[asterisk-users] Visual Dial Plan application: Recommendations?
I am found and application called Visual Dialplan - And the idea seems good, apart from it didn't read the dial plans from a freepbx setup. Are there any other applications that I may try, that you guys can recommend? ___ -- 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] Linksys SPA922 - hangup problem
dubravko caric wrote: Hi all, I'm testing Linksys SPA922 phone and I have strange issue. when call is finished on the phone I see CallEnded and normal silence for cca. 5 seconds and then I get fast busy for cca. 20 sec. So, this isn't automatic hangup as on other phones I have tried (Cisco 7940, grandstream, XLite,... ) and I have to manually hangup handset to finish a call. Is this normal behavior for this phones or have I misconfigured something (maybe on Asterisk, but other types of phones are working OK)? This is normal behaviour for the Linksys phones - it's either a carry over from their ATAs or a slightly misguided attempt to make phones behave similarly to analogue phones. It's one of the few, admittedly minor irritations these phones have. ___ -- 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] Add volume sip accounts
you should read the asterisk bible... http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf use templates, you can use them in ANY config file. for example sip.conf [model-user](!); the (!) told asterisk this is a template host=dynamic allow=gsm secret=1234 ;more parameters ;more parameters ;more parameters ;the template finish when other mark start in this case [6000] [6000](model-user); the (model-user) tell asterisk to use the template model-user [6001](model-user) [6002](model-user) secret=56789665 ; you change just this parameter, only to this user [6003](model-user) [6004](model-user) . . . . [6199](model-user) you can use openoffice calc to make the list if you are to lazy to make a bash script, i do that :) David 2008/12/6 Alex Balashov [EMAIL PROTECTED] Generate with script. Mike Li wrote: Hi, all I want to add more than 200 sip accounts into sip.conf, username from 6000 to 6199, password is the same, i remember there is a better way to do this case, however, i have not searched the method yet. Anybody can tell me this method, TIA. BR Mike Li ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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 is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Convert CallerID name to uppercase
if asterisk can handle it use asterisk.
2008/12/5 Eric Fort [EMAIL PROTECTED]
Would agi and perl be a good means of doing this?
-Eric
On Fri, Dec 5, 2008 at 3:06 PM, John Todd [EMAIL PROTECTED] wrote:
On Dec 5, 2008, at 4:30 PM, Sean Dennis wrote:
Our legacy PBX will not accept the callerID name in anything but
capital letters. (Harris 20-20) When I send a call to the legacy PBX
from asterisk I would like to have asterisk convert the callerID name
to uppercase letters. Is there a way to do this?
Thanks for any input.
-Sean
In TRUNK versions of Asterisk, there is a function called TOUPPER,
which converts strings to upper case. I don't know when, exactly, it
appeared but I expect if it's not in the version you're using it may
be portable backwards without too much difficulty if the version
you're using supports functions.
JT
*CLI core show function TOUPPER
-= Info about function 'TOUPPER' =-
[Synopsis]
Convert string to all uppercase letters.
[Description]
Example: ${TOUPPER(Example)} returns EXAMPLE
[Syntax]
TOUPPER(string)
[Arguments]
string
[See Also]
Not available
*CLI
---
John Todd
[EMAIL PROTECTED]+1-256-428-6083
Asterisk Open Source Community Director
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Re: [asterisk-users] Visual Dial Plan application: Recommendations?
i like very much the vi ide :) 2008/12/6 Mr Gabriel [EMAIL PROTECTED] I am found and application called Visual Dialplan - And the idea seems good, apart from it didn't read the dial plans from a freepbx setup. Are there any other applications that I may try, that you guys can recommend? ___ -- 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 is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Gosubs broken since r160626 (1.6.0 SVN) ?
