Re: [asterisk-users] How to program a 100ms delay between the ringing of queued calls w/ ringall
On Tue, Nov 22, 2011 at 5:34 PM, Douglas Mortensen wrote: > Hello, > > ** ** > > Does anyone have any idea of how I can program a 100ms delay in between > the ringing of 2 subsequent calls in a queue configured with a ringall > strategy? > Does the "wrapuptime" queue option do what you want? http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf -Scott. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting MixMonitor options from Queue
Hello, We are recording our calls to queues by putting the appropriate options in our "queue.conf". This is all working properly. We would now like to set the MixMonitor option to adjust the caller volume (which is very quiet). With the regular MixMonitor application, we would just add the "v4" option to make it much louder. I don't see a way to set this option when MixMonitor is started from the queue. Does anybody know of a way to set MixMonitor options from the queue.conf? Thanks, Scott. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Matching Originate action with its NewChannel event
Jose Arias writes:
> An application commanding asterisk with AMI is going to launch lots of
> concurrent calls in very few seconds using the Originate AMI command but it's
> also going to need to be able to cancel very quickly any call of them even
> before each OriginateResponse event comes in. All the calls will be done by
> the
> same trunk (a trunking enabled channel). But there's a problem for
> canceling any call: there's no way to know what channel to hangup to because
> all channel prefixes in the NewChannel event are the same (the trunking
> channel
> one) and although the Originate action has an ActionId property, it isn't
> available in the NewChannel event but only in the OriginateResponse event,
> being very late.
I had a similar problem with faxes, and at the suggestion of somebody
on the list, solved it like this: I sent a custom identifier variable
in with the AMI command, which was then available to the channel. I
directed the calls into a custom dialplan which used UserEvent to send
an event with the identifier variable and all of the channel
information. The script could then use this event to associate the
identifier it generated with the channel information for that call.
I just used the fax filename as a unique identifier, and I passed it
in with a Variable line to AMI, called FaxFile. The dialplan entry
was like this:
exten => s,1,UserEvent(FaxStarted|Channel:
${CHANNEL}|Uniqueid:${UNIQUEID}|FaxFile: ${FAXFILE})
I then matched everything up by lookin at the FaxFile part of the user
event. In your case, you could just make up a unique identifier of
your own to send.
Hope this helps,
Scott.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Emulating attended transfer through the dialplan
Hello, I'd like to implement something similar to an attended transfer, but with a little more control (I'd like to be able to use MixMonitor and StopMixMonitor to control the call recording, set the account code, etc. I'm on Asterisk 1.4.26. All of the ways I have seen to do this are complicated plans using MeetMe and applicationmap features, and playing with those over the weekend, none of them seem to get the CDRs right. Ideally I'd like the sort of attended transfer you can get with a Zap flash: First talk to transferee, then all talk together, then hang up to let remaining parties talk to each other. I could also live with a *2-style attended transfer. When I use *2, the CDRs are in a workable format, so it seems like a good start. Is there any way to implement these without the problems that come with MeetMe? Or any suggestions on how to get usable CDRs out of MeetMe? Thanks! ---Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe time doesn't show up in CDRs?
Hello,
I'm working on some dialplan rules to pull multiple users into a
conference call. I have some fairly straightforward rules which start
up a new MeetMe conference, allow escape with the * key to invite more
users, then use a features.conf sequence to bring the new user into
the conference with ChannelRedirect.
The problem I'm running into is the time in the MeetMe conference
doesn't seem to show up in the CDRs anywhere.
I tried creating the MeetMe conference, bringing one user in, bring
another in, then keeping the conference open for 4 more minutes. I
ended up with 4 CDR entries. None of the times is anywhere near 4
minutes; they don't even add up to 4 minutes.
Any idea how I can get that time to show up in a CDR entry, or any
details of how CDRs should work with MeetMe and ChannelRedirect?
