Re: [asterisk-users] Is it possible to get full callin number from E1?
thanks all
i found the telco only send me the normal number 87654321
i just want to start a fax service and people can direct dial some extend
num like 87654321...but it never send to me ... so the only thing i can do
is to provide a ivr and let the people enter the extend num then jump to fax.
thanks alot ~
在2009-03-13 03:42:54，D Tucny d...@tucny.com 写道：
2009/3/12 ssmax ss...@126.com
Hi all
i have just set up a asterisk in china, using DE410P and one E1 line
and get a phone number like: +86 020 87654321 from my sp
when somebody dial +86 020 87654321 , the asterisk will get the call in
number by ${EXTEN} variable, but it can only get 87654321, no area code .
As others have mentioned in this thread, this is pretty normal... Telco's all
have different policies on how many digits they will pass for the dialled
number, some may let you specify, but often each telco has an internal
standard... You don't need the rest of the number, but if you're doing
something where it would be useful to have it, you can easily add the prefix
yourself... It's also worth pointing out at this point, the prefix is either
020 or +86-20 for Guangzhou depending on whether it's written for national or
international use, the 0 is not dialled internationally...
when someone dial +86 020 87654321 , means 4 digits, the phone
can call in, and the ${EXTEN} is only 87654321 too ,
is it possible to get full call in number 87654321 in asterisk ?
thanks
This is confusing... Guangzhou numbers are only 8 digits long + prefixes, so,
+86 20 87654321 would be the number... Where do you get '' from? What would
be more normal would be that you'd by DDIs with your ISDN30 service, such that
you'd have for example, a range of 100 numbers say from 87654000 to 87654099,
though you can buy more... With this, the number dialled (or the last 8 digits
in your case) would be accessible through the ${EXTEN} variable... I guess it
could be possible that your telco is attempting to offer some other type of
service that allows them to issue you a single number, but, that they will
accept as many as four additional digits that they will pass to you in some
way, but, it's impossible to say what they are doing here and how it's been
implemented... I would further suspect that if this is the case that there
would be a very distinct chance that this would only work with inbound calls
from other users of the same telco, and in China, in the same city...
Within China, telco interoperability can be flaky at best and, from experience,
the fact the telco engineers don't entirely understand their huawei switches
for local problems and for problems over a wider area, the fact that the
national telcos are split into provincial and city organisations that don't
appear to communicate without going through some circuitous route via the
telco's head office both lead to some very long fix times if there ever is a
fix... That said, as long as you have cash, you can get them to provide almost
any service you want, though they won't often offer anything but the most basic
service, you have to ask for it...
d
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[asterisk-users] Initial silence during call
Hi all, I've got a problem where many times, there is silence at the first 1-2 seconds of a call. Then it clears up and it's crystal clear. I've not put a sniffer on it, yet, but I suspect that the media channel is still being set up. The server shouldn't be too overloaded. Can anyone give me some advise on how to solve/mitigate this problem? Mike. ___ -- 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] Initial silence during call
Check your echo can settings. On Fri, Mar 13, 2009 at 3:06 AM, Mike Diehl mdi...@diehlnet.com wrote: Hi all, I've got a problem where many times, there is silence at the first 1-2 seconds of a call. Then it clears up and it's crystal clear. I've not put a sniffer on it, yet, but I suspect that the media channel is still being set up. The server shouldn't be too overloaded. Can anyone give me some advise on how to solve/mitigate this problem? Mike. ___ -- 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 -- 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] Portech MV3770 Caller-ID
On Thu, Mar 12, 2009 at 09:53:48AM +0100, wrote:
On 3/11/2009, Hĺkan Källberg h...@simulina.se wrote:
On Wed, Mar 11, 2009 at 04:16:43PM +0100, Christian Victor wrote:
2009/3/11 Hĺkan Källberg h...@simulina.se
Does anyone of you have Caller Presentation working in the other
direction?? My mv370 is working well, execpt the Caller ID on outgoing
GSM calls. This works with the SIM card/Provider I am using if I put
All I had to do is to enable the Caller ID ind the Mobile-Settings dialog
for each SIM (something like presentation/revocation afair). I did NOT set
the GSM number anywhere nor do I send it from Asterisk.
That is what I'd expect too, but, no...
Mobile-Settings-CLID Presentation- Supression or Invocation
it makes no difference. (and yes - I do reboot:-) When I move the SIM
to a phone, it works well...
I'm not sure wether it's an operator-specific setting or not. But i
don't handle callerid supression/invocation with the MV370, i rather do
it with asterisk.
Try simply prefix the number with *31# for invocation and #31# for
supression. Example:
[...]
exten = _06[237]0NXX!,n,Dial(SIP/*31#${ext...@gsmgw)
[...]
Again: this method may be country or even operator specific, check with
the provider if the above works. If it does, simply use the prefixes
above and forget about the MV370's settings.
It worked like a charm!! Szabó András szu...@gmail.com came up with the
same solution too. Thank you *very* much, both of you!!
Håkan
pgpEnpa0HQABR.pgp
Description: PGP signature
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Re: [asterisk-users] an easy way to deal with/without leading 1 ?
sean darcy seandar...@gmail.com writes:
The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes that are toll free.
We try to canonicalize dialled numbers as soon as they enter the
system.
Something like this:
[fromphones]
exten = _XXX,1,Goto(canonical,+1555${EXTEN},1)
exten = _XX,1,Goto(canonical,+1${EXTEN},1)
exten = _00.,1,Goto(canonical,+${EXTEN:2},1)
[canonical]
exten = _+1800XXX,1,...
exten = _+1877XXX,1,...
We don't actually have any US locations yet, so the above is made up
from scratch without any testing.
Whether to use the + to indicate a number in e164 format is a topic of
some debate.
/Benny
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Re: [asterisk-users] UK ISDN-30 and ANI
On Thu, 12 Mar 2009 10:21:06 +, Julian Lyndon-Smith aster...@dotr.com wrote: Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 AFAIK (and our E1 doesn't go to * box) a) you mean CLI b) you have to pay BT extra for Calling Line Identity Presentation GBP7.50 / qtr on our last bill HTH ___ -- 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] Serving 120 concurrent calls
On Thu, 12 Mar 2009, Tarek Sawah wrote: Hello, a local prison contacted us regarding some calling card solution. they need 4 E1s to serve 120 rooms in that prison. we are planning on using 4 servers to serve the calls and one for the database servers' specifications are: 2.8 Dual Core Proccessors 2 GB Ram 160 Sata Drive each server will be provided with 1 E1 card I'm rather surprised that you're using 4 servers - especially when I have 1GHz boxes handling a full E1 lines without breaking into a sweat... Questions are: 1- will those servers be able to handle that ammount of calls?' Just ONE of those servers ought to be able to handle all those calls. You're not doing any transcoding, so it's just a data moving platform. They (Digium, etc.) make 4-port E1 cards... What sort of processor do you think those ought to be connected to? 2- the important issue is that they require call recording on all calls.. which means we will have to record ALL calls going out of the system .. which means we will need a call recroding.. will the four Asterisk servers handle the recording process or we will need external assistant? and if it was the second choice what is the best suggestion? is there a way to force an Asterisk server to record remote channels? Do the sums: 120 x 64Kb/sec x 2 = 15360Kb/sec or 1920KB/sec or just under 2MB/sec. Any PC built this decade can do that. Of-course multiple servers could be for some sort of redundancy setup... But if not, I'd be really surprised if just one box had any issues with that call volume. Gordon ___ -- 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] Initial silence during call
If there is NAT between the phone and * then that can be responsible. Also, Eyebeam (et al)'s ICE setting causes this. Steve On 3/13/09, Mike Diehl mdi...@diehlnet.com wrote: Hi all, I've got a problem where many times, there is silence at the first 1-2 seconds of a call. Then it clears up and it's crystal clear. I've not put a sniffer on it, yet, but I suspect that the media channel is still being set up. The server shouldn't be too overloaded. Can anyone give me some advise on how to solve/mitigate this problem? Mike. ___ -- 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 -- Sent from my mobile device ___ -- 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] Serving 120 concurrent calls
On Thu, 12 Mar 2009 21:42:28 +, Tarek Sawah tareksa...@hotmail.com wrote: Hello, a local prison contacted us regarding some calling card solution. they need 4 E1s to serve 120 rooms in that prison. If there's only one person per room, then I'm not sure that they need *4* E1s if you think about it... ___ -- 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] Timeout for Queue
You should look at the queue() command invocation. Thanks l. 2009/3/12 Darrin Henshaw dhens...@ignition.bm Hello, We had an incident recently where a call was in queue for an extended period of time. We use queuemetrics for reporting, and it reports that the call was waiting for 20 minutes. The different thing about it is that the disconnect reason is stated as Timeout. Is there a set maximum time a call will wait in the queue before being automatically disconnected? I tried looking through the code directly, but I humbly admit my programming skills are lax. I’m running Asterisk 1.2.31 on CentOS 4.7. Thanks. Cheers, [image: logo] Darrin Henshaw |* *IT Administrator | MCTS: Exchange 2007 | MCSE 2003 |LPIC Ignition Support Center* *|* *www.ignition.bm Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288 Atlanta | Bermuda | Cayman | Halifax -- This email and its attachments may be confidential and are intended solely for the use of the individual or parties’ to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com image001.jpg___ -- 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] log to cdr each dialpan action, not only one record for each call
I'm only half joking: what about parsing the full log looking for command inviocations and channel IDs? this would be completely transparent, albeit insane :) l. 2009/3/12 nik600 nik...@gmail.com Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know that exists ResetCDR and ForkCDR but they don't do what i need, expecially because they fill-in lastdata and lastapp with ResetCDR So, what can i do? Is it better to do some customization to generate a CDR event on each dialplan step or is better to parse the logfile and extract the information needed? I'm using Asterisk 1.4.23.1 TIA -- /*/ nik600 http://www.kumbe.it ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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] log to cdr each dialpan action, not only one record for each call
