[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' = 'unknown',
'Event' = 'Newchannel',
'Uniqueid' = '1248244247.11',
'Privilege' = 'call,all',
'Channel' = 'Zap/1-1',
'CallerIDNum' = 'unknown',
'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.
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Re: [asterisk-users] voicemail does not work from local calls!!!
Hi again,
I figured out the problem.
My dialplans were as follows..
8XXX are my asterisk number subnet..
other 3 numbers are my local numbers that works on an ericsson which is
connected by e1.
[local]
exten = _8XXX,1,Dial(SIP/${EXTEN})
exten = 1234,1,Dial(SIP/1234)
exten = 2345,1,Dial(SIP/2345)
exten = 3456,1,Dial(SIP/3456)
[everythingelse]
exten = _,1,Dial(DAHDI/g1/${EXTEN})
I changed the plan as follows; But still i wonder if there is any other
solution than i did here..
exten = _8XXX,1,Goto(default,${EXTEN:0},1)
exten = 1234,1,Goto(default,${EXTEN:0},1)
exten = 2345,1,Goto(default,${EXTEN:0},1)
exten = 3456,1,Goto(default,${EXTEN:0},1)
now by default it goes to macro-stdexten but still i need to add all
numbers out of asterisk subnet one by one.
Hello, i was trying to add call forwarding, lastcaller etc for my
asterisk 1.6.0.9+asterisk-gui
So i added some lines to my macro-stdexten.
Now, if i got a call from trunk, everything seems working well.
But if i get an inside call from asterisk clients, voicemail does not
work.
Here is the changes i made on macro-stdexten.
PS: As i notice now lastcaller does not work either..
exten = s,1,Set(__DYNAMIC_FEATURES=${FEATURES})
exten = s,2,Set(DB(lastcaller/${ARG1})=${CALLERID(num)}) ;added for
lastcaller
exten = s,3,GotoIf($[${FOLLOWME_${ARG1}} = 1]?8)
exten = s,4,Set(temp=${DB(CFIM/${ARG1})}) ;added for callforward
exten = s,5,GotoIf($[${temp} != ]?1000:6) ;added for call forward
exten = s,6,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,7,Goto(s-${DIALSTATUS},1)
exten = s,8,Macro(stdexten-followme,${ARG1},${ARG2})
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
exten = s,1000,Goto(DLPN_sadecelokal,${temp},1) ;added for call forward
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[asterisk-users] sip configuration masking the peers
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Re: [asterisk-users] how to use patgen and pattest for PRI card?
hello: I can set a environments to test the two pri cards. the patlooptest is ok. the result has no problem. how do i use patgen and pattest with two pri cards and the setting of zaptel/sys.conf? in the http://docs.tzafrir.org.il/man/pattest.8.html, there are no setting files and cablling for the two sides. please explan the settings and cabliing. thanks! Chris On Wed, Jul 22, 2009 at 1:10 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Jul 21, 2009 at 12:01:18PM +0800, Chris YM wrote: hello: I wan to use the test tools-patgen and pattest for pri cards. according to Tzafrir Cohen from http://docs.tzafrir.org.il/man/pattest.8.html, i still does not know how to use that. do i need to connect two pri cards with two servers, or use a cable to connect two cards in one server? please give me a more details in term of cables and configurations. the pat* and hdlc* test tools open dahdi channels on their own. Thus the dahdi channels that they use must not be used by Asterisk at the time. The spans need to be created (drivers loaded) and configured (ztcfg / dahdi_cfg run). Run patgen on one side with one specific b-channel and pattest on the corresponding b-channe on the other side. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip configuration masking the peers
Hi all,
I need to specify two groups of peers who are on two sub networks, the
case is as follows:
two groups of users (that are supposed to use the X-lite) group1 and group2,
each group is on a sub network net1, and net2, respectively, each group has
its own dial plan defined in the extension.conf,
we have defined the peers in the sip.conf for both groups, and successfully
made a call between two peers from the groups, but the idea is we need to
prevent users from network1 to register as peers of group1,
I suppose this would be a configuration solution, but I am afraid that do
know what are the right needed configurations:
here is definition of two peers each from different group:
[1010]
type=friend
host=dynamic
context=group1
secret=pass
host=dynamic
callerid=TestAccount1010
vm Extension=test 1010
mailbox=1...@default
nat=yes
[2003]
type=friend
context=group2
secret=pass
host=dynamic
callerid=Account2003
vm Extension=test 2003
mailbox=2...@default
nat=yes
each of group1 and group2 context are defined in the extension configuration
as follows :
exten = _2XXX,1,Dial(SIP/${EXTEN})
exten = _2XXX,n,Playback(unavailable)
exten = _2XXX,n,Hangup()
exten = _1XXX,1,Dial(SIP/${EXTEN})
exten = _1XXX,n,Playback(unavailable)
exten = _1XXX,n,Hangup()
in order the both groups can talk to each other,
currentlly users in network1 can register as peer 2003 which is supposed to
be allowed just for users from network2 , although this registration is
supposed to be failed, any suggestions plz!!
hope I made the scenario clear , any help would appreciated.
Thanks in advance.
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Re: [asterisk-users] AGI to announce temperature from weather.com XML file
It appears I opened some flood gates when I offered my 'alternative' version. So, rather than send hundreds of e-mails out - here's the link : http://www.dv-ip.com//downloads/files/misc/weather.txt Any questions - just 'yell'. Andrew Thomas Technical Services Manager a...@datavox.co.uk DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen Sent: 17 July 2009 18:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] AGI to announce temperature from weather.com XML file Trevor Hammonds wrote: I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. For a non-Asterisk project, I have a cron job that periodically pulls down an XML file from weather.com containing local weather data (TWC's user agreement requires that data be cached locally). Using sed, I also create a text file that contains only the numeric value of the current temperature, created from that XML file (e.g. tmp65/tmp in the XML file becomes a text file with 65 as its only contents). I am hoping someone on the list has an example of a lightweight AGI script that I may modify to either read the simple text file and set a dialplan variable to the current temperature, or hopefully a more-sophisticated one which will parse the XML file to set the dialplan variable. The end goal is to have Asterisk play the speech files temperature sixty five degrees or the equivalent non-English files per the channel's current language setting. Thank you. Any assistance will be greatly appreciated. Since your problem came up on the VoIP Users Conference today, it ended up being the basis for a blog post I wrote today. The blog post (which may solve your problem) is available here: http://leifmadsen.wordpress.com/2009/07/17/howto-read-a-value-from-a-fil e-and-say-it-back/ Let me know if that works for you -- just respond on the comments section since I don't always check this users list. Note: I haven't actually tested the dialplan yet, so if someone can test it for errors, let me know if you run into any, and I'll update the blog post with any that may be found. Thanks! Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- 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 ___ -- 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
On 22/7/09 7:24 PM, Scott Gifford wrote:
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.
