[asterisk-users] ChanIsAvail kills dialplan processing when no Zap available on 1.2.14.
Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro.
The logic is simple; try Zap/g1 (a group of two E1s), and if that
fails, try locating a channel via DUNDi. Here's a massively cut down version
to illustrate the problem I'm having.
macro dialout ( dest ) {
ChanIsAvail(Zap/g1);
noop(Value of AVAILCHAN is ${AVAILCHAN});
Dial(Zap/G1/${dest}||H);
};
Here's what happens when only the second bearer is connected:
-- Executing Macro(SIP/1210-082a9768, dialout|0800789456) in new stack
-- Executing Set(SIP/1210-082a9768, dest=0800789456) in new stack
-- Executing ChanIsAvail(SIP/1210-082a9768, Zap/g1) in new stack
-- Hungup 'Zap/32-1'
-- Executing NoOp(SIP/1210-082a9768, Value of AVAILCHAN is Zap/32-1) in
new stack
-- Executing Dial(SIP/1210-082a9768, Zap/G1/0800789456||H) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G1/0800789456
-- Zap/62-1 is making progress passing it to SIP/1210-082a9768
-- Zap/62-1 answered SIP/1210-082a9768
-- Hungup 'Zap/62-1'
== Spawn extension (macro-dialout, s, 4) exited non-zero on
'SIP/1210-082a9768' in macro 'dialout'
== Spawn extension (macro-dialout, s, 4) exited non-zero on
'SIP/1210-082a9768'
i.e. perfect - it picks the first available channel on the second
bearer - Zap/32. If only the first bearer is connected, it picks Zap/1
as I'd expect.
The killer is if /neither/ bearer is connected, I get this:
-- Executing Macro(SIP/1210-08299328, dialout|0800789456) in new stack
-- Executing Set(SIP/1210-08299328, dest=0800789456) in new stack
-- Executing ChanIsAvail(SIP/1210-08299328, Zap/g1) in new stack
== Spawn extension (macro-dialout, s, 3) exited non-zero on
'SIP/1210-08299328' in macro 'dialout'
== Spawn extension (macro-dialout, s, 3) exited non-zero on
'SIP/1210-08299328'
Processing does not continue to the NoOp or Dial - what am I doing
wrong? I've also tried with the 'j' option to 'jump to priority n+101'
even though I'm using AEL, but it's made no difference.
Cheers,
Gavin.
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Re: [asterisk-users] ChanIsAvail kills dialplan processing when no Zap available on 1.2.14.
On Wed, 24 Jan 2007 09:11:20 +
Gavin Hamill [EMAIL PROTECTED] wrote:
Processing does not continue to the NoOp or Dial - what am I doing
wrong? I've also tried with the 'j' option to 'jump to priority n+101'
even though I'm using AEL, but it's made no difference.
For the benefit of the archive
I got this working by using a 'catch h {...}' block
at the bottom of the macro rather than switch'ing on the variables
set by ChanIsAvail().
Cheers,
Gavin.
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Re: [asterisk-users] How to exit from console?
On Tue, 23 Jan 2007 22:36:12 +1100 Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried qui Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Run asterisk just by typing 'asterisk'. Using the -c option will cause the behaviour you are experiencing. Then when you connect with 'asterisk -r', you can use 'exit' (or just ctrl-c) to disconnect, but leave Asterisk running in the background. On the console, you can get the - behaviour with 'set verbose 4' Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL parse failure on 1.2.14
Am I doing something really stupid in this AEL macro, or is nesting an
'if' inside a 'switch', inside an 'if' not supported in the 1.2 AEL
parser?
macro stdexten( ext , dev ) {
// First determine if the SIP peer is registered here
Set(aretheyhere=${SIPPEER(${ext}:status)});
if(${aretheyhere:0:2}) == OK) {
MixMonitor(${UNIQUEID}.wav|b);
Dial(${dev},30);
switch(${DIALSTATUS}) {
case BUSY:
MailboxExists(${ext});
if(${VMBOXEXISTSSTATUS} != SUCCESS) {
Busy(5);
};
Voicemail(b${ext});
Hangup();
break;
default:
MailboxExists(${ext});
if(${VMBOXEXISTSSTATUS} != SUCCESS) {
Congestion(5);
};
Voicemail(u${ext});
Hangup();
break;
};
};
};
When I do an AEL reload, I get
2007-01-23 16:11:31 WARNING[10795]: pbx_ael.c:102 __grab_token: Syntax error at
line 370 of 'extensions.ael', too many closing braces!
-- Registered extension context 'macro-stdexten'
-- Added extension 's' priority 1 to macro-stdexten
-- Added extension 's' priority 2 to macro-stdexten
-- Added extension 's' priority 3 to macro-stdexten
-- Added extension 's' priority 4 to macro-stdexten
2007-01-23 16:11:31 WARNING[10795]: pbx_ael.c:102 __grab_token: Syntax error at
line 371 of 'extensions.ael', too many closing braces!
2007-01-23 16:11:31 NOTICE[10795]: pbx_ael.c:1146 handle_root_token: Unknown
root token '}'
Cheers,
Gavin.
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[asterisk-users] IGNORE: AEL parse failure on 1.2.14
Doh!
macro stdexten( ext , dev ) {
// First determine if the SIP peer is registered here
Set(aretheyhere=${SIPPEER(${ext}:status)});
if(${aretheyhere:0:2}) == OK) {
^^^ errant close-bracket
Sorry for the noise (twice).
gdh
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Re: [asterisk-users] Problems with rxfax
On Monday 22 January 2007 20:19, Marco Mouta wrote: I would install Hylafax(opensource too) with Asterisk via IAXModem, it worth! You can keep all the features, hylafax running in same server or a separated one, and IAXmodem will be a Modem on Hylafax, and a simple extension on you *. It works great for me, and there are more users using this architecture, you can setup as much IAXModem as your servers can handle, so it's very scalable. Just a 'me too' - we send about 3500 faxes per day via Hylafax (I appreciate this thread is about fax reception), until recently using the onboard DSPs of an Eicon Diva Server quad-BRI card, and the success rate we've achieved using 8 instances of IAXModem instead of the Diva Server card is just as high :) The thing that really impressed me was that we're not using any kind of dedicated (v)LAN for the link between the Hylafax and Asterisk servers. The IAX data is just shuffling across the same noisy LAN with chatter from a hundred Windows boxes.. HTTP traffic.. Samba file sharing.. all kinds of nonsense, and any problems are negligible! Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Backports to 1.2.14 of 1.4.0 app_queue features.
Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay in Call Distribution using the Queue Application
On Monday 15 January 2007 19:22, [EMAIL PROTECTED] wrote: Hello all, For example there are 10 callers in the queue, an Agent is finishing a call and it takes up to 30 seconds before his phone rings again. We're already set the wrapuptime parameter in queues.conf to 0, for my point of view an agent phone that becomes available again should ring immediately after hanging up a call. Try setting wrapuptime to 1 . Setting it to zero likely enables some default value. Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple users and a single extension
On Friday 05 January 2007 21:06, Phil Finkler wrote: Hi all, Quick question. Is there a way to have multiple people have an extension, say 900, to their polycom 501 SIP phones on one of the blue buttons to where when a call comes in, exten = 900,1,Set(CALLERID(name)=TechSupport) exten = 900,2,Dial(SIP/101SIP/102SIP/103) will most likely do what you want :) gdh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] over 200 queues, anyone?
On Thu, 04 Jan 2007 11:05:38 +0100
Lenz [EMAIL PROTECTED] wrote:
You are correct, this is more or less the scenario involved - the
problem is that people want to call a personalized line AND speak to
the same subset of agents preferably.
I have never seen such a setup myself - I have seen CCs with 30 or
40 queues, never 200 - so I was wondering if anybody ever trued
something on these lines; or if there are better solutions to the
same problem. Best regards
We planned to do this.. but using only one queue for the agents, but
depending on the incoming DDI dialled, we do this:
441616608981 = huntgroup(affiliates,Affiliate One);
441616608982 = huntgroup(affiliates,Affiliate Two);
441616608983 = huntgroup(affiliates,Affiliate Three);
macro huntgroup( queuename , friendlyname ) {
Ringing();
Set(CALLERID(name)=${friendlyname});
Set(CDR(userfield)=${queuename} ${friendlyname});
ResetCDR(wav);
Queue(${queuename},t);
...};
So there's one Asterisk queue called 'affiliates' but we are able to
display a different name on the SIP handset just by setting the CALLERID
(name), thus the agent knows what to answer the phone as :)
(The Set(CDR) + ResetCDR were to ease the accounting + billing of
calls..)
Cheers,
Gavin.
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Re: [asterisk-users] Re: Asterisk Queues
On Thursday 28 December 2006 13:33, Matt wrote: Hi, Well, in our case, it seems that the issue was being caused by announcements. That is, someone in QUEUE1 would be waiting 15 minutes.. and QUEUE2 would be waiting 5 minutes. Yep we noticed this too - it's a rather unfortunate side-effect; we found a very agreeable workaround, though :) Instead of making use of the announcement feature, we made a .WAV file which is 60 seconds long. The first 10 seconds are 'Sorry you have been waiting' and the remaining 50 seconds are the sound of telephone ringing... This way the customer gets the announcement and indication they are on hold, and Asterisk processes the queue more elegantly :) The only side-effect is sometimes a queue member will join in the middle of the announcement, but this is of little concern to us. Cheers, Gavin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] vzaphfc?