On Thu, Mar 12, 2009 at 4:22 PM, BJ Weschke bwesc...@gmail.com wrote: nik600 wrote: Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know that exists ResetCDR and ForkCDR but they don't do what i need, expecially because they fill-in lastdata and lastapp with ResetCDR So, what can i do? Is it better to do some customization to generate a CDR event on each dialplan step or is better to parse the logfile and extract the information needed? I'm using Asterisk 1.4.23.1 We generated a patch for a client probably about a year ago against the 1.4 branch that logged apps for each call, params, and exit status codes into a separate file. Like others have said, it generates a tremendous amount of data and probably does impact performance on very high load servers, but it was very useful to determine EXACTLY what happened with a given call. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ Any chance of sharing? I have several clients with complex IVRs. They (I) would like to see if there are logical or user loops, where people hang up to check for complexity or pure frustration. Sounds like your patch might help in this regard. -- 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] Serving 120 concurrent calls
On Thu, Mar 12, 2009 at 09:42:28PM +, Tarek Sawah wrote: Hello, a local prison contacted us regarding some calling card solution. they need 4 E1s to serve 120 rooms in that prison. 120 concurrent calls? (do you assume that most of those lines will be busy most of the time?) Normally they aren't. You also didn't mention what type of outgoing lines / trunks / whatever-you-call-it you had in mind. we are planning on using 4 servers to serve the calls and one for the database servers' specifications are: If you have any trunks: where are they from in any of those servers? 2.8 Dual Core Proccessors 2 GB Ram 160 Sata Drive each server will be provided with 1 E1 card Questions are: 1- will those servers be able to handle that ammount of calls?' Sure. I suspect you have a slight overkill (maybe with too many points of failure) -- 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
Re: [asterisk-users] AGX Asterisk Addon - Can't find app_fax.c withspandsp-0.0.4
You now need to compile and install SpanDSP-0.0.6pre3 at least (AGX has been changed). After you've done that - try AGX again. HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 11 March 2009 06:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AGX Asterisk Addon - Can't find app_fax.c withspandsp-0.0.4 Hi, I've installed spandsp-0.0.4pre16 With this: cd ~ svn co https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons agx-ast-addons cd agx-ast-addons/trunk ./build.sh I've got this: CMake Error in spandsp-0.0.4/CMakeLists.txt: Cannot find source file app_fax.c. Tried extensions .c .C .c++ .cc .cpp .cxx .m .M .mm .h .hh .h++ .hm .hpp .hxx .in .txx This is coherent with : A agx-ast-addons/trunk/spandsp-0.0.4 A agx-ast-addons/trunk/spandsp-0.0.4/app_rxfax.c A agx-ast-addons/trunk/spandsp-0.0.4/app_txfax.c A agx-ast-addons/trunk/spandsp-0.0.4/CMakeLists.txt A agx-ast-addons/trunk/spandsp-0.0.4/README My CMake knowledge is too short to propose a workaround. Maybe, this could come from a change in agx-ast-addons/trunk/spandsp-0.0.4/CMakeLists.txt as the content of this file is: project (app-fax-span4) # -- # Target # we use MODULE cause it build a shared object module # -- ADD_LIBRARY(app_fax MODULE app_fax.c) # # We remove the lib prefix from the libmodule.so filename # SET_TARGET_PROPERTIES(app_fax PROPERTIES PREFIX ) # # We add library dependencies to use those modules # TARGET_LINK_LIBRARIES(app_fax spandsp tiff) # # override default INSTALL rules # INSTALL(TARGETS app_fax DESTINATION lib/asterisk/modules) Could you help ? 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
[asterisk-users] Silence suppression problem with DECT phones and g729 codec
Hello, I have been experiencing audio problems when accessing the Voicemail application using DECT phones and the g729 codec. The issue is that whereas the vm-password is always played correctly by the DECT phone, the rest of audio files, randomly, are played or not by the DECT phone. Everything works correctly if another codec (alaw,ulaw) is used. I have noticed that asterisk doesn't send RTP with silence, but stop sending them and I think the problems is that the DECT phones are having problems with that. To check that this was the problem I have implemented a simple dialplan exten = *91,1,Set(CHANNEL(language)=es) exten = *91,n,Answer() exten = *91,n,Wait(4) exten = *91,n,Playback(vm-tmpexists) exten = *91,n,Wait(4) exten = *91,n,Playback(vm-tomakecall) exten = *91,n,Wait(4) exten = *91,n,Playback(vm-goodbye) exten = *91,n,Hangup ...and I have verified that if there is a pause between the playbacks the problem occurs, otherwise the audio is played correctly by the DECT phones I know it looks like a problem with the phones but, is there a way to configure asterisk so it sends RTP during silent periods? Thanks. Best regards, Santi ___ -- 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] DAHDI and B410P (BRI)
That's at least 2 of us then Paul ;). -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Paul Hales -- Sent: 11 March 2009 00:04 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] DAHDI and B410P (BRI) -- -- -- I wish it was available too - I have just had to back dahdi out of a -- system and revert to misdn after a whole day of testing. -- -- PaulH -- -- -- Andrew Thomas wrote: -- I have LibPri installed and working (.../wPRI). -- -- So, if I understand Tzafrir correctly - DAHDI support for the B410P -- isn't available in 1.4 at all. -- -- Looks like I'm going back to mISDN. -- -- Cheers -- Andy -- -- -- -- -- -Original Message- -- -- From: asterisk-users-boun...@lists.digium.com -- [mailto:asterisk-users- -- -- boun...@lists.digium.com] On Behalf Of Jose Luis Villalon -- -- Sent: 09 March 2009 18:07 -- -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- -- Subject: Re: [asterisk-users] DAHDI and B410P (BRI) -- -- -- -- Hi -- -- -- -- What it's the result of execute -- -- -- -- strings /usr/lib/asterisk/modules/chan_dahdi.so | grep '^DAHDI -- -- Telephony' -- -- -- -- It's LibPri install before of Dahdi package? -- -- -- -- JL. -- -- -- -- El Lun, 9 de Marzo de 2009, 6:36 pm, Andrew Thomas escribió: -- -- Hi all, -- -- -- -- -- -- I am having trouble setting the signalling method for the -- B410P -- -- using -- -- DAHDI. Asterisk complains that it has never heard of -- 'bri_cpe' or -- -- 'bri_net' - but it doesn't mind having 'pri_cpe' etc. -- -- -- -- -- -- ERROR[4294]: chan_dahdi.c:11327 process_dahdi: Unknown -- signalling -- -- method -- -- 'bri_net' -- -- -- -- -- -- Dahdi - dahdi-linux-complete-2.1.0.4+2.1.0.2 -- -- Asterisk - 1.4.23.1 -- -- Libpri - 1.4.9 -- -- -- -- -- -- I have set the spans up with no problems (well, dahdi_cfg -- doesn't -- -- complain) - it's just my chan_dahdi.conf file I need to fix -- now. -- -- -- -- Thanks -- -- Andy -- -- -- -- -- -- -- -- ___ -- -- -- 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 -- -- -- -- ___ -- -- 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] Silence suppression problem with DECT phones and g729 codec
On 13 Mar 2009, at 09:51, Santiago Gimeno wrote: I know it looks like a problem with the phones but, is there a way to configure asterisk so it sends RTP during silent periods? Asterisk.conf transmit_silence_during_record = yes ___ -- 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] UK ISDN-30 and ANI
Please explain (in English) what you mean by ANI. Thanks -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith -- Sent: 12 March 2009 10:21 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: [asterisk-users] UK ISDN-30 and ANI -- -- Has anyone in the UK got ANI to work on an inbound call ? -- -- Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 -- -- Julian -- -- __ -- This email has been scanned by the MessageLabs Email Security System. -- For more information please visit http://www.messagelabs.com/email -- __ -- -- ___ -- -- 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] Silence suppression problem with DECT phones and g729 codec
Yes, I had already tried that and it didn't work. Asterisk doesn't send any RTP. Regards, Santi On Fri, Mar 13, 2009 at 11:06 AM, Steve Howes st...@geekinter.net wrote: On 13 Mar 2009, at 09:51, Santiago Gimeno wrote: I know it looks like a problem with the phones but, is there a way to configure asterisk so it sends RTP during silent periods? Asterisk.conf transmit_silence_during_record = yes ___ -- 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] UK ISDN-30 and ANI
Hi Andrew Andrew Thomas wrote: Please explain (in English) what you mean by ANI. http://www.tech-faq.com/ani-automatic-number-identification.shtml Julian Thanks -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith -- Sent: 12 March 2009 10:21 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: [asterisk-users] UK ISDN-30 and ANI -- -- Has anyone in the UK got ANI to work on an inbound call ? -- -- Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 -- -- Julian -- -- __ -- This email has been scanned by the MessageLabs Email Security System. -- For more information please visit http://www.messagelabs.com/email -- __ -- -- ___ -- -- 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 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] UK ISDN-30 and ANI
David Quinton wrote: On Thu, 12 Mar 2009 10:21:06 +, Julian Lyndon-Smith aster...@dotr.com wrote: Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 AFAIK (and our E1 doesn't go to * box) a) you mean CLI a) No I don't. CLI is different to ANI b) you have to pay BT extra for Calling Line Identity Presentation GBP7.50 / qtr on our last bill See a). We already have CLI. I need ANI ;) HTH ___ -- 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 __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- 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] UK ISDN-30 and ANI
On 13 Mar 2009, at 10:43, Julian Lyndon-Smith wrote: We already have CLI. I need ANI ;) Why? Just out of interest.. If people withold CLI its usually for a reason.. ___ -- 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 and WebIntegration