For doing fax broadcasting we use the UserEvent function.
exten =
h,n,UserEvent(SmoothTorque|SmoothTorqueFax:${PHASEESTATUS}-${campaignid}-${phonenumber})
Then in the back end we parse the results.
--
Cheers,
Matt Riddell
Director
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Re: [asterisk-users] sip configuration masking the peers
'host=dynamic' is your problem - as this allows any IP address to register as
that friend - assuming they know the password/username combination.
Why not simply have group 1 as 'secret=pass123' and group2 as 'secret=pass456'?
Just don't tell group 1 uses the password for group 2 - and vice-versa!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of peace keeper
Sent: 22 July 2009 09:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip configuration masking the peers
Hi all,
I need to specify two groups of peers who are on two sub networks, the
case is as follows:
two groups of users (that are supposed to use the X-lite) group1 and group2,
each group is on a sub network net1, and net2, respectively, each group has
its own dial plan defined in the extension.conf,
we have defined the peers in the sip.conf for both groups, and successfully
made a call between two peers from the groups, but the idea is we need to
prevent users from network1 to register as peers of group1,
I suppose this would be a configuration solution, but I am afraid that do know
what are the right needed configurations:
here is definition of two peers each from different group:
[1010]
type=friend
host=dynamic
context=group1
secret=pass
host=dynamic
callerid=TestAccount1010
vm Extension=test 1010
mailbox=1...@default
nat=yes
[2003]
type=friend
context=group2
secret=pass
host=dynamic
callerid=Account2003
vm Extension=test 2003
mailbox=2...@default
nat=yes
each of group1 and group2 context are defined in the extension configuration as
follows :
exten = _2XXX,1,Dial(SIP/${EXTEN})
exten = _2XXX,n,Playback(unavailable)
exten = _2XXX,n,Hangup()
exten = _1XXX,1,Dial(SIP/${EXTEN})
exten = _1XXX,n,Playback(unavailable)
exten = _1XXX,n,Hangup()
in order the both groups can talk to each other,
currentlly users in network1 can register as peer 2003 which is supposed to be
allowed just for users from network2 , although this registration is supposed
to be failed, any suggestions plz!!
hope I made the scenario clear , any help would appreciated.
Thanks in advance.
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Re: [asterisk-users] Waiting for a call to complete with AMI Originate
Scott Gifford schrieb: 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): 'Application' = 'txfax', Any suggestions? Probably not what you are looking for but you could use Iaxmodem + HylaFax. Alternatively have a look at the SendFax() application in Asterisk 1.6. -- -= Info about application 'SendFAX' =- [Synopsis] Send a FAX [Description] SendFAX(filename[|options]): Send a given TIFF file to the channel as a FAX. ... This application sets the following channel variables upon completion: FAXSTATUS - status of operation: SUCCESS | FAILED FAXERROR- Error when FAILED -- Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] Callin Numbers.
Hello, I lookin' for a call in number from UK or USA. Can somebody offers me a peering for this or specify any sip provider that offers this thing? Thank you very much, Jonson. ___ -- 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] Callin Numbers.
On Wednesday, July 22, 2009, Catalin S. wrote: I lookin' for a call in number from UK or USA. Can somebody offers me a peering for this or specify any sip provider that offers this thing? There are several providers who offer UK or US regional geographical numbers for little or no cost if you only use them inbound. For example, I have UK geographicals from Sipgate (http://www.sipgate.co.uk/user/index.php) and VoipCheap (http://www.voipcheap.com/en/index.html - *not voipcheap.co.uk*). The latter, I had to install their client to a Windows host and inspect the configuration to obtain the info necessary to connect my Asterisk server. However, those are only examples and there are a lot more to be found if you look around. HTH, -- Geoff ___ -- 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] Inquiry abount Asterisk extensions.conf
Curious - Why? What is the peer switch and why does it have this requirement? John Novack hadi motamedi wrote: Dear All Can you please let us know how we can modify our Asterisk extensions.conf file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as 665 so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one 665 packet . Your reply is very welcome Regards H.Motamedi ___ -- 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 -- Dog is my co-pilot ___ -- 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] Inquiry abount Asterisk extensions.conf
John Novack wrote: Can you please let us know how we can modify our Asterisk extensions.conf file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as 665 so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one 665 packet . Curious - Why? What is the peer switch and why does it have this requirement? That's a funny way of answering the question :) I *think* what he wants is overlap dialing. Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- 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] german voiceprompts
Hello ! Are there any plans at Digium to include also german voice prompts ? Thanks regards Hans ___ -- 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] ExecIf and empty variables (early evaluation)
Imagine that you have this code:
exten = _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an
argument: queuename
The obvious solution:
exten = _X!,n,ExecIf($[${QueueName} !=
]?Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
However, this doesn't actually work! Functions and variables on the
right hand side are evaluated BEFORE it is decided whether it needs to
be executed at all!
This is fairly harmless in this case, in that it just causes a warning.
However, what about this case?
exten = _X!,n,ExecIf($[${bar} 10]?Set(foo=${INC(bar)}))
Which you can argue that this code is in poor taste, it is definitely
surprising that INC is evaluated in this case, changing ${bar} even if
${bar} = 10.
It probably isn't possible to do something about this, but now you have
all been warned... This could be a good reason for avoiding side effects
in functions, and thereby banning ${INC()}
/Benny
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Re: [asterisk-users] ExecIf and empty variables (early evaluation)
You should submit this as a bug. It may or may not get fixed, but it
definitely won't until someone reports it or takes it upon themselves to fix
it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Benny Amorsen
Sent: Wednesday, July 22, 2009 7:44 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ExecIf and empty variables (early evaluation)
Imagine that you have this code:
exten = _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an
argument: queuename
The obvious solution:
exten = _X!,n,ExecIf($[${QueueName} !=
]?Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
However, this doesn't actually work! Functions and variables on the
right hand side are evaluated BEFORE it is decided whether it needs to
be executed at all!
