[asterisk-users] Manipulating REGISTER messages
Hello, I would like to add SIP headers to the REGISTER messages Asterisk (1.6) sends to an external proxy. Also, I want to be able to reorder the lines. Is it possible? If yes, how? 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] Digium and Sangoma Cards PCI express compatibility
Ricardo Melendez schrieb: Hi to All, I dont know much about PCI express slots in newer Servers, my doubt is if the Digium and Sangoma PCI express cards, are compatible with the x8 PCI express slots that come in the HP Proliant ML150 G5 server. http://lists.digium.com/pipermail/asterisk-users/2009-March/228930.html Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- 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] Simple(?) dialplan question.
Hi List,
I have a nice simple dialplan question for you Currently, I have
definitions similar to the following in my extensions.conf file, to allow me
to dial out using a variety of channels:
; Direct dial (number starts with zero), use 0151 xxx :
exten = _0.,1,Set(CALLERID(number)=0845xxx)
exten = _0.,n,Dial(SIP/${ext...@sipgate,90,t)
exten = _0.,n,Playback(invalid)
exten = _0.,n,Hangup[/code]
(I've munged some of the numbers, hence the x's)
Now, this works fine provided the person answers in 90 seconds or less: If
not, I get that option is invalid announced, and it hangs up. I want to do
this:
If DIAL fails because the other party is engaged, I'd like Asterisk to
automatically re-try the number, for as long as I've got the handset off the
hook or until the other party starts ringing. As there'll be no ring tone,
it'd be nice it it could play music until DIAL succeeds in getting a ring
tone; at which point it makes ring ring noises (this will serve as my
prompt that - hopefully - someone's going to answer soon).
If DIAL fails because I got the number wrong, then a PLAYBACK to that effect
would be useful... I can record my own soundfile if there isn't a standard
one. By wrong, I mean the exchange would return number unavailable, rather
than I get the wrong person!
If DIAL fails after it's been ringing for ages (e.g. when calling the local
Post Office sorting office, who only answer 1 in 5 calls), I'd like it to
retry, ala the busy response.
IF DIAL exits because the other party hung up, I'd want it to simply hang up
on me like it does now. I suspect this is standard behaviour? But maybe it
tries to read the invalid announce to a closed channel with my dialplan, I'm
not sure.
If the above can be achieved in extensions.conf, that's great, as I've not
done any AEL... but if AEL (or AGI, even) is the only way, so be it...
FWIW, I'm using Asterisk 1.4.18 on Ubuntu 4.1.2 (kernel version
2.6.20-15-server).
Cheers!
Ade.
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Re: [asterisk-users] Simple(?) dialplan question.
On Sun, 22 Mar 2009, Asterisk wrote:
Hi List,
I have a nice simple dialplan question for you Currently, I have
definitions similar to the following in my extensions.conf file, to allow me
to dial out using a variety of channels:
; Direct dial (number starts with zero), use 0151 xxx :
exten = _0.,1,Set(CALLERID(number)=0845xxx)
exten = _0.,n,Dial(SIP/${ext...@sipgate,90,t)
exten = _0.,n,Playback(invalid)
exten = _0.,n,Hangup[/code]
(I've munged some of the numbers, hence the x's)
Now, this works fine provided the person answers in 90 seconds or less: If
not, I get that option is invalid announced, and it hangs up. I want to do
this:
It's easy to do in dialplan - you just need to know the status codes
returned by Dial (and hope that sipgate return the correct codes too)
You can simply:
exten = _0.,n,Goto(${DIALSTATUS})
(before the playback)
Use the labels as the destinations - eg.
exten = _0.,n(BUSY),Noop()
exten = _0.,n(CONGESTION),Noop()
then you could at this point, insert a Wait(1) then a Goto back to your
Dial line.
See the helpfile on the Dial command for all possible status messages -
ie.
DIALSTATUS - This is the status of the call:
CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL
DONTCALL | TORTURE | INVALIDARGS
If DIAL fails because I got the number wrong, then a PLAYBACK to that effect
would be useful... I can record my own soundfile if there isn't a standard
one. By wrong, I mean the exchange would return number unavailable, rather
than I get the wrong person!