I reverse the inbound calls so they appear as outbound calls for agents, all of our calls are managed by the dialer i've written and integrate directly to our CRM, essentially asterisk is only providing the SIP/IAX functionality to me everything else is done via php... so... inbound call comes in and gets parked in a php script stores in database as an outbound call, agents screen then pops and checks the database for the CLI so we can try to guess who's calling us and opens up all of their details. php script that is parking the inbound call then dials the allocated agents extension and connects the call. also on the dial command i have used Dial(SIP/1234,,A(beep)) so that the agent hears a beep when they get a call. Hope this enlightens you a bit on handling inbounds in this situation :) Cheers 2009/3/12 Kurian Thayil kurianmtha...@gmail.com Hi Geriant, My apologies for the delay in reply. We won't be using php but Perl and there is an AGI module for perl Asterisk::AGI. I may be using Manager API for sending Hangup signal. Im planning to write a bash script which perl invokes when hangup button is pressed in the web interface. Bash script telnets and sends Hangup signal to the manager API. I am not yet able to acheive sending commands via bash script using telnet. But I am trying. One thing that's confusing me is if in future, incoming facility needs to be activated and since Auto Answer feature in EyeBeam/Twinkle is ON, don't you think that would be a problem? I think for that, the possible work around will be using 2 softphones, say EyeBeam and Xlite together in the same PC. Configuring one extension in EyeBeam to make outbound calls (with Auto Answer enabled) and configuring Xlite with an extension which receives inbound calls. Do you have any suggestion on that? Regards, Kurian Mathew Thayil. On Tue, Mar 10, 2009 at 7:32 PM, Geraint Lee gera...@gmail.com wrote: If you're using a php i'd take a look at phpagi - there are others around for various different languages too. our agents use twinkle with auto-answer, the only reason they need to look at twinkle is if they need to perform a transfer (that too will soon be done from the web browser), you can do pretty much anything with the asterisk manager (originate the call and hangup the call and a load of other useful stuff) Cheers 2009/3/10 Kurian Thayil kurianmtha...@gmail.com Hi Steve, That worked beautifully. Thank you so much. But one question though. Imagine if I keep a Hangup Button in the interface and it should terminate the call. Will that be possible? This scenario happens when the user gets connected to an invalid phone number where the user have to manually disconnect. I don't plan to confuse the user by asking them to use eyebeam to disconnect the call. If it could be integrated to the web interface they just have to stick on to that alone. Is there any way? Regards, Kurian Mathew Thayil. On Tue, Mar 10, 2009 at 4:51 PM, Steve Totaro stot...@first-notification.com wrote: On Tue, Mar 10, 2009 at 6:40 AM, Kurian Thayil kurianmtha...@gmail.com wrote: Hi All, Is there a way that I can include call dialing functionality in a webinterface. I have EyeBeam configured with a SIP user say 8440. Will I be able to design an inteface which agent can choose a number and the Dial without punching in the number in Eyebeam. I tried using the .call file. ie The agent can choose which number to dial from a web interface. Then, a .call file is created with the following informations. Channel: Zap/g2/9444204943 Context: inbound_support Extension: 8440 Priority: 0 Now, in the extensions.conf file, I mentioned the following under inbound_support context. [inbound_support] exten =8440,1,Dial(SIP/8440,55,tTo) exten =8440,2,Answer exten =8440,3,Hangup But, here the call gets connected only when the receiver end receives the call. When the receiver end picks up the phone, SIP/8440 rings. Is there any other way to implement this. I am not ready to use Vicidial (AstGUIClient) because the interface to be designed is too custom and the agent should have the list of numbers in front of them while they dial which cannot be done using Vicicial. Regards, Kurian Mathew Thayil. The following will ring the internal support personnel (8440) first, after answered, it will then dial the customer (14109850123) (Are you in Maryland?) Turn on auto-answer and it should be seamless. Stolen from Wiki: To create a call to 14109850123 on a SIP phones called bt101, here's the file you'd create in /var/spool/asterisk/outgoing (whatever name is good, of course must be accessible and deletable by asterisk GNU/Linux user): Channel: SIP/8440 MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [outgoing] # Context: outgoing # Extension: 14109850123 # Priority: 1 -- Thanks, Steve Totaro
[asterisk-users] Outbound routing
Dear All, I have a small call center in which I have to define least cost routing for outbound calls. For now I have always done this by routing numbers to different providers according to the number prefix. However, a new law became effective now which allows people to switch between providers without changing their telephone numbers. This makes least cost routing based on number prefixes much less effective. Are there any known solutions for this? Thanks in advance, Alex ___ -- 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] Trying to get sample applicationmap to work (*1.4)
Mark Michelson wrote: You can work around the bug, although it's not exactly optimal. What you can do is to modify your dialplan as follows: exten = 301,n,Set(DYNAMIC_FEATURES=monkey) Couldn't you just set _DYNAMIC_FEATURES here and have it get automatically inherited to the outbound channel? exten = 301,n,Dial(SIP/DavidR1,,M(dynamic_features)) [macro-dynamic_features] exten = s,1,Set(DYNAMIC_FEATURES=monkey) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] UK ISDN-30 and ANI
I think I understand what you mean now. The biggest difference between CLI and ANI is that ANI can't be blocked/withheld (like you can with CLI by using 141). It also uses different signalling. This is mainly used by law enforcement agencies to trace calls etc. So, you want the number - regardless of what the user dials. I presume you are some sort of 'carrier' then. You'll be lucky to get the information otherwise as it throws up all sorts of privacy laws (ie. you have to have a damn good reason for wanting it). BT are the main people to ask I suppose (unless your calls go through another main carrier). I'm not even sure if ANI signalling is implemented in Asterisk - one for the config file writers ;). Cheers -- -Original Message- -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- -- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith -- Sent: 13 March 2009 10:43 -- To: Asterisk Users Mailing List - Non-Commercial Discussion -- Subject: Re: [asterisk-users] UK ISDN-30 and ANI -- -- David Quinton wrote: -- On Thu, 12 Mar 2009 10:21:06 +, Julian Lyndon-Smith -- aster...@dotr.com wrote: -- -- -- Has anyone in the UK got ANI to work on an inbound call ? -- -- Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 -- -- -- -- AFAIK (and our E1 doesn't go to * box) -- a) you mean CLI -- -- a) No I don't. CLI is different to ANI -- b) you have to pay BT extra for Calling Line Identity Presentation -- GBP7.50 / qtr on our last bill -- -- See a). We already have CLI. I need ANI ;) -- HTH -- -- -- ___ -- -- 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 -- -- -- -- -- __ -- This email has been scanned by the MessageLabs Email Security System. -- For more information please visit http://www.messagelabs.com/email -- __ -- -- ___ -- -- 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] UK ISDN-30 and ANI
2009/3/13 Andrew Thomas a...@datavox.co.uk I think I understand what you mean now. The biggest difference between CLI and ANI is that ANI can't be blocked/withheld (like you can with CLI by using 141). It also uses different signalling. This is mainly used by law enforcement agencies to trace calls etc. So, you want the number - regardless of what the user dials. I presume you are some sort of 'carrier' then. You'll be lucky to get the information otherwise as it throws up all sorts of privacy laws (ie. you have to have a damn good reason for wanting it). BT are the main people to ask I suppose (unless your calls go through another main carrier). I'm not even sure if ANI signalling is implemented in Asterisk - one for the config file writers ;). Cheers I am sure of one thing that i can do a sip trunk with ANI in our billing system, not sure how it works, but the option is there ;) ___ -- 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] Outbound routing
If it's anything like the UK, it won't make a difference... for example: o2 mobile number ported to orange mobile... On most providers you still pay the o2 rate. three mobile ported to o2... you still pay the three rate (which isn't so good since it's far more expensive than o2). Cheers 2009/3/13 Asterisk aster...@abraxas.si Dear All, I have a small call center in which I have to define least cost routing for outbound calls. For now I have always done this by routing numbers to different providers according to the number prefix. However, a new law became effective now which allows people to switch between providers without changing their telephone numbers. This makes least cost routing based on number prefixes much less effective. Are there any known solutions for this? Thanks in advance, Alex ___ -- 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] Initial silence during call
If there is NAT between the phone and * then that can be responsible. Also, Eyebeam (et al)'s ICE setting causes this. STUN server settings also contribute on eyeBeam. You have to turn off ICE and if you're not using a STUN server check the Use a specified STUN server checkbox while leaving the actual name of the STUN server empty if you are not using one. For some reason they haven't put in an explicit do not use STUN server checkbox. Regards, Mike ___ -- 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] Initial silence during call
I believe it's echo and/or jitter being measured when the call is connected as I recall it being explained. This issue has existed for a long time and I'm not sure there's much you can do about it except to wait for a second before speaking when a call is connected. I think maybe I have trained myself over the years to deal with this by waiting until I actually hear sound on the other end of the line before speaking. For example, when I answer a call, I don't say, Hello until I hear a bit of noise on the channel which takes a second. If it's longer than about a second maybe you have some other issues to deal with. Mike Diehl wrote: Hi all, I've got a problem where many times, there is silence at the first 1-2 seconds of a call. Then it clears up and it's crystal clear. I've not put a sniffer on it, yet, but I suspect that the media channel is still being set up. The server shouldn't be too overloaded. Can anyone give me some advise on how to solve/mitigate this problem? Mike. ___ -- 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] an easy way to deal with/without leading 1 ?
You've had some good suggestions so far but honestly the brute force
method is not that difficult. I have been in the process of trying to
make my dialplan more concise (fewer statements) but haven't tried to do
anything about this one:
; Toll-Free
exten = _1800NXX,1,Macro(dial-avail-tf,${EXTEN})
exten = _1866NXX,1,Macro(dial-avail-tf,${EXTEN})
exten = _1877NXX,1,Macro(dial-avail-tf,${EXTEN})
exten = _1880NXX,1,Macro(dial-avail-tf,${EXTEN})
exten = _1881NXX,1,Macro(dial-avail-tf,${EXTEN})
exten = _1882NXX,1,Macro(dial-avail-tf,${EXTEN})
exten = _1888NXX,1,Macro(dial-avail-tf,${EXTEN})
exten = _800NXX,1,Macro(dial-avail-tf,1${EXTEN})
exten = _866NXX,1,Macro(dial-avail-tf,1${EXTEN})
exten = _877NXX,1,Macro(dial-avail-tf,1${EXTEN})
exten = _880NXX,1,Macro(dial-avail-tf,1${EXTEN})
exten = _881NXX,1,Macro(dial-avail-tf,1${EXTEN})
exten = _882NXX,1,Macro(dial-avail-tf,1${EXTEN})
exten = _888NXX,1,Macro(dial-avail-tf,1${EXTEN})
Benny Amorsen wrote:
sean darcy seandar...@gmail.com writes:
The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes that are toll free.
We try to canonicalize dialled numbers as soon as they enter the
system.
Something like this:
[fromphones]
exten = _XXX,1,Goto(canonical,+1555${EXTEN},1)
exten = _XX,1,Goto(canonical,+1${EXTEN},1)
exten = _00.,1,Goto(canonical,+${EXTEN:2},1)
[canonical]
exten = _+1800XXX,1,...
exten = _+1877XXX,1,...
We don't actually have any US locations yet, so the above is made up
from scratch without any testing.
Whether to use the + to indicate a number in e164 format is a topic of
some debate.
/Benny
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Re: [asterisk-users] Outbound routing
Alex, What country is your call center located? Thanks, Al On Fri, Mar 13, 2009 at 7:36 AM, Asterisk aster...@abraxas.si wrote: Dear All, I have a small call center in which I have to define least cost routing for outbound calls. For now I have always done this by routing numbers to different providers according to the number prefix. However, a new law became effective now which allows people to switch between providers without changing their telephone numbers. This makes least cost routing based on number prefixes much less effective. Are there any known solutions for this? Thanks in advance, Alex ___ -- 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] an easy way to deal with/without leading 1 ?