This is fairly harmless in this case, in that it just causes a warning.
However, what about this case?
exten = _X!,n,ExecIf($[${bar} 10]?Set(foo=${INC(bar)}))
Which you can argue that this code is in poor taste, it is definitely
surprising that INC is evaluated in this case, changing ${bar} even if
${bar} = 10.
It probably isn't possible to do something about this, but now you have
all been warned... This could be a good reason for avoiding side effects
in functions, and thereby banning ${INC()}
/Benny
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Re: [asterisk-users] Latest chan_mobile
Carlos Ruiz Diaz wrote: @Steve: I considered the hardware separation between servers but when I exposed the idea it was immediately discarded because it is mandatory to have all in a box. Well, I'll start the migration then. Thank you. I doubt this helps anyone, but today I built the newest stable kernel (2.6.30.2) and the latest bluez libs (bluez-4.46) and obviously rebuilt dahdi and asterisk-addons. Without any config changes chan_mobile is working for incoming calls, picking up the handset is answeing the calls, and there is 2 way audio (which wasn't working before). Oddly when a call finishes, the mobile disconnects for a while and then reconnects again and there is terrible audio with outgoing calls, (scratchy and with a few seconds delay). This is definite progress (and doesn't require a separate box). This is all with a Cambridge Silicon Radio USB2 dongle and a nokia e61. ___ -- 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] Latest chan_mobile
That is exactly what happens to me. Still looking for a solution. On Wed, Jul 22, 2009 at 9:44 AM, Thomas Kenyon dig...@sanguinarius.co.ukwrote: Carlos Ruiz Diaz wrote: @Steve: I considered the hardware separation between servers but when I exposed the idea it was immediately discarded because it is mandatory to have all in a box. Well, I'll start the migration then. Thank you. I doubt this helps anyone, but today I built the newest stable kernel (2.6.30.2) and the latest bluez libs (bluez-4.46) and obviously rebuilt dahdi and asterisk-addons. Without any config changes chan_mobile is working for incoming calls, picking up the handset is answeing the calls, and there is 2 way audio (which wasn't working before). Oddly when a call finishes, the mobile disconnects for a while and then reconnects again and there is terrible audio with outgoing calls, (scratchy and with a few seconds delay). This is definite progress (and doesn't require a separate box). This is all with a Cambridge Silicon Radio USB2 dongle and a nokia e61. ___ -- 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 ___ -- 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] CallerPres SIP headers Analog Phone
hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this? ___ -- 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] Latest chan_mobile
Carlos Ruiz Diaz wrote: That is exactly what happens to me. Still looking for a solution. Well, it's a step forward from what I was getting before. Have you tried with different USB adapters and handsets? ___ -- 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] Latest chan_mobile
Yes, I tried with: Dell Computer Corp. Wireless 355 Bluetooth, built-in Encore, USB adapter. Always with: Nokia N80 Kernel: 2.6.27.21-0.1-pae. On Wed, Jul 22, 2009 at 10:46 AM, Thomas Kenyon dig...@sanguinarius.co.ukwrote: Carlos Ruiz Diaz wrote: That is exactly what happens to me. Still looking for a solution. Well, it's a step forward from what I was getting before. Have you tried with different USB adapters and handsets? ___ -- 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 ___ -- 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] A reason TO run Asterisk as root
I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. Thus, you may get some warning when everything non-root starts failing and give you a chance to free up some space before Asterisk is affected. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ExecIf and empty variables (early evaluation)
Benny Amorsen schrieb:
Imagine that you have this code:
exten = _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
If ${QueueName} happens to be unset, this will cause a warning:
[Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187
queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an
argument: queuename
The obvious solution:
exten = _X!,n,ExecIf($[${QueueName} !=
]?Set(foo=${QUEUE_WAITING_COUNT(${QueueName})}))
However, this doesn't actually work! Functions and variables on the
right hand side are evaluated BEFORE it is decided whether it needs to
be executed at all!
This is fairly harmless in this case, in that it just causes a warning.
You could split it up into multiple statements:
if (${QueueName} != ) {
Set(foo=${QUEUE_WAITING_COUNT(${QueueName})});
} else {
Set(foo=-1); // or whatever
}
(don't remember how to write that in extensions.conf format)
Pros:
- conditional evaluation
- more readable (ExecIf() looks ugly)
Cons:
- more statements
- less readable then a ternary conditional expression in real
programming languages:
foo = ($queuename != ? queue_waiting_count($queuename) : -1)
However, what about this case?
exten = _X!,n,ExecIf($[${bar} 10]?Set(foo=${INC(bar)}))
Which you can argue that this code is in poor taste, it is definitely
surprising that INC is evaluated in this case, changing ${bar} even if
${bar} = 10.
It probably isn't possible to do something about this, but now you have
all been warned... This could be a good reason for avoiding side effects
in functions, and thereby banning ${INC()}
Ban ExecIf(). Use AEL. Use if(){} blocks. :-)
In order to use control structures like if .. else/switch .. case
it's almost necessary to write your dialplan in AEL because the same
thing is so incredibly hard to read and write in extensions.conf
format (GotoIf()).
Although very basic dialplans look good in extensions.conf my
suggestion is to use AEL and let the AEL compiler figure out how to
translate that into an Asterisk dialplan.
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
--
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Re: [asterisk-users] CallerPres SIP headers Analog Phone
Ketema Harris schrieb: hello all...I have been trying to get a handle on CallerPres with an analog handset. I have usecallingpres=yes in my chan_dahdi.conf file and when I dial *67 on my analog handset I see Disabling Caller*ID on DAHDI/4-1 but when the call is then forwarded to my outbound SIP provider the RPID header is not correct privacy=off;screen=no instead of full and yes how can I correct this? I don't know if/how Asterisk handles/stores CLIR for analog handsets but SetCallerPres(prohib_passed_screen) does the trick when dialing to a SIP channel. Remote-Party-ID: ...;privacy:full;screen:yes You could add a *67 extension to your dialplan and store the CLIR state in AstDB for example. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] A reason TO run Asterisk as root
On Wed, Jul 22, 2009 at 11:31 AM, Steve Edwardsasterisk@sedwards.com wrote: I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. Hehe, sounds like a reason to standardize on ReiserFS ___ -- 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] A reason TO run Asterisk as root
tune2fs -m 0 [device] :) not anymore ;p David Backeberg wrote: On Wed, Jul 22, 2009 at 11:31 AM, Steve Edwardsasterisk@sedwards.com wrote: I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. Hehe, sounds like a reason to standardize on ReiserFS ___ -- 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 -- *Singer X.J. Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: w...@pythian.com MSN:pythianw...@hotmail.com Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] OT - Do analog gateways detect a phone is plugged in or out ?