I used to play messages back, but resorted to just returning the codes to
the phone.
If DIAL fails after it's been ringing for ages (e.g. when calling the local
Post Office sorting office, who only answer 1 in 5 calls), I'd like it to
retry, ala the busy response.
IF DIAL exits because the other party hung up, I'd want it to simply hang up
on me like it does now. I suspect this is standard behaviour? But maybe it
tries to read the invalid announce to a closed channel with my dialplan, I'm
not sure.
that's standard, or implement a 'h' priority.
Does sipgate let you change outgoing caller ID these days now?
Gordon___
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Re: [asterisk-users] Simple(?) dialplan question.
Gordon Henderson wrote:
It's easy to do in dialplan - you just need to know the status codes
returned by Dial (and hope that sipgate return the correct codes too)
Not just Sipgate: I've got an account with CallCentric and a BT line to deal
with too...
You can simply:
exten = _0.,n,Goto(${DIALSTATUS})
(before the playback)
Use the labels as the destinations - eg.
exten = _0.,n(BUSY),Noop()
exten = _0.,n(CONGESTION),Noop()
I've never seen that before, does that definitely work in 1.4.x? If so,
cool...
then you could at this point, insert a Wait(1) then a Goto
back to your
Dial line.
See the helpfile on the Dial command for all possible status
messages -
ie.
DIALSTATUS - This is the status of the call:
CHANUNAVAIL | CONGESTION | NOANSWER |
BUSY | ANSWER | CANCEL
DONTCALL | TORTURE | INVALIDARGS
Thanks, I'll give that a go.
Does sipgate let you change outgoing caller ID these days now?
No - it's a legacy left over from when I hoped it would let me do it. I
just haven't got around to removing it yet.
Cheers,
Ade.
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Re: [asterisk-users] Simple(?) dialplan question.
Asterisk wrote:
You can simply:
exten = _0.,n,Goto(${DIALSTATUS})
(before the playback)
Use the labels as the destinations - eg.
exten = _0.,n(BUSY),Noop()
exten = _0.,n(CONGESTION),Noop()
I've never seen that before, does that definitely work in 1.4.x? If so,
cool...
That's been a part of the standard extension macro I've been using
forever, as follows...
[macro-stdexten]
exten = s,1,Set(__DYNAMIC_FEATURES=${FEATURES})
exten = s,2,GotoIf($[${FOLLOWME_${ARG1}} = 1]?5:3)
exten = s,3,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,4,Goto(s-${DIALSTATUS},1)
exten = s,5,Macro(stdexten-followme,${ARG1},${ARG2})
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
(the syntax is marginally different, but not significantly. Note the
_s-. extension to catch any odd/unexpected return status codes)
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[asterisk-users] Looking for Prepaid Solution
All; I am looking for a solution a motel that is switching to VoIP. They are especially interested in prepaid services. That is, a resident will come to the office and pre-pay for phone services for their room. When the money runs out, service is shut off, and they need to come down to the office again. Accurate reports are very important. A search on voip-info.org shows several solutions out there. Does anyone have expertise in this area? What do people recommend? Any insight at all would be greatly appreciated. Thanks FSD _ Hotmail® is up to 70% faster. Now good news travels really fast. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_70faster_032009___ -- 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] Simple(?) dialplan question.
On Sun, Mar 22, 2009 at 4:09 AM, Asterisk aster...@solutionengineers.comwrote:
Hi List,
I have a nice simple dialplan question for you Currently, I have
definitions similar to the following in my extensions.conf file, to allow
me
to dial out using a variety of channels:
; Direct dial (number starts with zero), use 0151 xxx :
exten = _0.,1,Set(CALLERID(number)=0845xxx)
exten = _0.,n,Dial(SIP/${ext...@sipgate,90,t)
exten = _0.,n,Playback(invalid)
exten = _0.,n,Hangup[/code]
(I've munged some of the numbers, hence the x's)
Now, this works fine provided the person answers in 90 seconds or less: If
not, I get that option is invalid announced, and it hangs up. I want to
do
this:
If DIAL fails because the other party is engaged, I'd like Asterisk to
automatically re-try the number, for as long as I've got the handset off
the
hook or until the other party starts ringing. As there'll be no ring tone,
it'd be nice it it could play music until DIAL succeeds in getting a ring
tone; at which point it makes ring ring noises (this will serve as my
prompt that - hopefully - someone's going to answer soon).