Here are my 4 CDR entries:
calldate=2009-07-26 01:05:32
clid="PPI/SWG"
src=SWG
dst=*
dcontext=cob-meetme-escape
channel=SIP/SWG-c80008c0
dstchannel=Local/92345...@callout-d564,1
lastapp=Dial
lastdata=Local/92345...@callout||g
duration=20
billsec=3
uniqueid=1248584732.888
calldate=2009-07-26 01:05:39
clid="PPI/SWG"
src=SWG
dst=92345678
dcontext=callout
channel=Local/92345...@callout-d564,2
dstchannel=Zap/24-1
lastapp=Dial
lastdata=Zap/G0/92345678|30|W
duration=10
billsec=0
uniqueid=1248584739.891
calldate=2009-07-26 01:05:52
clid="PPI/SWG"
src=SWG
dst=*
dcontext=cob-meetme-escape
channel=SIP/SWG-c80008c0
dstchannel=Local/91234...@callout-fc48,1
lastapp=Dial
lastdata=Local/91234...@callout||g
duration=37
billsec=20
uniqueid=1248584732.888
calldate=2009-07-26 01:06:00
clid="PPI/SWG"
src=SWG
dst=91234567
dcontext=callout
channel=Local/91234...@callout-fc48,2
dstchannel=Zap/23-1
lastapp=Dial
lastdata=Zap/G0/91234567|30|W
duration=9
billsec=0
uniqueid=1248584760.896
My dialplan rules look like this; I come in to cob-meetme from an
extension earlier in the plan, with MY_ACCOUNTCODE and COB_CONFNO
already set. The callout context places an outgoing call.
[cob-meetme]
exten => _XXX,1,Answer
exten => _XXX,n,Set(CDR(accountcode)=${MY_ACCOUNTCODE})
exten => _XXX,n,Set(MEETME_EXIT_CONTEXT=cob-meetme-escape)
exten => _XXX,n,MeetMe(${EXTEN},d1qMX)
exten => _XXX,n,Hangup
[cob-meetme-escape]
exten => *,1,Set(CDR(accountcode)=${MY_ACCOUNTCODE})
exten => *,n,Read(DEST,,0,,1,0)
exten => *,n,Set(DYNAMIC_FEATURES=cob-join#cob-nojoin)
exten => *,n,Dial(Local/${de...@callout,,g)
exten => *,n,Set(DYNAMIC_FEATURES=)
exten => *,n,Goto(cob-meetme,${COB_CONFNO},1)
[macro-cob-join]
exten => s,1,Set(CDR(accountcode)=${MY_ACCOUNTCODE})
exten => s,n,ChannelRedirect(${BRIDGEPEER},cob-meetme,${COB_CONFNO},2)
[macro-cob-nojoin]
exten => s,1,Set(CDR(accountcode)=${MY_ACCOUNTCODE})
exten => s,n,SoftHangup(${BRIDGEPEER})
I have these in features.conf and enabled in the dialplan:
cob-join => *33,self/both,Macro,cob-join
cob-nojoin => *34,self/both,Macro,cob-nojoin
Thanks!
---Scott.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reasons to use AEL
Philipp Kempgen writes: > Miguel Molina schrieb: >> Philipp Kempgen escribió: > >>> Use macros in AEL so you don't have to care about the underlying >>> implementation. :-) scnr > >> Right now for every implementation I made, I didn't have the need to >> program in AEL, only plain extensions, some AMI and AGI. But well, it >> seems to have a lot of advantages. Please tell me some, I may take a >> look to it too see if it's worth spending the time to learn and get the >> best out of it. > > I'd say control structures (and proper indentation) are one of the > most important reasons to use AEL (conditionals: if .. else, switch > .. case, ..., loops: for, while) because they look so familiar. > Imagine nested "control structures" in extensions.conf with Goto(), > GotoIf(), While(), EndWhile(), ExitWhile(), ContinueWhile() and > priorities - such code is not what I call maintainable. Can you recommend a good tutorial or book that covers AEL? I tend to use extensions.conf because most of the examples I come across use it, and it's covered by my first edition of the O'Reilly Asterisk book. Do later editions of the O'Reilly book cover AEL thoroughly? Thanks, Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Goto from a feature macro is not working?
Hello,
I'm trying to implement multi-party calls according to these
instructions:
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
They are almost working, except that the Goto at the end of
[dynamic-nway-start] doesn't seem to work. When I turn verbosity up a
bit, I get something like this in my error log:
== Channel 'SIP/SWG-0085a180' jumping out of macro 'nway-start'
and then the SIP line is hung up, with no further dialplan steps
logged for that line.