Hi, I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send qualify queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Regards ___ -- 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] A reason TO run Asterisk as root
2009/7/22 Steve Edwards asterisk@sedwards.com I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. Do you imply this default can (and should) be changed ? Is it the same for other filesystems ? Thus, you may get some warning when everything non-root starts failing and give you a chance to free up some space before Asterisk is affected. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astmanproxy?
2009/7/21 James Green james.gr...@mjog.com Hi, We currently fire multiple HTTP requests (via multi-curl) to the AJAM interface in order to place calls. We are finding Asterisk has it's limits however, and I've found astmanproxy recommended for helping maintain the connections. This would prove particularly useful with multiple servers of course. However, in testing astmanproxy crashes with buffer overflows. This leads to the inevitable question: Is astmanproxy still recommended for use or are we missing some knowledge here? Platform is 64 bit Intel, Ubuntu 9.04, Asterisk 1.6 with latest trunk of astmanproxy from github. We successfully used astmanproxy here with 1.6.1.1 (intel 32 bits ). Thanks, James No virus found in this outgoing message. Checked by AVG - www.avg.com Version: 8.5.392 / Virus Database: 270.13.21/2252 - Release Date: 07/21/09 05:58:00 ___ -- 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 ___ -- 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] A reason TO run Asterisk as root
On Wed, Jul 22, 2009 at 10:31 AM, Steve Edwardsasterisk@sedwards.com wrote: I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. Thus, you may get some warning when everything non-root starts failing and give you a chance to free up some space before Asterisk is affected. Couldn't you get the same effect using quotas? Also, using separate partitions for various parts of the filesystem is a nice addition. Having your /var/log somewhere besides the same partition as / helps keep runaway logs at bay, just as an example. -jonathan ___ -- 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] OT - Do analog gateways detect a phone is plugged in or out ?
Olivier wrote: Hi, I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send qualify queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Unless there is a call in progress the switch has no idea what phones might be plugged or unplugged. Nothing happens on the line what it could detect. Steve ___ -- 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] A reason TO run Asterisk as root
On Wed, 22 Jul 2009, Olivier wrote: 2009/7/22 Steve Edwards asterisk@sedwards.com I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. Do you imply this default can (and should) be changed ? Is it the same for other filesystems ? No - I think you are all getting his intention wrong. He is saying that it is a GOOD thing, and that you get a warning before the disk fills and processes start crashing. If you run asterisk as 'asterisk', then this holdover percentage (I actually thought the default was 10%) is not accessible by the asterisk process, and once the filesystem hits 100% the process might crash. So to rephrase it: One GOOD reason to run asterisk as root is that you get to take advantage of the default filesystem overflow space reserved for root. j Thus, you may get some warning when everything non-root starts failing and give you a chance to free up some space before Asterisk is affected. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astmanproxy?
On Tue, Jul 21, 2009 at 10:15 AM, James Green james.gr...@mjog.com wrote: Hi, We currently fire multiple HTTP requests (via multi-curl) to the AJAM interface in order to place calls. We are finding Asterisk has it's limits however, and I've found astmanproxy recommended for helping maintain the connections. This would prove particularly useful with multiple servers of course. However, in testing astmanproxy crashes with buffer overflows. This leads to the inevitable question: Is astmanproxy still recommended for use or are we missing some knowledge here? Platform is 64 bit Intel, Ubuntu 9.04, Asterisk 1.6 with latest trunk of astmanproxy from github. Thanks, James When faced with this same problem, creating and FTPing .call files to the outgoing spool directory freed up the AMI for other functions. Plain, simple, and Just Worked I looked at, but never tried AstManProxy because I prefer to eliminate levels of complexity and points of failure rather than add, whenever possible. Not saying that this is your best solution, just what I found to be much more reliable than pounding the AMI. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] A reason TO run Asterisk as root
Jeff LaCoursiere schrieb: 2009/7/22 Steve Edwards asterisk@sedwards.com I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. So to rephrase it: One GOOD reason to run asterisk as root is that you get to take advantage of the default filesystem overflow space reserved for root. That could just as well be a reason NOT to run Asterisk as root. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] A reason TO run Asterisk as root
On Wed, 22 Jul 2009, Jonathan Moore wrote: On Wed, Jul 22, 2009 at 10:31 AM, Steve Edwardsasterisk@sedwards.com wrote: I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. Thus, you may get some warning when everything non-root starts failing and give you a chance to free up some space before Asterisk is affected. Couldn't you get the same effect using quotas? Also, using separate partitions for various parts of the filesystem is a nice addition. Having your /var/log somewhere besides the same partition as / helps keep runaway logs at bay, just as an example. This is real sysadmin territory And it's a dying art, I fear. Too many people just creating one big partition, doing stupid (IMO) tricks like tune2fs -m 0 ... and so on. It's something you can't/won't ever learn from just doing a modern Linux install, or (worse, I reckon), installing something like pbxinaflash, etc. although to their credit, most of these pre-canned installs do seem to work well. Until they break. Then you need a sysadmin... Gordon (once a sysadmin, always a sysadmin) ___ -- 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] A reason TO run Asterisk as root
Yeah, and GOTO's are a good reason not to use COBOL. But they both still LIVE!! (Wah ha ha ha!!!) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Wednesday, July 22, 2009 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] A reason TO run Asterisk as root Jeff LaCoursiere schrieb: 2009/7/22 Steve Edwards asterisk@sedwards.com I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. So to rephrase it: One GOOD reason to run asterisk as root is that you get to take advantage of the default filesystem overflow space reserved for root. That could just as well be a reason NOT to run Asterisk as root. :-) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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 ___ -- 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] sip configuration masking the peers
On Wednesday 22 July 2009 03:44:22 peace keeper wrote: currentlly users in network1 can register as peer 2003 which is supposed to be allowed just for users from network2 , although this registration is supposed to be failed, any suggestions plz!! hope I made the scenario clear , any help would appreciated. Thanks in advance. What you're missing in your configuration are permit and deny lines, e.g. [peer1] ... deny=0.0.0.0/0 permit=192.168.1.0/24 [peer2] ... deny=0.0.0.0/0 permit=192.168.2.0/24 -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- 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] german voiceprompts
On Wednesday 22 July 2009 07:20:41 Johann Steinwendtner wrote: Are there any plans at Digium to include also german voice prompts ? There are no plans currently, but we do accept translation contributions from community members, wanting to ensure that prompts make sense for various languages. If you look in trunk in the subdirectory doc/lang, you'll see translations for both Hebrew and Urdu (Open Document format). -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- 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] ExecIf and empty variables (early evaluation)
On Wednesday 22 July 2009 08:30:03 Danny Nicholas wrote: You should submit this as a bug. It may or may not get fixed, but it definitely won't until someone reports it or takes it upon themselves to fix it. Don't bother. It's not fixable. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- 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 as a gateway
Greeting everyone, I'm trying to connect an old PBX to a Asterisk box with a 4 BRI card. The idea is for the PBX to follow asterisk's dialplan rules such as calling through VoIP when possible, ISDN when needed, etc, and all incoming calls being redirected to the PBX. The odd part is that incoming calls work perfectly, while when I make a call from a phone connected to the PBX through ISDN, I can hear the other party but they can't hear me and when the call is made through VoIP, I can't ear the ringing nor the other party (neither can the other party ear me), but the call is placed. I'm using alaw codec on every call. Does anyone have any idea what this problem could be? Thanks in advance, Best regards, Paulo Santos ___ -- 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] OT - Do analog gateways detect a phone is plugged in or out ?
