Re: [asterisk-users] OT, I'm looking for SIP/register-enabled softphone
On Wed, Aug 08, 2007 at 10:28:05AM +0600, Kate Kretz wrote: sorry, I meant RFC 3856, sip presence, not sip regitration Twinkle 1.1 (new in that version. Released only about a month ago), linphone. Kphone should also support it, but I so far failed to get it authenticated with my Asterisk server. -- Tzafrir Cohen icq#16849755jabber:[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
[asterisk-users] Query
Hi, I am running asterisk PBX ( digium TE120P card configured) on one system. It is connected to E1 card running application on the other system. After establishing sync between two card, I am able to place call from sip phone to E1 card running application. I want to pass the callerid, when calling from sip phone to E1 card running application. Which all configuration files is to be changed in the asterisk. I am doing the following changes in extensions.conf exten=115,1,Dial(ZAP/g1/115,20) So, extension 115 is received at other end as callerid. Is it correct. Can any body help in how to configure for callerid with digium card. thanks and regards sanchal ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Callto:// tags inside web pages
2007/8/7, mitcheloc [EMAIL PROTECTED]: Ollvier, You could use the Firefox plug-in for Snap. It will auto detect numbers on a webpage and make them dialable. Cheers, Mitchel I'm waiting for Snap internationalized version to use it again ;-))) Anyway, auto detect implies some pattern matching. Beside being dependent from local usage (visual separators, digits number, ...), I think it's better to reply on embedded and invisible HTML code to detect phone, if you have the chance to modify the web application. What I mean is instead of looking for patterns to match, I will first look into HTML code for callto: or tel: or sip: or whatever tag. The trouble is I'm not aware yet of any widely adopted usage for that. Which tag would you yourself recommend for that ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Callto:// tags inside web pages
2007/8/7, Dean Collins [EMAIL PROTECTED]: Mitchel, he's not looking for a click to dial solution - he wants to implement some form of click on his website so people can call him. At the end of the day most people aren't going to have it configured correctly etc and you should really use web page based softphone. Regards, Dean, In fact, I'm extending an existing Web directory application to use it as an operator console to welcome and forward incoming calls. So it should simply be able to dial local extension or outside numbers. To minimize modifications, I'm wondering if I could just : - append some tags around displayed extensions and numbers in existing web pages - configure with browser options the script and parameter to use when operator click on those web pages Ideally, it should be usable from both Firefox and IE. How would you proceed to avoid duplicating development effort ? 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] Siemens Gigaset DECT base provisioning
Hello, My goal is to provision C450IP or S450IP models. Has anyone a hint to provision them from configuration files ? Usually, we use dedicated menu embedded inside Gigaset handset. An http server also exists but I couldn't find any dhcp-tftp combination to configure them. Any clue ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Learn some terminalogy before mountingthistask.
Flash Operator Panel would do it. Also the Aastra 55i phones with the expansion module, which has 36 lines on it should work, but you will need to cofigure your Asterisk for Shared Line Appearances (also called Bridged Line Appearance) for the Busy Lamp Field (BLF) to work. The Aastra 55i would show you if they are talking or not. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de James R. Stevens Enviado el: lunes, 06 de agosto de 2007 5:39 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Learn some terminalogy before mountingthistask. All, In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 02, 2007 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mounting thistask. On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote: All, It's been some time since this thread was alive but we are now seeing some progress in this project. Which I will document. We have ordered a T1 for the new building which we are moving (We are getting 14 channels of the T1.) and have a Sangoma A101 card for a 3U rack server. The T1 will have B8ZF decoding and ESF framing which the sangoma card should handle. They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ? Yes. You want NI2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Marceau Sent: Tuesday, April 10, 2007 11:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Learn some terminalogy before mounting this task. James, I'm sorry that I can't add anything but just wanted you to know that I am watching this thread with great interest and suspect that many others will too. Thanks in advance for posting lots of details as you go thru the process. Pierre [EMAIL PROTECTED] 4/10/2007 10:41:36 PM Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS handsets) ports, etc. Would I be better off getting a Zapata T1 card and forgetting the Channel bank all together(Use the digital signal)? You could do that. Personally, the easiest approach I would say would be to order a PRI. They've probably considerably gone down in prices,
Re: [asterisk-users] Query
You can set the caller-id in many different ways but the easiest in by setting it in the sip.conf profile for the extension. So you can just add a line like this to your sip.conf under the extension: callerid=Your Name 5554441212 Hope this helps.. Regards, Todd R. -- Prestige Messaging Live Answering Services SIP or Toll-Free Connectivity Light Accounts From $14.95/mo http://www.PrestigeMessaging.com On 8/8/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I am running asterisk PBX ( digium TE120P card configured) on one system. It is connected to E1 card running application on the other system. After establishing sync between two card, I am able to place call from sip phone to E1 card running application. I want to pass the callerid, when calling from sip phone to E1 card running application. Which all configuration files is to be changed in the asterisk. I am doing the following changes in extensions.conf exten=115,1,Dial(ZAP/g1/115,20) So, extension 115 is received at other end as callerid. Is it correct. Can any body help in how to configure for callerid with digium card. thanks and regards sanchal ___ --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] Free sitting
When I tried it, when a user login at a phone, it replaced any previously logged one. hope that help Implant them with RFIDs. Thanks, Steve Tattoos and barcode scanners. 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] OT - Callto:// tags inside web pages