Mark Michelson wrote: In a fit of wild curiosity, I decided to double-check to be sure that the problem was an AEL parser issue and not one of my own. I actually discovered a bug introduced by my changes. I have fixed this bug in revision 161494 of the 1.6.0 branch. I suspect this will fix the problem you were seeing, too. I've just tested with this revision and all seems to be well again. Thanks for finding and fixing the bug! Gary H ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip Node w/ 4 wire audio AT command set call supervision
I have several discontinued Sierra Wireless MP775 mobile GSM/EDGE radio modems. These devices were originally installed in emergency vehicles to provide data and voice access along with GPS reporting. They have external RF connections for GSM/EDGE and GPS signals. The baseband side includes four wire audio, RS-232 serial, USB and some parallel digital I/O for panic switches etc. GSM voice connections are established by sending AT commands via the RS-232 serial or USB ports and utilizing a handset connected to the four-wire audio jack. Here is a link to the Sierra Wireless web pages for the MP775. http://www.sierrawireless.com/support/mp775.aspx My interest comes in finding an inexpensive way to connect an Asterisk PBX or similar system to the PSTN via GSM when POTS and Internet service isn't available or is too costly to connect. In my case, I'm considering building a house at the end of a long unpopulated stretch of dead end road and the cost of trenching and cabling from the nearest telco POP is prohibitive. I would like to find a way to connect these modems to my network so they appear as a SIP FXS device. This would require a device generating/reading AT commands and passing baseband audio on the front end and SIP emulation on the backend. I'm sure there must be a way to do this with a pc but to minimize power consumption; I would prefer to use something like a small single board computer with a Geode processor. The latter, having multiple RS-232 and audio ports, ought to be capable of handling at least two of the MP775's. Has anyone seen any hardware, software or combination that would allow me to accomplish such an interface? Regards, George ___ -- 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] Oslec issue
Hi Joseph.
I spent some time to understand what's missing in the OSLEC patch for
dahdi... I can confirm the same problem you reported some days ago and I
need OSLEC for home personal use.
For what I've understood looking at the code, there is some missing in
the dahdi_echocan_oslec.c file you can find in the
dahdi-linux/drivers/dahdi. I can list below what I did to have it
working. Actually I'm using the trunk revision 5443.
1. In the function echo_can_create i've modified the line
*ec = (struct echo_can_state *)oslec_create(ecp-tap_length,0);
with
*ec = (struct echo_can_state *)oslec_create(ecp-tap_length,
ECHO_CAN_USE_ADAPTION | ECHO_CAN_USE_NLP | ECHO_CAN_USE_CLIP |
ECHO_CAN_USE_TX_HPF | ECHO_CAN_USE_RX_HPF);
This instructs OSLEC to have a working modality properly set.
2. I've replaced the function echo_can_update with the code below:
static void echo_can_update(struct echo_can_state *ec, short *iref,
short *isig)
{
unsigned int SampleNum;
for (SampleNum = 0; SampleNum DAHDI_CHUNKSIZE; SampleNum++,
iref++)
{
short iCleanSample;
iCleanSample = (short) oslec_update((struct oslec_state
*)ec, *iref, *isig);
*isig++ = iCleanSample;
}
}
This lets the OSLEC to work on complete DAHDI_CHUNKSIZE buffer.
Please, if you have time, let me know if this solves your problem and,
if yes, I'll appreciate to have it public on trunk. I never did a commit
on asterisk svn so I need some hints on how to do it.
Thank you and best regards.
Marco Signorini.
Joseph L. Casale wrote:
Yesterday I pulled in the latest svn of Dahdi and added the files
from a recent kernel in the drivers/staging/echo structure and modified
the Kbuild file so it would compile without error. I insmod'ed the module
in, and modified my system.conf has echocanceller=oslec.
cat /proc/dahdi/1 shows:
Span 1: WCTDM/0 Wildcard TDM410P Board 1 (MASTER)
IRQ misses: 1
1 WCTDM/0/0 FXSKS (In use) (EC: OSLEC)
2 WCTDM/0/1
3 WCTDM/0/2
4 WCTDM/0/3
With the reco's from http://www.rowetel.com/ucasterisk/oslec.html#install on
configuring the chan_dahdi.conf file, the system behaves exactly as if there
is no ec enabled at all?