2009/7/22 Steve Underwood ste...@coppice.org Olivier wrote: Hi, I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send qualify queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Unless there is a call in progress the switch has no idea what phones might be plugged or unplugged. Nothing happens on the line what it could detect. Steve OK Thanks! ___ -- 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 ___ -- 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] ExecIf and empty variables (early evaluation)
I see your name enough to know this must be a true statement; Can you elaborate a little on why? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Wednesday, July 22, 2009 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ExecIf and empty variables (early evaluation) On Wednesday 22 July 2009 08:30:03 Danny Nicholas wrote: You should submit this as a bug. It may or may not get fixed, but it definitely won't until someone reports it or takes it upon themselves to fix it. Don't bother. It's not fixable. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- 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 ___ -- 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] OT - Do analog gateways detect a phone is plugged in or out ?
On Wednesday 22 July 2009 11:07:32 Steve Underwood wrote: Olivier wrote: Hi, I've got a general question about analog gateways (Xorcom, Audiocodes, Patton, ...) . Is it usual for analog gateways to detect when an analog phone is plugged in or out ? If positive, would it be then useful to send qualify queries for each connect phone (I'm implying here that an analog gateway would then reply appropriately for qualify query. Unless there is a call in progress the switch has no idea what phones might be plugged or unplugged. Nothing happens on the line what it could detect. Yes, but on the other side of the switch, a station can detect battery. This is what the Digium analog cards do in order to report whether a line is in red alarm status or not. Whether the analog gateways do this or not is another question entirely, but it's certainly possible. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- 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] Attended transfer and 'pbx-invalid' - 1.4.26
Hi,
I've created a tiny dialplan to test the return of a call on transfers,
like this: (I had to use the DEVSTATE backport here)
[phones]
exten = _12XX,1,Dial(SIP/${EXTEN},6,tT)
exten = _12XX,n,GotoIf($[ x${BLINDTRANSFER} = x ]?noBT)
exten = _12XX,n,Set(DIALRET=${CUT(BLINDTRANSFER,-,1)});
exten = _12XX,n,Goto(dRet)
exten = _12XX,n(noBT),GotoIf($[ x${TRANSFERERNAME} = x ]?sai)
exten = _12XX,n,Set(DIALRET=${CUT(TRANSFERERNAME,-,1)});
exten = _12XX,n,GotoIf($[ ${DEVSTATE(${DIALRET})} = INUSE ]?sai);
exten = _12XX,n(dRet),Set(CALLERID(all)=RET_${EXTEN} ${CALLERID(num)})
exten = _12XX,n,Dial(${DIALRET},,mTt)
exten = _12XX,n(sai),Hangup()
It all works like a charm, except that when I do an atxfer and dial another
SIP and it rings, but dont answer, asterisk plays the 'pbx-invalid' sound,
that is a bit confusing, because the phone is there and actually rang . Here
is the CLI output
*CLI
-- Executing [1...@irrestrito-user:1] Dial(SIP/1202-08330f80,
SIP/1201|6|tT) in new stack
-- Called 1201
-- SIP/1201-08335530 is ringing
-- SIP/1201-08335530 answered SIP/1202-08330f80
-- Started music on hold, class 'default', on SIP/1202-08330f80
-- SIP/1201-08335530 Playing 'pbx-transfer' (language 'en')
-- Executing [1...@irrestrito-user:1]
Dial(Local/1...@irrestrito-user-70b2,2, SIP/1203|6|tT) in new stack
-- Called 1203
-- SIP/1203-08325260 is ringing
-- Local/1...@irrestrito-user-70b2,1 is ringing
Ring and no answer...
-- Nobody picked up in 6000 ms
-- Executing [1...@irrestrito-user:2]
GotoIf(Local/1...@irrestrito-user-70b2,2, 1?noBT) in new stack
-- Goto (irrestrito-user,1203,5)
-- Executing [1...@irrestrito-user:5]
GotoIf(Local/1...@irrestrito-user-70b2,2, 0?sai) in new stack
-- Executing [1...@irrestrito-user:6]
Set(Local/1...@irrestrito-user-70b2,2, DIALRET=SIP/1201) in new stack
-- Executing [1...@irrestrito-user:7]
GotoIf(Local/1...@irrestrito-user-70b2,2, 1?sai) in new stack
-- Goto (irrestrito-user,1203,10)
-- Executing [1...@irrestrito-user:10]
Hangup(Local/1...@irrestrito-user-70b2,2, ) in new stack
== Spawn extension (irrestrito-user, 1203, 10) exited non-zero on
'Local/1...@irrestrito-user-70b2,2'
-- Stopped music on hold on SIP/1202-08330f80
? -- SIP/1201-08335530 Playing 'pbx-invalid' (language 'en')
am I doing something wrong?