Zoiper is using the callto: tag (see http://www.zoiper.com/downloads/Zoiper_2.0_Biz_Manual.pdf page 46.) It also works without the extension= in there. (I will update the information today to show all the ways we support) the SIP wouldnt work if the user wants to use IAX instead. (Depending on if you put just a number in there, or a number + hostname for the server to use) Zoa Olivier wrote: 2007/8/7, mitcheloc [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Ollvier, You could use the Firefox plug-in for Snap. It will auto detect numbers on a webpage and make them dialable. Cheers, Mitchel I'm waiting for Snap internationalized version to use it again ;-))) Anyway, auto detect implies some pattern matching. Beside being dependent from local usage (visual separators, digits number, ...), I think it's better to reply on embedded and invisible HTML code to detect phone, if you have the chance to modify the web application. What I mean is instead of looking for patterns to match, I will first look into HTML code for callto: or tel: or sip: or whatever tag. The trouble is I'm not aware yet of any widely adopted usage for that. Which tag would you yourself recommend for that ? ___ --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] Free sitting
Am Dienstag, den 07.08.2007, 07:47 +0200 schrieb Olivier: So no proper logoff between logins, right ? As I will apply free sitting in school environment, chances are phones would then remain logged-in several hours or days between another user logs in. My thoughts are focused on finding the right balance between cost control and ease of use requirements. Maybe, we should program something like 3 states logins : - normal status : user receives call or can call cheap destinations - enhanced status : user can call expensive destinations - logged off status : no incoming calls Downgrading from enhanced to normal status is automatic : if a teacher is working during off hours, he will still receive incoming calls even after being downgraded to normal status. To elevate to enhanced status, you just have to enter your PIN code. What do you think of this ? has anyone tried something approaching ? This somehow reminds me of how sudo works: For the first time you want to run a root command, you have to enter your password. After that, the password will stick (not be asked again) for a few minutes. You surely could put together something like that (time based): The first time you want to place an expensive call, enter your pin: The phone will be granted access for this call +15 minutes, and every next usage of the phone (incoming or outgoing) appends additional time. Same for follow-me function: Keep the person logged in for incoming calls for 90 minutes after the last time he used the phone, or until he logs out. I would probably implement in like that in an environment like a school office, where people share desks: They still _can_ logout, but there will not be much harm if they do not. An intelligent system could also couple the login to the logout of the previous teacher (if that is reasonable in that environment), and auto-login a teacher to the phone adjacent to the PC standing on the desk... BR Anselm ___ --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] ISDN30 card for UK : sanity check
Price. They are good cards, just bells and whistles plus the Echo cancellation on the a101d. Ask Sangoma, their must have a reason for still selling them ;-) Gavin. On 07/08/07, Rory Campbell-Lange [EMAIL PROTECTED] wrote: Hi Gavin Many thanks for the note. For what reason do you recommend the old a101 though? Regards Rory On 07/08/07, Gavin Henry ([EMAIL PROTECTED]) wrote: Very good. Sangoma cards are great. Get the a101d though. Nice wee review: http://www.smithonvoip.com/new-voip-products/sangoma-a101d-single-port-t1e1-card/ Voipon are great guys too. We resell for them. On 07/08/07, Rory Campbell-Lange [EMAIL PROTECTED] wrote: We will be connecting our Asterisk server to ISDN 30 and intend using the Sangoma A101 card. The install location is in London (UK). Sangoma card at Voipon http://www.voipon.co.uk/sangoma-a101-pri-isdn-card-p-132.html?gclid=CI32vJz22I0CFQXklAodIgjHaA I would be grateful to hear if this is the right choice of card. Usage reports would be helpful. -- Rory Campbell-Lange Director Campbell-Lange Workshop Ltd. [EMAIL PROTECTED] www.campbell-lange.net ___ --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] OT - P-asserted-identity and remote id
Hi, The case I'm working on is : - a call comes from PSTN to a given extension (say 122) - a user picks the call up (dialing *8122) from another extension (say 240) using a SIP hardphone - the hardphone (he one with 240 extension) displays the dialed string (here *8122) instead of original caller-id. This is logical but I would like to change this default behaviour so that original caller-id is displayed along or instead of dialed string. SIP hardphone vendor says it could be either done with : 1. SIP MESSAGE 2. SIP P-asserted-identity (Unfortunately both way are not supported yet by this phone vendor but that's another story). Anyway, I tried to understand what SIP P-asserted-identity is and how it relates to my case. I can't see any relation. Can anyone explain ? My thoughts are : - when replying 200 OK to SIP INVITE (from 240 extension), Asterisk server has to add a field P-Asserted-Identity in SIP header filled with original caller-id (using SipAddHeader ?) - when receiving such SIP message, P-Asserted-Identity-enabled SIP phone would display P-Asserted-Identity field data instead of what it uses to display - is this correct ? - and what if phone also supports caller-name display (phone has embedded directory and when caller-id matches with a directory entry, it replaces caller-id with caller-name) ? Shall I understand it will use P-Asserted-Identity data for directory lookup instead of other SIP header field ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk1.2 to 1.4 g711a fax
i have problem with pass-through faxing
with this scenario
hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen
virtual) - linksys ATA
i can fax to fax2mail on hylafax
but after upgrade asterisk2 to 1.4 faxing is not working
hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen
virtual) - linksys ATA
configuration is same
do you hava any idea what is changed in 1.4 in g711 pass-through faxing?
thanks
Jitterbuffer behavior, maybe?
jbenable=yes or no has no effect
BUT i'm discover that with clear DIAL command fax works but if i use AGI
(like a2billing etc) then fax FAIL
any ideas?
can you someone confirm that faxing with this simple AGI script is
working? (phpagi is from phpagi.sf.net)
#!/usr/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
$agi-answer();
$dialstr = SIP/asterisk1/1|300|HgL(61:61000);
$myres = $agi-exec(DIAL $dialstr);
$agi-hangup();
?
thanks!