Writing a small test case to see what's going on, I get the same
behavior:
; extensions.conf
[macro-test1]
exten => s,1,Goto(macro-test1a,s,1)
[macro-test1a]
exten => s,1,NoOp
; features.conf
macro-test1 => *1,self/both,Macro,test1
When I activeate this feature with *1, I get:
-- Executing [...@macro-test1:1]
Goto("SIP/SWG-007f9280","macro-test1a|s|1") in new stack
-- Goto (macro-test1a,s,1)
== Channel 'SIP/SWG-007f9280' jumping out of macro 'test1'
then nothing else about this (though the line isn't disconnected).
The documentation for Macro on voip-info.org seems to say that using
Goto from a macro is fine:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Macro
This is with Asterisk 1.4.26.
Any idea why this is happening, or any ideas how to work around it?
What I'm really looking for is an attended transfer where all three
parties are on the line together at the end, then the attending
operator can drop off when everything is confirmed. Maybe there's an
easier way to do that?
Thanks!
Scott.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Waiting for a call to complete with AMI Originate
Matt Riddell writes:
> On 22/7/09 7:24 PM, Scott Gifford wrote:
[...]
>> In this case, I don't seem to have enough information to tell when the
>> call has failed and I should give up. I do get a Hangup event, but I
>> don't see a way to distinguish it from other hang-up events from other
>> calls.
>
> For doing fax broadcasting we use the UserEvent function.
>
> exten =>
> h,n,UserEvent(SmoothTorque|SmoothTorqueFax:${PHASEESTATUS}-${campaignid}-${phonenumber})
>
> Then in the back end we parse the results.
Thanks, that worked great! I didn't know about the UserEvent app, it
will be a very useful trick to have up my sleeve. :-)
Scott.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Waiting for a call to complete with AMI Originate
Hello,
I'm using an AMI Originate command to send a fax. The fax is sent by
a script, and I'd like my script to send the fax, wait until it has
succeeded or failed, then exit with an appropriate error code (it is
driven by a mail system, so the exit code will tell the mail system
whether to retry the fax later).
The script works great if the fax succeeds, or if the line is busy or
doesn't pick up. The problem I'm having is that when a fax is sent
and the line picks up but doesn't accept the fax (for example if I
call a voice line).
In this case, I don't seem to have enough information to tell when the
call has failed and I should give up. I do get a Hangup event, but I
don't see a way to distinguish it from other hang-up events from other
calls.
Here is an example of a recent fax I sent (the format of the request/
response lines is a dump of the variables in Perl, hopefully it makes
sense):
REQUEST: {
'MaxRetries' => 0,
'Channel' => 'Zap/g0/91234567,
'WaitTime' => 20,
'Action' => 'Originate',
'Application' => 'txfax',
'ActionID' => '1248244247.1814',
'Priority' => 1,
'Data' => '/home/sgifford/prog/faxscripts/testfax4.tif',
'Variable' => ''
};
RESPONSE: {
'Message' => 'Originate successfully queued',
'ActionID' => '1248244247.1814',
'Response' => 'Success'
};
EVENT: {
'CallerIDName' => '',
'Event' => 'Newchannel',
'Uniqueid' => '1248244247.11',
'Privilege' => 'call,all',
'Channel' => 'Zap/1-1',
'CallerIDNum' => '',
'State' => 'Rsrvd'
};
...
EVENT: {
'Event' => 'Hangup',
'Uniqueid' => '1248244250.12',
'Privilege' => 'call,all',
'Channel' => 'Zap/2-1',
'Cause-txt' => 'Unknown',
'Cause' => ''
};
...
EVENT: {
'Event' => 'Hangup',
'Uniqueid' => '1248244247.11',
'Privilege' => 'call,all',
'Channel' => 'Zap/1-1',
'Cause-txt' => 'Unknown',
'Cause' => ''
};
I see the same behavior in Asterisk 1.4.18 and 1.4.26.
Any suggestions?