Thanks,
Gabriel
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Re: [asterisk-users] A reason TO run Asterisk as root
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Wednesday, July 22, 2009 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] A reason TO run Asterisk as root Jeff LaCoursiere schrieb: 2009/7/22 Steve Edwards asterisk@sedwards.com I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. So to rephrase it: One GOOD reason to run asterisk as root is that you get to take advantage of the default filesystem overflow space reserved for root. That could just as well be a reason NOT to run Asterisk as root. :-) Philipp Kempgen I think Jeff just rephrased it, not sure he endorsed it. Seems like if an Asterisk implementation stumbles 'cause of low disk issues, it would be good to still have the system healthy enough to be able to use remote-management tools to recover. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax ___ -- 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] A reason TO run Asterisk as root
on Wednesday 07/22/2009 Gordon Henderson(gordon+aster...@drogon.net) wrote On Wed, 22 Jul 2009, Jonathan Moore wrote: On Wed, Jul 22, 2009 at 10:31 AM, Steve Edwardsasterisk@sedwards.com wrote: I finally found a reason TO run Asterisk as root. By default, ext[23] file systems reserve 5% of the filesystem for root. Thus, you may get some warning when everything non-root starts failing and give you a chance to free up some space before Asterisk is affected. Couldn't you get the same effect using quotas? Also, using separate partitions for various parts of the filesystem is a nice addition. Having your /var/log somewhere besides the same partition as / helps keep runaway logs at bay, just as an example. This is real sysadmin territory And it's a dying art, I fear. Too many people just creating one big partition, doing stupid (IMO) tricks like tune2fs -m 0 ... and so on. It's something you can't/won't ever learn from just doing a modern Linux install, or (worse, I reckon), installing something like pbxinaflash, etc. although to their credit, most of these pre-canned installs do seem to work well. Until they break. Then you need a sysadmin... I do agree, but I do change the reserved blocks to 0, otherwise even as root the DF numbers are wrong and I have a number of partitions, even one for /tmp, so I figure its not so bad. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- 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] german voiceprompts
On Wed, Jul 22, 2009 at 8:20 AM, Johann Steinwendtnersteinwendt...@gmx.net wrote: Hello ! Are there any plans at Digium to include also german voice prompts ? I cannot speak on behalf of Digium, but I suspect that if somebody: * made cogent and sensible German translations of the English prompts * found a German voice talent to record those prompts * released them under an open-source license That the community would be grateful. I have no idea what it would cost to do a project like that for name_your_non-English_language I find it surprising that somebody hasn't already made at least a subset of German language prompts. The problem would probably be proper licensing of somebody else's effort. ___ -- 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] Iphone setup
I think siax -from cydia- could also be an alternative as they stated to use natively 3g. I only test WIFI. SIAX on WIFI works SIAX on WIFI works great so far. I don't have a router that i can secure my network with so I haven't tested it over 3G yet. I plan on doing that soon. Putting SIAX in the background only works for a little while. Also it does hangup the call if a call comes in on the regular number. That is of course a problem. ___ -- 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] ExecIf and empty variables (early evaluation)
While I can't be sure this is correct, I'd assume there are 2 pieces to executing a line of code, the first one does all the expansion and variable replacement, and the second one actually executes the line. From the behavior I'd have to guess that INC() is handled by first part and not the second. Ira I see your name enough to know this must be a true statement; Can you elaborate a little on why? On Wednesday 22 July 2009 08:30:03 Danny Nicholas wrote: You should submit this as a bug. It may or may not get fixed, but it definitely won't until someone reports it or takes it upon themselves to fix it. Don't bother. It's not fixable. ___ -- 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] ExecIf and empty variables (early evaluation)
On Wednesday 22 July 2009 13:56:39 Ira wrote: Danny Nicholas wrote: Tilghman Lesher wrote: On Wednesday 22 July 2009 08:30:03 Danny Nicholas wrote: You should submit this as a bug. It may or may not get fixed, but it definitely won't until someone reports it or takes it upon themselves to fix it. Don't bother. It's not fixable. I see your name enough to know this must be a true statement; Can you elaborate a little on why? While I can't be sure this is correct, I'd assume there are 2 pieces to executing a line of code, the first one does all the expansion and variable replacement, and the second one actually executes the line. From the behavior I'd have to guess that INC() is handled by first part and not the second. That is precisely the reason. I could not have said it better myself. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- 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] how to use patgen and pattest for PRI card?
On Wed, Jul 22, 2009 at 04:33:28PM +0800, Chris YM wrote: hello: I can set a environments to test the two pri cards. the patlooptest is ok. the result has no problem. how do i use patgen and pattest with two pri cards and the setting of zaptel/sys.conf? in the http://docs.tzafrir.org.il/man/pattest.8.html, there are no setting files and cablling for the two sides. please explan the settings and cabliing. thanks! Chris Nothing special. E.g. you should be able to call with Asterisk from one to the other. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] grandstream and jitter buffer
Hi guys, I have a bunch grandstream phones using ulaw and my users are complaining they are jittery when I use canreinvite=yes. The data connection is an ADSL link dedicated for phone traffic. At any given time, I have at most 2 calls in parallel. I'm not a huge fan of asterisk being in media path doing buffering because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and sometimes my users complain that are you on a sat phone? Any suggestions? - Kelvin Chan ___ -- 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] german voiceprompts
Here's what Philipp Kempgen wrote on this topic back in January. Nice summary, I believe. Klaus Darilion schrieb: http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international# German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). (Note: I might be a bit biased here as I work for Amooma but let me tell you something about it anyway.) http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international# German : === Greenable === Didn't try them yet. === Westany === Commercial (all others are free). marilda female german doesn't sound very German to me. ;-) Their web site is in English. I suspect they are selling to people who don't speak German themselves and who could never tell if the files sound ok. Didn't give them a try though. === Amooma === * http://www.amooma.de/asterisk/sprachbausteine/#prompts-tts These files are generated by our web-based text-to-speech engine. Pros: If you need additional custom prompts, just go to http://www.amooma.de/tts/ and generate them and the voice will match. Cons: Some of them don't sound right - yet. (But you can go to the web interface any time and regenerate them with better pronunciation in SAMPA alphabet.) I know that some work is going on here to improve them. * http://www.amooma.de/asterisk/sprachbausteine/#prompts-gabi Professional recordings. About 5 prompts are missing. * 3rd option: Gabi plus :-) Download the ones we use for Gemeinschaft ( http://www.amooma.de/gemeinschaft/gemeinschaft-installation-trunk.html#installation-trunk-single-debian-gemeinschaft) $ cd /usr/src $ svn checkout https://svn.amooma.com/asterisk-sounds-de/trunkasterisk-sounds-de-trunk $ cd /var/lib/asterisk/sounds/ $ ln -s /usr/src/asterisk-sounds-de-trunk de The recordings are the same as Gabi but none are missing and they are in WAV format. = This is what I use personally. === Aegee === Out of date. (2004, Asterisk 1.0) === Schwärzl === These happen to be the well known soundfiles released by Stadt- verwaltung Pforzheim. Unfortunately they are for Asterisk 1.2 and the upstream is gone. (Stadt Pforzheim doesn't have them on their web site any more.) === AsteriskFreaks === Don't know. A bit out of date (Jan 2006). === Debian: asterisk-prompt-de === That's the same thing as the Stadt Pforzheim prompts. Out of date, upstream is gone. Philipp Kempgen ___ -- 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] OT - Do analog gateways detect a phone is plugged in or out ?