Marek Cervenka
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[asterisk-users] Monitor doohicky got event Event 160 on channel..
Hi all, I am seeing on my logs this message: Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 (repeated much more then what I will show here). I see that it comes from static void* do_monitor(void *data)in chan_zap.c, but I do not understand what does it mean, and now why is it spamming my logs. Can anyone give me a hint...? - diego ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk wait for traling digits
Dear all
I have asterisk setup now what happend when i dial 4 digit
number my asterisk wait for few digit why when i press # key it is dialing fast
but without # wait for few number is there any configuration for dialplan
I have setup asterisk with avaya system i have 5 avaya system on 5
location i use 16XX,22XX,33XX,44XX,20XX to reach avaya extentions but when i
press 1627 then it is wait for 5 second and then rining start alternative press
'#' what is the method to break this space of waiting after dialing
my extention.conf
;North Delhi NOC Extention
exten = _16XX,1,Dial(SIP/mediant/${EXTEN},60)
exten = _16XX,2,Playback(vm-nobodyavail)
exten = _16XX,102,Playback(all-allbusy)
;Mumbai NOC extention
exten = _22XX,1,Dial(SIP/mediant/${EXTEN},60)
exten = _22XX,2,Playback(vm-nobodyavail)
exten = _22XX,102,Playback(all-allbusy)
exten = _17XX,1,Dial(SIP/mediant/${EXTEN},60)
exten = _17XX,2,Playback(vm-nobodyavail)
exten = _17XX,102,Playback(all-allbusy)
exten = _20XX,1,Dial(SIP/mediant/${EXTEN},60)
exten = _20XX,2,Playback(vm-nobodyavail)
exten = _20XX,102,Playback(all-allbusy)
exten = _33XX,1,Dial(SIP/mediant/${EXTEN},60)
exten = _33XX,2,Playback(vm-nobodyavail)
exten = _33XX,102,Playback(all-allbusy)
exten = _44XX,1,Dial(SIP/mediant/${EXTEN},60)
exten = _44XX,2,Playback(vm-nobodyavail)
exten = _44XX,102,Playback(all-allbusy)
exten = _79XX,1,Dial(SIP/mediant/${EXTEN},60)
exten = _79XX,2,Playback(vm-nobodyavail)
exten = _79XX,102,Playback(all-allbusy)
exten = _08XXX,1,Dial(SIP/mediant/${EXTEN},60)
exten = _08XXX,2,Playback(vm-nobodyavail)
exten = _08XXX,102,Playback(all-allbusy)
exten = _0.,1,Dial(SIP/mediant/${EXTEN:1})
exten = _0.,2,Congestion
exten = _11.,1,Dial(SIP/mediant/${EXTEN:2})
exten = _11.,2,Congestion
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[asterisk-users] Siemens Openstage Asterisk ?
Hi, is anyone on the list using the Siemens Openstage phones together with asterisk? If yes, is it possible to use the programmable keys of these phones together with Asterisk? Thanks for any hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound dialing
Hi Drew. Thanks for the tips. My Line 1 works as I'd like it to, and I could be
wrong, but I don't think changing the dialplan there will help. I really just
want to be able to dial local phone calls (7 digits) and have it go out the
SPA3102, without having to dial twice. This is a snip what I have so far.
extentions.conf
exten = _NXX,1,Dial(SIP/201/${EXTEN},20)
exten = _NXX,2,Hangup
sip.conf
[201]
type=friend
username=x
secret=x
host=dynamic
context=sip
nat=yes
canreinvite=yes
qualify=yes
subscribecontext=localextensions
dtmfmode=rfc2833
vmexten=voicemail
disallow=all
allow=ulaw
allow=gsm
On the SPA (in the PSTN Line tab)
Dial Plan 1: (xxxS0:@gw0)
Dial Plan 2: S0:255
DialPlan 1 is just what I have for now
DialPlan 2 is my extention 255, with a PSTN call comes in it rings my SIP phone.
I set my VoIP Answer Delay to 0 (from 1, I also tried 5 some time ago) and I
set the SPA To PSTN Gain to 5 and now 15.
With things the way I have them now, when I dial a local number, I get a single
DTMF tone on the phoneline, not sure what digit it is.
Tim
- Original Message -
From: Drew Gibson
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, August 07, 2007 5:55 PM
Subject: Re: [asterisk-users] Outbound dialing
Tim,
If the Asterisk stuff below doesn't fix it, try the docs at
http://www.jmgtechnology.com.au/spa_3000_guide.pdf
Ensure you enable VoIP to PSTN gateway mode and that PSTN Line is
registered with Asterisk. This is probably OK as you appear to get dialtone
back from the SPA. If you are calling from the phone on Line 1, make all
calls go through Asterisk. See above docs for details.
In case you are dialing from a phone on Line 1, here is the Line 1
dialplan from my home SPA3102...
(*xx|[3469]11|0|00|[29]x|1xxx[2-9]xx|2[01]x|50[01]|.)
I can't remember if that is default or if I tweaked it. Works in Ontario.