Thanks,
Scott.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stop recording on SIP attended transfer
"Danny Nicholas" writes: > I don't know the full details, but I think if the Dial command(s) have the W > and/or w options on them, you can activate/deactivate recording via DTMF. Thanks, that's a good idea, I might be able to rig something up with that! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stop recording on SIP attended transfer
Hello, We have an application where operators will sometimes take an incoming call from a queue, then contact an outside line, do a consultation, and finally do a SIP attended transfer to join the two parties together. We'd like to record the incoming caller's conversation with the operator and the attended part of the outgoing call, but not the unattended part, after the transfer has completed and the incoming caller is talking on the outgoing line (this part of the call may be confidential). We start the recording with a MixMonitor command when the outgoing call is placed. However, I don't see anything in the dialplan that gets run when the SIP attended transfer happens, where I could issue a command to stop recording. Any suggestions? Thanks! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unique id used for call recording missing from CDR data for transferred call
Hello,
I have an application that needs to record outgoing calls. It's
running on Asterisk 1.4.18, with CDR data stored in MySQL.
Outgoing calls are recorded based on their uniqueid. When outgoing
calls are placed, there is a line like this on my extensions.conf:
exten => _.,n,MixMonitor(/var/spool/asterisk/monitor/${UNIQUEID}.gsm)
For regular outgoing calls, this works fine. The call is recorded in
a file named for its uniqueid, and if I need the recording I can pull
the information out of the CDR table, find the uniqueid, then pull up
the recording. We have a Web application that does this
automatically.
When an incoming is transfered to an outgoing call via an attended SIP
transfer, however, the uniqueid assigned to the outgoing call does not
seem to end up in the CDR table at all.
Here's what happens:
1. User A calls in to the queue on Zap/1-1 uniqueid 1247686911.203
2. Operator picks up call.
3. Queue application starts recording in 1247686911.203.gsm
4. Operator talks for awhile, decides to transfer call
5. Operator switches Line 2, calls User B on channel Zap/24-1 with
uniqueid 1247686911.205
6. Dialplan command MixMonitor(1247686911.205.gsm) starts recording
7. Operator talks to User B for awhile
8. Operator transfers call, connects User A to User B, and hangs up
This generates these 4 CDRs:
1. 2009-07-15 15:41:51
channel Zap/1-1
dstchannel SIP/DEV-0078ebd0
duration 175
uniqueid 1247686911.203
2. 2009-07-15 15:43:17
channel SIP/DEV-0078ebd0
dstchannel (none)
duration 89
uniqueid 1247686997.204
3. 2009-07-15 15:43:47
channel Zap/24-1
dstchannel Zap/1-1
duration 80
uniqueid 1247687027.206
4. 2009-07-15 15:43:47
channel SIP/DEV-c4018700
dstchannel Zap/24-1
duration 59
uniqueid 1247686911.203
The uniqueid for the outgoing call, 1247686911.205, doesn't end up in
any of the CDRs, so is effectively lost.
Any ideas on how to handle this? Is this something that's likely to
be fixed in a later version of Asterisk?
Thanks,
Scott.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API in PHP
Olivier writes: [...] > What is your opinion regarding PHP AMI API and Asterisk 1.6.1 ? > I'm referring here to http://code.google.com/p/asterisk-php-api/. In my experience, asterisk-php-api works OK, but it's a bit slow. It determines when Asterisk has finished sending its responses by waiting until it doesn't send anything for a few seconds, which means you wait those few seconds fairly often. It also doesn't do much to filter out AMI messages that are unrelated to your query. We modified it slightly to avoid the pause, and we filter out the AMI messages we don't care about in our code. But in the end, AMI is a very simple protocol, and you might find it easier to just speak it directly over the socket. Hope this is helpful, Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open source SIP client
DHAVAL INDRODIYA writes: > can anybody help me to give Opensource SIP client information which > can be modified as per our requirment Hello Dhaval, We have tried several open-source SIP phones on Linux. We have had the best luck with Twinkle Phone: http://www.xs4all.nl/~mfnboer/twinkle/index.html It has lots of hooks where you can stick your own scripts to modify its behavior. We also had pretty good luck with SFLphone: http://www.sflphone.org/ There is a list of open source clients on voip-info that includes these two. It might be a good starting point: http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software Good luck! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ignoring time spent waiting in queue in CDR