On Thursday 28 December 2006 23:27, Tzafrir Cohen wrote: vzaphfc is not a complete replacement of bristuff. It replies on most of it. Rather, it replaces the zaphfc subdirectory with an improved ZapBRI driver for HFC-s-based PCI cards. Further, if you're looking for 'something else' re: cheapo ISDN cards, definately give Asterisk 1.4 and mISDN a look - no BRIStuff, no huge patches, no wacky stuff.. all Asterisk-core support that worked really well in the brief time I tested it. The key difference is rather than generating 8000 interrupts per second, the mISDN kernel driver (which itself can be thought of 'isdn4linux' version 2.0) polls the card, leading to much lower system load, and no 'wanted 8 bytes, read 7!' errors from dmesg. Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallbackLogin() deprecated in 1.4
On Wed, 20 Dec 2006 14:39:42 +0100
Markus Bönke [EMAIL PROTECTED] wrote:
Hello all,
The other thing is, that show agents
doesn't show me which agents are logged in and if I use show queue
I can see local channels attached to a queue but no agents. For my
point of view there is some functionality lost with the new concept.
Snap! I've been designing an * system for our call centre and fallen
into exactly the same trap. I ended up coding my own agent login/logout
procedures using astdb functions to store the extension at which an
agent is sitting...
However what I'm missing most is a 'wallboard' for 'number of agents
on Do-Not-Disturb' / number of waiting calls / average wait time, so I'm
considering QueueMetrics, but E 2500 is a lot of cash for that one
feature. I'll probably get one of the codies here to knock something
together.
If I want to program a realtime display to show agentstates in queues
based on the output from show queue, what's the concept to map
agents to the local channels? How can I configure agents in future?
Well, you might want to make use of the 'pre-queue AGI' facility, and
use that to set that agent as 'on call' (Postgres/MySQL or just AstDB)
in that.. then after the Queue application exits, use ${UNIQUEID} to
change the state of the agent to 'free' via func_odbc or another AGI.
If you specify setinterfacevar=yes in queues.conf, then
you can do 'GET VARIABLE MEMBERINTERFACE' in the AGI to find the name
of the Local/ channel that the caller is about to be connected to.
Then look that up in AstDB... here's what I do - it's heavily based on
the agi-test.agi that comes with Asterisk.
# Which queue member was this incoming caller about to speak to?
print GET VARIABLE MEMBERINTERFACE\n;
my $result = STDIN;
checkresult($result);
# Incoming string is 200 result=1 (Local/[EMAIL PROTECTED]) so we need to
# trim the fat
$aid=$result;
$aid =~ s/.*Local\///;
$aid =~ s/[EMAIL PROTECTED]//;
chop $aid; # drop the end carriage return
# This /has/ to work because this is the same logic that the 'agents'
# context uses in the dialplan!
print DATABASE GET LRCC $aid\n;
my $result = STDIN;
checkresult($result);
# More trimmings.
$ext=$result;
$ext =~ s/.*\(//;
$ext =~ s/\).*//;
chop $ext;
You can then go on and do...
$sql=UPDATE agent_status SET status = 'on call', uniqueid='.$AGI
{'uniqueid'}.', extension='.$AGI{'ext'}.' WHERE agentid='$aid';
$dbh-do($sql);
.. then just view the contents of the agent_status table. I hope that
makes sense - it was a bit of a ramble :) 1.4 has been a lot of fun so
far - I'm using a lot of the new features and doing stuff that I
couldn't have thought of with 1.2 :)
Cheers,
Gavin.
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Re: [asterisk-users] IBM Server / USB Ports
On Tue, 19 Dec 2006 10:35:32 -0500 Matt [EMAIL PROTECTED] wrote: Ok so that 'unknown' is infact the Digium card then? I suspected that. The Vendor ID is 'D161' which is supposed to look a bit like the first four letters of 'Digium' :) gdh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup Party
On Tue, 12 Dec 2006 15:27:06 +0200 Idris AVCI [EMAIL PROTECTED] wrote: Hello, Is there a way to find out which party hanged up the call. Generally this is reported as Local disconnet/Remote disconnect in callcenter environments. This is already written to the queue_log e.g. 1165572107|1165572085.354|french|Local/[EMAIL PROTECTED]|COMPLETEAGENT|20|2 or 1165495361|1165495218.23|french|SIP/1337-08234748|COMPLETECALLER|6|137 gdh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RINGNOANSWER on 1.2
Hi, I've been trying to implement this [1] on 1.2.13 and whilst my twiddlings seem to work, I just wanted confirmation that I'm not doing something really stupid which could cause a segfault under certain conditions. My chan_queue.c addition is this one line: ast_queue_log(queue, qe-chan-uniqueid, outgoing-chan-name, RINGNOANSWER, %d, orig); The output in queue_log is of the format 1165076773|asterisk-21332-1165076763.17|ccuk|Local/[EMAIL PROTECTED],1| RINGNOANSWER|1 Is there a way I can just have 'Local/[EMAIL PROTECTED]' without the other stuff after it? [1] http://lists.digium.com/pipermail/asterisk-commits/2006-May/004096.html Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_sql_postgres gone in 1.4
Hi,
I'm putting together a system to manage agents with Realtime, and
without chan_agent. In 1.2.13, there's a handy (although marked as deprecated
in apps/Makefile) PGSQL application to let me do this:
macro queue-addremove(queuename,penalty) {
switch(${MACRO_EXTEN:0:1})
{
case I: // Login
PGSQL(Connect connid host=XXX user=XXX password=XXX dbname=XXX);
PGSQL(Query resultid ${connid} INSERT INTO queue_member_table
VALUES (\'${queuename}\'\,\'Local/${MACRO_EXTEN:[EMAIL
PROTECTED]'\,${penalty}));
PGSQL(Clear ${resultid});
PGSQL(Disconnect ${connid});
break;
I do this because AddQueueMember does not INSERT the new agent into the table
defined in extconfig.conf (I even have ReadOnly in odbc.ini set to No). In this
way, I can preserve the state of agents between Asterisk restarts.
However, I notice in 1.4 beta3, this application has gone. Can anyone suggest
what would be the best alternative?
I have thought of System(psql -h xxx -U -P xxx ...) but that's just
horrendous :)
gdh
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[ignore] Re: [asterisk-users] app_sql_postgres gone in 1.4
On Fri, 1 Dec 2006 16:26:49 + Gavin Hamill [EMAIL PROTECTED] wrote: slump Just found func_odbc.conf - problem solved. :) Hurrah, it's Friday afternoon, and not a moment too soon gdh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallbackLogin deprecated?
On Tue, 28 Nov 2006 17:57:04 -0600 Octavio Ruiz (Ta^3) [EMAIL PROTECTED] wrote: Is there an isolated example somewhere of how to use existing dialplan logic and dynamic queue membership to simulate the current behaviour? http://svn.digium.com/view/asterisk/trunk/doc/queues-with-callback-members.txt Thanks for that - didn't realise the mainline docs contained such useful and comprehensive information these days! Why? Seems that reinventing the well was the agentcallbacklogin implementation, when it could be happend in dialplan logic. Cool, in conjunction with the one-line patches at http://bugs.digium.com/view.php?id=7736 I think I have the ACD functionality I need without bothering with chan_agent :) Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallbackLogin deprecated?
On Tue, 28 Nov 2006 17:57:04 -0600 Octavio Ruiz (Ta^3) [EMAIL PROTECTED] wrote: Why? Seems that reinventing the well was the agentcallbacklogin implementation, when it could be happend in dialplan logic. Hm, now that I have examined this in more depth, I still seem to be missing one vital piece of the puzzle. The queues-with-callback-members.txt tutorial assumes that one agent (as a specific human being) is always reachable at a specific phone. This is not the case, and why I investigated chan_agent in the first instance. Our agents sit at any phone and log in, so their ACD groups follow them. This is what I really meant about re-inventing the wheel, since with AgentCallbackLogin removed, surely I'll have to maintain my own database tables of which agent is available at which extension? I'm hoping I've just overlooked something really obvious :) Cheers, Gavin, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallbackLogin deprecated?
On Tue, 28 Nov 2006 10:27:27 -0600 (CST) Jason Parker [EMAIL PROTECTED] wrote: Yes, AgentCallbackLogin is deprecated, but it will not be removed until after 1.4. Is there an isolated example somewhere of how to use existing dialplan logic and dynamic queue membership to simulate the current behaviour? What about generation of statistics for callcentre monitoring? If this is not taking place through chan_agent, won't it be reinventing the wheel to have to simulate this behaviour, too? Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Doubling up; redunancy with DUNDi
Hi :) We currently have a single * box with 4-port E1 card terminating
60 channels:
[PSTN]
| | 2 x E1
[Asterisk]
| | 2 x E1
[Legacy PBX]
What I'd like to have is this:
[PSTN]
| \__
||
[*1]- - - -[*2] - DUNDi peering between 2 * boxes
||
[Legacy PBX]
Whereby a call in either direction would be routed either 'straight
through' to/from the PSTN from/to the Legacy PBX, or in the case where
all channels were in use (we max out at about 40-45 channels usage), it
would connect via IAX to a free channel on the other * box.
This is slightly different from the DUNDi tutorials I have found on the
web (including markster's excellent one on voip-magazine.com) whereby
DUNDi is only consulted if 'is this extension local?' fails. In my
case, both sets of extensions are local to both boxes, but I want to
utilise any spare capacity on the other machine. Also, I am not using
any SIP devices here, only Zaptel ones.
Should this be as straightforward as replacing our current 'Dial'
entries in extensions.conf :
exten = _31.,1,Dial(Zap/G2/${EXTEN})
with a 'DUNDi-aware' version like:
exten = _31.,1,Dial(Zap/G2/${EXTEN})
switch = DUNDi/priv
How does this deal gracefully with the scenario of 'no free channels' ?
Is there much variance in DUNDi between 1.2.X and 1.4.X ?
Cheers,
Gavin.