If DIAL fails because I got the number wrong, then a PLAYBACK to that
effect
would be useful... I can record my own soundfile if there isn't a standard
one. By wrong, I mean the exchange would return number unavailable, rather
than I get the wrong person!
If DIAL fails after it's been ringing for ages (e.g. when calling the local
Post Office sorting office, who only answer 1 in 5 calls), I'd like it to
retry, ala the busy response.
IF DIAL exits because the other party hung up, I'd want it to simply hang
up
on me like it does now. I suspect this is standard behaviour? But maybe it
tries to read the invalid announce to a closed channel with my dialplan,
I'm
not sure.
If the above can be achieved in extensions.conf, that's great, as I've not
done any AEL... but if AEL (or AGI, even) is the only way, so be it...
You can do it all three ways.
In AEL, you'd do something like this
context internalexten
{
_0. = {
Set(prevstatus=NOANSWER); /* set up a prevstatus */
Set(CALLERID(number)=0845xxx);
while(${prevstatus} == NOANSWER || ${prevstatus} ==
BUSY)
{
Dial(SIP/${ext...@sipgate,90,tm); /* transfers and moh
*/
switch(${DIALSTATUS})
{
case CHANUNAVAIL:
Playback(bad_num);
hangup();
break;
case CONGESTION:
Playback(congested);
hangup();
break;
case BUSY:
case NOANSWER:
break; /* BUSY will fall thru into NOANSWER */
default:
break;
}
Set(prevstatus=${DIALSTATUS});
}
hangup();
}
}
The above code should (I haven't tested it or anything) give you most of the
behavior you
specified, but it will play MOH up to the time someone answers. No
ringing/moh mixture...
Dial doesn't do that. You may have to correct some typos, etc. that I've
made above!
A hangup from the remote end will end the Dial app, and the result should be
ANSWER,
which should drop you out of the loop and end the call.
Also, a hangup from the dialing exten should just terminate the dialplan
execution.
I might note that the above code should be easier to read than the equiv
extenstions.conf
code! But, I guess I'm biased!
murf
--
Steve Murphy
ParseTree Corp
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Re: [asterisk-users] H323plus homepage down?
Sebastian wrote: Anybody knows why is down? Or if has been moved to another page?? ___ -- 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 It's up for me ___ -- 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] Looking for Prepaid Solution
At the hotels we service we have produced prepaid calling cards that they sell in the lobby. They dial '8' to get the card program, which is a custom AGI that maintains a mysql database of PIN numbers and balances... j On Sun, 22 Mar 2009, cbbs...@hotmail.com wrote: All; I am looking for a solution a motel that is switching to VoIP. They are especially interested in prepaid services. That is, a resident will come to the office and pre-pay for phone services for their room. When the money runs out, service is shut off, and they need to come down to the office again. Accurate reports are very important. A search on voip-info.org shows several solutions out there. Does anyone have expertise in this area? What do people recommend? Any insight at all would be greatly appreciated. Thanks FSD _ Hotmail? is up to 70% faster. Now good news travels really fast. http://windowslive.com/online/hotmail?ocid=TXT_TAGLM_WL_HM_70faster_032009 ___ -- 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] H323plus homepage down?
It's up today :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell Sent: domingo, 22 de marzo de 2009 01:42 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] H323plus homepage down? Sebastian wrote: Anybody knows why is down? Or if has been moved to another page?? ___ -- 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 It's up for me ___ -- 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: 270.11.23/2016 - Fecha de la versión: 03/21/09 17:58:00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CID when using WaitExten?