Are there any additional steps needed to enable oslec under dahdi, I am
guessing
I have missed something?
Thanks,
jlc
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[asterisk-users] Pika FAX
Hi, On the voip-info.org page for Asterisk FAX, someone has added a note saying Pika have stopped selling their soft FAX add on for Asterisk. Can any confirm or deny this? I found it strange, as it appears Pika have recently licence a V.34 FAX modem from Commetrex. I assumed they were going to try to carve out a niche for high end soft FAX with this. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Oslec issue
I spent some time to understand what's missing in the OSLEC patch for dahdi... I can confirm the same problem you reported some days ago and I need OSLEC for home personal use. Wow, Appreciate the info! I will need a few days to get this done. Out of curiosity, how do you find this ec's quality compared to the shipped modules and hpec? Thanks! jlc ___ -- 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] top posting again [was: Re: CDR Design]
On Sat, Dec 06, 2008 at 05:25:59AM +0200, Atis Lezdins wrote: GMail webinterface does automatically hides quotations. I expect that other mail clients are following. apt-get install t-prot The /ignore for top-posters :-) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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] Sip Node w/ 4 wire audio AT command set call supervision
On Dec 6, 2008, at 7:55 AM, George Bean wrote: I have several discontinued Sierra Wireless MP775 mobile GSM/EDGE radio modems. These devices were originally installed in emergency vehicles to provide data and voice access along with GPS reporting. They have external RF connections for GSM/EDGE and GPS signals. The baseband side includes four wire audio, RS-232 serial, USB and some parallel digital I/O for panic switches etc. GSM voice connections are established by sending AT commands via the RS-232 serial or USB ports and utilizing a handset connected to the four-wire audio jack. Here is a link to the Sierra Wireless web pages for the MP775. http://www.sierrawireless.com/support/mp775.aspx My interest comes in finding an inexpensive way to connect an Asterisk PBX or similar system to the PSTN via GSM when POTS and Internet service isn’t available or is too costly to connect. In my case, I’m considering building a house at the end of a long unpopulated stretch of dead end road and the cost of trenching and cabling from the nearest telco POP is prohibitive. I would like to find a way to connect these modems to my network so they appear as a SIP FXS device. This would require a device generating/reading AT commands and passing baseband audio on the front end and SIP emulation on the backend. I’m sure there must be a way to do this with a pc but to minimize power consumption; I would prefer to use something like a small single board computer with a Geode processor. The latter, having multiple RS-232 and audio ports, ought to be capable of handling at least two of the MP775’s. Has anyone seen any hardware, software or combination that would allow me to accomplish such an interface? Regards, George I've not heard of any such hardware/software combination to do this, so I apologize for the semi-useless reply. However, I'd be very interested in any solutions you come up with, since I've got a few similar older WaveCom modem devices which probably do mostly the same thing - I think I picked them up for about $6 each on eBay, and I see a few there now for $30-$50 range. I bought them just for SMS, but they are fully-functioning GSM phones if you want to wire up the right pins on the connector to speaker/microphone parts. You also should take a look at eBay which has a number of SIP-to- mobile (SIP to GSM, SIP to CDMA) conversion units. Type sip