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Re: Subject: [asterisk-users] Slow dialing from PBX via E1
On Thu, 3 Aug 2006 08:58:47 +0800 Hi Gavin, This is the default of setting of the Asterisk. If you wish to adjust the timing, please edit the source file of the Asterisk name call chan_zap.c, And look for the static int matchdigittimeout line to change the setting. The timing is in millisecond. Remember, compile your Asterisk after the changes. Ouch I was really hoping to avoid a recompile :) I can understand why Asterisk would wait up to 3 seconds in case more digits were received, and this makes sense when listening for digits from an analogue handset via a channel bank, but this is one digital PBX talking directly to another - is the 'matchdigittimeout' still valid here? I guess what I'm saying is, is there a non-chan_zap.c way of affirming that extensions receieved from chan_zap are complete and non-ambiguous? Our legacy PBX already does its own 3-second timeout to wait for extra digits.. so 6 seconds in total is a huge wait for end users :( Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Slow dialing from PBX via E1
Hi :) I have a 'slow dialing' problem. When I dial 200# for the 'echo test' application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 so Asterisk has accepted the call setup from the PBX. Then exactly 3 seconds elapses, and finally: -- Executing Playback(Zap/65-1, demo-echotest) in new stack -- Playing 'demo-echotest' (language 'en') at which point Allison announces 'You are about to enter an echo test..' How can I remove this 3 second pause? It's really annoying, and it doesn't happen when I dial out from the legacy PBX via an ISDN30 bearer not connected to Asterisk (nor does it happen with SIP phones on Asterisk). Even with debug + verbose both at 99, I see no extra information The extensions.conf is trivial: [general] static=yes writeprotect=yes [fromaxxess] exten = 200,1,Playback(demo-echotest) ; Let them know what's going on exten = 200,2,Echo ; Do the echo test exten = 200,3,Playback(demo-echodone) ; Let them know it's over This is with Asterisk 1.2.4 and Zaptel 1.2.3, on a Sangoma A104u (Sangoma support say their driver does no buffering and can't understand why this is happening) As ever, any advice warmly welcomed :) Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pre-made E1 crossver cables for the UK
Hi, just a note to let people know that I had NetShop make me some E1 crossover cables to replace my own dodgy crimpings, and they work perfectly =) The 3 metre version is £5.58 and their order code is CS000111/3 (I guess the /3 refers to the length..). They're at www.netshop.co.uk - 01753 691661. Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] interfacing w/ a legacy InterTel PBX
Erik Anderson wrote: In Italy, alphanumeric caller IDs are not used, only numbers and I'm able to use them between two systems. Great - thanks! ___ Just as thought .. If your Axxess has T1/E1 switchable cards in it (as ours do), consider running the Axxess - Asterisk link in E1 mode since you'll then get 30 usable voice channels as a 'freebie' :) My Asterisk + Axxess experiences have been a world of pain, but it's all working now with a Sangoma card - I gave up on the Digium stuff, but looking back I think it was more a bad set of coincidences (lots of badly made cables) and Dell hardware :) gdh ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agents in 1.2.1
Hi, I've used Agents + Queues before with success, but I can't figure
out why this trivial setup is not functioning...
stage*CLI show agents
1306 (gdh) available at '[EMAIL PROTECTED]' (musiconhold is 'default')
1 agents configured [1 online , 0 offline]
and the internal context is simply:
[internal]
exten = _13XX,1,Dial(SIP/${EXTEN},,h)
Now, taking this line...
exten = 123454,1,Dial(SIP/1306)
(Legacy PBX On Zaptel interface dials 123454)
-- Starting simple switch on 'Zap/66-1'
-- Accepting overlap call from '1010' to '123454' on channel 0/4, span 3
-- Executing Dial(Zap/66-1, SIP/1306) in new stack
-- Called 1306
-- SIP/1306-f498 is ringing
-- Channel 0/4, span 3 got hangup request
== Spawn extension (fromaxxess, 123454, 1) exited non-zero on 'Zap/66-1'
-- Hungup 'Zap/66-1'
Great - the phone rings - hurrah! BUT... :O
exten = *11,1,AgentCallbackLogin(${CALLERIDNUM}||[EMAIL PROTECTED])
exten = 123455,1,Dial(Agent/1306)
(SIP phone 1306 dials *11)
-- Executing AgentCallbackLogin(SIP/1306-d752,
1306||[EMAIL PROTECTED]) in new stack
== Setting global variable 'AGENTBYCALLERID_1306' to '1306'
-- Playing 'agent-loginok' (language 'en')
== Callback Agent '1306' logged in on [EMAIL PROTECTED]
-- Playing 'vm-goodbye' (language 'en')
== Spawn extension (fromip, *11, 1) exited non-zero on 'SIP/1306-d752'
(Legacy PBX On Zaptel interface dials 123455)
-- Starting simple switch on 'Zap/66-1'
-- Accepting overlap call from '1010' to '123455' on channel 0/4, span 3
-- Executing Dial(Zap/66-1, Agent/1306) in new stack
== Everyone is busy/congested at this time (1:1/0/0)
-- Hungup 'Zap/66-1'
Why am I being told that 'everyone is busy' on this Agent, when it is
clearly 'available', and calling the SIP device directly does work?
I'm assuming it's because of something I'm doing wrong, but I can't see
what :(
gdh
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[Asterisk-Users] Info request from Sangoma users
Hi :) I have an A104 and wondered if other owners could confirm the strange behaviour I'm seeing.. it's best seen on an idle system, thus eliminating asterisk or other factors.. Very simply, just let 'vmstat 1' run for a few minutes and watch the output, specifically the 'sy' column... On the 2.4G Xeon machine I'm using, the system CPU usage sits very low for a minute or two, and then spikes up to 100 for a few seconds, before tailing off again - this happens all the time :( Interestingly, the 'load average' as reported with 'w' always stays at zero even with this high 'system load'... I moved the card to another PCI slot (and bus) and get the same thing, but now much more frequently, but for a much shorter length of time... Now bringing Asterisk into the picture, I can't use Monitor() because once I get even 5 simultaneous recordings, the real 'load average' on the machine spikes up to 2 and greater, and calls become stuttered as the machine fails to keep up with whatever it's doing.. The machine is SCSI, with a decent LSI Logic onboard controller and fast disks - so it's nothing to do with enabling DMA - hdparm shows 70MB/sec with minimal load increase. Can anyone confirm this behaviour? Cheers, Gavin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info request from Sangoma users
On Tue, 2005-12-13 at 07:24 -0500, Matt Florell wrote: Hello, Can you post what firmware your board is and what wanpipe driver version you are using? Hi Matt :) I've already been through all this with Sangoma's support - just looking for external opinions from real-life installs - so thank you for the response :) I've seen this behaviour with everything from the first 2.3.2 Asterisk-compatible wanpipe to the latest 2.3.3-beta18. We do up to 50 concurrent recordings on our systems and they do not have recording issues. We use MegaRAID 320-1 cards as well. That's what I thought - I mean the amount of disk IO is absolutely nothing at all :( What kind of CPUs are you using? Also, single or dual (or a single with hyperthreading ?) What onboard L2 cache do they have? My last hope is to try a P4 machine with 1MB cache, since the others I've used have 512K.. They're all Dell machines - and I know the reaction that usually evokes when dealing with Digium hardware (been there, seen that...) - I thought someone like Sangoma with many more years in the business would be more immune to things like this :( Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info request from Sangoma users
On Tue, 2005-12-13 at 14:32 +0100, Patrick wrote: Been a while since I used Asterisk on a Dell box but I remember I had to turn off HT. Have you tried that? For sure, I've tried HT on and off, with 2.4 and 2.6 kernels :) or booting the kernel with noht. On Dell boxes I have also seen some funky NMI received for unknown reason. Dazed and confused messages in /var/log/messages. Yes I had those with the Digium card (before I returned it, obviously :), although Digium support managed to solve those in the driver. While you are at it reseat everything you can find :) Feel the build quality :)) As a test you can also disable the onboard nic and stick in a quality nic on its own interrupt to see if that helps. And off course disable in the BIOS everything that you do not use (serial/parallel/usb etc.). All very sage advice - I have another box to try it on yet before curling up in a corner and crying - I'll report back if I find anything spectacularly wrong :) Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attack dialing
On Mon, 2005-12-12 at 06:39 +1100, Eric Bishop wrote: Anyone have eny elegant dial plan config for attack dialing? Basically I just want to automatically and continuously dial a busy until it is answered and then hand it over to a SIP hanset. Hi Eric :) The terminology I've always understood for this feature is called 'Camp on Hold' ... and google says this about 'asterisk camp on hold' http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+reverse+hold Good luck! :) gdh ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change time when * is running
Julian Lyndon-Smith wrote: Kevin P. Fleming wrote: Can I change the time when * is running ? I don't want to try just in case it causes * some grief. It can cause some repercussions. I wouldn't recommend changing the time backwards by such a large amount while Asterisk is running with active calls... Julian, can you check and make sure that your system is configured to use the correct timezone? Usually just typing # ls -l /etc/localtime will be enough - to ensure that the symlink is pointing at the right area of the world, since that will automatically keep the clock up to date with daylight savings etc. Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Change time when * is running
Julian Lyndon-Smith wrote: Hmm, ls -l /etc/localtime -rw-r--r-- 1 root root 1323 Nov 25 12:43 /etc/localtime there's no symlink that I can see. This is CentOS 4.2 OK, I just had a look in /usr/share/zoneinfo and the only files which were 1323 bytes were for UK + Ireland, so if that's where you are, then you're all set - looks like you will indeed need a clock jump - best schedule that in a quiet time, or just do it and blame BT :) Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running asterisk within screen
Tzafrir Cohen wrote: /usr/bin/screen -L strace -f -o /tmp/trace /usr/sbin/asterisk -v and I have no screen session running and I also have no asterisk CLI to connect to. I can't explain the behaviour and the screenlog is empty. permissions? If that is what you suspect, strace the whole screen session. Can this be as simple as needing to run 'asterisk -c' to keep a console open (which then 'screen' will manage) ? Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Load spikes with 1.0.10
Hi, I have a trivial setup on a 2.4GHz Xeon Dell PE 1750 SCSI machine
dealing with 4 ports of E1 in an 'inline PBX' arrangement.