If that is OK, try increasing the gain SPA to PSTN. If the gain is too low,
the DTMF may not be recognised by the CO. I found this out whilst
troubleshooting echo problems.
regards,
Drew
Nicholas Blasgen wrote:
Not specific to the SPA3102, but just normal outbound dialing is as follows:
exten = _1NXXNXX,1,Dial(trunk type/name/${EXTEN})
or if you want to require people to dial 9, then:
exten = _91NXXNXX,1,Dial(trunk type/name/${EXTEN})
or if you're like me and you're used to a cell phone and don't like dialing
the 1:
exten = _NXXNXX,1,Dial(trunk type/name/1${EXTEN})
On 8/7/07, Tim Johnson [EMAIL PROTECTED] wrote:
Hello all. I am just getting back into Asterisk and I am setting up my
Linksys SPA3102. I have incoming calls working fine, as is the phone
plugged into the unit. My problem is I cannot get the SPA3102 to dial
a phone number automatically. I can call the extention of the PSTN and
I get a second dialtone, and I can then manually dial. I'd like to be
able to have Asterisk pass the number I dialed to the SPA and have it
dialout. I've played with dialplans on the SPA I've found during my
googling, but I think it might be something I am missing in my
extentions.conf file. Any ideas?
Tim
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/Nick
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Drew Gibson
Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com
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[asterisk-users] Asterisk AND Cisco Phones in H323 cloud...problems with some models.
Hi to all, I'm using asterisk 1.4.9 with chan_h323. When someone in the H323-VoIP cloud dial 1234 this number is assigned to my asterisk-machine, so the VoiceGW forward the flow to my machine, asterisk though the dialplan can delivery the call to a particular SIP phone...this is ok... I can also dial from my sip phone every phone in the H323-VoIP cloud like siemensBUT...when I call to a cisco phone (model 7912) this start ringing asterisk*CLI -- Executing [EMAIL PROTECTED]:1] Dial(SIP/user-08219f40, H323/[EMAIL PROTECTED]|60)|Ttm) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called [EMAIL PROTECTED] -- Started music on hold, class 'default', on SIP/user-08219f40 -- H323/XXX.XXX.XXX.XXX-10 is ringing -- H323/XXX.XXX.XXX.XXX-10 is ringing I answer and == Everyone is busy/congested at this time (1:0/0/1) -- Stopped music on hold on SIP/user-08219f40 == Auto fallthrough, channel 'SIP/user-08219f40' status is 'CHANUNAVAIL' asterisk*CLI but when I call to cisco 7940 all thinghs function very well...problems only with 7912... any ideas??? bye -- Alessandro 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] OT, I'm looking for SIP/register-enabled softphone
I believe X-Lite v3 (and EyeBeam) from Counterpath both support 3856 as presence user agents. N. Kate Kretz wrote: sorry, I meant RFC 3856, sip presence, not sip regitration On 8/7/07, *Tzafrir Cohen* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On Tue, Aug 07, 2007 at 08:16:35PM +0600, Kate Kretz wrote: Can You please advice me free softphone which supports SIP registrations ? twinkle? ekiga? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] mailto:jabber:[EMAIL PROTECTED] +972-50-7952406 mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir http://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 ___ --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] pick sip channel whn two party talking
Dear all i need this feature in asterisk whn 2 party calling that time i pickup call and listen conversation of that party spoofing like is it possible in asterisk Rgds satish patel - Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Callto:// tags inside web pages
I'm confused is it for a single installation? Why bother messing around - just install a softphone and set it up right. If it's for a deployment to multiple sites is the web app commercial? If so then buy a,license for one of the java softphone solutions - there's a few free and non free versions out there that will do what you need. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Wednesday, 8 August 2007 3:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT - Callto:// tags inside web pages 2007/8/7, Dean Collins [EMAIL PROTECTED]: Mitchel, he's not looking for a click to dial solution - he wants to implement some form of click on his website so people can call him. At the end of the day most people aren't going to have it configured correctly etc and you should really use web page based softphone. Regards, Dean, In fact, I'm extending an existing Web directory application to use it as an operator console to welcome and forward incoming calls. So it should simply be able to dial local extension or outside numbers. To minimize modifications, I'm wondering if I could just : - append some tags around displayed extensions and numbers in existing web pages - configure with browser options the script and parameter to use when operator click on those web pages Ideally, it should be usable from both Firefox and IE. How would you proceed to avoid duplicating development effort ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound dialing
On Wednesday August 08 2007 8:28 am, Tim Johnson wrote:
Hi Drew. Thanks for the tips. My Line 1 works as I'd like it to, and I
could be wrong, but I don't think changing the dialplan there will help. I
really just want to be able to dial local phone calls (7 digits) and have
it go out the SPA3102, without having to dial twice. This is a snip what I
have so far.
extentions.conf
exten = _NXX,1,Dial(SIP/201/${EXTEN},20)
exten = _NXX,2,Hangup
sip.conf
[201]
type=friend
username=x
secret=x
host=dynamic
context=sip
nat=yes
canreinvite=yes
qualify=yes
subscribecontext=localextensions
dtmfmode=rfc2833
vmexten=voicemail
disallow=all
allow=ulaw
allow=gsm
On the SPA (in the PSTN Line tab)
Dial Plan 1: (xxxS0:@gw0)
Dial Plan 2: S0:255
DialPlan 1 is just what I have for now
DialPlan 2 is my extention 255, with a PSTN call comes in it rings my SIP
phone.
I set my VoIP Answer Delay to 0 (from 1, I also tried 5 some time ago) and
I set the SPA To PSTN Gain to 5 and now 15.
With things the way I have them now, when I dial a local number, I get a
single DTMF tone on the phoneline, not sure what digit it is.
snip
I believe you want to put the dial plan in line 1 and reference gw0 there.