gsm in the search term box. There's even one for $150 that seems to use bluetooth to connect SIP to Bluetooth phones (which can be done directly with Asterisk, though it may take some significant fist- slamming to get the drivers all set up correctly.) Maybe you have an older Bluetooth phone to use the lower-cost option, or maybe the integrated SIP/GSM system makes sense. If you're going to spend the money for a purpose-built solution, you might want to make sure that whatever gateway you get also supports SMS in some fashion. Some of us are waiting for the drivers to get Asterisk to talk directly to the PORTech devices for send/receive SMS capability. (hint, hint) If you take the modem/soundcard route, perhaps there are some useful cards or things here - http://www.ramseyelectronics.com/ - I haven't dug through, but they seem to have lots of small-end hacker-friendly bits and pieces which might do some of the line-level conversions you would require. You might connect directly to an Asterisk-driven soundcard, or you might use an FXO/FXS type conversion as well and connect to a single-line analog PC card that works with DAHDI/Zaptel. Get out the soldering iron! JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Recording - Asterisk
Hello folks, I wanted to setup Oreka to monitor calls on a trixbox box I have setup. Oreka doesn't seem to be catching all of the calls though I have port mirroring setup on the port that trixbox is connected to, mirrored to the port Oreka is connected to. I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if this meant that some phones, after checking in with Asterisk would simply communicate via RTP between each other, without going media transport going through trixbox itself? If this is the case then I guess I'd need to mirror the full VoIP VLAN to the Oreka port wouldn't I? Or is there another reason that I'm missing here? Just trying to get this sussed out in my head! Thanks for your time. Chris ___ -- 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] questions regarding DAHDI
Just moved over to DAHDI today and have a few questions. My environment is: Asterisk 1.4.22 Using a TE410P 4 PRI's 1. Now when calls come in on those PRI's I get this message in the console: [Dec 6 19:42:18] WARNING[31557]: chan_dahdi.c:1481 dahdi_enable_ec: Unable to enable echo cancellation on channel 41 (No such device) How can I resolve this? 2. Do I need to be running echo cancellation on TDM circuits? (PRI's?) 3. I compiled DAHDI tools and I seem to have all the scripts, however I do not have or can not seem to find dahdi_tools. How can I get this? I believe this is the zttools replacement. 4. Is there a way for me to verify that I am receiving clock source on these PRI's from the telco? I need to verify this because it seems that if I use my internal clock source my HDLC errors seem to go away. (Still testing) Thanks in advance, Jon ___ -- 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 trunk mixing
Hi List,
Help me pls, or you think this can be an asterisk bug and should i make
a bug report?
thanks,
Csaba
Tóth Csaba írta:
hi,
i have a problem, and i am completely stuck with it, i hope someone can
point out where is my config wrong.
I have three server, connect together with IAX trunking. The server are
at romania (10.0.4.23, V1.4.22), hungary (10.0.1.23, V1.4.20) and serbia
(10.0.3.4, V1.4.22). I have a hardphone (6251) connected to the romanian
server, i dial a hungarian telephone number, the call goes to the
hungarian server well, but that server recognise the call come from
serbia.. and everything is mixed inside..
the phone starts at context do-phoning on the romanian server.
i called 003620XXX from the phone, and as you see, the romanian
server starts the call in good IAX trunk, but the hungarian server
identifies it badly..