Cary,
You also forgot 880, 881, 882 although I'm not sure I've ever even come
across one of those.
Cary Fitch wrote:
In my previous reply, I may be wrong, 877 is probably a valid toll free
NPA, add it in the mix.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Thursday, March 12, 2009 7:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] an easy way to deal with/without leading 1 ?
I posted this before, but it didn't show up. So if it's a dup...
I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.
The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes that are toll free.
I tried
exten=exten = _!877NXX,2,Dial(${pstnline}/ww1${EXTEN:-10})
but that matches everything. I'd hoped it would only match strings
that had zero or more characters, followed by the 877 pattern.
sean
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Re: [asterisk-users] Asterisk 1.6.0.7-rc1 Now Available
Did these announcements stop coming on the [asterisk-announce] group? I only seem to get sporadic announcements there. Asterisk Development Team wrote: The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 1.6.0.7, tagged as version 1.6.0.7-rc1. Release candidate 1.6.0.7-rc1 is available for immediate download at http://downloads.digium.com/ In addition to other bug fixes, this release candidate resolves an issue where IMAP voicemail message retrieval and Message Waiting Indication (MWI) would not work properly with the same mailbox name in multiple voicemail contexts. This release also fixes a couple of issues with RFC2833 DTMF, and corrects an issue with compiling on CentOS 64-bit platforms. Issues found in this release candidate can be reported at http://bugs.digium.com/. For a full list of changes in this release candidate, please see the ChangeLog: http://svn.digium.com/view/asterisk/tags/1.6.0.7-rc1/ChangeLog?view=co Thank you for your continued support of 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] Trying to get sample applicationmap to work (*1.4)
Kevin P. Fleming wrote: Mark Michelson wrote: You can work around the bug, although it's not exactly optimal. What you can do is to modify your dialplan as follows: exten = 301,n,Set(DYNAMIC_FEATURES=monkey) Couldn't you just set _DYNAMIC_FEATURES here and have it get automatically inherited to the outbound channel? Yes, that would be another suitable workaround. I sometimes forget that you can let variables inherit like that. Mark Michelson ___ -- 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] Trying to get sample applicationmap to work (*1.4)
David Ruggles wrote: The patch doesn't work for me. Here's what I did: Changed to my asterisk-1.4.23.1 directory Executed the wget / patch command from the link you provided make saw that res_features.so was recompiled Moved /usr/lib/asterisk/modules/res_features.so to res_features.so.old make install Confirmed that res_features.so was recreated in /usr/lib/asterisk/modules asterisk -r -- I never shut asterisk down module unload res_features.so module load res_features.so After this there was no change, it worked using the macro but using the Set(DYN... on the caller only. Thanks, All right. Let's continue this discussion on the bug report I opened. To start with, could you upload console output from an attempt at using the dynamic feature with my patch attached? For the console output, it would help if the verbose and debug levels were both set to at least 4. That way I can hopefully see what the problem is. Thanks. Mark Michelson ___ -- 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] Calling id problem on outgoing call
Umm, I don't think a called number sends any callerid info as there's probably not even a protocol for that. Maybe you need to post a sample CDR. The only thing I could think of is if you are calling an internal extension and asterisk is posting the callerid you have defined for that extension but I've never seen this issue. Artifex Maximus wrote: Hi all! On outgoing call sometimes Asterisk use/give back the caller id sent back by called number instead of number called by me. This is annoying and misleading statistics if other side use some exotic number. For example I have called number 12345678 and CDR include the number 333 as callerid which was sent back by called number/set/switch/whatever. Normally it cannot be an issue but I have found a lot of record like this. How should I change this behavior? I am using a pretty old Asterisk 1.2.26 with zaptel 1.2.22.1 and libpri 1.2.7. Bye, a ___ -- 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] Trying to get sample applicationmap to work (*1.4)
I'm sorry, but it looks like it's working correctly now. I will update the bug if I am able to verify any problems. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson Sent: Friday, March 13, 2009 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4) David Ruggles wrote: The patch doesn't work for me. Here's what I did: Changed to my asterisk-1.4.23.1 directory Executed the wget / patch command from the link you provided make saw that res_features.so was recompiled Moved /usr/lib/asterisk/modules/res_features.so to res_features.so.old make install Confirmed that res_features.so was recreated in /usr/lib/asterisk/modules asterisk -r -- I never shut asterisk down module unload res_features.so module load res_features.so After this there was no change, it worked using the macro but using the Set(DYN... on the caller only. Thanks, All right. Let's continue this discussion on the bug report I opened. To start with, could you upload console output from an attempt at using the dynamic feature with my patch attached? For the console output, it would help if the verbose and debug levels were both set to at least 4. That way I can hopefully see what the problem is. Thanks. Mark Michelson ___ -- 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.10/1996 - Release Date: 03/13/09 05:59:00 ___ -- 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
[asterisk-users] Asterisk to Ericsson MD110 on E1 with ISDN-USR (not QSIG)?
I have been asked by a potential customer whether we can connect an Asterisk box to an Ericsson MD110 that has an E1 port with ISDN-USR. They are unable or unwilling to upgrade their E1 port to QSIG. Has anyone here had experience of successfully making such a connection? I have found a couple of hits on Google that suggest it should work, but I'm after something a little more definitive, based on actual experience, if possible. Can anyone tell me what the USR part of ISDN-USR actually means? Thanks Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org ___ -- 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] an easy way to deal with/without leading 1 ?
The ERC NPAs for Toll free are:
Toll-Free Special Use NPAs
* 800-NXX-
* 888-NXX-
* 877-NXX-
* 866-NXX-
* 855-NXX-
Those are the US toll free NPAs.
The Easily Recognizable Codes (ERC) NPAs have identical last two digits.
NOT 881 for instance, but like, 211, 311, 511, 611, 911, 800, 888, 877, 866,
855, 900,
Original area codes had the middle digit as a 1 or 0, like 512 or 201, but
as they ran out of those, other digits have been allowed, such as 361 or
832.
Authoritative info is available at www.nanpa.com
Cary
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of M Hulber
Sent: Friday, March 13, 2009 8:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] an easy way to deal with/without leading 1 ?
Cary,
You also forgot 880, 881, 882 although I'm not sure I've ever even come
across one of those.
Cary Fitch wrote:
In my previous reply, I may be wrong, 877 is probably a valid toll free
NPA, add it in the mix.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Thursday, March 12, 2009 7:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] an easy way to deal with/without leading 1 ?
I posted this before, but it didn't show up. So if it's a dup...
I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.
The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes that are toll free.
I tried
exten=exten = _!877NXX,2,Dial(${pstnline}/ww1${EXTEN:-10})
but that matches everything. I'd hoped it would only match strings
that had zero or more characters, followed by the 877 pattern.
sean
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[asterisk-users] No reply to our critical packet
Hi, I’ve installed Asterisk for use as a SIP server. I can call people, but one strange thing happens: if I call someone with a SIP account outside my server (for example, sip:enum-echo-t...@sip.nemox.net) everything is fine, if I call any Asterisk extension it also works, but the call gets disconnected in about 20 seconds. To be exact, audio is turned off but the SIP client still thinks it’s connected. Logs say “no reply to our critical packet”. tcpdump shows that the packet does arrive at the destination. sip set debug shows this is what the packet contains: Retransmitting #6 (NAT) to 77.239.189.223:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 77.239.189.223;branch=z9hG4bK-d8754z-db899ced94cc7fd3-1---d8754z-;received=77.239.189.223 From: Romasip:r...@qwertty.com;transport=UDP;tag=01785d5e To: sip:e...@qwertty.com;transport=UDP;tag=as068592d2 Call-ID: ZTkzNjYxNzZmOWMzY2ZhOTdjMWIwYTEwZTYxZmUyZTY. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:e...@78.46.49.80 Content-Type: application/sdp Content-Length: 285 v=0 o=root 25952 25952 IN IP4 78.46.49.80 s=session c=IN IP4 78.46.49.80 t=0 0 m=audio 30606 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv There’s NAT: computer (192.168.1.2) behind a router (77.239.189.223), the server (78.46.49.80) doesn’t have any NAT. I have even set DMZ host to 192.168.1.2, so I’m sure all packets reach it. As far as I understand, Asterisk expects the SIP client to reply to that packet with an ACK, the client receives the packet but does not reply. What have I configured incorrectly? In sip.conf I have nat=yes (otherwise I don’t hear anything), whatever I do with NAT settings of SIP clients does not help. Maybe there’s something wrong with the headers of the packet that makes the client think the packet is misaddressed? Twinkle says, “you have the following registrations sip:r...@192.168.1.2” while I’d expect sip:r...@qwertty.com. So how do I make sure the client sends its ACK? -- TIA Roman. smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Trying to get sample applicationmap to work (*1.4)
David Ruggles wrote: I'm sorry, but it looks like it's working correctly now. I will update the bug if I am able to verify any problems. Thanks, Heh, no reason to be sorry for it working :) When you say it works now, was this with or without the patch applied? Mark Michelson ___ -- 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] VoIP Users Conference today at 12 Noon EDT
The USA is on DST now, but Europe is not. If you are in Europe, be aware that the VoIP Users Conference conference will start one hour early today. In Paris, that translates to GMT+1 or 5PM, in the UK 4PM. Grand Central is about to be re-branded as Google Voice. http://www.google.com/voice Changes should be announced soon. I logged in but see no difference yet. FWIW, Google says it'll still be FREE. Fred Tweeted an interesting asterisk)related story: http://bit.ly/ddablocksvoip using Asterisk to get Parking Space Availability from Ann Arbor garages. The response from the VoIP community was fantastic! We received great comments and feedback from people like Jason Goecke, Dug Song, Dave Michels, Evan Cooke, and more! People not only responded, they even showed different ways of providing access to this information. And everyone shared their work in an open forum — truly a great example of open source coding inspiring innovation (albeit with Parking Spaces). Then the idiots in the government decided to block access! Read the link above if you want to learn more. Don't forget if you are in Europe in May, AMOOCON in Rostock, Germany with Mark Spencer, Kevin Fleming, Jim Van Meggelen and a bunch of other people worth meeting. More on AMOOCON at http://amoocon.de/ Enjoy, it's spring! /r ___ -- 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] No reply to our critical packet
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky Sent: Friday, March 13, 2009 9:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] No reply to our critical packet Hi, I've installed Asterisk for use as a SIP server. I can call people, but one strange thing happens: if I call someone with a SIP account outside my server (for example, sip:enum-echo-t...@sip.nemox.net) everything is fine, if I call any Asterisk extension it also works, but the call gets disconnected in about 20 seconds. To be exact, audio is turned off but the SIP client still thinks it's connected. Logs say no reply to our critical packet. tcpdump shows that the packet does arrive at the destination. sip set debug shows this is what the packet contains: Retransmitting #6 (NAT) to 77.239.189.223:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 77.239.189.223;branch=z9hG4bK-d8754z-db899ced94cc7fd3-1---d8754z-;received=7 7.239.189.223 From: Romasip:r...@qwertty.com;transport=UDP;tag=01785d5e To: sip:e...@qwertty.com;transport=UDP;tag=as068592d2 Call-ID: ZTkzNjYxNzZmOWMzY2ZhOTdjMWIwYTEwZTYxZmUyZTY. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:e...@78.46.49.80 Content-Type: application/sdp Content-Length: 285 v=0 o=root 25952 25952 IN IP4 78.46.49.80 s=session c=IN IP4 78.46.49.80 t=0 0 m=audio 30606 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv There's NAT: computer (192.168.1.2) behind a router (77.239.189.223), the server (78.46.49.80) doesn't have any NAT. I have even set DMZ host to 192.168.1.2, so I'm sure all packets reach it. As far as I understand, Asterisk expects the SIP client to reply to that packet with an ACK, the client receives the packet but does not reply. What have I configured incorrectly? In sip.conf I have nat=yes (otherwise I don't hear anything), whatever I do with NAT settings of SIP clients does not help. Maybe there's something wrong with the headers of the packet that makes the client think the packet is misaddressed? Twinkle says, you have the following registrations sip:r...@192.168.1.2 while I'd expect sip:r...@qwertty.com. So how do I make sure the client sends its ACK? -- TIA Roman. -- Two thoughts (both could be wrong) 1. Do you have the incoming 1-2 holes in your firewall so the remote server can get it's reply back to *? 2. If #1 is ok, try putting an Answer command in front of your Dial Command. Danny Nicholas ___ -- 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] Trying to get sample applicationmap to work (*1.4)
It was with the patch applied, but after I restarted asterisk. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson Sent: Friday, March 13, 2009 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4) David Ruggles wrote: I'm sorry, but it looks like it's working correctly now. I will update the bug if I am able to verify any problems. Thanks, Heh, no reason to be sorry for it working :) When you say it works now, was this with or without the patch applied? Mark Michelson ___ -- 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 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.11.10/1996 - Release Date: 03/13/09 05:59:00 ___ -- 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
[asterisk-users] Ast/Hyla/IAX Scalability?