On Wed, Jul 22, 2009 at 12:40:23PM -0500, Tilghman Lesher wrote:
Yes, but on the other side of the switch, a station can detect battery. This
is what the Digium analog cards do in order to report whether a line is in
red alarm status or not. Whether the analog gateways do this or not is
another question entirely, but it's certainly possible.
Xorcom gateways certainly do. Again, that's for FXOs
And generally
perl -MDahdi -e '
my @chans = map {$_-chans()} Dahdi::spans();
foreach (@chans) {
next unless $_-type() eq 'FXO';
printf %3d %s\n,$_-num, $_-battery;
}'
The battery method ATM only implemented on Astribanks in the perl
modules, however this should be soon fixed.
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http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] grandstream and jitter buffer
jbmaxsize=80 is way overkill. If your jitter is really close to 80ms then you have some serious issues on your link and it's not suitable for VoIP at all. Try jbmaxsize=40. Vinícius Fontes www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP - Kelvin Chan kelv...@positronics.com escreveu: Hi guys, I have a bunch grandstream phones using ulaw and my users are complaining they are jittery when I use canreinvite=yes. The data connection is an ADSL link dedicated for phone traffic. At any given time, I have at most 2 calls in parallel. I'm not a huge fan of asterisk being in media path doing buffering because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and sometimes my users complain that are you on a sat phone? Any suggestions? - Kelvin Chan ___ -- 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 ___ -- 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] Callin Numbers.
On Wed, Jul 22, 2009 at 2:41 PM, Geoff Lanege...@gjctech.co.uk wrote: On Wednesday, July 22, 2009, Catalin S. wrote: I lookin' for a call in number from UK or USA. Can somebody offers me a peering for this or specify any sip provider that offers this thing? There are several providers who offer UK or US regional geographical numbers for little or no cost if you only use them inbound. For example, I have UK geographicals from Sipgate (http://www.sipgate.co.uk/user/index.php) and VoipCheap (http://www.voipcheap.com/en/index.html - *not voipcheap.co.uk*). The latter, I had to install their client to a Windows host and inspect the configuration to obtain the info necessary to connect my Asterisk server. However, those are only examples and there are a lot more to be found if you look around. HTH, -- Geoff ___ -- 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 ___ -- 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] Callin Numbers.
Hello sorry for earlier message, I push send before write something. Anyway I tried that sites and also lowratevoip.com. All gives me the follwing message: Sorry – at this moment there are no VoIP-In numbers available for your country (yet). We will inform you as soon as there are (new) numbers available for your region. Click to go back. Do you have some tested sites please? Thank you. On Wed, Jul 22, 2009 at 2:41 PM, Geoff Lanege...@gjctech.co.uk wrote: On Wednesday, July 22, 2009, Catalin S. wrote: I lookin' for a call in number from UK or USA. Can somebody offers me a peering for this or specify any sip provider that offers this thing? There are several providers who offer UK or US regional geographical numbers for little or no cost if you only use them inbound. For example, I have UK geographicals from Sipgate (http://www.sipgate.co.uk/user/index.php) and VoipCheap (http://www.voipcheap.com/en/index.html - *not voipcheap.co.uk*). The latter, I had to install their client to a Windows host and inspect the configuration to obtain the info necessary to connect my Asterisk server. However, those are only examples and there are a lot more to be found if you look around. HTH, -- Geoff ___ -- 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 ___ -- 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] A reason TO run Asterisk as root
On Wed, 2009-07-22 at 16:52 +, Jeff LaCoursiere wrote: So to rephrase it: One GOOD reason to run asterisk as root is that you get to take advantage of the default filesystem overflow space reserved for root. It might be A reason, but it certainly isn't a GOOD one. A GOOD system would be to set up proper disk monitoring, log rotation, purging/archiving, etc. Using the crashing of applications to notify you when a disk is full is NOT good. There might be a good reason to run Asterisk as root, but this isn't it. -Justin ___ -- 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 CSTA
does Asterisk suppoet CSTA protocol for CTI applications?___ -- 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] CallerPres SIP headers Analog Phone
Yes. I am able to match the *67 and appropriately set the
SetCallerPres when SIP phones make calls because the *67 is passed
through and can be matched.