Miguel Molina writes: [...] > Why don't you just make your billing statistics from the queue log? > Assuming that you upload the queue log to a database, it would be very easy. I took a look at this option and it looks like it will work. Thanks to all who responded for your advice! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ignoring time spent waiting in queue in CDR
Hello, I'm working on an Asterisk configuration for a call center, and they bill based on the time spent talking to an agent, but not for any time spent waiting in a queue. The CDR information contains the entire duration of the call as billable seconds, including time spent waiting in the queue. I would like the billable seconds to only include the time spent actually talking to an agent. I am using Asterisk 1.4.18. The only way I have found so far is to correlate the CDRs with the "CONNECT" queue records, figure out the end time of the call by adding the CDR start time to the duration, then figure out the actual duration by subtracting the time of the queue "CONNECT" record. That seems messy and error-prone, and I'm hoping there's a better way. I also looked at using the ResetCDR() or ForkCDR() dialplan functions, but I don't see a way to cause code to run immediatly after the agent answers a call from the queue. Any suggestions? Am I missing some easy way of doing this? Thanks! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk queues sending calls to members on the phone
Hello, We're using Asterisk to manage call queues. Queue members are connected via IAX2 using the Zoiper softphone, and Zoiper is configured with 2 lines. We're finding that calls are routed to queue members even when they are on the phone, on their softphone's other line. For example, if a queue member makes an outgoing call on line 1 or is handling a queue call on line 1, the queue will often route calls to them on line 2. When this happens, "queue show" shows the member's status in the queue as "In use", so Asterisk seems to know that this member is busy. We are using Asterisk 1.4.18, configuring our IAX users and queues with the realtime extension. Is this expected behavior, and if so is there any way to turn it off? Or is this unusual behavior, and if so any tips on troubleshooting it? Does anybody happen to know how Asterisk determines whether a queue member is available to have a call routed their way? Thanks! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call recording problems from queue
"Ex Vito" <[EMAIL PROTECTED]> writes: > I don't have access to an asterisk system right now > (nor any other sort of information source) but I seem > to recall that from 1.4 onwards the config option for > recording queue calls is named differently... > > Is it mixmonitor ? Check you 1.4 queues.conf sample. > > PS: I'm not really sure about this one! Hi exvito, Mysteriously it started working today. Maybe Asterisk just needed a restart after playing with the configuration all day, I'll see if it keeps working. Thanks! ---Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call recording problems from queue
Hello, I'm trying to set up call recording for a queue. Right now the recording appears to work correctly, but when I call and chatter for a minute or so, at the end of the call I end up with a very small file (less than 100 bytes), which contains about .06 seconds of silence. If I talk for another minute, this file will get up to 200 bytes or so. In my queue configuration, I have: [testq] monitor-format = gsm monitor-type = MixMonitor ... I can see what looks like MixMonitor starting and stopping at the right time: -- IAX2/sgifford-3 answered Zap/1-1 == Begin MixMonitor Recording Zap/1-1 -- Hungup 'IAX2/sgifford-3' == Spawn extension (incoming, 3772, 3) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' == End MixMonitor Recording Zap/1-1 I have tried turning debugging up very high (like 50) and I still don't see any clues. I'm using Asterisk 1.4.18 built from source. The incoming lines are Zap on a Sangoma card. The queue members are IAX clients. The calls are being sent as GSM. Does anybody have an idea what could be going wrong, or where to look to debug this problem? Thanks! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_txfax
Hello, I'm setting up Asterisk to send outgoing faxes over a PRI line. I installed app_txfax and its prerequisites and astfax to submit email messages to Asterisk. This all seems to work fine, but I get some error messages in my logs I don't understand. Whenever I send a fax it goes through fine, but I get these messages in the logs: [Jan 17 11:21:07] WARNING[2413] chan_zap.c: Unable to request echo training on channel 1 [Jan 17 11:21:13] WARNING[2413] pbx.c: Zap/1-1 already has a call record?? [Jan 17 11:21:35] WARNING[2413] /home/sgifford/src/agx-ast-addons/app_txfax.c: Transmission loop error When I send a fax to a line that's busy, I get: [Jan 17 11:40:29] NOTICE[2439] pbx_spool.c: Call failed to go through, reason (0 ) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) while I would expect a simple BUSY. Also, app_txfax will retry the fax a few times before giving up. I'd like to know when it gives up, so I can let the fax sender know that it didn't go through. Is there a way to hook into that? I'm using Asterisk 1.4.17 with spandsp 0.0.4, tx_fax from agx-ast-addons 1.4.3, and astfax 1.0. It's running on Debian Sarge (4.0) with Debian-supplied kernel 2.6.18. I'm using a Digium TE110P card with zaptel driver 1.4.7.1. Thanks for any ideas! Scott. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo problem