Hello everyone- I recently read the thread entitled Faxing Success Rate on PRI which dealt with Asterisk/HylaFax/IAXmodem. I'm successfully using this 'recipe' in a few instances on systems with only a couple of analog lines all the way up to a full PRI worth of Iaxmodems. However, I'm finding that I'll need to scale upwards in the coming months and would like to know if anyone has experience with a system containing more than 1 PRI, 5+ PRIs or even a full DS3? Are there any limitations of the Asterisk/Hylafax/Iaxmodem set that would prevent it from performing on a scale of this magnitude? A system like this would obviously be fault tolerant and have backup mechanisms in place, but I'm simply wondering if it would be possible to use this much connectivity on a single platform. I've also noticed that IAXmodem is compiled statically against a version of spandsp included with the iaxmodem source. For a large installation, would it be better to compile iaxmodem dynamically to reduce the per-instance size of each iaxmodem? Or, would it be better to simply throw more RAM at it? Are there any concurrency issues when receiving a large number of faxes on a system using IAXmodems? I can only assume the system load would increase in a linear fashion for each active/inuse iaxmodem on the system, not including the addtional processing of faxrcvd/FaxDispatch after reception. All ideas/thoughts/experiences/etc welcome! Marshall ___ -- 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] No reply to our critical packet
Hi, thanks for the quick reply. 1. Do you have the incoming 1-2 holes in your firewall so the remote server can get it's reply back to *? This was what I checked first. Both firewalls let everything through. 2. If #1 is ok, try putting an Answer command in front of your Dial Command. Doesn’t help, alas. Also, it works the same (disconnect after 20 seconds) both for Dial and Echo, regardless of presence of Answer. -- TIA Roman. smime.p7s Description: S/MIME cryptographic signature ___ -- 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] No reply to our critical packet
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky Sent: Friday, March 13, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No reply to our critical packet Hi, thanks for the quick reply. 1. Do you have the incoming 1-2 holes in your firewall so the remote server can get it's reply back to *? This was what I checked first. Both firewalls let everything through. 2. If #1 is ok, try putting an Answer command in front of your Dial Command. Doesn't help, alas. Also, it works the same (disconnect after 20 seconds) both for Dial and Echo, regardless of presence of Answer. -- TIA Roman. Next Step would be to check/update the firmware on your phones or router. Regards, Danny Nicholas ___ -- 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] Ast/Hyla/IAX Scalability?
On Fri, Mar 13, 2009 at 11:38 AM, Marshall Henderson marshall...@gmail.com wrote: I recently read the thread entitled Faxing Success Rate on PRI which dealt with Asterisk/HylaFax/IAXmodem. I'm successfully using this 'recipe' in a few instances on systems with only a couple of analog lines all the way up to a full PRI worth of Iaxmodems. Then you have probably seen that YMMD, and that some people claim great success with VoIP fax. Other people claim that the only way to go is a hardware fax solution, like the dedicated multi-modem fax cards. The only way you're going to find a solution that will work for you is to try it, scale it, build your own expertise with your solution, load test it, and watch your error rate. The other consideration is your budget and your cost of dropping a fax. The faxmodem cards are not cheap compared to a voip solution. But if the faxes have a high value to the business the hardware cards are probably justified. Again, you'll find people arguing that their voip solution has as low of a failure rate as a hardware solution. I'm jealous. My voip fax solution does not yet have that low of a failure rate, but I'm hopefully getting closer to working out the last bugs. I've also noticed that IAXmodem is compiled statically against a version of spandsp included with the iaxmodem source. For a large installation, would it be better to compile iaxmodem dynamically to reduce the per-instance size of each iaxmodem? Or, would it be better to simply throw more RAM at it? I'm not sure what difference RAM makes. What breaks a fax on voip is latency and dropped packets. You solve both of those problems if you go the hardware solution route with a faxmodem card. The in-between solution is using a proprietary telco - fax gateway, like a Cisco box that terminates a PRI and provides FXO ports that you plug into a single-pair faxmodem or a 'real' fax machine. That solution quickly becomes ridiculous when you try to scale it. Are there any concurrency issues when receiving a large number of faxes on a system using IAXmodems? File system contention, but fax files aren't very large, and I would call that a non-issue. Most people don't want a piece of paper; they want a PDF that they can turn into paper once in a while. ___ -- 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] BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai
2009/3/12 Paulo Santos paulo.r.san...@sapo.pt: Gavin Henry wrote: Hi All, We've got msidn configured: Port 1: TE-mode BRI S/T interface line (for phone lines) - Protocol: DSS1 (Euro ISDN) - childcnt: 2 I don't know if it depends on the card, but in my case I need to set the termination jumper on TE mode for lines from PSTN. Mind to check the TE/NT jumper as well. te_ptmp=1 (...) [isdn] ports=1 context=from-pstn msns=* Everything worked first time, so 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
Re: [asterisk-users] BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai
2009/3/12 Giorgio Incantalupo gincantal...@fgasoftware.com: Hi Gavin, if you can make and receive calls it works...do not worry if your line is shown as DOWN, some telco turns it off but it works without problem. Remember to ask your telco for the right signalling and set it the right way (PTP or PMP). Thanks. It's all working with above, I just hadn't tested an inbound call. Pretty lucky really ;-) ___ -- 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 Ericsson MD110 on E1 with ISDN-USR (not QSIG)?