Hi, all. My autoattendant looks like this: exten = s,1,Answer() exten = s,n,Background(corporate-greeting) exten = s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds exten = s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds exten = s,n,WaitExten(30) When the call gets forwarded to the destination extension, however, there's no caller ID (instead, calls are from Asterisk). What am I doing wrong? Thanks! -Ken P.S. Apologies if this is a duplicate; sent originally from an account the Asterisk mailing doesn't/shouldn't know about. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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] CID when using WaitExten?
Ken D'Ambrosio wrote: When the call gets forwarded to the destination extension, however, there's no caller ID (instead, calls are from Asterisk). What am I doing wrong? That would normally be the case if there was no caller-id on the inbound call. What does your console show? What does your dial statement look like? 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
[asterisk-users] I need a country, state, city database
I need a country, state, city database for a web application. Anyone have a free version they can email (or drop.io) for me? Looking for something like this at $197 but may as well ask in case you know of a free source. http://www.globixdata.com/pop.cfm?db=worldv1=lv2=sv3=apricing=99 Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- 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] 1.6.2 beta 1 crash
Tilghman Lesher tilgh...@mail.jeffandtilghman.com writes: The useragent field should have been there previously. Now, Asterisk warns you, instead of the query silently failing. The field lastms is new (numeric). What is the recommended type for the useragent field with MySQL? I don't see it in the recommendations on voip-info.org, but maybe I'm just blind. /Benny ___ -- 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] Global videoconferencing solution.
Hello everybody, i am searching a solution for a videoconferencing, Any solution (Free/commercial). Asterisk is a great software, but recently we have more and more demands about videoconferencing of 3 or more peoples, Existing solutions are heavy and costly, around 2500€ for 1 client. This is insane. Is there any solutions out there for non millionaires ? Or even Free ? I remember a company who sold his software called cu see mee There were some conference rooms, used webcams 12 ppl max as remember. It could be perfect. Thank you for giving me advices. ___ -- 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] Global videoconferencing solution.
Google Red5, it’s a toolkit so not a turn key solution but it does what you are looking for. Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grygoriy Dobrovolskyy Sent: Sunday, March 22, 2009 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Global videoconferencing solution. Hello everybody, i am searching a solution for a videoconferencing, Any solution (Free/commercial). Asterisk is a great software, but recently we have more and more demands about videoconferencing of 3 or more peoples, Existing solutions are heavy and costly, around 2500€ for 1 client. This is insane. Is there any solutions out there for non millionaires ? Or even Free ? I remember a company who sold his software called cu see mee There were some conference rooms, used webcams 12 ppl max as remember. It could be perfect. Thank you for giving me advices. ___ -- 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] make script 1.6.0.6 breaks up, need help!
Hi people! I need help according getting asterisk 1.6.0.6 installed. I posted to digium, but it seems to be that it is not an error, but either I am not getting smart what I have to do, to get it solved (configured and installed as well). ./configure make gets me this output: In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file or directory In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t' /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here make[1]: *** [extconf.o] Error 1 make: *** [utils] Error 2 for any help and support, I would thank you people! Tamer ___ -- 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] make script 1.6.0.6 breaks up, need help!
