but couldn't figure out where to obtain fxstest from as it's not in the yum repos and Google doesn't come up with any answers either?? -Vladimir Regards, Kaya -- _ -- 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] SET SIP_CODEC and Video issues
Of course you are disabling the video maybe also include the video protocols in the sip_codec -Original Message- From: Tarek Sawah tareksa...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Sat, 19 May 2012 17:33:57 To: Asterisk Usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SET SIP_CODEC and Video issues -- _ -- 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