Tony Mountifield wrote: I have been asked by a potential customer whether we can connect an Asterisk box to an Ericsson MD110 that has an E1 port with ISDN-USR. They are unable or unwilling to upgrade their E1 port to QSIG. Has anyone here had experience of successfully making such a connection? I have found a couple of hits on Google that suggest it should work, but I'm after something a little more definitive, based on actual experience, if possible. Can anyone tell me what the USR part of ISDN-USR actually means? Thanks Tony I would assume that USR is the user side of the connection, where the other side is the NET. The * term is CPE. But that does not describe which protocol they are using (EuroISDN, QSIG, etc..). But thats only a guess. Best 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
Re: [asterisk-users] No reply to our critical packet
On Friday, 13.03.2009 17:50:57 Danny Nicholas wrote: Next Step would be to check/update the firmware on your phones or router. I don’t think the router is to blame, it does deliver all the packets. And there are no hardware phones, only numerous software SIP clients. Which (GNU/Linux) software clients are known to have maximum compatibility with Asterisk? -- TIA Roman. smime.p7s Description: S/MIME cryptographic signature ___ -- 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] No reply to our critical packet
I have the same situation. My scenario is weird: I have a DID with IPkall that points to my asterisk server, and I have it play a message with Playback() after about 20 seconds call drops and give me the same message you get: no reply to our critical packet BUT I have a DID from Vitelity, and that one works fine no drops. I have no idea why. On Fri, Mar 13, 2009 at 12:37 PM, Roman Odaisky r...@qwertty.com wrote: On Friday, 13.03.2009 17:50:57 Danny Nicholas wrote: Next Step would be to check/update the firmware on your phones or router. I don’t think the router is to blame, it does deliver all the packets. And there are no hardware phones, only numerous software SIP clients. Which (GNU/Linux) software clients are known to have maximum compatibility with Asterisk? -- TIA Roman. ___ -- 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] Trying to get sample applicationmap to work (*1.4)
David Ruggles wrote: It was with the patch applied, but after I restarted asterisk. Thanks, Fix committed to Asterisk 1.4 in revision 181990. Mark Michelson ___ -- 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] No reply to our critical packet
--- On Fri, 3/13/09, Pascal Bruno tipas...@gmail.com wrote: I have the same situation. My scenario is weird: Well, I've experienced the same symptoms but in a whole different context. It's happening in my LAN (no firewalls, no NAT) and only with specific IP phones + early dial + pedantic=yes. http://bugs.digium.com/view.php?id=14652 I guess it's the client's fault in this case. ___ -- 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] Recording calls and SLA
I'm trying to record all calls including calls that are part of a SLA. Using both monitor and mixmonitor the recording appears to happen (that is, asterisk logging shows it happening) however the file is never written to. It doesn't seem to be possible to use the recording that is part of the meetme app because when the SLA app creates the conference it passes NULL in the options field (discovered this reading the source). Has anyone encountered similar issues? My next thought is to use ztmonitor to record the channel directly but this would be a bit of a hack that I'm not excited about. Thanks, Norbert ___ -- 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] No reply to our critical packet
I had this issue with SIP Phones (Cisco 7961) to local voicemail; the issue was resolved by adding a Ringing() followed by Wait(1) before the VoicemailMain() in the dial plan... it seems like there should be a better way, and I feel it's rather crude to force the user to listen to a second of ringback before launching into voicemail, but it solved the problem for me (and yes, I did try just Ringing() with no wait with no such luck) -- maybe it would work in your case with a SIP call as well? HTH, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 | F: 440.729.0884 | I:http://www.thecontrolworks.com/ Crestron Authorized Independent Progammers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky Sent: Friday, March 13, 2009 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No reply to our critical packet Hi, thanks for the quick reply. 1. Do you have the incoming 1-2 holes in your firewall so the remote server can get it's reply back to *? This was what I checked first. Both firewalls let everything through. 2. If #1 is ok, try putting an Answer command in front of your Dial Command. Doesn't help, alas. Also, it works the same (disconnect after 20 seconds) both for Dial and Echo, regardless of presence of Answer. -- TIA Roman. ___ -- 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] No reply to our critical packet
Not a better hack but perhaps more palatable to the listener Playback(please-wait) Wait(1) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lincoln King-Cliby Sent: Friday, March 13, 2009 1:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] No reply to our critical packet I had this issue with SIP Phones (Cisco 7961) to local voicemail; the issue was resolved by adding a Ringing() followed by Wait(1) before the VoicemailMain() in the dial plan... it seems like there should be a better way, and I feel it's rather crude to force the user to listen to a second of ringback before launching into voicemail, but it solved the problem for me (and yes, I did try just Ringing() with no wait with no such luck) -- maybe it would work in your case with a SIP call as well? HTH, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 | F: 440.729.0884 | I:http://www.thecontrolworks.com/ Crestron Authorized Independent Progammers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky Sent: Friday, March 13, 2009 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No reply to our critical packet Hi, thanks for the quick reply. 1. Do you have the incoming 1-2 holes in your firewall so the remote server can get it's reply back to *? This was what I checked first. Both firewalls let everything through. 2. If #1 is ok, try putting an Answer command in front of your Dial Command. Doesn't help, alas. Also, it works the same (disconnect after 20 seconds) both for Dial and Echo, regardless of presence of Answer. -- TIA Roman. ___ -- 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] Ast/Hyla/IAX Scalability?
On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg dbackeb...@gmail.com wrote: On Fri, Mar 13, 2009 at 11:38 AM, Marshall Henderson marshall...@gmail.com wrote: I recently read the thread entitled Faxing Success Rate on PRI which dealt with Asterisk/HylaFax/IAXmodem. I'm successfully using this 'recipe' in a few instances on systems with only a couple of analog lines all the way up to a full PRI worth of Iaxmodems. Then you have probably seen that YMMD, and that some people claim great success with VoIP fax. Other people claim that the only way to go is a hardware fax solution, like the dedicated multi-modem fax cards. The only way you're going to find a solution that will work for you is to try it, scale it, build your own expertise with your solution, load test it, and watch your error rate. I certainly understand the value of building the solution, testing, patching, and fixing problems as they arise. It was my hope however that others would have large-scale experience with these technologies and could share some pointers. I'm about to perform some bulk testing between two servers to see how the system reacts. I'm more than happy to post my findings here if anyone has interest. The other consideration is your budget and your cost of dropping a fax. The faxmodem cards are not cheap compared to a voip solution. But if the faxes have a high value to the business the hardware cards are probably justified. Again, you'll find people arguing that their voip solution has as low of a failure rate as a hardware solution. I'm jealous. My voip fax solution does not yet have that low of a failure rate, but I'm hopefully getting closer to working out the last bugs. Do you have any specifics to share about the problems you're finding? I've also noticed that IAXmodem is compiled statically against a version of spandsp included with the iaxmodem source. For a large installation, would it be better to compile iaxmodem dynamically to reduce the per-instance size of each iaxmodem? Or, would it be better to simply throw more RAM at it? I'm not sure what difference RAM makes. What breaks a fax on voip is latency and dropped packets. Agreed. I was simply inquiring about the efficiency of IAXmodem at the system resource level. Latency and packet drops will be minimal or nonexistent at all in this environment. You solve both of those problems if you go the hardware solution route with a faxmodem card. I've found hardware fax boards aren't a 100% fix either. Many of the boards are buggy. However, I will have to say that certain manufacturers like Mainpine are near 100%. The in-between solution is using a proprietary telco - fax gateway, like a Cisco box that terminates a PRI and provides FXO ports that you plug into a single-pair faxmodem or a 'real' fax machine. That solution quickly becomes ridiculous when you try to scale it. Are there any concurrency issues when receiving a large number of faxes on a system using IAXmodems? File system contention, but fax files aren't very large, and I would call that a non-issue. Most people don't want a piece of paper; they want a PDF that they can turn into paper once in a while. The purpose of such a system as I'm inquiring about is for digital archival. Very little 'paper' will be in use. Buffering aside, each fax could be written at the speed at which it is received correct? So, if I'm receiving 50 faxes at 14.4kbps each, assuming a direct receive frame--block write, I'd be looking at roughly 90KBps written to disk. Is my logic sound here? Thank you for the response and ideas. Marshall ___ -- 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] No reply to our critical packet
My best guess at the root cause of the problem after looking at the packet capture was that the phone was not happy seeing the call connected before any of the intermediate states (trying, ringing, etc.) and Ringing() generated the session progress (e.g. in addition to the in-band ringback it also generates the SIP message to tell the phone that the phone is ringing... or maybe it just generates the SIP message and the phone generates the ringback) necessary to make the phone happy; I don't think Playback() does the same thing. -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 | F: 440.729.0884 | I:http://www.thecontrolworks.com/ Crestron Authorized Independent Progammers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, March 13, 2009 2:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] No reply to our critical packet Not a better hack but perhaps more palatable to the listener Playback(please-wait) Wait(1) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lincoln King-Cliby Sent: Friday, March 13, 2009 1:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] No reply to our critical packet I had this issue with SIP Phones (Cisco 7961) to local voicemail; the issue was resolved by adding a Ringing() followed by Wait(1) before the VoicemailMain() in the dial plan... it seems like there should be a better way, and I feel it's rather crude to force the user to listen to a second of ringback before launching into voicemail, but it solved the problem for me (and yes, I did try just Ringing() with no wait with no such luck) -- maybe it would work in your case with a SIP call as well? HTH, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 | F: 440.729.0884 | I:http://www.thecontrolworks.com/ Crestron Authorized Independent Progammers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky Sent: Friday, March 13, 2009 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No reply to our critical packet Hi, thanks for the quick reply. 1. Do you have the incoming 1-2 holes in your firewall so the remote server can get it's reply back to *? This was what I checked first. Both firewalls let everything through. 2. If #1 is ok, try putting an Answer command in front of your Dial Command. Doesn't help, alas. Also, it works the same (disconnect after 20 seconds) both for Dial and Echo, regardless of presence of Answer. -- TIA Roman. ___ -- 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] No reply to our critical packet
Correct you are. Playback just plays a file back to the caller, Ringing sends a ringing to over the channel (to the user). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lincoln King-Cliby Sent: Friday, March 13, 2009 1:42 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] No reply to our critical packet My best guess at the root cause of the problem after looking at the packet capture was that the phone was not happy seeing the call connected before any of the intermediate states (trying, ringing, etc.) and Ringing() generated the session progress (e.g. in addition to the in-band ringback it also generates the SIP message to tell the phone that the phone is ringing... or maybe it just generates the SIP message and the phone generates the ringback) necessary to make the phone happy; I don't think Playback() does the same thing. -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 | F: 440.729.0884 | I:http://www.thecontrolworks.com/ Crestron Authorized Independent Progammers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, March 13, 2009 2:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] No reply to our critical packet Not a better hack but perhaps more palatable to the listener Playback(please-wait) Wait(1) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lincoln King-Cliby Sent: Friday, March 13, 2009 1:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] No reply to our critical packet I had this issue with SIP Phones (Cisco 7961) to local voicemail; the issue was resolved by adding a Ringing() followed by Wait(1) before the VoicemailMain() in the dial plan... it seems like there should be a better way, and I feel it's rather crude to force the user to listen to a second of ringback before launching into voicemail, but it solved the problem for me (and yes, I did try just Ringing() with no wait with no such luck) -- maybe it would work in your case with a SIP call as well? HTH, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC V: 440.729.4640 x1107 | F: 440.729.0884 | I:http://www.thecontrolworks.com/ Crestron Authorized Independent Progammers -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roman Odaisky Sent: Friday, March 13, 2009 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No reply to our critical packet Hi, thanks for the quick reply. 1. Do you have the incoming 1-2 holes in your firewall so the remote server can get it's reply back to *? This was what I checked first. Both firewalls let everything through. 2. If #1 is ok, try putting an Answer command in front of your Dial Command. Doesn't help, alas. Also, it works the same (disconnect after 20 seconds) both for Dial and Echo, regardless of presence of Answer. -- TIA Roman. ___ -- 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 ___ -- 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] Ast/Hyla/IAX Scalability?
Marshall Henderson wrote: Hello everyone- I recently read the thread entitled Faxing Success Rate on PRI which dealt with Asterisk/HylaFax/IAXmodem. I'm successfully using this 'recipe' in a few instances on systems with only a couple of analog lines all the way up to a full PRI worth of Iaxmodems. I've never gone beyond 23 instances of iaxmodem on our fax server, but the load on those 23 active (Outgoing faxes) wasn't very high at all. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ast/Hyla/IAX Scalability?