On the SPA (in the PSTN Line tab)
Should be in line 1 tab
Dial Plan 1: (xxxS0:@gw0)
Dial plan should be in line 1 dial plan:
normal-dial-plan| xxx:@gw0|[49]11:@gw0|some-more-dial-plan-if-needed
notice where the is in relation to the digits
this will send all seven digit calls out the PSTN and also all 411 and 911
calls out the PSTN line and also 411 and 911 calls. If you leave the PSTN
dial plan as factory default it should work.
If memory serves.
JohnM
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[asterisk-users] Buddy watch and the hint priority - brain teaser
Apologies if this is a resend, but I've sent this 12 hours ago and still
can't see it on the list.
Hi,
I've just started to setup my phones with Buddy watch. Basically, it all
works fine when using the simple example on the wiki:
exten = 123,hint, SIP/some_sip_reg
exten = 123,1,SIP/some_sip_reg
BUT, what I need to do is dynamically decide where the hint checks for buddy
status, because I am using patterns in this context.
In other words, I need to find out the values of ${some_sip_reg} before the
using the hint priority. Ideally, something sort of like this:
exten = _XXX,hint,Set(hint_reg=${EXTEN}-reg}
exten = _XXX,hint,SIP/${hint_reg}
exten = _XXX,SIP/${EXTEN}-reg}
Or, even easier (if it can even be done) is write a function:
exten = _XXX,hint,SIP/ReturnCorrectRegistration()
What's the best way to approach my problem?
Mike
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Re: [asterisk-users] Query
Hi Sanchal, 115 in your case is just DIALLED NUMBER and it will be searched by you E1 trunk. If you want change your CALLERID, you would insert one default or would insert one to each user. the command is the same sendt by Todd: callerid=Your Name 5554441212 but you can work with function callerid and set up it in the same extensions. more informations about it, you have in http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid all the best and good luck, Thiago Maluf. 2007/8/8, [EMAIL PROTECTED] [EMAIL PROTECTED]: Hi, I am running asterisk PBX ( digium TE120P card configured) on one system. It is connected to E1 card running application on the other system. After establishing sync between two card, I am able to place call from sip phone to E1 card running application. I want to pass the callerid, when calling from sip phone to E1 card running application. Which all configuration files is to be changed in the asterisk. I am doing the following changes in extensions.conf exten=115,1,Dial(ZAP/g1/115,20) So, extension 115 is received at other end as callerid. Is it correct. Can any body help in how to configure for callerid with digium card. thanks and regards sanchal ___ --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 -- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: [EMAIL PROTECTED] ___ --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] Help : problem in SLA (Shared Line Apperence
On 8/7/07, raviprakash sunkara [EMAIL PROTECTED] wrote:
Hello Russell,
Nice To meet U and Good Morning. I got u r mail-Id from
http://www.asterisk.org/node/48325
Recently i started the SLA configuration. But i didn't understand the
Flow of its Functionality
One of the My Client Ask to have do deploySLA feature
He Using the Aastra 55i, when users is busy , Aastra 55i will blink lamps
in SLA.conf
slatest]
type=trunk
device=SIP/1001
autocontext=slatest
[slatest1]
type=trunk
device=SIP/1003
autocontext=slatest1
[slateststation]
type=station
device=SIP/1002
autocontext=slateststation
trunk=slatest
trunk=slatest1
sip.conf
[1001]
type=friend
username=1001
secret=1001
host=dynamic
;context=slatest
context=slatest
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1002]
type=friend
username=1002
secret=1002
host=dynamic
;context=default1
context=slateststation
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1003]
type=friend
username=1003
secret=1003
host=dynamic
;context=default1
context=slatest1
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
Dialplan
[testing]
exten = _100X,1,Dial(SIP/${EXTEN}/${EXTEN})
exten = 101,1,Goto(slateststation|102|1)
exten = 102,1,Goto(slatest|1|1)
exten = 103,1,Goto(slatest1|1|1)
exten = h,1,Hangup()
[slatest]
exten = 1,1,SLATrunk(slatest)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slatest1]
exten = 1,1,SLATrunk(slatest1)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slateststation]
exten = 102,1,SLAStation(slateststation)
Thanks Regards
Ravi Prakash Sunkara
India
--
Thanks Regards
Ravi Prakash Sunkara
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[asterisk-users] Zap Bridge Question
asterisk*CLI show channels Channel Location State Application(Data) Zap/3-1 (None) Up Bridged Call(Zap/47-1) Zap/47-1 [EMAIL PROTECTED] Up Dial(ZAP/g1/2105||TWK) Zap/25-1 (None) Up Bridged Call(Zap/1-1) Zap/1-1 [EMAIL PROTECTED]:2 Up Dial(Zap/g2/4999||twk) Zap/26-1 (None) Up Bridged Call(Zap/2-1) Zap/2-1 [EMAIL PROTECTED]:2 Up Dial(Zap/g2/4999||twk) Can I assume those calls are truly bridged above? If so why does zap show channel show me the Echo Cancellation is active when I have requested it not be active on bridged calls? System is a 2x Digium T1 card, one connects to PSTN the other to a Nortel phone system. Zapata.conf follows, if I'm missing something to ensure zap channel bridging please let me know. [trunkgroups] [channels] language=en context=default switchtype=national rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes cidsignalling=bell hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 immediate=no switchtype=national signalling=pri_cpe context=from-pri channel=1-23 group=2 signalling=pri_net context=from-nortel channel=25-47 signalling=fxo_ks channel=49 signalling=fxs_ks channel=52 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- 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] asterisk wait for traling digits
satish patel said I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan This part of the dial plan looks like it should dial without the wait. Could there be another part of your dial plan that starts with '16'? If not have you reloaded extenions.conf either by restarting asterisk or doing an 'extensions reload'? Don Pobanz ___ --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] Method for scripting options specified in make menuconfig