"Steve Davies" <[EMAIL PROTECTED]> writes: > "Scott Gifford" <[EMAIL PROTECTED]> writes: [...] > 1.5 to 2 seconds. That is a HUGE delay. echo delay is normally > measured in tens or perhaps hundreds of milliseconds, and you are > unlikely to find a software EC that can deal with a 1.5 to 2 second > delay! > > This sounds as if there is something very broken in the voice network, > causing huge amounts of delay. As suggested above, check the > intermediate switch. What's interesting is the lines come in via 2 PRI lines, and most calls go out via analog lines to people's desks and a voicemail system. These lines all work fine. So the problem likely isn't in the PSTN and isn't an inherent flaw with the switch, though it could be the T1 card connected to our Asterisk server or its configuration. It seems the problem is either on the Tadiran switch or the Asterisk server. Unfortunately we don't have a good way to determine which, since we don't have another switch to try, or another device to replace the Digium server. > [snip] > >> > >> > We have done loopback tests with the Digium card with a loop plug in >> > it. > > What were the results? Oh, sorry, I should have said: These tests were successful. Scott. ___ --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] Echo problem
pixiesfr <[EMAIL PROTECTED]> writes: > Hi, > > Did you try to increase echotraining ?? > echo training = 800 .. Yes, I tried 800, 1200, and 2000; none seemed to make any difference. Thanks! ---Scott. ___ --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] Echo problem
Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds; none of these adjustments made a difference, except adjusting gain made the echo quieter. We followed these instructions in trying to eliminate echo: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html Our lines come in from the telco in a PRI, then connect to a Tadiran switch which hands the lines off to Asterisk over a T1 card. We have done loopback tests with the Digium card with a loop plug in it. We're a bit stumped as to what to try next. Any suggestions or advice? Thanks ---Scott. ___ --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] Using Asterisk/Digium card with Tadiran switch
Hello, We've got an Asterisk server with a Digium TE110P card, connected to a Tadiran Coral Flexicom IPX 500 switch using a T1 card. We are having echo problems on the lines coming in from the Digium card. I was wondering if anybody is successfully using a Digium card and Asterisk with a Tadiran switch, and if so whether they could share some configuration information? Thanks! Scott. ___ --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] Help with JIAXClient
"Enrique Sanchez" <[EMAIL PROTECTED]> writes: > I'm trying to make a little example program for register to an > Asterisk PBX and dial a softphone, but i just can't register to the > PBX. Buenos noches Enrique, I've been trying to get JIAXClient working for over a month with no success. It fails mysteriously at different places with no exceptions, and my efforts to figure out why have so far failed. So I can't offer helpful advice, only sympathy. :) Does this work for other people? If so, what's your build environment like? ---Scott. ___ --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] JIAX status
"Rubens Zupelli Filho" <[EMAIL PROTECTED]> writes: > Scott, > > Could you point me some step-by-step instructions? I hadn't figure out > what I'm doing wrong. I started over several times and did not find > where I lost it. I just did configure and make, then fixed all the problems that came up. I don't have any step-by-step instructions, and I don't have my small fixes into any kind of publishable form. Scott. ___ --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] JIAX status
"Rubens Zupelli Filho" <[EMAIL PROTECTED]> writes: > You are compiling in Linux or Windows? Both. It works on Linux, but not yet on Windows. > The package the java compiler is not founding is: > > net.sourceforge.iaxclient.jni That's part of the source package; probably the classpath just needs to be tweaked. ---Scott. ___ --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] JIAX status
"Rubens Zupelli Filho" <[EMAIL PROTECTED]> writes: > Anyone knows the current status of JIAXclient? I have been playing with jiaxclient 0.0.6, and it seems to mostly work if you have a working copy of the C iaxclient library. I would test iaxclient with the command-line tools that come with it and make sure that all works satisfactorily before moving on to JIAXClient, since the Java code is built on top of the C library. > I tried to recompile the sources available in sourceforge but > they reference a old java package that I was not able to find. What package? Scott. ___ --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] Announcement Haiku