Hi, I know i doesn't make practical difference, but often it is the far end that is atually buggy, not out end. A lot of the work in spandsp to increase success rate is to do with workarounds for issues in the remote machine, Steve On 3/13/09, Marshall Henderson marshall...@gmail.com wrote: On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg dbackeb...@gmail.com wrote: On Fri, Mar 13, 2009 at 11:38 AM, Marshall Henderson marshall...@gmail.com wrote: I recently read the thread entitled Faxing Success Rate on PRI which dealt with Asterisk/HylaFax/IAXmodem. I'm successfully using this 'recipe' in a few instances on systems with only a couple of analog lines all the way up to a full PRI worth of Iaxmodems. Then you have probably seen that YMMD, and that some people claim great success with VoIP fax. Other people claim that the only way to go is a hardware fax solution, like the dedicated multi-modem fax cards. The only way you're going to find a solution that will work for you is to try it, scale it, build your own expertise with your solution, load test it, and watch your error rate. I certainly understand the value of building the solution, testing, patching, and fixing problems as they arise. It was my hope however that others would have large-scale experience with these technologies and could share some pointers. I'm about to perform some bulk testing between two servers to see how the system reacts. I'm more than happy to post my findings here if anyone has interest. The other consideration is your budget and your cost of dropping a fax. The faxmodem cards are not cheap compared to a voip solution. But if the faxes have a high value to the business the hardware cards are probably justified. Again, you'll find people arguing that their voip solution has as low of a failure rate as a hardware solution. I'm jealous. My voip fax solution does not yet have that low of a failure rate, but I'm hopefully getting closer to working out the last bugs. Do you have any specifics to share about the problems you're finding? I've also noticed that IAXmodem is compiled statically against a version of spandsp included with the iaxmodem source. For a large installation, would it be better to compile iaxmodem dynamically to reduce the per-instance size of each iaxmodem? Or, would it be better to simply throw more RAM at it? I'm not sure what difference RAM makes. What breaks a fax on voip is latency and dropped packets. Agreed. I was simply inquiring about the efficiency of IAXmodem at the system resource level. Latency and packet drops will be minimal or nonexistent at all in this environment. You solve both of those problems if you go the hardware solution route with a faxmodem card. I've found hardware fax boards aren't a 100% fix either. Many of the boards are buggy. However, I will have to say that certain manufacturers like Mainpine are near 100%. The in-between solution is using a proprietary telco - fax gateway, like a Cisco box that terminates a PRI and provides FXO ports that you plug into a single-pair faxmodem or a 'real' fax machine. That solution quickly becomes ridiculous when you try to scale it. Are there any concurrency issues when receiving a large number of faxes on a system using IAXmodems? File system contention, but fax files aren't very large, and I would call that a non-issue. Most people don't want a piece of paper; they want a PDF that they can turn into paper once in a while. The purpose of such a system as I'm inquiring about is for digital archival. Very little 'paper' will be in use. Buffering aside, each fax could be written at the speed at which it is received correct? So, if I'm receiving 50 faxes at 14.4kbps each, assuming a direct receive frame--block write, I'd be looking at roughly 90KBps written to disk. Is my logic sound here? Thank you for the response and ideas. Marshall ___ -- 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 -- Sent from my mobile device ___ -- 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] No reply to our critical packet
Ringing() followed by Wait(1) I made it exten = echo,1,Ringing() exten = echo,2,Wait(1) exten = echo,3,Playback(abandon-all-hope) exten = echo,4,Echo() to no avail. This looks like a client issue, though all of my clients fail. Which clients are the most standards conforming? Also, maybe Asterisk isn’t what I need? I need a server with which several people would register accounts, they’ll be able to place calls among themselves and also call other SIP accounts, as well as calling PSTN numbers using predefined accounts (so that it would be possible to share one paid account between several users). So a SIP server + possibility for dialplans through 3rd party SIP servers. Maybe something like SER would suffice? Or SER as a proxy in front of Asterisk is the way to go? -- TIA Roman. smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Ast/Hyla/IAX Scalability?
On Fri, 13 Mar 2009, David Backeberg wrote: [various snippage] Again, you'll find people arguing that their voip solution has as low of a failure rate as a hardware solution. I'm jealous. My voip fax solution does not yet have that low of a failure rate, but I'm hopefully getting closer to working out the last bugs. The proposed platform being enclosed in one machine should make it as good as a hardware solution IMO. Certainly cheaper. Don't have to worry about a board frying and taking the whole service down. I've also noticed that IAXmodem is compiled statically against a version of spandsp included with the iaxmodem source. For a large installation, would it be better to compile iaxmodem dynamically to reduce the per-instance size of each iaxmodem? Or, would it be better to simply throw more RAM at it? I'm not sure what difference RAM makes. What breaks a fax on voip is latency and dropped packets. There won't be any latency or dropped packets because there are no physical network links involved, though if the machine becomes too loaded or starts swapping for lack of RAM, more RAM would make all the difference. RAM is so cheap anyway you may as well load it up! I don't think compiling dynamically would save enough RAM to make a difference anyway... a few MB? My iaxmodems seem to reserve 3MB each, and their binary size is only .5MB, so some piece of that .5MB could be saved per instance. The upper limit saved over 45 instances would be something less than 23MB then... Someone will probably point out the error in my calculations, though :) Are there any concurrency issues when receiving a large number of faxes on a system using IAXmodems? File system contention, but fax files aren't very large, and I would call that a non-issue. Most people don't want a piece of paper; they want a PDF that they can turn into paper once in a while. File system contention won't break faxes, though, as at that point the fax is already received. I think at some point CPU contention will start to appear, as each instance is doing some hefty (?) DSP work. Just guessing though. I run this same setup, but have never seen more than four or five at once. Damn stable and error free though! I lose about one page in a thousand, and that seems to always be the same senders, so I suspect a prob on their end. j ___ -- 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] work around the 64 pickupgroups limit
At 16:10 3/10/2009, Matt Riddell wrote:
On 7/03/2009 4:58 a.m., Klaus Darilion wrote:
Hi!
What are the typical ways to work around the 64 groups limit?
What we actually do is store a pickup group with a caller id.
So the AsteriskDB has ${DB/pickup/${CALLERID(num)}} and we set
pickupmark to the same.
That way when someone dials 29 (what we use for pickup) it just checks
that group - no limitations on number of groups that way.
Hey Matt,
Would share some config file code with us?
___
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Re: [asterisk-users] AGX Asterisk Addon - Can't find app_fax.c withspandsp-0.0.4
2009/3/13 Andrew Thomas a...@datavox.co.uk You now need to compile and install SpanDSP-0.0.6pre3 at least (AGX has been changed). Yes, you're right but I thought that to compile with AGX Asterisk Addon with either 0.0.4 or 0.0.5 or 0.0.6 spandsp version, you should just just have to edit the previously mentioned file. Is this true ? After you've done that - try AGX again. HTH -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: 11 March 2009 06:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] AGX Asterisk Addon - Can't find app_fax.c withspandsp-0.0.4 Hi, I've installed spandsp-0.0.4pre16 With this: cd ~ svn co https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addonsagx-ast-addons cd agx-ast-addons/trunk ./build.sh I've got this: CMake Error in spandsp-0.0.4/CMakeLists.txt: Cannot find source file app_fax.c. Tried extensions .c .C .c++ .cc .cpp .cxx .m .M .mm .h .hh .h++ .hm .hpp .hxx .in .txx This is coherent with : Aagx-ast-addons/trunk/spandsp-0.0.4 Aagx-ast-addons/trunk/spandsp-0.0.4/app_rxfax.c Aagx-ast-addons/trunk/spandsp-0.0.4/app_txfax.c Aagx-ast-addons/trunk/spandsp-0.0.4/CMakeLists.txt Aagx-ast-addons/trunk/spandsp-0.0.4/README My CMake knowledge is too short to propose a workaround. Maybe, this could come from a change in agx-ast-addons/trunk/spandsp-0.0.4/CMakeLists.txt as the content of this file is: project (app-fax-span4) # -- # Target # we use MODULE cause it build a shared object module # -- ADD_LIBRARY(app_fax MODULE app_fax.c) # # We remove the lib prefix from the libmodule.so filename # SET_TARGET_PROPERTIES(app_fax PROPERTIES PREFIX ) # # We add library dependencies to use those modules # TARGET_LINK_LIBRARIES(app_fax spandsp tiff) # # override default INSTALL rules # INSTALL(TARGETS app_fax DESTINATION lib/asterisk/modules) Could you help ? 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 ___ -- 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] SendFAX/T.38 question
I have some questions about the T.38 faxing capability. I have been able to successfully setup the inbound receive fax. However, I am having problems tracking down the format of the outbound extensions.conf SendFAX command. I have looked at the code and it looks like it only takes a single parameter, a file name. But the attempts I have tried seem unsucessful. I have tried dialing out and then calling SendFAX and calling SendFAX before the dial. No success. Can someone please provide me with an extensions.conf example of how to use SendFAX? Thank you. Jonathan Augenstine ___ -- 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] Realtime dialplan application versus REALTIME dialplan function
Hi All,
I'm upgrading some PBX's from 1.2 to 1.4 and having a bit of trouble with
converting the Realtime application to the REALTIME function. I have the
method down and understand simplistically what is going on, at least enough
to get my old 1.2 apps to run in 1.4 functions. I do not understand why
change from the app to the func? What the benefits?
To me, the app seemed so elegant with appending a variable name to each
extracted data field within a row. Really convenient and easy, one priority
in the dialplan and all data is extracted and easily used further down the
line.
Initial take on the function is increased priorities in the dialplan to
extract the data then cut it up into specified variable, then cut the
resulting variable into further bits to get the usable data into another
variable so it can be used for the real work.
For example here is the 1.2 dialplan:
exten = 326,1,Realtime(cfwd|exten|${EXTEN}|cf_)
exten = 326,n,GotoIf($[${cf_active} = yes ]?:326|20)
exten = 326,n,Goto(cfaccess,${cf_cfnum},1)
Here is the 1.4 dialplan to accomplish the same thing:
exten = 326,1,Set(row=${REALTIME(cfwd,exten,${EXTEN})})
exten = 326,n,Set(column3=${CUT(row,|,3)})
exten = 326,n,Set(column4=${CUT(row,|,4)})
exten = 326,n,Set(cf_cfnum=${CUT(column3,=,2)})
exten = 326,n,Set(cf_active=${CUT(column4,=,2)})
exten = 326,n,GotoIf($[${cf_active} = no ]?:326|20)
exten = 326,n,Goto(cfaccess,${cf_cfnum},1)
So I have in mind that maybe the function is a bit more versatile than the
old app, but I don't really see it just yet.