On 8/8/07, arkda [EMAIL PROTECTED] wrote: I've been digging around and I haven't found a way to do this, but I have a feeling I'll feel like an idiot because it's something I'm over looking. Normally if I need to specify an additional option (such as different language sound files) or I'm building an Asterisk server with a lean configuration and need to remove some modules I do so with 'make menuconfig'. I've ran into a need however to install Asterisk entirely from the command line, so I'm looking for the method of accomplishing what I've normally done through 'make menuconfig' solely from the command line. Anyone know how this is accomplished? After you run make menuselect, you'll have a file 'menuselect.makeopts' in your asterisk source dir. Copy that to /etc/asterisk.makeopts (or ~/.asterisk.makeopts) and it will be used for future builds. Once you've copied the file over, do a 'make distclean ; ./configure ; make' to check that it worked. It's the same idea for asterisk-addons, except you copy its menuselect.makeopts to /etc/asteriskaddons.makeopts. -- 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] asterisk wait for traling digits
i have only one single 16XX dialplan for reached to avaya system then why i have to wait for more digit satish patel Don Pobanz [EMAIL PROTECTED] wrote: satish patel said I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan This part of the dial plan looks like it should dial without the wait. Could there be another part of your dial plan that starts with '16'? If not have you reloaded extenions.conf either by restarting asterisk or doing an 'extensions reload'? Don Pobanz ___ --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 - Pinpoint customers who are looking for what you sell. ___ --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 wait for traling digits
This is part f the phones dial plan. Our aastra phones do the same thing. Most phones allow you to configure the dial plan on them. satish patel wrote: i have only one single 16XX dialplan for reached to avaya system then why i have to wait for more digit satish patel */Don Pobanz [EMAIL PROTECTED]/* wrote: satish patel said I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan This part of the dial plan looks like it should dial without the wait. Could there be another part of your dial plan that starts with '16'? If not have you reloaded extenions.conf either by restarting asterisk or doing an 'extensions reload'? Don Pobanz ___ --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 Pinpoint customers http://portal.mxlogic.com/redir/?atTQSjhOUqenT3qtXTvhvp7ndw0SWt53ySAWRVvfcPeoujvLw1g0tfSdyqKNa_ek2f5J9RHO-r5rablxiIvgF-NIj5j9EVU8AGD1cojjjsqIGIs1Z9RGRqpAUgmy30RGxM7qECsd3rh0V-VK_nLt6WtQXTdTdXivNBgGnrFYq5O5mUm-wafBitegAhASHOVJNdwQsCQknD7TAm1P1JZAS2_id41FrSA_zaxkKTjUQdbFEwSA_zaxkQg6dBcQgeRyq89NQ-k29EwgAhBexKvxYYmfSk3q9J4SDtBxBwQszDC3vZCceKlBwho are looking for what you sell. ___ --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 -- Michael Rice Systems Administrator Office: 210-366-2500 Ext. : 231 Direct: 210-293-6231 McClelland and Hine, Inc. ___ --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] Siemens Openstage Asterisk ?
Hi, I don't have this answer but would be curious to know its price for reseller. Any clue ? 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] PRI Reset
Is it normal for a PRI to reset the inactive B channels periodically(like once every hour). I'm seeing on my asterisk console successful restarts, just curious as this is all new to me. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- 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] Which spandsp unicall version to use with 1.2?
Patrick wrote: Hi all, Anyone have an idea which version of spandsp, libunicall, libmfcr2, libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the latest asterisk 1.2? Would that be the ones listed below? http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/ http://www.soft-switch.org/downloads/unicall/unicall-0.0.5pre1/ http://www.soft-switch.org/downloads/snapshots/unicall/asterisk-1.2.x-20060205/ For * 1.2 use: spandsp-0.0.2 and the apps that accompany it. unicall-0.0.3pre11 and the chan_unicall that accompanies it. 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] PRI Reset
On Wed, 2007-08-08 at 09:29 -0500, Jeremy Mann wrote: Is it normal for a PRI to reset the inactive B channels periodically(like once every hour). I’m seeing on my asterisk console successful restarts, just curious as this is all new to me. Yes, that's normal. You can disable it (as I usually do) by setting resetinterval=never in zapata.conf. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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] PRI Reset
Absolutely normal, yes. -Darren -- Darren Nickerson Senior Sales Support Engineer Telephony Depot www.telephonydepot.com +1.215.825.8710 ext 8106 (office) +1.215.243.8335 (fax) - Original Message - From: Jeremy Mann To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, August 08, 2007 10:29 AM Subject: [asterisk-users] PRI Reset Is it normal for a PRI to reset the inactive B channels periodically(like once every hour). I'm seeing on my asterisk console successful restarts, just curious as this is all new to me. -- This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- 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___ --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] How to write a function with a return value in Asterisk
Hi, Is it possible to write a function in Asterisk, that returns a value? Sort of like any programming language allows? For example, I`d like function ReturnSipReg to return the right SipRegistration to dial, based on some value so that I could use it in my dial plan: i.e: exten = 1234,1,Dial(SIP/ReturnSipReg(John)) ; would dial John's extension, which I don't know at this point to which Sip Registration it's associated. ReturnSipReg would find that out for me. Unfortunately, doing it in two steps (by setting a variable and using it after) can't be done, I need it to all be done in the same Asterisk priority. See my previous email for background (Buddy watch and the hint priority - brain teaser). Any help is extremely appreciated. 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
[asterisk-users] Order of matching SIP packet to sections in sip.conf