Here is the message on the HUNGARIAN asterisk console about it:
-- Accepting AUTHENTICATED call from 10.0.4.23:
requested format = speex,
requested prefs = (gsm),
actual format = gsm,
host prefs = (),
priority = caller
-- Executing [EMAIL PROTECTED]:1]
MixMonitor(IAX2/telsrv-husrb-1541, om_1228466966.19588_6251.wav) in
new stack
== Begin MixMonitor Recording IAX2/telsrv-husrb-1541
-- Executing [EMAIL PROTECTED]:2]
Macro(IAX2/telsrv-husrb-1541, kitelco|0620XXX) in new stack
-- Executing [EMAIL PROTECTED]:1] Set(IAX2/telsrv-husrb-1541,
telszam=0620XXX) in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(IAX2/telsrv-husrb-1541,
ZAP/g2/0620XXX) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/0620XXX
-- Zap/37-1 is proceeding passing it to IAX2/telsrv-husrb-1541
here is ROMANIAN console:
[Dec 5 08:51:34] -- Executing [EMAIL PROTECTED]:1]
Set(SIP/6251-00c888c0, telszam=0620XXX) in new stack
[Dec 5 08:51:34] -- Executing [EMAIL PROTECTED]:2]
Set(SIP/6251-00c888c0, ~~EXTEN~~=s) in new stack
[Dec 5 08:51:34] -- Executing [EMAIL PROTECTED]:3]
Dial(SIP/6251-00c888c0, IAX2/telsrv-huro/0620XXX) in new stack
[Dec 5 08:51:34] -- Called telsrv-huro/0620XXX
[Dec 5 08:51:34] -- Call accepted by 10.0.1.23 (format gsm)
[Dec 5 08:51:34] -- Format for call is gsm
[Dec 5 08:51:35] -- IAX2/telsrv-huro-16384 is proceeding passing it
to SIP/6251-00c888c0
[Dec 5 08:51:35] -- Hungup 'IAX2/telsrv-huro-16384'
[Dec 5 08:51:35] == Spawn extension (macro-kitelsrvhu, s, 3) exited
non-zero on 'SIP/6251-00c888c0' in macro 'kitelsrvhu'
[Dec 5 08:51:35] == Spawn extension (macro-kitelsrvhu, s, 3) exited
non-zero on 'SIP/6251-00c888c0'
here are the snippets of the config files:
ROMANIAN server
iax.conf:
[telsrv-huro]
type=friend
host = 10.0.1.23
user = telsrv-huro
secret = xxx
bandwidth=low
qualify=yes
trunk=yes
timezone=Europe/Budapest
context=incoming-hu
[telsrv-rosrb]
type=friend
host = 10.0.3.4
user = telsrv-rosrb
secret = xxx
bandwidth=low
qualify=yes
trunk=yes
timezone=Europe/Bucharest
context=incoming-srb
extensions.ael:
context do-phoning {
includes {
do-nationalcall;
}
}
abstract context do-nationalcall {
_0036. = kitelsrvhu(06${EXTEN:4});
_6[2-8]XX = kitelsrvhu(${EXTEN});
_7[2-8]XX = kitelsrvhu(${EXTEN});
_00381. = kitelsrvsrb(${EXTEN:4});
_51[567]X = kitelsrvsrb(${EXTEN});
}
context incoming-hu {
includes {
template-companynumbers;
template-spec;
template-helyi;
template-mobil;
template-orszagos;
}
}
context incoming-srb {
includes {
template-companynumbers;
template-spec;
template-helyi;
template-mobil;
template-orszagos;
}
}
macro kitelsrvhu(telszam) {
Dial(IAX2/telsrv-huro/${telszam});
switch(${DIALSTATUS}) {
case CHANUNAVAIL:
Playback(/var/lib/asterisk/sounds/beeperr);
case CONGESTION:
Playback(/var/lib/asterisk/sounds/beeperr);
case BUSY:
Busy();
Wait(5);
};
Hangup();
}
macro kitelsrvsrb(telszam) {
Dial(IAX2/telsrv-srbro/${telszam});
switch(${DIALSTATUS}) {
case CHANUNAVAIL:
Playback(/var/lib/asterisk/sounds/beeperr);
case CONGESTION:
Playback(/var/lib/asterisk/sounds/beeperr);
case BUSY:
Busy();
Wait(5);
};
Hangup();
}
HUNGARIAN server
iax.conf:
[telsrv-huro]
type=friend
host = 10.0.4.23
user = telsrv-huro
secret = xxx
bandwidth=low
qualify=yes
Re: [asterisk-users] Convert CallerID name to uppercase
In TRUNK versions of Asterisk, there is a function called TOUPPER, which converts strings to upper case. I don't know when, exactly, it appeared but I expect if it's not in the version you're using it may be portable backwards without too much difficulty if the version you're using supports functions. JT *CLI core show function TOUPPER This looks like exactly what I need. I see that it's available in 1.6 so I will upgrade and let you know how it goes. Thank You. -Sean Dennis ___ -- 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] questions regarding DAHDI
the echo cancelation sintaxis has change in dahdi read the documents about it. David 2008/12/6 Jon Weisman [EMAIL PROTECTED] Just moved over to DAHDI today and have a few questions. My environment is: Asterisk 1.4.22 Using a TE410P 4 PRI's 1. Now when calls come in on those PRI's I get this message in the console: [Dec 6 19:42:18] WARNING[31557]: chan_dahdi.c:1481 dahdi_enable_ec: Unable to enable echo cancellation on channel 41 (No such device) How can I resolve this? 2. Do I need to be running echo cancellation on TDM circuits? (PRI's?) 3. I compiled DAHDI tools and I seem to have all the scripts, however I do not have or can not seem to find dahdi_tools. How can I get this? I believe this is the zttools replacement. 4. Is there a way for me to verify that I am receiving clock source on these PRI's from the telco? I need to verify this because it seems that if I use my internal clock source my HDLC errors seem to go away. (Still testing) Thanks in advance, Jon ___ -- 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 is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] CLI and choice of messages