My extensions.conf is simply:
[general]
static=yes
writeprotect=yes
[frompstn]
exten = _31.,1,Dial(Zap/g2/${EXTEN})
exten = _31.,2,Congestion
[fromaxxess]
exten = _13.,1,Dial(SIP/${EXTEN},,h)
exten = _13.,2,Congestion
exten = _31.,1,Dial(Zap/g2/${EXTEN})
exten = _31.,2,Congestion
include = outbound
[outbound]
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Congestion
We have a full 30-channel PRI and a 4-channel partial PRI and are
experiencing load spikes that I can't find the source of.
The machine Debian sarge on the default 2.6.8-2-686 kernel, and no other
daemons are running than sshd.
The machine is doing no IP work - purely TDM, yet on a Xeon 2.4GHz
machine, the load average is sitting at 0.6 with 40 active Zap channels
(i.e. 20 live calls) and will randomly jump to 2 (with call quality
starting to stutter)
A few seconds of vmstat:
procs ---memory-- ---swap-- -io --system--
cpu
r b swpd free buff cache si sobibo incs us sy
id wa
7 0 0 223560 1276 22304000 310 8394 1 2
97 0
0 0 0 223552 1284 22304000 016 5128 3461 1 0
98 1
0 0 0 223552 1284 22304000 0 0 5094 3319 10 9
81 0
0 0 0 223552 1284 22304000 016 5130 2955 1 10
89 0
0 0 0 223552 1292 22304000 060 5121 2918 0 1
97 2
0 0 0 223552 1292 22304000 0 0 5031 2936 1 0
99 0
Does this sound about normal for what is just shuffling data between
ports of the Sangoma A104? I want to record the call data with the
'Monitor' application but this just causes the load to increase even
more (even though 'hdparm' shows 70MB/sec disk transfer with low
user+system CPU usage)
/proc/interrupts is
CPU0
0: 423253622IO-APIC-edge timer
1:175IO-APIC-edge i8042
9: 0 IO-APIC-level acpi
11: 0 IO-APIC-level ohci_hcd
12: 58IO-APIC-edge i8042
15: 13IO-APIC-edge ide1
177: 50 IO-APIC-level ioc0
185: 29 IO-APIC-level ioc1
193: 1311243931 IO-APIC-level wanpipe1, wanpipe2, wanpipe3, wanpipe4
201: 13289965 IO-APIC-level eth0
217:5420038 IO-APIC-level eth2
NMI: 0
LOC: 423311408
ERR: 0
MIS: 0
Help! :)
Cheers,
Gavin.
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[Asterisk-Users] 1.2.0 using 1G of RAM
Hi, this morning I've switched our 'in-line' Asterisk system (between legacy PBX and PSTN) live after a few false starts with the PBX configuration. I've been executing 'show channels' probably hundreds of times, and I wanted to see show channel Zap/64-1 So I start with 'show cha' and pressed TAB. asterisk then segfaulted, but I was able to immediately reconnect to the console with -r. A minute or two later, response from the console was very sluggish, and calls stopped were dropped from the PBX. I ctrl-c'd back to the console and found that every byte of 640M swap space was being used as well as all of the this 512MB RAM machine, and asterisk's entry in 'top' was: 32021 root 25 0 1112m 473m 7476 S 0.0 93.8 0:00.10 asterisk I have not yet killed this process and I wondered how I could determine where the memory leak is occurring. It's 1.2.0 final, compiled and installed with 'make install' - I have not stripped any symbols, etc. Any help would be greatly welcomed since I dearly want a stable and modern asterisk system! :) Cheers, Gavin. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Slow dialling from PBX into * via E1
Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and finally -- Executing Playback(Zap/65-1, demo-echotest) in new stack -- Playing 'demo-echotest' (language 'en') at which point Allison's 'sultry' voice announces 'You are about to enter an echo test...' Can I remove this 3 second pause? It's rather annoying, and it doesn't happen when I dial out from the PBX to a normal PSTN line or to a PBX-provided extension, or indeed to any local ISDN2 device. Even with debug + verbose both at 99, I see no extra information I'm using a Sangoma A104u with wanpipe-beta15-2.3.3.tgz and Asterisk/Zaptel/Libpri 1.2.0-beta1. The extensions.conf is trivial [general] static=yes writeprotect=yes [fromaxxess] exten = 200,1,Playback(demo-echotest) ; Let them know what's going on exten = 200,2,Echo ; Do the echo test exten = 200,3,Playback(demo-echodone) ; Let them know it's over Sangoma's support can't understand how it can be their drivers / cards causing the issue since there is no buffering at all in Zaptel (and let's face it that makes sense :) Cheers, Gavin. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1
Yep, another list posting on this topic :) All the messages I've read on this are from people experiencing these errors in quiet times - I get them as soon as I plug a port on our TE410P to an Inter-Tel AXXESS PBX.. and I get them continuously... I'm just sticking an * box in between ISDN30e (we're in the UK so euroisdn) and the PBX.. and whilst the telco ISDN30e side works like a charm [1] I simply can't get a reliable link to the PBX.. I've tried two different T1 crossovers (1-4, 2-5) with identical results and zapata.conf is indeed using signalling=pri_cpe for the telco ISDN30e and pri_net for the PBX Digium support have taken me through loopback testing which came out perfect, and the card is not sharing any IRQ, yet this error renders the card useless :( Digium are reluctant to accept a return and replace the card since they don't believe it to be at fault - and neither do I. I see the same behaviour with 1.0.9 asterisk / libpri and 1.0.9.1 zaptel... and CVS-HEAD versions of everything. Any ideas/advice would be warmly received right now! Cheers, Gavin. [1] http://www.voip-info.org/tiki-index.php?page=UK+Asterisk+Details ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) onPrimary D-channel of span 1
On Wednesday 03 August 2005 17:33, Jens von Bülow wrote: Gavin, Any ideas/advice would be warmly received right now! You are not going to like my response... Erk :) The only way I could get this to work (luckily I had 2 identical sites and was busy with the upgrade to the gen2 card) was to downgrade to zaptel 1.0.7. Alas no - just moved down to zaptel, libpri and asterisk 1.0.7 with identical behaviour, both with span=1,0,0,ccs,hdb3,crc4 and span=1,1,0,ccs,hdb3,crc4 - I don't have any other active spans in the system :/ Tim Panton: As above, I've already tried timing source twiddles (and even changing the build-out length values, even though the cable is 2 metres :)) My whole zaptel.conf is span = 1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk With zapata.conf snippet: switchtype=euroisdn immediate=no overlapdial=yes pridialplan=unknown prilocaldialplan=unknown group = 1 signalling=pri_net context = fromaxxess channel = 1-15 Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Full T38 sip Faxing now Available
On Wednesday 03 August 2005 18:46, Michael D Schelin wrote: Why do you put me down? I have not done a thing to you and I'm not a spammer. Please stop this activity It's not professional. If I were to give you bad service please feel free to comment negatively but I've never dealt with you nor do you have an account with us. I guess the point Kevin was making was that whilst your post wasn't advertising a l0w m0rtg4g3 r4t3 or selling v!ag.rAah... it was still touting a commercial service on a non-commerical mailing list... After all, just take a look at the To: header on each post to the list :) asterisk-biz would have been a suitable vehicle for your posting - but of course it wouldn't have reached 10,000 subscribers :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stupid hold music
On Sunday 24 July 2005 22:44, Steve Gladden wrote: OK was actually able to pull it out of the archives! It's now at http://stuff.michiganbroadband.com/asterisk I'll leave it there for about a week or two then remove it. T othe best of my knowledge it's public domain, if anyone needs more info please contact me offlist. Sorry, this ain't public domain - it's from the soundtrack for the 1996 PC game 'The Neverhood' - the whole soundtrack was composed by Terry S. Taylor and is full of this kind of stuff and much better :) So, a commercial release, alas... well worth a search on your favourite peer-to-peer music service to 'sample' though :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell Hardware
On Friday 22 July 2005 14:48, Anton Krall wrote: Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models? http://www.digium.com/index.php?menu=compatibility Digium's recommendation is quite clear: 'Don't use Dell hardware' And it's a great shame Digium hardware has such problems on Dell kit, since there's so much of it about :( Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A104c vs. A104u
On Friday 15 July 2005 21:12, Mike M wrote: I'm just trying to decide if the extra ?200 for the A104u is worth it :) Isn't it the other way around? c u? Yes you're quite right. I think I must have just taken the headstaggers last Friday :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A104c vs. A104u
On Friday 15 July 2005 21:12, Mike M wrote: I'm just trying to decide if the extra ?200 for the A104u is worth it :) Isn't it the other way around? c u? The c version has channelized HDLC which means board does HDLC instead of main CPU. That mirrors what their UK reseller 'MyPhonecall.co.uk' said: The A104c is the channelised card and is not tested with Asterisk whereas the A104u is the unchannelised card and IS tested with Asterisk. Unfortunately, the A104u is GBP 1199, whereas the A104c is GBP 999 - that's quite a difference. Less bit banging on main CPU is good. 200 of what currency? The ? was meant to be UK Pounds Sterling, so about US $350 difference.. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma A104c vs. A104u
Hi, Just a quickie - if I want to implement an * solution purely for voice (well, and physical fax machines / dialup modems..) on EuroISDN E1s, is there any benefit to the A104u over the A104c? I'm just trying to decide if the extra £200 for the A104u is worth it :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers
On Thursday 07 July 2005 19:55, Russell Horn wrote: I can call non-geographic numbers from my land line in the US, my mobile phone and from any calling card I have tried. This isn't an issue with BT but with broadvoice and those they contract to supply connections to the UK PSTN. nod If BroadVoice don't let you call national rate numbers, then use a second ITSP for those routes, or switch completely. They're cheap and nasty, but they do use IAX... http://www.call1899.co.uk/voip.php http://www.call1899.co.uk/voiprates.php No signup fee, no contract, cheap rates, lukewarm reliability :) Suck it and see - better than no service at all. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers
On Thursday 07 July 2005 20:42, Mark Phillips wrote: My take on this is that they are protecting themselves against fraud. Discounting the freefone numbers for a while, the national rate numbers are charged at variying rates and so how is a company to know just what they are gonna get charged. In the UK, the charging bands for each band of numbers is publically available information - the service provider can know accurately what they will be billed for each charge band. i.e. http://www.serviceview.bt.com/list/current/docs/Call_Charges.boo/16321.htm http://www.serviceview.bt.com/list/current/docs/Call_Charges.boo/00165.htm Of course these are BT retail rates but I fully expect wholesale rates based on call prefix will be available for carriers / ITSP Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial more then 9 digits
On Wednesday 15 June 2005 12:40, altus wrote: exten = _OO.,1,Dial(H323/[EMAIL PROTECTED]) Sorry, I couldn't help but notice this... Is that really meant to be _OO (capital letter 'Oh') rather than _00 as the double-zero international prefix? Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 190: dial tone without registration?