Hi, When Asterisk receives SIP INVITE packets, it tries to match the packet to a section on sip.conf, so that it can know what context of the dialplan should be used, what codec's are allowed, etc. (what else does it do here?) I would like to know what is exactly the order for this matching considering Asterisk 1.4. I guess it's something like this: 1. It tries to find type=peer sections where the host=... setting is the same as the Host: header on the SIP packet. 2. It tries to find type=user sections where the username or the thing in [...] is the same as the authenticated username on the SIP packet. 3. It tries to find domain=... on the [default] section, where the configured domain is the same as the @... part on the To: header on the SIP packet. I guess it's more or less like this, but I'm not certain of the details... could someone tell me exactly how it's done? If you can point me to the code that does it, it would be fine. Also, regarding authentication, as far as I know, usually SIP INVITE packets are sent unauthenticated, then Asterisk will reply with an 403 Proxy Auth Required, and then the original UAC will retry, now sending a SIP INVITE packet with authentication information. Where, in the algorythm above, will Asterisk know that the user should authenticate and issue the 403 response? Thanks, Filipe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
On Wednesday 08 August 2007 11:24:45 am Mike wrote: Is it possible to write a function in Asterisk, that returns a value? Sort of like any programming language allows? Digium has taken the stance that it's better to set arbitrary variable names to arbitrary values rather than allow what many would consider the perfectly accepted method of using a $? type of return code in addition to any application-specific variables. It's been a long-standing sore point with me for sure, since there is no standard way to see if an application returned successfully or not; you have to consult the individual application to see what it sets. -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
Re: [asterisk-users] How to write a function with a return value inAsterisk
But what if I wanted to write my own custom application for one specific
purpose, I can't set a return value? It's not possible at all?
Let me put it this way then, if I needed to have some processing all done in
the same Asterisk priority (in my case, I want to use the hint priority
but I need to find the value of a variable and use it in the same line).
i.e.:
1) First find out some value ${A}
2) Use ${A} in my Dial command (ex: Dial(SIP/${A})
But this has to be done in the same priority...since hint seems to be an
atomic priority that can only have one line.
Exten = 12345,hint,Dial(SIP/${A}) ; I need to know ${A} first, but I can't
know before this line is called (it's very DB driven).
What can I do? Am I dead in the water here?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, August 08, 2007 11:35
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to write a function with a return value
inAsterisk
On Wednesday 08 August 2007 11:24:45 am Mike wrote:
Is it possible to write a function in Asterisk, that returns a value?
Sort of like any programming language allows?
Digium has taken the stance that it's better to set arbitrary variable names
to arbitrary values rather than allow what many would consider the perfectly
accepted method of using a $? type of return code in addition to any
application-specific variables.
It's been a long-standing sore point with me for sure, since there is no
standard way to see if an application returned successfully or not; you have
to consult the individual application to see what it sets.
-A.
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Re: [asterisk-users] OT - Callto:// tags inside web pages
Dean, It's for tens of single user : a couple of users at a time on as many locations I can get ! I've got a contact with an ISV with sells directories (with coporate charting capabilities). Today, its software is mainly used to edit and display charts and directories. In directory use, it displays extensions and phone numbers with convenient browsing facilities. I now have the opportunity to ask the ISV to extend its software to include click2call facilities. But given ISV background, soft will remain independant from phone infrastructure as attendant console usage is not the widest spread usage. So bottom line is if infrastructure provides click2call, that's fine and if doesn't, it doesn't really matter. So I'm looking for something as non-intrusive and general as possible : for an attendant welcoming calls all day long, click2call feature is a must and you can pick whatever browser is best for that task. But the operator uses this software from time to time, browser compliance is mandatory. Using a softphone or not for answering call, is not my main concern today (nor having this software installed or downloaded is not my focus) as I know it can somehow be done, at least with an hardphone. But what really matters for me now, is I could trigger click2dial feature. Skype Internet Exporer plugin detects phone numbers and replace them with active callto: tag enabled buttons Shall I aim the same with Asterisk instead of Skype ? Your suggestion to both use sip and callto seems very smart to me. How shall I then link those tags with my click2call or softphone software ? It looks like a Configuration Panel option, but I'm not sure. Your opinion ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - P-asserted-identity and remote id
You can add the header the vendor is suggesting in asterisk as follows;
exten = #,1,SipAddHeader(P-Asserted-Identity:
sip:${CALLERID(num)[EMAIL PROTECTED])
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Wednesday, August 08, 2007 2:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] OT - P-asserted-identity and remote id
Hi,
The case I'm working on is :
- a call comes from PSTN to a given extension (say 122)
- a user picks the call up (dialing *8122) from another extension (say
240) using a SIP hardphone
- the hardphone (he one with 240 extension) displays the dialed string
(here *8122) instead of original caller-id.
This is logical but I would like to change this default behaviour so
that original caller-id is displayed along or instead of dialed string.
SIP hardphone vendor says it could be either done with :
1. SIP MESSAGE
2. SIP P-asserted-identity
(Unfortunately both way are not supported yet by this phone vendor but
that's another story).
Anyway, I tried to understand what SIP P-asserted-identity is and how it
relates to my case.
I can't see any relation.
Can anyone explain ?