This extremely useful dialplan requires the standard Asterisk sounds, plus the additional ones in the asterisk-sounds package. Scott. [haiku] exten => s,1,Playback(privacy-please-dial) exten => s,n,Playback(letters/a) exten => s,n,Playback(high) exten => s,n,Playback(letters/q) exten => s,n,WaitExten(60) exten => 01,1,Playback(a-collect-charge-of) exten => 01,n,Wait(1) exten => 01,n,Playback(attention-required) exten => 01,n,Wait(1) exten => 01,n,Playback(shall-i-try-again) exten => 01,n,Wait(3) exten => 01,n,Goto(s,1) exten => 02,1,Playback(at) exten => 02,n,Playback(midnight-tonight) exten => 02,n,Wait(1) exten => 02,n,Playback(doing-enum-lookup) exten => 02,n,Wait(1) exten => 02,n,Playback(do-not-disturb) exten => 02,n,Wait(3) exten => 02,n,Goto(s,1) exten => 03,1,Playback(message-from) exten => 03,n,Playback(detroit) exten => 03,n,Wait(1) exten => 03,n,Playback(chicago) exten => 03,n,Playback(houston) exten => 03,n,Playback(letters/l) exten => 03,n,Playback(letters/a) exten => 03,n,Wait(1) exten => 03,n,Playback(for-english-press) exten => 03,n,Playback(plugh) exten => 03,n,Wait(3) exten => 03,n,Goto(s,1) ___ --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] Can I recreate a Fax from a recorded file?
Colin Anderson <[EMAIL PROTECTED]> writes: >>Why is this hard to fake at all? You send a different fax to your >>system, and replace the Asterisk audio file with the one from the >>altered fax. Additionally, the client has no realistic way of >>verifying the correctness of your audio-to-fax translation tool; it >>could just as easily output a TIFF file completely different from the >>one that was actually faxed. > > That's interesting, I hadn't thought of it that way. I was thinking in terms > of subtly modifying the original audio stream not outright replacing the > recording and faking the datestamp! Given that, essentially recording the > audio is the *same* as retaining the TIFF in terms of integrity > vulnerability. > > How about this: (theoretical of course) > > 1. Fax comes in > 2. Audio is recorded > 3. A checksum of the audio is generated then relayed somehow to a seperate, > secure system > 4. In the event of a dispute, the checksum is retrieved, compared with the > original audio file, then the original audio is "replayed" and the fax is > regenerated. I don't see the advantage to this; the client still has to trust that all of this is done correctly, and if they don't trust the fax recipient to put the correct fax in the paper file or keep the correct TIFF, why would they trust them to do this? Using a third party to receive and relay the fax, one which is trusted by both the client and the fax recipient, would solve the problem; the third party could create a document with the caller information (ideally from ANI, which is harder to forge), the time, and the message itself, then digitally sign it. This might even be an interesting business plan, for some applications where confirmed document transmittal is important. But it's hard for me to imagine this isn't overkill; if a client and a service provider distrust each other so thoroughly that they have to communicate through a third party to verify integrity, probably they just shouldn't do business with each other. Scott. ___ --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] Can I recreate a Fax from a recorded file?
Colin Anderson <[EMAIL PROTECTED]> writes: >>Why not capture the faxes (in or out) in tiff format, instead of audio >>format? Setup your asterisk box to relay faxes! > > I think in this case the impact on the client would be much greater if you > can show them a recreation of the image from the raw data; you could always > claim that a TIFF file was altered (which it can be, trivially) but it's > pretty much impossible to change the raw audio to your ends unless you are > in a Tom Clancy novel. Why is this hard to fake at all? You send a different fax to your system, and replace the Asterisk audio file with the one from the altered fax. Additionally, the client has no realistic way of verifying the correctness of your audio-to-fax translation tool; it could just as easily output a TIFF file completely different from the one that was actually faxed. Scott. ___ --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 applications when a queued call isanswered
"Alexander Lopez" <[EMAIL PROTECTED]> writes:
>>From the queues.com file.
[...]
> ;announce = queue-markq
>
>
> This allows you to have one announcement per queue, You could use a
> script to change the file, but it does not lock the file so if you
> change it while it is being read the other person (agent) hears the
> change
Yup, I saw this.
> Do you want a dynamic message played to each agent before he picks up??