On Monday 13 June 2005 16:42, alan wrote: Hello. I'm currently evaluating the Sipura SPA-841, and snom 190 phones for use in an Asterisk PBX/call center environment. How about tackling this with iptables and matching specific IP addresses on specific MAC addresses? Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AT-320 + supervised transfer
On Tuesday 07 June 2005 09:44, Giordano Grandis wrote: Ok, just a thing...cuold is see a sample peer in tuou extensions.conf I'm newly testing the atxfer and i always the same question: if i transfer a call to a peer that don't answer me, ho can i re-take the call. Actually i got the call hanged up without the possibility the speack back with my first caller. I have the same problem now that I've actually tried this... and so have other people - check this thread which has been running at the same time: http://lists.digium.com/pipermail/asterisk-users/2005-June/110856.html The 'hook flash' certainly doesn't have any effect - it just puts the call on hold (even though it's already on hold because of the atxfer..) sigh Cheers, Gavin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: 4 port BRI options ?
On Monday 06 June 2005 17:12, Louis-David Mitterrand wrote: With kernel 2.6 you only get ISDN tty devices through the capidrv module, no more analog modem login support with the obsoleted diva2i4l module. I've never used it on 2.6, and only with the official Eicon supplied drivers with the 'Divatty' kernel module. You get benefits from the zaptel interface (zapscan, zapbarge, etc.), closer to the metal, less latency. Yep, that's true. It works, but when ISDN channels are busy faxing, asterisk has no way to know which outgoing channel is free: users get a congestion when trying to call out (unless you program some channel testing logic/loop inside your dialplan). This is a major bummer in production. Have a look at the MTPX 'adapter pool'support that comes as part of the Eicon driver set - it gives a single eight B-channel device via CAPI, and both the TTY and CAPI interfaces will do a 'find any free channel' via this, so as long as you have one channel, it actually doesn't make any difference whether you've opened /dev/ttyds01 or /dev/ttyds08 :) I expect the same would then be true of multiple CAPI applications. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium G729 licensing - is it worth the trouble?
Yeh, I know what you mean re segfaults.. I just worked through the directory until I found one that worked and had good translation times (9ms on a P4-celeron).. show translation recalc 10 is the magic command. The codec works fine with the ATCOM -320 phones - in a test environment it worked fine with a few UK providers. gdh -Original Message- From: Nir Simionovich[EMAIL PROTECTED] Sent: 06/06/05 09:26:32 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Digium G729 licensing - is it worth the trouble? Cool, so you have satisfied yourself that you are licensed to use the G.729 codec and not get your ass sued by the IP holders. Now you can simply use the no-license-required codecs from here... http://kvin.lv/pub/Linux/Asterisk/ I've tried using these codecs in the past, but usually ended up with a pile of SegFaults and Machine Halts. Question is, are these codecs maintained by Someone, and if they had ever been interoped with other equipment. It's one Thing to implement a codec, there is another interoping it with other vendors. Nir S ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: R: R: R: [Asterisk-Users] AT-320 + supervised transfer
Hi again :) I'm afraid I simply don't have any more suggestions... It 'works for me' ... gdh -Original Message- From: Giordano Grandis[EMAIL PROTECTED] Sent: 06/06/05 17:40:52 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: R: R: R: [Asterisk-Users] AT-320 + supervised transfer Hi Gawin, I'm newly testing the atxfer and i always the same question: if i transfer a call to a peer that don't answer me, ho can i re-take the call. Actually i got the call hanged up without the possibility the speack back with my first caller. Thanks Giordano ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium G729 licensing - is it worth the trouble?
On Sunday 05 June 2005 16:31, Chris Mason (Lists) wrote: I have purchased 50 licenses at $10 each from Digium, Cool, so you have satisfied yourself that you are licensed to use the G.729 codec and not get your ass sued by the IP holders. Now you can simply use the no-license-required codecs from here... http://kvin.lv/pub/Linux/Asterisk/ I'm probably massively over-simplifying the problem.. but I see this as the same problem as having installed the same MS Office 2000 CD + key on 50 PCs, but as long as you have 50 unique certificates of authenticity for Office 2000, it doesn't matter a crap (in the real world) what actual key got used on each PC. gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 port BRI options ?
On Friday 03 June 2005 14:28, Nardis Dome wrote: --- Brett, Gary [EMAIL PROTECTED] wrote: Is the Eicon that much better ? sorry, i have only experience with Eicon... maybe someone else is able to give a feedback... Aside from paying for a recognised brand name, with Eicon you get on-board DSPs and firmware, as well as software to make the card appear as a series of /dev/tty devices (and/ or CAPI) under Linux. The Junghanns BRI boards have no such on-board logic, they are pure telephone interfaces driven by the host - and as such they are /perfect/ for Asterisk. We use an Eicon Diva Server 4BRI for our fax server because of the TTY interface for Hylafax - the DSPs themselves do all the fax negotiation / compression and it works extremely well. However, we're also running * on that machine via CAPI, and the board gives us the flexibility to share 8 channels between * and Hylafax really easily :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 port BRI options ?
On Friday 03 June 2005 15:19, Remco Barende wrote: Hmmm, the sirrix isn't even that much cheaper than the Junghanns boards. The difference is only EUR 100 Telephony is an expensive game to be in :) But my ISDN card was only 20 EUR! suddenly doesn't mean an awful lot when you have to manufacture in very low volume.. The Junghanns and Sirrix devices are not consumer kit, and you should not expect to simply multiply the cost of a single-ISDN card by the number of ports :) Asterisk has /vastly/ lowered the entry cost for telephony - ask around to find the average prices of, say, an E1 module for a proprietary PBX or a 16-channel BRI ISDN module... then suddenly 600 EUR for 8 channels will seem like a very good deal indeed. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Built-In Transfer Questions
On Wednesday 01 June 2005 06:45, Jennifer Hales wrote: Hello Matthew, You need to put exten = o,1,Hangup underneath your voicemail macro, then if your dial zero the call will come back to you, however it does read back an error in your ear. It still works. ... or alternatively, if you add the 'h' option to the Dial command, you will be able to hang up by pressing the * key on your phone (or if you use CVS, any sequence you define as disconnect in the [featuremap] section of features.conf) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: R: R: [Asterisk-Users] AT-320 + supervised transfer
On Wednesday 01 June 2005 11:01, Giordano Grandis wrote: Hi Gavin, I'm testing atxfer and it looks work fine, but i have a small problem: A call B B answer, dial atxfer extension and then the new peer (C) If C does not answer the phone, A and B got hangup and cannot speak again I set canreinvite to no. Can u help me ? Hm, this is the same response as I've posted to the list earlier today :) if you add the 'h' option to the Dial command, you will be able to hang up by pressing the * key on your phone ... If C's phone does not answer, pressing * should return you to talking to A. You can change the '*' button by changing the 'disconnect' line in features.conf. Be sure to STOP NOW and restart asterisk when changing features.conf - a 'reload' is /not/ enough. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AT-320 + supervised transfer
On Wednesday 01 June 2005 12:43, Giordano Grandis wrote: No...maybe i don't explain u well. After that B call C andC not answer (go in timeout), B hear first the beeperr and then, together A the busy tone. Now i can't re-take the call :| I'm afraid I don't have any more suggestions to offer - anyone else? Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] AT-320 + supervised transfer
On Wednesday 01 June 2005 13:04, Giordano Grandis wrote: Ok, thanks for all. Just a thingh: how do u set DTMF on your phones ? We have them set to RFC2833. I think I've noticed some cases where the remote party hears the tones, but it's not an issue that bothers me :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: [Asterisk-Users] AT-320 + supervised transfer
On Wednesday 01 June 2005 14:15, Giordano Grandis wrote: This is what happen when i call a peer that not answer: Jun 1 13:45:57 WARNING[25325]: res_features.c:858 builtin_atxfer: Unable to create channel Local/[EMAIL PROTECTED]/n do you have chan_local? I don't like the look of this part at all. Please try to rm /usr/lib/asterisk/modules/* then 'make clean; make install' on a fresh checkout of CVS HEAD :) Also, there should be no need for the 'r' option to Dial since SIP already supports all the progress indication necessary. gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: R: [Asterisk-Users] AT-320 + supervised transfer
On Tuesday 31 May 2005 14:41, Giordano Grandis wrote: Hi Gavin, But...how atxfer work ? Ehm, just the way I explained yesterday :) Just make sure you include the 't' option to the Dial application, in the same way you need for the old-style '#' blind-transfer to function. gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AT-320 + supervised transfer