However on my analog handset its as if the *67 is processed and
discarded. Here is my chan_dahdi.conf and a snippet of console output:
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=1.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
context=from_telco
group=0
echocancel=yes
signalling=fxs_ks
channel = 1
context=from_telco
group=0
echocancel=yes
signalling=fxs_ks
channel = 2
context=fax
group=1
echocancel=yes
signalling=fxo_ks
channel = 3
context=analog_phone
group=1
echocancel=yes
signalling=fxo_ks
channel = 4
extensions.conf snippet
[analog_phone]
immediate=no ;This line tells asterisk to wait for input from the
analog phone then continues on
include = default
[default]
exten = _*XX.,1,Goto(outbound,${EXTEN},1)
exten = _*XX.,n,Hangup()
exten = _X.,1,Goto(outbound,${EXTEN},1)
exten = _X.,n,Hangup()
[outbound]
exten = _*67NXXNXX,n,SIPAddHeader(Remote-Party-ID:
sip:xxx...@sipprovider:5060
\;user=phone\;party=calling\;screen=yes\;privacy=full)
exten = _*67NXXNXX,n,Noop(${CALLERID(num)})
exten = _*67NXXNXX,n,Dial(SIP/provider/${EXTEN:3})
Again for SIP handset the above works fine, but here is what an analog
phone does:
-- Starting simple switch on 'DAHDI/4-1' --THIS WHEN THE PHONE GOES
OFFHOOK
-- Disabling Caller*ID on DAHDI/4-1 --THIS IS AS SOON AS *67 is
PRESSED
-- Executing [xxx...@analog_phone:1] Goto(DAHDI/4-1,
outbound,XX,1) in new stack --THE REMAINING DIGITS ARE
PASSED TO OUTBOUND
-- Goto (outbound,XX,1)
--BUT CAN'T MATCH
because the *67 is STRIPPED
-- Executing [xxx...@outbound:1] Dial(DAHDI/4-1, SIP/
provider/XX) in new stack
I'd like to know if this is because of the phone handset, the analog
card, or something else. Obviously there has to be a way to either
capture the *67 from the handset or utilize the fact that apparently
the card is detecting that *67 was pressed, and get it to properly set
the CallerPres()
Thanks
On Jul 22, 2009, at 11:36 AM, Philipp Kempgen wrote:
Ketema Harris schrieb:
hello all...I have been trying to get a handle on CallerPres with an
analog handset. I have usecallingpres=yes in my chan_dahdi.conf file
and when I dial *67 on my analog handset I see Disabling Caller*ID on
DAHDI/4-1 but when the call is then forwarded to my outbound SIP
provider the RPID header is not correct privacy=off;screen=no
instead of
full and yes how can I correct this?
I don't know if/how Asterisk handles/stores CLIR for analog handsets
but SetCallerPres(prohib_passed_screen) does the trick when dialing
to a SIP channel.
Remote-Party-ID: ...;privacy:full;screen:yes
You could add a *67 extension to your dialplan and store the CLIR
state in AstDB for example.
Philipp Kempgen
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Re: [asterisk-users] grandstream and jitter buffer
If the users' understanding of jitter is technically correct, and they're complaining about quality issues (due to jitter or packet loss), then lowering the jitter buffer isn't going to help. An ADSL link, depending on the sync rate, can have 40+ msec of latency between the DSL modem and DSLAM. If the link quality is good, ask your DSL provider to change the mode from INTERLEAVED to FAST; that should drop the latency at least 10 msec. But before all that, you may want to ascertain how much of the issue is packet loss versus jitter. Regards, Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vinícius Fontes Sent: Wednesday, July 22, 2009 3:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] grandstream and jitter buffer jbmaxsize=80 is way overkill. If your jitter is really close to 80ms then you have some serious issues on your link and it's not suitable for VoIP at all. Try jbmaxsize=40. Vinícius Fontes www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP - Kelvin Chan kelv...@positronics.com escreveu: Hi guys, I have a bunch grandstream phones using ulaw and my users are complaining they are jittery when I use canreinvite=yes. The data connection is an ADSL link dedicated for phone traffic. At any given time, I have at most 2 calls in parallel. I'm not a huge fan of asterisk being in media path doing buffering because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and sometimes my users complain that are you on a sat phone? Any suggestions? - Kelvin Chan ___ -- 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 ___ -- 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 ___ -- 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] Asterisk to PBX
Can I assume that your project has stalled? PaulH logan wrote: Thanks Paul. Your help is much appreciated here. I don't really understand this question - Asterisk can make calls over phone lines. And it does it well. Surely, Asterisk does that well, but Asterisk needs to have multiple phone lines for that. I thought that a traditional switchboard made that happen without multiple phone lines. BTW, in Asterisk terminology a phone line means different PSTN connections to the operator, right? Why would you guess this? We had 16 phone lines in the first business I worked in. Yeah, that's fine, but even 16 phone lines don't mean you can have 16 desk phones only or 16 simultaneous calls? Thanks I will take a look at asteriskdocs. Best Regards, Hitesh ___ -- 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 ___ -- 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] odd behaviour with AGI and dial agent
Hi, I have come across an odd problem. Basically I am transferring a call to an agent. The agent is logged in and set as paused. In order to find which agent to call I am using a fastagi script to just set a variable. When it falls through the agi script and dials the agent (using the variable) it doesn't connect the call properly to the agent. I get the beep but no audio (along with some other strange behaviour with the channel not hanging up properly, core show channels doesn't work properly). Now if I just set the variable in the dialplan (ie. no agi), or just hardcode the agent being called then it works fine. It seems that calling the fastagi is doing something to the channel which means that it doesn't work properly afterwards. I have also tried calling the agent in the agi with the same problems. Does anyone have any idea what the agi script could be doing to the channel/call, what it could be changing and how I can make it work properly. Keiron ___ -- 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] Asterisk CSTA
On Wed, Jul 22, 2009 at 4:03 PM, gergis.rasmygergis.ra...@gmail.com wrote: does Asterisk suppoet CSTA protocol for CTI applications I'd never heard of it, so I googled it. http://en.wikipedia.org/wiki/Computer-supported_telecommunications_applications So, ummm, I can't think of a good synonym for this, but it sounds like it's a high-level abstraction of how people may choose to communicate using 'computers' and 'technology'. While I've never heard of Asterisk explicitly supporting this, it sounds like: * partial implementations of features is good enough to be considered 'implementing' the protocol * because you can use asterisk to make phone calls, asterisk could be coerced to meet this model if there is some reason that it does not now meet the requirements that I could find in my ten minute googling. The stuff about XML and high-level ways of initiating communications independent of a channel sounds cool, but also kind of iconoclastic and against the traditional way of how PBX systems are designed and thought of. I think of 'phones' people talking about PRIs and DSPs and TLAs, and being knee-deep in signaling protocols. My thoughts reading the documents for some CSTA conference: * oh no, another protocol * and we thought people had trouble with the learning curve of phones and asterisk _before_ * interoperability is already a problem with simple, fairly straightforward protocols. Good luck building a high level interface that makes the underlying implementation incompatibilities go away. ___ -- 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