Thanks, for some reason I had completely missed that cmd. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: Friday, December 05, 2008 17:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CLI and choice of messages On Fri, Dec 5, 2008 at 6:00 PM, Mike [EMAIL PROTECTED] wrote: Is there a way, for debugging purpose, to have a level where only Noop() cmds are shown in the CLI but nothing else in the dialplan appears (except for errors and warnings or course)? Replace NoOp(something) with Verbose(something) and it will be printed out with Verbosity of 0. That's default verbosity you see in CLI. NoOp really does nothing as opposed to Verbose(), so you will see it only in -- Executing message which has verbosity 2. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] CLI and choice of messages
I am almost ashamed not to have thought of it . Thanks a lot, that will do perfectly. Mike From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Friday, December 05, 2008 17:06 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CLI and choice of messages I log verbose to a file and tail it. tail f /var/log/asterisk/asterisk-verbose | grep Noop From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, December 05, 2008 13:44 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] CLI and choice of messages Youd think theyd actually have something like this. But nope, they dont. Only for debug, but no verbose output filtering. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: December 5, 2008 11:00 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] CLI and choice of messages Is there a way, for debugging purpose, to have a level where only Noop() cmds are shown in the CLI but nothing else in the dialplan appears (except for errors and warnings or course)? 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] ISDN PRI settings for Telus BC network
Gondar Monn wrote: Hi there! Does anyone deal with Telus in BC ? We have some PRI lines that were used for dialup, would like to convert them for pbx system, talked with some technicians @ Telus, but the information given was not clear, kind of: try this see if it works Does anyone here have the settings required to talk to there equipment ? A few years ago I had a PRI from TELUS. The winning zaptel.conf line: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 And in zapata.conf it was switchtype=national pridialplan=unknown prilocaldialplan=national signalling=pri_cpe These are both from the asterisk 1.2 days so a lot may have changed between now and then... Hope this helps, Trevor ___ -- 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] CLI and choice of messages
On Saturday 06 December 2008 17:05:28 Mike wrote: Thanks, for some reason I had completely missed that cmd. Verbose was added in 1.2. New to 1.4 was the Log command, designed specifically for exceptional dialplan conditions where you might wish to log a set of output for later perusal (like if it happens at 2am and you weren't awake to catch it). -- 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] ISDN PRI settings for Telus BC network
Thanks a lot, will let you know how things are going when I get them to turn on two way dialing . By the way any pointers on how to connect to Portmaster ? Looks like we are going to have to share some PRIs lines with portmaster (dialup) FYI, I am sticking with Asterisk 1.4 for now ... Thanks! Gondar On Sat, Dec 6, 2008 at 3:49 PM, Trevor Peirce [EMAIL PROTECTED] wrote: Gondar Monn wrote: Hi there! Does anyone deal with Telus in BC ? We have some PRI lines that were used for dialup, would like to convert them for pbx system, talked with some technicians @ Telus, but the information given was not clear, kind of: try this see if it works Does anyone here have the settings required to talk to there equipment ? A few years ago I had a PRI from TELUS. The winning zaptel.conf line: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 And in zapata.conf it was switchtype=national pridialplan=unknown prilocaldialplan=national signalling=pri_cpe These are both from the asterisk 1.2 days so a lot may have changed between now and then... Hope this helps, Trevor ___ -- 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] Question on queue terms