Hi, 1.0.7 does not support atxfer. You nees to use CVS. I have a bunch of the ATCOM cheapos and only CVS will take notice of 'atxfer' in features.conf. Otherwise , consider this scenario... Call comes in, press HOLD, dial other party to see if they wish to speak to the caller. If so, press * to hang up, then HOLD to swap back to the incoming caller. Announce you are going to transfer them, and now dial the final extension and press FWD to do a blind transfer. This works for me with the SIP 1.43 firmware. The IAX fw still has some way to go... The phone seems slow to boot? Ensure you have the IP/hostname of a valid NTP time server at the bottom of the web config page. If you don't, it will take ages and eventually fall back on 'time.windows.com' Cheers, Gavin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] AT-320 + supervised transfer
On Monday 30 May 2005 16:19, Giordano Grandis wrote: Hi, Thanks for yuor answer. The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time. Well, I run a local NTP server, so it's as fast plus has the correct time at the end :) I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer I will try also to use CVS, but i am skeptic to utilize asterisk to emprove atxfer...how asterisk emprove the atxfer ? :| When Asterisk does the transfer natively, the procedure is like this: Call comes in, hold on I'll try to transfer you. you dial *2 (or any sequence you define), speak to the remote party. If they want to speak to the caller, YOU hang up. If they don't, THEY hang up and you are returned to the original caller :) How do u set your sip.conf for the at-320 ? Did u set the canreinvite option ? [1300] type=friend username=1300 secret=ahem host=dynamic context=from-ip nat=yes canreinvite=no Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID for UK
On Monday 30 May 2005 07:22, Vassilis Konstantinou wrote: Hmmmyes but last time I played with my FXO module on the TDM400 could not detect hangup properly (that is on a London BT line). Has this been fixed? I keep an eye on the CVS but I have not seen any fixes for that. Maybe I missed it. No idea... I didn't say the TDM400P was actually any good (I don't own one), just that it is better supported than the X100P :) gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: [Asterisk-Users] AT-320 + supervised transfer
On Monday 30 May 2005 17:22, Giordano Grandis wrote: The procedure that will do asterisk is very nice ;) but whe it was available ? Asterisk's atxfer support is only in CVS. Currently is there any way to emprove the transfer? I tryied the scenario that u suggest me but it doesn't work :| and i don't why. You *must* be using a new firmware for the phone. Download 1.43 from http://www.aredfox.com/edownloadssip.htm (the AT-320 needs PA186S code) Here my sip.conf for the phone, can u say me if there is somethingh wrong ? Looks fine to me.. I think is ok, maybe i have some problem on phone settings.Can I see your exmple phone setting ? They're at work so I can't see the config right now... but they're just the defaults with the DTMF changed to RFC2833 and the NTP server set... Try resetting to defaults using the procedure at http://www.voip-info.org/wiki-ATCOM+AT-320 Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK DID providers
On Saturday 28 May 2005 14:31, Tom Fanning wrote: Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. Yeh, Sipgate's price is good (hey you can't argue with £0 setup and £0 per month...) but the service is lukewarm- they drop off the 'net a fair amount and some calls just result in an unavailble tone to the caller (no incoming SIP activity). Try www.gradwell.com for an 0161... http://www.gradwell.com/voip/ddi-inbound.php They offer IAX2 recently as well as SIP. gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CRM integration (was RE: CallerID)
On Saturday 28 May 2005 20:21, Rusty Shackleford wrote: D'oh! I had misread the PP's statement and assumed he meant a bareback browser window. You are, of course, quite right. A Java app could handle this, but we are still left with the issue of having to install SOMETHING, even if it is a small Java app, on the client to make this work. What about this 'Ajax' stuff that's terribly trendy right now? It'd be a horrible polling implementation, but you could use a javascript Timer object to fire an XmlHTTPRequest every couple of seconds to check for new callerID at the IP address of the current browser? Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID for UK
On Saturday 28 May 2005 20:33, [EMAIL PROTECTED] wrote: Hi, Can anyone tell me if UK CallerID support has been added to the CVS for the x100p card ??? http://www.voip-info.org/wiki-Asterisk+and+UK+Caller+ID Please, please always try the wiki first :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID for UK
On Saturday 28 May 2005 21:41, [EMAIL PROTECTED] wrote: Hi Gavin, Thanks for the reply. I did look at the wiki and read the page you pointed at. According to the wiki mark didn't want to add caller id for the x100p card. I found patches for the x100p card and just this minute re-compiled and they work a treat. Was just wondering if the patches had been put into the CVS and if not was there plans to put the patches in. Would be a shame to waste the work and to deny caller id for these cards in the UK. Well, the official line is as Mr. Spencer has made in that bugtracker entry... Digium sell the TDM400P which supports polarity detection. CVS supports UK CallerID on that card. Digium no longer sell the X100P so it's not supported any more. The X100P is a fairly crappy choice for the UK since it has a hardcoded 600ohm impedance, suitable really only for the USA... But yes, 'it was only £10 on eBay' been there done that, wasted hours playing with txgain/rxgain/echo cancellation... :) Cheers, Gavin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID for UK
On Saturday 28 May 2005 23:31, [EMAIL PROTECTED] wrote: Hi Gavin, Can you recommend a good card/modem for the UK? Looks like I'm going down the same track as you. At first I was going for the TDM400 dev kit but they are not certified or available yet in the UK and I wanted to play :-) The cheapo X100P I have at home works adequately. We have such a low call volume it's barely relevant, so it 'works for me'... I don't know anything about the other affordable options like the Grandstream ATA-486 FXO port, etc. Aside from that, I'd just sign up a new geographic number with sipgate.co.uk and use that for incoming, thereby eliminating any analogue problems :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID when transferring calls.
On Saturday 28 May 2005 23:54, Adam Vocks wrote: So if my receptionist answers the phone and transfers it to one of my techs, I want my techs phone to display the caller ID of the person who called the receptionist. Does anyone have a solution to this problem? Hi :) show application dial Have a look at the 'o' option - is this what you're after? gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attended Transfer failing with Agents
using CVS HEAD :) Some config snippets:
extensions.conf:
[from-ip]
exten = 51,1,Dial(SIP/1301,20,t)
exten = 52,1,Queue(ddi831,t)
exten = 53,1,Queue(marketing,t)
[internal]
exten = _13XX,1,Dial(SIP/${EXTEN},20,Tt)
queues.conf:
[ddi831]
strategy=roundrobin
timeout=10
announce-frequency=0
announce-holdtime=no
member = SIP/1301
[marketing]
strategy=roundrobin
timeout=10
announce-frequency=0
announce-holdtime=no
member = Agent/1600
qax*CLI show agents
1600 (Smith.Suzanne) available at '[EMAIL PROTECTED]' (musiconhold is
'default')
qax*CLI show features
Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #1
Attended Transfer *2
One Touch Monitor *1
Disconnect Call * *0
If I dial 51, the phone rings and atxfer works by *2
If I dial 52, the phone rings and atxfer works by *2
If I dial 53, the phone rings but hangs up the instant I press '*'.
Any suggestions would be warmly received, because 'set verbose 99' and 'set
debug 99' print nothing whatsoever on this :/
Cheers,
Gavin.
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Re: [Asterisk-Users] CTI
On Tuesday 24 May 2005 20:51, Jean-Denis Girard wrote: Have you tried MozPhone ? It has rough corners but is usable, and as a Firefox extension, it will give you: . popup with callerid on incoming calls (you can accept or reject call), . open a web page when accepting call by using url parameter of asterisk's dial or queue command (so asterisk can include callerid or other info in url), . outgoing call from web page (using the tel:// protocol in href). Very interesting! Is it possible to use the three features you highlight above WITHOUT actually seeing the built-in soft phone? i.e. when I call an agent, I would actually call both their SIP hardphone and their MozPhone client? If you choose to try it, be aware that there is no automatic uninstall procedure. Can you briefly describe the manual procedure here? Of course this is a shameless plug, as I'm the developper of MozPhone... :)) gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Deleting Monitor Files After 2 Months
On Thursday 19 May 2005 13:51, Steve Totaro wrote:
Does anyone know the best way to automate the deletion of monitor files
after they age two months?
How about ...
$ find /path/to/files -ctime +60 -exec rm {}\;
Cheers,
Gavin.
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[Asterisk-Users] SIP transfers failing
Hullo :)
I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using CAPI for
ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from
sipgate.co.uk to any other extension.
My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to blind
transfer, simply dial the number you want to transfer to, and press 'FWD'...
This is what happens when I start the sip debug after the initial call
setup...
01618313800 is the callerID of the person making the call, 1301 is the
internal SIP extension logged in as Agent 1600 at 10.0.0.82.
10.0.0.242 and 194.24.251.3 are the same machine, just two IPs on the same
eth0.
All I'm doing is answering the SIP phone, tapping 500 and pressing FWD to
transfer the incoming caller to the screaming monkeys gsm. If I dial 500 from
the phone directly, I immediately hear the monkeys, so assumed that a
transfer should be possible. e.g. in [from-ip] I have:
exten = 500,1,Playback(tt-monkeys)
and the sip.conf section is...