Can anyone shed some light on this and expound on the benefits or reasoning
behind the switch in the application usage?
Thanks.
JR
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Re: [asterisk-users] Realtime dialplan application versus REALTIME dialplan function
On Friday 13 March 2009 17:41:42 JR Richardson wrote: I'm upgrading some PBX's from 1.2 to 1.4 and having a bit of trouble with converting the Realtime application to the REALTIME function. I have the method down and understand simplistically what is going on, at least enough to get my old 1.2 apps to run in 1.4 functions. I do not understand why change from the app to the func? What the benefits? The specific reason was one of interface consistency. Dialplan applications do specific things with the channel, whereas functions are designed to either retrieve or set information on the channel. Applications may be interactive with the telephone user; functions should never be. You are not the first person to have trouble with this conversion. There are two new functions, REALTIME_FIELD() and REALTIME_HASH(), which will be in 1.6.2, which return some of this functionality. REALTIME_FIELD() permits the retrieval of any single field within a realtime database, while REALTIME_HASH() retrieves an entire record, though it encapsulates it into a container, to avoid possible clashes with existing channel variables. -- Tilghman ___ -- 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] TRANSFER EVENT ON QUEUE_LOG
Hi, Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6. I make an attended transfer (asterisk feature), and I cant see the event. Any idea? Should I submit a bug report? ___ -- 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] TRANSFER EVENT ON QUEUE_LOG
Sebastian wrote: Hi, Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6. I make an attended transfer (asterisk feature), and I cant see the event. Any idea? Should I submit a bug report? If you do, be sure to headline it in all caps. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] TRANSFER EVENT ON QUEUE_LOG
Forget about this. Is still working. From: Sebastian [mailto:s...@adinet.com.uy] Sent: viernes, 13 de marzo de 2009 10:05 p.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: TRANSFER EVENT ON QUEUE_LOG Hi, Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6. I make an attended transfer (asterisk feature), and I cant see the event. Any idea? Should I submit a bug report? ___ -- 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] TRANSFER EVENT ON QUEUE_LOG
I made another post, it is working, I have queue_log to mysql db and I have a trigger that made the insert fail. Sorry for the post!. Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: viernes, 13 de marzo de 2009 10:14 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TRANSFER EVENT ON QUEUE_LOG Sebastian wrote: Hi, Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6. I make an attended transfer (asterisk feature), and I cant see the event. Any idea? Should I submit a bug report? If you do, be sure to headline it in all caps. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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 Se certificó que el correo entrante no contiene virus. Comprobada por AVG - www.avg.es Versión: 8.5.278 / Base de datos de virus: 270.11.13/1999 - Fecha de la versión: 03/13/09 05:59:00 ___ -- 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
Re: [asterisk-users] work around the 64 pickupgroups limit
On 14/03/2009 10:29 a.m., Doug wrote:
At 16:10 3/10/2009, Matt Riddell wrote:
On 7/03/2009 4:58 a.m., Klaus Darilion wrote:
Hi!
What are the typical ways to work around the 64 groups limit?
What we actually do is store a pickup group with a caller id.
So the AsteriskDB has ${DB/pickup/${CALLERID(num)}} and we set
pickupmark to the same.
That way when someone dials 29 (what we use for pickup) it just checks
that group - no limitations on number of groups that way.
Hey Matt,
Would share some config file code with us?
in the standard extension macro we add a line:
exten = s,n,Set(_PICKUPMARK=${DB(pickupgroup/${ARG1})})
Where ARG1 is the extension about to be called (i.e. 201)
When someone dials 29 to pickup:
exten = 29,1,Pickup(${DB(pickupgroup/${CALLERID(number)})}...@pickupmark)
So to make extension 201 in pickup group 1 just do:
asterisk -rx 'database put pickupgroup 201 1'
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Kind Regards,
Matt Riddell
Director
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Re: [asterisk-users] Ast/Hyla/IAX Scalability?
On Fri, Mar 13, 2009 at 2:30 PM, Marshall Henderson marshall...@gmail.com wrote: On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg dbackeb...@gmail.com wrote: Again, you'll find people arguing that their voip solution has as low of a failure rate as a hardware solution. I'm jealous. My voip fax solution does not yet have that low of a failure rate, but I'm hopefully getting closer to working out the last bugs. Do you have any specifics to share about the problems you're finding? Sure. I can't disagree with the poster who said that problems they've seen are really the other side's fault. But assigning blame doesn't make me any happier. I have fax receiving problems I can't reproduce. When I load test it, I don't have problems. When I send 'real' inbound faxes from outside the network, over the real phone system, I don't have problems. I'm in New Haven, CT. One sender that messes up the most is in Kansas City, KS. They are a legitimate client, really sending a fax. I get occasional fax receipts that say: 'The call dropped prematurely' There will sometimes be a cluster of these, followed by a successful receipt. When I load tested, and send from real fax machines out and back in on POTS, I get 100% success. I've successfully load-tested around 175 simultaneous inbound faxes. I slowed down the simulation to about 5 simultaneous faxes, and left that running over a long weekend, generating something like 30,000 faxes and something like 1GB of received fax files. Again, the success rate was 100%. A problem with my simulation was that I used sending faxes that speak the protocol correctly. Does anybody have some faxes that send garbage? Then I put it into production with a limited amount of real fax traffic for our clients. I'm talking fewer than 10 calls per day most days. But it seems like the reality of the speed of light over continental long-distance, combined with the reality of crappy fax machines that don't speak protocols correctly result in occasional failures. I've made some adjustments that I think anecdotally have solved the silly problems, but that one with the faxes dropping early is the one that (maybe) hasn't gone away. I'd like a success rate around 99%. I'm getting around 63% if you count individual failed calls that eventually result in a success. I can't tell if I'm having bad luck with this phase of my pilot or if my failure rate is going to remain constant as I add clients. I need more data points to get statistical significance. What I really need is a failing fax I can control, then tune parameters on my side, and see if the failure rate gets worse or better. Seriously considering breaking down and asking for the cooperation of the client in that endeavor. People who have been following my posts on this topic know that I'm using: PRI(s) - Cisco voip gateway hardware - T.38 / SIP / g711 - Asterisk-1.6 with ReceiveFax (depends on SpanDSP, but does NOT use IAX or IAXmodem) What I've been 'tuning' most recently have been arguments to the Cisco setup fax and SIP translation. I did try out IAXModem with Hylafax and 1.4 and had lots of problems that all went away when I switched to using the approach I use now. I never tried 1.6 with IAXModem and Hylafax, so I can't tell you how well they work together. ___ -- 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] Ast/Hyla/IAX Scalability?
David Backeberg wrote: On Fri, Mar 13, 2009 at 2:30 PM, Marshall Henderson marshall...@gmail.com wrote: On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg dbackeb...@gmail.com wrote: Again, you'll find people arguing that their voip solution has as low of a failure rate as a hardware solution. I'm jealous. My voip fax solution does not yet have that low of a failure rate, but I'm hopefully getting closer to working out the last bugs. Do you have any specifics to share about the problems you're finding? Sure. I can't disagree with the poster who said that problems they've seen are really the other side's fault. But assigning blame doesn't make me any happier. I have fax receiving problems I can't reproduce. When I load test it, I don't have problems. When I send 'real' inbound faxes from outside the network, over the real phone system, I don't have problems. I'm in New Haven, CT. One sender that messes up the most is in Kansas City, KS. They are a legitimate client, really sending a fax. I get occasional fax receipts that say: 'The call dropped prematurely' There will sometimes be a cluster of these, followed by a successful receipt. When I load tested, and send from real fax machines out and back in on POTS, I get 100% success. I've successfully load-tested around 175 simultaneous inbound faxes. I slowed down the simulation to about 5 simultaneous faxes, and left that running over a long weekend, generating something like 30,000 faxes and something like 1GB of received fax files. Again, the success rate was 100%. A problem with my simulation was that I used sending faxes that speak the protocol correctly. Does anybody have some faxes that send garbage? Then I put it into production with a limited amount of real fax traffic for our clients. I'm talking fewer than 10 calls per day most days. But it seems like the reality of the speed of light over continental long-distance, combined with the reality of crappy fax machines that don't speak protocols correctly result in occasional failures. I've made some adjustments that I think anecdotally have solved the silly problems, but that one with the faxes dropping early is the one that (maybe) hasn't gone away. I'd like a success rate around 99%. I'm getting around 63% if you count individual failed calls that eventually result in a success. I can't tell if I'm having bad luck with this phase of my pilot or if my failure rate is going to remain constant as I add clients. I need more data points to get statistical significance. What I really need is a failing fax I can control, then tune parameters on my side, and see if the failure rate gets worse or better. Seriously considering breaking down and asking for the cooperation of the client in that endeavor. People who have been following my posts on this topic know that I'm using: PRI(s) - Cisco voip gateway hardware - T.38 / SIP / g711 - Asterisk-1.6 with ReceiveFax (depends on SpanDSP, but does NOT use IAX or IAXmodem) What I've been 'tuning' most recently have been arguments to the Cisco setup fax and SIP translation. I did try out IAXModem with Hylafax and 1.4 and had lots of problems that all went away when I switched to using the approach I use now. I never tried 1.6 with IAXModem and Hylafax, so I can't tell you how well they work together. Fully open-to-the-public FAX servers tend to get just get a lot of bad calls, many of them wrong numbers, or voice users. FAX servers for closed user groups tend to get few bad calls, unless the phone number gets included on some unfortunate list. This is one of the things which made early real world testing of spandsp and iaxmodem tough. We have to capture every failure, and analyse them by hand whether it was our fault or the far end's. Without knowing the nature of your system I have no clue what kind of failure rate might be expected. You can find a bit more about these issues and our results at http://www.soft-switch.org/spandsp-soft-fax-performance.html Your differing failure rates between using ReceiveFAX and using iaxmodem seem to indicate your results relate to issues in your own system, rather than the nature of the callers, but we can't really tell. A minor change in usage pattern may have resulted in a big change in the results. What I can say is that a properly set up iaxmodem + HylaFAX setup, with an IAX connection that does not loose packets (don't assume LANs don't loose packets), will have a true failure rate (i.e. a rate of calls failing which had the potential to succeed) well below 1%. The results for the latest spandsp used on its own (i.e. as a full FAX machine, rather than a FAX modem for HylaFAX) should be approaching this figure, as its back end processing has matured. Right now I think the maturity of HylaFAXes handling of buggy FAX machine probably still puts it slightly ahead. Success with FAX has a lot to do with tolerating the