Are you installing on a 64bit OS?? Which Os are you using?? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tamer Higazi Sent: domingo, 22 de marzo de 2009 05:59 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] make script 1.6.0.6 breaks up, need help! Hi people! I need help according getting asterisk 1.6.0.6 installed. I posted to digium, but it seems to be that it is not an error, but either I am not getting smart what I have to do, to get it solved (configured and installed as well). ./configure make gets me this output: In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file or directory In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:103: error: conflicting types for 'uint64_t' /usr/include/stdint.h:56: error: previous declaration of 'uint64_t' was here make[1]: *** [extconf.o] Error 1 make: *** [utils] Error 2 for any help and support, I would thank you people! Tamer ___ -- 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: 270.11.24/2017 - Fecha de la versión: 03/22/09 17:51:00 ___ -- 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] music-on-hold kicks in and disconnects/interrupt the call
Joseph wrote: I'm using Asterisk 1.4.22.1 When I'm on active call it happens many times the call gets interrupted by music-on-hold without my pressing any button. MOH just kicks in and int erupt the call and I have no way of getting the call back. Did anybody experienced anything like this? No - do you have any dialplan code or cli output to show for this excitement? PaulH ___ -- 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] I need a country, state, city database
I don't, but it out to be out there. We needed a list of all (valid) bank routing numbers for a check writing program and a former associate found that for free. I suggest you look in the direction of the US Department of Commerce. They have to have a list of what you want, and the basic information is pretty static. (We know all the states, ;-), there are no new counties either, and not much changes in towns.) Map makers, GPS manufacturers, Census, Post Office, FCC, 1000 other govt. agencies, UPS,. FedEx etc all use such a list. I would almost think Google search could come up with such a list. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Collins Sent: Sunday, March 22, 2009 2:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] I need a country, state, city database I need a country, state, city database for a web application. Anyone have a free version they can email (or drop.io) for me? Looking for something like this at $197 but may as well ask in case you know of a free source. http://www.globixdata.com/pop.cfm?db=world http://www.globixdata.com/pop.cfm?db=worldv1=lv2=sv3=apricing=99 v1=lv2=sv3=apricing=99 Regards, Dean Collins Cognation Inc mailto:d...@cognation.net d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- 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-biz] OpenBTS chat with David A. Burgess
On Sat, Mar 21, 2009 at 4:17 PM, Steve Kennedy steve-aster...@gbnet.net wrote: On Sat, Mar 21, 2009 at 09:39:47AM +0100, randulo wrote: Hi, The OpenBTS Project is an effort to construct an open-source Unix application that uses the Universal Software Radio Peripheral (USRP) to present a GSM air interface (Um) to standard GSM handset and uses the Asterisk software PBX to connect calls. The combination of the ubiquitous GSM air interface with VoIP backhaul could form the basis of a new type of cellular network that could be deployed and operated at substantially lower cost than existing technologies in greenfields in the developing world. This looks like a great project, sorry I missed the call. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net Careful with this rabbit hole, it goes very deep and then logically branches in different directions then people become Of Interest, die In an Accident or Natural Causes, or disappear altogether. -- 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] I need a country, state, city database
You may want to check this link http://www.geodatasource.com/cities-free.html it may help you On Sun, Mar 22, 2009 at 8:14 PM, Cary Fitch ca...@usawide.net wrote: I don’t, but it out to be “out there”. We needed a list of all (valid) bank routing numbers for a check writing program and a former associate found that for free. I suggest you look in the direction of the US Department of Commerce. They have to have a list of what you want, and the basic information is pretty static. (We know all the states, ;-), there are no new counties either, and not much changes in towns.) Map makers, GPS manufacturers, Census, Post Office, FCC, 1000 other govt. agencies, UPS,. FedEx etc all use such a list. I would almost think Google search could come up with such a list. Cary Fitch -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dean Collins *Sent:* Sunday, March 22, 2009 2:12 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] I need a country, state, city database I need a country, state, city database for a web application. Anyone have a free version they can email (or drop.io) for me? Looking for something like this at $197 but may as well ask in case you know of a free source. http://www.globixdata.com/pop.cfm?db=worldv1=lv2=sv3=apricing=99 Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- 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] field lastms in 1.4.24