[1301]
type=friend
username=1301
secret=1301
host=dynamic
context=from-ip
nat=no
canreinvite=no
In extensions.conf's [internal] context (used by AgentCallbackLogin) I have
exten = _13XX,1,Dial(SIP/${EXTEN},20,t)
so that the agent has the ability to transfer calls (I also tried 'Tt' for
completeness)
-- Executing SetCIDName(SIP/217.10.79.218-40b8edd8, CCUK) in new stack
-- Executing Queue(SIP/217.10.79.218-40b8edd8, ccuk|r) in new stack
-- outgoing agentcall, to agent '1600', on 'Local/[EMAIL PROTECTED],1'
-- Called Agent/1600
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1301|20|t) in new
stack
-- Called 1301
-- SIP/1301-9ebb is ringing
-- Agent/1600 is ringing
-- SIP/1301-9ebb answered Local/[EMAIL PROTECTED],2
-- Agent/1600 answered SIP/217.10.79.218-40b8edd8
qax*CLI sip debug
SIP Debugging Enabled
qax*CLI
Sip read:
REFER sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.82:5060;branch=z9hG4bK95BVFY7yDu90YX0A
Max-Forwards: 7
User-Agent: PA168S
From: sip:[EMAIL PROTECTED]:5060;tag=g5VVthPSslPbjLib
To: CCUK sip:[EMAIL PROTECTED];tag=as0eb5392e
Call-ID: [EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]:5060
CSeq: 1 REFER
Refer-To: 500 sip:[EMAIL PROTECTED]
Referred-By: sip:[EMAIL PROTECTED]
Content-Length: 0
12 headers, 0 lines
Looking for 500 in from-ip
Looking for 1301 in from-ip
Transmitting (no NAT):
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.0.0.82:5060;branch=z9hG4bK95BVFY7yDu90YX0A
From: sip:[EMAIL PROTECTED]:5060;tag=g5VVthPSslPbjLib
To: CCUK sip:[EMAIL PROTECTED];tag=as0eb5392e
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 10.0.0.82:5060
set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send
to
set_destination: set destination to 10.0.0.82, port 5060
Reliably Transmitting:
NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 194.24.251.3:5060;branch=z9hG4bK08ef2ac1
From: CCUK sip:[EMAIL PROTECTED];tag=as0eb5392e
To: sip:[EMAIL PROTECTED]:5060;tag=g5VVthPSslPbjLib
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Event: refer;id=1
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14
SIP/2.0 200 OK (no NAT) to 10.0.0.82:5060
set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port to send
to
set_destination: set destination to 10.0.0.82, port 5060
Reliably Transmitting:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 194.24.251.3:5060;branch=z9hG4bK56da28c8
From: CCUK sip:[EMAIL PROTECTED];tag=as0eb5392e
To: sip:[EMAIL PROTECTED]:5060;tag=g5VVthPSslPbjLib
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 104 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 10.0.0.82:5060
monitor executing ( nice -n 19 soxmix
/var/spool/asterisk/monitor/agent-1600-asterisk-3407-1115714965-75-in.wav
/var/spool/asterisk/monitor/agent-1600-asterisk-3407-1115714965-75-out.wav
/var/spool/asterisk/monitor/agent-1600-asterisk-3407-1115714965-75.wav
rm -f
/var/spool/asterisk/monitor/agent-1600-asterisk-3407-1115714965-75-* )
after that it's just a series of shutdown SIP messages...
I can't understand that Subscription-state: terminated;reason=noresource
message from * to the phone - any ideas would be warmly welcomed!
Oh, I've tried all combinations of nat=yes/no and canreinvite=yes/no :/
Cheers,
Gavin.
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Re: [Asterisk-Users] IAX Timeout
On Monday 02 May 2005 16:07, Dan Levine wrote: Hello Everyone, How can I control the time Asterisk reregisters with the IAX Provider. The PPPoE ISP IP address sometimes address changes and the system doesn't reregister and incoming calls are disabled. Right now the only thing I'm able to do is Restart the server, that seems to solve the problem, but I know there is a better way. http://www.voip-info.org/wiki-asterisk+manager+events Connect to the Manager interface as part of the PPP script executed when you get a new IP address, and then issue an Event: Reload gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intel 536EP
On Saturday 30 April 2005 18:09, Jeff wrote: Will the Intel 536EP function as a FXO? No. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Tone
On Wednesday 27 April 2005 12:12, Henry Jensen wrote:
Hi,
User wants to call the number 12345
1. User picks up the hook
2. User dials 0 - hears dial tone
3. User dials 1 - dial tone stops
4. User dials 2345 - phone 12345 is ringing
We're using chan_capi and had this same problem... The following really hacky
solution works OK with Asterisk 1.0.7, but not with CVS - I don't know why :)
[default]
exten = _120.,1,Goto(s,1) ; fax extensions are 1201 - 1208
exten = s,1,NoOp( incoming call from ISDN )
exten = s,2,Answer
exten = s,3,PlayTones(dial); Give the caller a familiar noise.
exten = s,4,DigitTimeout(0.1)
exten = s,5,WaitExten(0.1)
; next section captures the next digit and stops the dialtone
exten = _X,1,NoOp( Got a digit! It was ${EXTEN})
exten = _X,2,StopPlaytones()
exten = _X,3,SetVar(Predigits=${EXTEN}) ; Put that digit aside for
use later...
exten = _X,4,Goto(s-gathermoredigits,1)
exten = s-gathermoredigits,1,NoOp( Now looking for the rest of the number)
exten = s-gathermoredigits,2,DigitTimeout,3
exten = s-gathermoredigits,3,WaitExten(8) ; and give the caller 8
seconds overall to do their thing
; log + dial the composite number of Predigits + the remainder
exten = _X.,1,NoOp(${TIMESTAMP} ok, now we're going to dial
${Predigits}${EXTEN})
exten = _X.,2,Goto(outbound,${Predigits}${EXTEN},1)
exten = t,1,Goto(#,1) ; If they take too long, give up
exten = i,1,Playback(invalid) ; That's not valid, try again
The [outbound] context is jsut full of the normal exten = _01.,1,Dial(blaaah)
call routing
If someone has a better way of doing this, I'd be interested to hear it!
Cheers,
Gavin.
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Re: [Asterisk-Users] X100P delayed ring on incoming calls?
On Friday 22 April 2005 10:45, Dave Cotton wrote: On Fri, 2005-04-22 at 10:22 +0100, bam wrote: Incoming calls seem to ring three or four times before asterisk then skips to do what it is supposed to do. If the caller drops the call before the extensions have started ringing asterisk seems not to pick this up and carries on regardless. I had this problem, and I think I tracked the problem down to the order I had the commands in my zapata.conf. Here is my working one which passes CallerID and causes * to pickup the call immediately: [channels] signalling=fxs_ks usecallerid=yes cidsignalling=v23 cidstart=usehist language=en context=from-landline echotraining=yes echocancelwhenbridged=yes echocancel=yes rxgain=1.0 txgain=-6.0 channel=1 immediate=no Previously I had the three CallerID directives *AFTER* the 'channel=1' and this seemed to confuse it.. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P delayed ring on incoming calls?
On Friday 22 April 2005 12:07, Peter Corlett wrote: Joseph Gutowski [EMAIL PROTECTED] wrote: [...] Either way, the best I've ever managed on the X100P's was 1 ring before Asterisk picks up and starts doing its thing. Well, when you think about it, it's hardly going to pick up after zero rings, is it? :) In the UK it's entirely possible - the CallerID info comes through as encoded data before the first ring has taken place :) Polarity change, a burst of V23 data, then the normal rings Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debugging zaphfc + PBX integration
On Friday 22 April 2005 14:46, Frank Sautter wrote: Gavin Hamill wrote: it seems as if this is exactly your problem. Sorry Frank, but this one isn't as simple as cabling... I've made reference in this thread already that I do have both straight + ISDN crossover (3/4 and 5/6 swapped) cables, and none of them work... one will get further than the other, i.e. using 'dmesg' I see a TEI request *FROM* the PBX, but I don't see any output from the HFC card going back to the PBX to tell it what TEI to use... Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How suppress echo
On Thursday 21 April 2005 13:07, Michaël Delvoye wrote: Hi all I try to implement an asterisk solution in my company. I use Digium card and sip phone Budgetone 102. When i call an external number, I hear myself during 3 or 4 sec and then it diseapear progressivly. Do you have any idea so as to solve my problem This is a really common question. Please start at http://www.voip-info.org/wiki-Asterisk+echo+cancellation and then spend several hours reading through the Wiki - it is a goldmine of information. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail 2 Email
On Thursday 21 April 2005 16:31, Peter Bowyer wrote: Many similar systems (webmail, bulletin boards etc) are configurable to use a local MUA (/sbin/sendmail etc) or talk SMTP directly to an MTA, either locally or remote. Asterisk voicemail unfortunately is not one of those systems (AFAICT) - you're stuck with having to use a local MUA. This is a good time to look at 'dummy mailers' like http://untroubled.org/nullmailer/ I usually use 'ssmtp' from Debian for this task - and since they purely exist to shuffle mail to a real MTA - they're very easy to configure - generally all you need is the IP address of the real MUA :) Cheers, Gavin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting SIP username for CallerID
Hi :) When I send an incoming call to a queue, I'm doing this: exten = 6608140,1,SetCallerID(CCUK) exten = 6608140,2,SetCIDName(CCUK) exten = 6608140,3,Queue(ccuk,r) I want the phone to say 'CCUK' - the queue name is more important to know than the incoming Caller ID :) Unfortunately the SIP phone (a cheapy using the PA168S chip and 1.42 firmware) displays the caller ID of asterisk when I do this, and it's clear why: --- -- outgoing agentcall, to agent '1601', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/1601 -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/1301|20|t) in new stack We're at 10.0.0.242 port 15334 12 headers, 12 lines Reliably Transmitting (NAT) to 10.0.0.82:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.242:5060;branch=z9hG4bK70ccd454;rport From: CCUK sip:[EMAIL PROTECTED];tag=as13d91518 To: sip:[EMAIL PROTECTED]:5060 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 20 Apr 2005 08:34:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 260 If I SetCallerID(12345678) then it is changed to sip:[EMAIL PROTECTED] as I'd expect, but if I use a string value, it stays at 'sip:[EMAIL PROTECTED]' So my question is, how can I change the sip username from sip:[EMAIL PROTECTED] to sip:[EMAIL PROTECTED] ? Am I doing something mind-bogglingly stupid? Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting SIP username for CallerID
On Wednesday 20 April 2005 10:32, Ronald Wiplinger wrote: So my question is, how can I change the sip username from sip:[EMAIL PROTECTED] to sip:[EMAIL PROTECTED] ? Shouldn't be there a quote mark and two values, like: SetCallerID(Ronald 123456789) Just tried a few combinations of that, and using the precise command above, the phone shows only the number. If I put a string inside the , * will still generate sip:[EMAIL PROTECTED]'... if I just put SetCallerID(CCUK) alone, I still get sip:[EMAIL PROTECTED] I am using CVS HEAD as of yesterday :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting SIP username for CallerID