Hello, I'm trying to setup a very simple queue with 5 SIP phones. I do NOT want the agents to have to be on the phone to get calls, but I want those 5 SIP phones to ring (according to the strategy chosen in queue.conf) to dispatch calls. Is this a call back queue, or is a callback queue a queue that calls back the customer? There is conflicting info when searching for callback queue. 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] CLI and choice of messages
New to 1.4 or 1.6? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Saturday, December 06, 2008 20:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CLI and choice of messages On Saturday 06 December 2008 17:05:28 Mike wrote: Thanks, for some reason I had completely missed that cmd. Verbose was added in 1.2. New to 1.4 was the Log command, designed specifically for exceptional dialplan conditions where you might wish to log a set of output for later perusal (like if it happens at 2am and you weren't awake to catch it). -- 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
Re: [asterisk-users] Sip Node w/ 4 wire audio AT command set callsupervision
Google Multitech CallFinder GSM out of Minnesota if you want a common off the shelf product. GSMA.org was using their product with FXO/FXS for backup purposes. I recall they have a GSM to FXO/FXS, and I thought they had GSM to H323. I also found a European company that made high end (24+) GSM banks to FXO/FXS but can't recall the name. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of George Bean Sent: Saturday, December 06, 2008 8:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sip Node w/ 4 wire audio AT command set callsupervision I have several discontinued Sierra Wireless MP775 mobile GSM/EDGE radio modems. These devices were originally installed in emergency vehicles to provide data and voice access along with GPS reporting. They have external RF connections for GSM/EDGE and GPS signals. The baseband side includes four wire audio, RS-232 serial, USB and some parallel digital I/O for panic switches etc. GSM voice connections are established by sending AT commands via the RS-232 serial or USB ports and utilizing a handset connected to the four-wire audio jack. Here is a link to the Sierra Wireless web pages for the MP775. http://www.sierrawireless.com/support/mp775.aspx My interest comes in finding an inexpensive way to connect an Asterisk PBX or similar system to the PSTN via GSM when POTS and Internet service isn't available or is too costly to connect. In my case, I'm considering building a house at the end of a long unpopulated stretch of dead end road and the cost of trenching and cabling from the nearest telco POP is prohibitive. I would like to find a way to connect these modems to my network so they appear as a SIP FXS device. This would require a device generating/reading AT commands and passing baseband audio on the front end and SIP emulation on the backend. I'm sure there must be a way to do this with a pc but to minimize power consumption; I would prefer to use something like a small single board computer with a Geode processor. The latter, having multiple RS-232 and audio ports, ought to be capable of handling at least two of the MP775's. Has anyone seen any hardware, software or combination that would allow me to accomplish such an interface? Regards, George - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.___ -- 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] config from DB
Hi Everyone, Sorry, if this has been already discussed, but maybe someone encountered interesting issue. I have an * Dialplan configured with MySQL db. Everything works excellent, except, I can't specify the ex-girlfriend logic. For example Context exten Priorityapp appdata Default 400 1 Wait20 Default 400/100 1 Wait10 So, it does not matter what is my callerid, it will always go in wait(20) If user with callerID 100 will try to dial x400, it will go to wait(20) as well, and never wait(10). In another words Asterisk will disregard this logic. If I place this logic in the extensions.conf file it will work as a charm - no problem. Thank you for your help. Sincerely, --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