Hi all, I found that a new field lastms is used in 1.4.24. What is the usage of that field and the datatype of it? ango ___ -- 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] I need a country, state, city database
http://www.geonames.org/ is a great free resource, though not sure if it's what you're looking for. The maintainer is pretty approachable, try the forums if you don't see what you need. -- Paul Dean Collins wrote: I need a country, state, city database for a web application. Anyone have a free version they can email (or drop.io) for me? Looking for something like this at $197 but may as well ask in case you know of a free source. http://www.globixdata.com/pop.cfm?db=worldv1=lv2=sv3=apricing=99 http://www.globixdata.com/pop.cfm?db=worldv1=lv2=sv3=apricing=99 Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- 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] I need a country, state, city database
Problem solved, I was just sent this link by a friend. It has the full database and you can download the full 33mb database for free Country Code ASCII City Name City Name State/Region Population Latitude Longitude http://www.maxmind.com/app/worldcities http://www.maxmind.com/download/worldcities/worldcitiespop.txt.gz Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Chambers Sent: Sunday, March 22, 2009 10:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] I need a country, state, city database http://www.geonames.org/ is a great free resource, though not sure if it's what you're looking for. The maintainer is pretty approachable, try the forums if you don't see what you need. -- Paul Dean Collins wrote: I need a country, state, city database for a web application. Anyone have a free version they can email (or drop.io) for me? Looking for something like this at $197 but may as well ask in case you know of a free source. http://www.globixdata.com/pop.cfm?db=worldv1=lv2=sv3=apricing=99 http://www.globixdata.com/pop.cfm?db=worldv1=lv2=sv3=apricing=99 Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- 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] field lastms in 1.4.24
On Sunday 22 March 2009 21:40:14 Rilawich Ango wrote: Hi all, I found that a new field lastms is used in 1.4.24. What is the usage of that field and the datatype of it? It's an integer field used to ensure that realtime qualify continues to function across a reload event. -- 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
Re: [asterisk-users] IAX trunktimestamps and AST_CONTROL_SRCUPDATE
On Thursday 19 March 2009 05:29:00 Steve Davies wrote: I have just discovered (a year after it was implemented) a possibly undocumented incompatability between IAX in Asterisk 1.4 and any version of Asterisk pre-March 2008. It seems an AST_CONTROL_SRCUPDATE frame type was added (in March '08), but no mechanism to negotiate whether it can be sent to the remote end, so if a new IAX endpoint sends it, and the remote end ignores it, I believe it can cause the call to fail. Am I being overdramatic? I have a scenario which seems to be showing a 1.2 box talking to a 1.4 box dropping calls sometimes, and the error message on the 1.2 box is showing that it does not like the unrecognised AST_CONTROL_SRCUPDATE frame that it receives. This issue may be exagerated by the fact that the Asterisk 1.2 box has trunktimestamps=no set to ensure compatability with an old Asterisk 1.0.x service. Help? Is there a workaround? Might it be enough to enable trunktimestamps in this instance? It will have no effect. The issue has always been that if the stream source changed during a call, the sequence numbers could be reset, sometimes causing audio weirdness. What has changed is that we're now able to tell the other side to expect such a reset, thus preventing audio weirdness (basically, audio would drop until the remote end decided that it was okay that it was missing a bunch of frames and could continue on). If your calls are breaking, they would have broken, regardless of whether this frame was sent or not. In other words, this is a long-standing bug that was recently solved. If either one of your Asterisk servers cannot send the frame or interpret the frame, then the old familiar behavior is the result. -- 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
Re: [asterisk-users] 1.6.2 beta 1 crash
On Sunday 22 March 2009 14:19:50 Benny Amorsen wrote: Tilghman Lesher tilgh...@mail.jeffandtilghman.com writes: The useragent field should have been there previously. Now, Asterisk warns you, instead of the query silently failing. The field lastms is new (numeric). What is the recommended type for the useragent field with MySQL? I don't see it in the recommendations on voip-info.org, but maybe I'm just blind. CHAR(30) should be sufficient for most UAs, though a UA string may be longer (up to 255 characters in Asterisk and much longer in the SIP spec). -- 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
Re: [asterisk-users] field lastms in 1.4.24
Tilghman, Thanks. Can you elaborate the usage about it? What is the meaning of each valid value in this field? ango On Mon, Mar 23, 2009 at 11:24 AM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Sunday 22 March 2009 21:40:14 Rilawich Ango wrote: Hi all, I found that a new field lastms is used in 1.4.24. What is the usage of that field and the datatype of it? It's an integer field used to ensure that realtime qualify continues to function across a reload event. -- 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 ___ -- 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