On Wednesday 20 April 2005 11:15, Arunachala wrote: Try using SetCIDNum(CCUK) Nope, the most I can ever extract from any combination of the three 'CID' commands is this in the SIP messages :( From: CCUK sip:[EMAIL PROTECTED] Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting SIP username for CallerID
On Wednesday 20 April 2005 11:55, Arunachala wrote: Hi Gavin, Just went through the code. There is a check in the code to check whether the CIDNum is a phone number (0-9,#,*) or no. If it is not a phone number, it is replaced with the default CIDNum asterisk. Hm, really smart :) If the SIP username can be alpha-numeric, I wonder what's prompted this check? If you really want to fix this, you can do the following in the code: Thanks for the tip - I'll be sure to give that a go :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AT-320 phones with IAX2
Hi :) I've just received a couple of these units and they're working nicely as a basic unit for making/receiving calls. Alas, neither the HOLD, FWD or FLASH buttons do anything so I'm probably going to end up using Park and speed-dials ( #700 etc.) to implement these basic features unless of course anyone has experience of these phones to implement this natively / via clever config? I'm on the 1.42 firmware. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debugging zaphfc + PBX integration
On Saturday 16 April 2005 09:33, Tim Robinson wrote: So I do not believe power is the issue...I strongly suspect you need the 100 ohm terminators and you need to check that your ISDN BRI crossover cable is wired correctly. i.e. the RX pins on one RJ45 connect to the TX pins of the other. Have a look at Tim that's just what I wanted to hear :) I'd rather do a dodgy 'wrapping CAT5 strands around resistors' effort than mess about with powered NT1s :)) I know the cables themselves are wired correctly because our local PBX support made them, and they work perfectly when plugged into a real BT ISDN2e wallbox =) So, I'll pop down to Maplin on Monday and find myself some resistance :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Debugging zaphfc + PBX integration
Howdy - just got meself a cheapy Cologne chip card to play with in the hopes that it will prevail where a £1000 Eicon Diva Server failed... I'm trying to run in NT mode and appear as a trunk to a PBX (it's an Inter-Tel Axxess unit). If I hook a specific port on the BRI board of the PBX directly to a BT ISDN2e wallbox and dial 95000, I get a dialtone and can complete calls. Any time I connect to the Diva Server or the zaphfc card, the handset tells me 'TEST ISDN2 IS UNPLUGGED'... I installed asterisk on a completely virgin box using the install.sh contained in the bristuff 0.2.0-RC7k tarball. /etc/asterisk/zapata.conf [channels] switchtype = euroisdn signalling = bri_net pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=demo channel = 1-2 /etc/zaptel.conf loadzone=uk defaultzone=uk span=1,1,3,ccs,ami bchan=1-2 dchan=3 Zapata Telephony Interface Registered on major 196 PCI: Enabling device :04:01.0 ( - 0003) ACPI: PCI interrupt :04:01.0[A] - GSI 17 (level, low) - IRQ 185 zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xd002af00 fifo 0xcb3d8000 (0xb3d8000) IRQ 185 HZ 1000 zaphfc: Card 0 configured for NT mode zaphfc: 1 hfc-pci card(s) in this box. Registered tone zone 4 (United Kingdom) zaphfc: card 0 layer 1 state = G2 plugs cable into PBX zaphfc: card 0 layer 1 state = G3 zaphfc: empty HDLC frame or bad CRC received (framelen = 3, stat = 0xfe). zaphfc: empty HDLC frame or bad CRC received (framelen = 40, stat = 0xff). zaphfc: card 0 RX [ 0xfc 0xff 0x3 0xf 0xff 0x6d 0x1 0xff ] 8 bytes zaphfc: card 0 RX [ 0xfc 0xff 0x3 0xf 0xf6 0x3e 0x1 0xff ] 8 bytes zaphfc: card 0 RX [ 0xfc 0xff 0x3 0xf 0x5a 0x1f 0x1 0xff ] 8 bytes I'm told that the RX bytes are the PBX requesting a TEI from the provider (i.e. the * box) but I can't understand why no reply is being given. This is precisely the same problem I had with the Eicon Diva Server card... Is this likely to be as simple as needing the 100 ohm resistors, or can it be fixed with software? Cheers Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ilbc codec in Asterisk
On Friday 15 April 2005 15:28, Stephen wrote: Hi all, How can I use ilbc codec in asterisk? been trying to set allow=ilbc at my sip.conf and iax.conf but the call is format to ulaw, though I put allow=ilbc above allow=ulaw. I have been tried for almost two weeks and now seeking experts for help. : ( . need ilbc codec to reduce the total bandwidth used by asterisk. Anyone can share their experience ? Stephen, you would be much much better off paying the tiny sum of $10 for a G.729 license... not only is this a lot more 'industry standard' but the latency is lower as are the conversion times. Best still, the bandwidth usage is even lower than ILBC, I believe :) You can also get a usable implementation at [1] but please be aware this is non-licensed and as such you will be infringing on the intellectual property of patent holders by using it in a live environment. Cheers, Gavin. [1] http://kvin.lv/pub/Linux/Asterisk/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debugging zaphfc + PBX integration
On Friday 15 April 2005 18:28, Rob Scott wrote: If you your board into an ISDN wall socket and it works then you are acting as a terminal so you are in terminal mode. Now, how are you connecting to the PBX? If you are connecting to an ISDN extension on the PBX, then still you have to match the kind of connection, whether it is point-2-point or point-2-multipoint. Then it should work. Thanks for the suggestion, I'm happy with the ISDN cabling of straight through versus 3/4 and 5/6 crossed, and understand the differences in PTMP and PTP and also know what I should be configuring... TE mode on one port works perfectly, NT mode on another port doesn't =) From the advice I've been taking, I'm now quite convinced this problem is simply due to the lack of termination so I'll have to get some resistors and see what happens :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debugging zaphfc + PBX integration
On Friday 15 April 2005 22:50, Olivier MONNET wrote: Hello, It can be that you need power on your ISDN bus. I had this problem with a SIEMENS 3070 connected to a HFC-S card in NT mode. I have just added an ISDN NT to power the bus: just follow the instructions here: http://home.foni.net/~jolly1/download/PBX4Linux-2.5.html 2.2 Connect ISDN telephones to your ISDN card. Olivier, thank you. This is very interesting... but it begs the question... where does one go to find broken NT1 boxes?! We have three ISDN2e NT1s but they are all active and working, I don't think I would be very popular if I removed one 'to play with' :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 x ISDN2 hardware...?
On Monday 11 April 2005 08:29, Peer Oliver Schmidt wrote: Marc wrote: Is it possible to use hylafax and asterisk with only the AVM C4? Or do I need a separete fax modem? Works fine and dandy with a single AVM C4 here. Just wanted to chip in to say that Eicon's Diva Server 4BRI-8M is working great in a combined CAPI + TTY mode ... Asterisk listens with CAPI, HylaFAX uses the TTY interface.. Of course, we only have a need to send faxes so that simplifies the setup :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK CallerID patch with 1.0.7 / 1-0 CVS
Hullo :) I've been trying to use a stable 1.0.7 codebase against the patches at http://www.lusyn.com/asterisk/patches.html - but am having no joy. Even if I copy-paste the instructions on that site verbatim, everything compiles perfectly, but simply no incoming number is received. If I then go back to a CVS checkout (even including make clean, make install...) that I did at the end of February, everything works as it did before. (hurrah!) Does anyone know what might have changed before I start wading through the CVS logs for chan_zap.c ? Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK CallerID patch with 1.0.7 / 1-0 CVS
On Monday 11 April 2005 10:06, Gavin Hamill wrote: Hullo :) Bah, I got bitten by my own hacks. Things are now working much better than before :) I'd forgotten that I'd commented out the line: if (p-use_callerid p-cid_start == CID_START_USEHIST) in my previous CVS version, and this made CID work. The source of the confusion was I'd put the three 'usecallerid' commands in zapata.conf AFTER a 'channel=1' statement, so they were getting lost... I've moved them up the config file and now not only does incoming CID work, but the phone is answered immediately rather than waiting for the Bellcore CID to not find any data :) Another happy ending :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel bank replacement
On Friday 08 April 2005 16:35, Peter Hoppe wrote: Hello, I am working for a charity in the UK and I am projecting a new phone system. So - would there be any other way to connect 40+ telephones (two wire) into an asterisk box? Are there any voip gateways that actually conform to SIP standard (unlike what I heard from the Mediatrix voip gateways 1124 and 1204 which seem to use non standard SIP and have pay-as-you-upgrade)? Thank you very much for your consideration! Hi Peter, I'm not sure how you are getting PSTN lines into your * box, but if it's not ISDN30, you might want to consider some of the cheap IAX phones on the market now rather than trying to soldier on with old analogue kit? e.g. http://www.iaxtalk.com/product_info.php?cPath=1products_id=29 Shipping for 30 units and UK power supplies was $340, and with the weak dollar right now, that works out at just over 40 quid per phone - I'm sure there's movement on the unit price when buying in bulk... Now remove the need for an Asterisk Quad-E1 / T1 interface card and you've dropped the cost by nearly a grand food for thought :) They also sell a single-ethernet-port version of the phone for $10 less if you have enough ethernet sockets. Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK ISDN with Asterisk
On Friday 08 April 2005 23:33, Henry Owens wrote: Hi all, My question is: can Asterisk work well as a small office (8 extensions) PBX, with a mixture of analogue and IP phones, on an ISDN2e telephone line from BT? Sure, no problem at all.. Get yourself a 'Fritz!Card PCI' - also marketed by BT themselves as a 'BT Speedway ISDN' adapter - these seem to be the most cheap and supported of low-end ISDN2 adapters. chan_capi will deal with things like both B-channels so you can happily receive two calls on the same number, and deal with MSNs (Multiple Subscriber Numbers) gracefully since these are more likely on UK ISDN2e service than true DDIs. gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice controlled calling?
On Thursday 07 April 2005 22:17, Alex Vishnev wrote: Magnus, Also, compression gives voice recognition quite a challenge, as the speech samples arriving at the voip voice recognition engine is not the same as it was spoken using regular 64kbits pstn connection (as an example). If http://www.shazam.com/ can identify music tracks over a GSM mobile, then it must surely be possible for voice with a quiet background to be discerned :) Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