Yes. I'd like to do something like:
Ringing()
SendURL(http://example.com/${EXTEN}.html)
SayDigits(${EXTEN})
Wait(5)
That's close to what you suggest, but Asterisk on its own announces
first then sends the URL with no wait, so the agent is left scrambling
to see who the call is for as the Web page and the call come at the
same time. Also SayDigits() sounds much nicer than what I can get
with Festival creating a WAV file.
> That can also be done, but it is not as simple.
If you could point me in the right direction, that would be very
helpful; I don't expect any handholding, but I'm not seeing how to get
ahold of the channel to the agent after they've answered but before
the caller is transferred there.
Thanks! By the way, your first message did answer my original
question, but I've learned more about Asterisk in the meantime and my
expectations have gone up. :-)
Scott.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running applications when a queued call is answered
"Alexander Lopez" <[EMAIL PROTECTED]> writes:
> Use the Local channel and add the agents using that IE:
>
> Member Local/[EMAIL PROTECTED]
Thanks Alexander,
That works, but it's backwards. That is, when I give a command like:
SayDigits(${EXTEN})
it says it to the caller, not the agent. Is there a way to make this
work the other way around, so I can ring the agent's phone, send
things to the agent's phone (like announcements or URLs), then when
I'm done connect the agent to the caller?
Thanks!
Scott.
> Snip
>
>> Hello,
>>
>> I'm experimenting with Asterisk for possible use in a call center.
>> I'm trying to figure out how to run applications when an agent answers
>> a call in the queue.
[...]
___
--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] SendURL
Jean-Denis Girard <[EMAIL PROTECTED]> writes: > Scott Gifford a écrit : [...] >> * Does anybody know of a softphone that works with Asterisk's >> SendURL command? Cross-platform would be nice, open source ideal. > > May I suggest MozIAX: it's a Mozilla / Firefox extension, so it does > natively support receiving URL from Asterisk. It also adds "tel:" > protocol to Firefox, so you can call from the web page. MozIAX also > supports receiving / sending text messages to / from Asterisk, for > chat sessions. It is open source, runs on windows and linux, and was > reported to work on OS X. More info at: http://moziax.mozdev.org/. Thanks Jean-Denis, that works great! Scott. ___ --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] Running applications when a queued call is answered
Hello,
I'm experimenting with Asterisk for possible use in a call center.
I'm trying to figure out how to run applications when an agent answers
a call in the queue. I see that the queue itself supports a very
limited range of applications; for example, I can give a URL to the
Queue() application to SendURL(), or an announcement to read to the
agent. I'd like to do some slightly more sophisticated things, like
run an external application with System().
When I was using normal extensions and routing the call to one person,
I could do something like this:
exten => 3772,1,Ringing()
exten => 3772,2,System(/home/sgifford/ircsay sgifford "Call for ${EXTEN} at
${DATETIME}")
exten => 3772,3,Wait(2)
exten => 3772,4,Dial(SIP/sgifford)
to run an external application and wait 2 seconds while the caller
still heard ringing. Is there a way to do something similar when a
queued call is delivered? Maybe with AGI?
I've seen some recommendations to tail the logfile, but that seems
kludgey...
I'm currently using the "1.0.7-BRIstuffed-0.2.0-RC7k" Asterisk package
included with Debian 3.1 (Sarge), but I'd be happy to upgrade to a
newer version if that would help.
Thanks for any tips or ideas!
Scott.
___
--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] SendURL
Hello, I just started working with Asterisk about a month ago, and so far I've had great luck with it! Things I expected to be hard were easy, and it's easy to customize. Thanks! I'm trying to send a URL with a queue; that is, when an agent picks up the phone, I'd like a particular URL to be displayed on the agent's screen, depending on the queue or the dialed number (DNIS). The Queue() application supports this via a URL parameter, which is exactly what I want. But I can't seem to find a client that will do anything with the URL. I tried creating an extension that just uses SendURL, and nothing seemed to work there, either. I'm testing on Linux; the actual application will probably run on Windows initially, then hopefully move to Linux over the next few months, so something cross-platform would be ideal. So, a few questions on this: * Does Asterisk support SendURL over SIP, or only over IAX? Is there support in the SIP protocol for sending URLs or similar? * Does anybody know of a softphone that works with Asterisk's SendURL command? Cross-platform would be nice, open source ideal. * If not, can anybody recommend a good open-source softphone that I could add URL support to? Thanks! Scott. ___ --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