My thoughts are :
- when replying 200 OK to SIP INVITE (from 240 extension), Asterisk
server has to add a field P-Asserted-Identity in SIP header filled with
original caller-id (using SipAddHeader ?)
- when receiving such SIP message, P-Asserted-Identity-enabled SIP phone
would display P-Asserted-Identity field data instead of what it uses to
display
- is this correct ?
- and what if phone also supports caller-name display (phone has
embedded directory and when caller-id matches with a directory entry, it
replaces caller-id with caller-name) ? Shall I understand it will use
P-Asserted-Identity data for directory lookup instead of other SIP
header field ?
Regards
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[asterisk-users] FSK callerid
Hello, I installed Asterisk on Dell Precision workstation and configured with sample configuration. I have two TDM400 board and one with 4xFXO and second one 4xFXS module installed. I made call to telephone line connected to FXO port and never seen callerid on those lines. I tested cidsignalling and cidstart types and all doesn't work. /var/log/messages: Aug 3 13:42:51 towerpbx kernel: Module 0: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 1: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 2: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 3: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: input: ImPS/2 Logitech Wheel Mouse as /class/input/input2 Aug 3 13:42:51 towerpbx kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Aug 3 13:42:51 towerpbx kernel: ACPI: PCI Interrupt :02:08.0[A] - GSI 17 (level, low) - IRQ 185 Aug 3 13:42:51 towerpbx kernel: Freshmaker version: 71 Aug 3 13:42:51 towerpbx kernel: Freshmaker passed register test Aug 3 13:42:51 towerpbx kernel: Module 0: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 1: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 2: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 3: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Aug 3 13:42:51 towerpbx kernel: ACPI: PCI Interrupt :02:09.0[A] - GSI 18 (level, low) - IRQ 177 Aug 3 13:42:51 towerpbx kernel: FALC version: 0005 Aug 3 13:42:51 towerpbx kernel: TE110P: Setting up global serial parameters for T1 FALC V2.2 Aug 3 13:42:51 towerpbx kernel: TE110P: Successfully initialized serial bus for card Aug 3 13:42:51 towerpbx kernel: Found a Wildcard: Digium Wildcard TE110P T1/E1 Aug 3 13:42:51 towerpbx kernel: Registered tone zone 0 (United States / North America) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 2 (France) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 0 (United States / North America) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 2 (France) zapata.conf usecallerid=yes callerid=asreceived cidsignalling=dtmf cidstart=ring sendcalleridafter=1 Is there any to support FSK callerid? How can I to debug callerid detection process on asterisk? Regards, Balgaa ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
Mike wrote: Hi, Is it possible to write a function in Asterisk, that returns a value? Sort of like any programming language allows? For example, I`d like function ReturnSipReg to return the right SipRegistration to dial, based on some value so that I could use it in my dial plan: i.e: exten = 1234,1,Dial(SIP/ReturnSipReg(John)) ; would dial John's extension, which I don't know at this point to which Sip Registration it's associated. ReturnSipReg would find that out for me. Unfortunately, doing it in two steps (by setting a variable and using it after) can't be done, I need it to all be done in the same Asterisk priority. See my previous email for background (Buddy watch and the hint priority - brain teaser). Any help is extremely appreciated. 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 AGI ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
I can be a bit slow sometimes, but you said that it's not possible, and on
the other hand told me to write my own function (which appears to contradict
the first statement).
Your example of the use of a function is exactly what I need (Create a
function and Dial(SIP/${MY_FUNKY_NEW_FUNC(ooga)})) , what I don't know is
how to actually write the function with a return value (and Googling this
doesn't get me any relevant result, apparently).
I'd be most thankful for some link to a page that shows how to write such a
function in Asterisk.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, August 08, 2007 11:59
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to write a function with a return
valueinAsterisk
On Wednesday 08 August 2007 11:41:38 am Mike wrote:
But what if I wanted to write my own custom application for one
specific purpose, I can't set a return value? It's not possible at all?
Not possible, to my knowledge.
Let me put it this way then, if I needed to have some processing all
done in the same Asterisk priority (in my case, I want to use the hint
priority but I need to find the value of a variable and use it in the
same line).
Create a function and Dial(SIP/${MY_FUNKY_NEW_FUNC(ooga)})
Exten = 12345,hint,Dial(SIP/${A}) ; I need to know ${A} first, but I
can't know before this line is called (it's very DB driven).
Give the function method a try; that's about the only way I can think of
doing something like that... Note that if it's a very DB driven system, you
can use func_odbc to do what you want by declaring an SQL statement as a
function.
-A.
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Re: [asterisk-users] Which spandsp unicall version to use with 1.2?
2007/8/8, Steve Underwood [EMAIL PROTECTED]: Patrick wrote: Hi all, Anyone have an idea which version of spandsp, libunicall, libmfcr2, libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the latest asterisk 1.2? Would that be the ones listed below? http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/ http://www.soft-switch.org/downloads/unicall/unicall-0.0.5pre1/ http://www.soft-switch.org/downloads/snapshots/unicall/asterisk-1.2.x-20060205/ For * 1.2 use: spandsp-0.0.2 and the apps that accompany it. unicall-0.0.3pre11 and the chan_unicall that accompanies it. Steve Hello Steve, I don't want to bother but this http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz file appears, from ftp server, to be modified the 02-Aug-2007. But when you look at ChangeLog file( between AUTHORS and config-h.in), last modification concerns 0.0.3 version and dates from 06.05.23 (see bellow): 06.05.23 - 0.0.3 - Steve Underwood [EMAIL PROTECTED] - T.38 now implemented, though it needs further polishing. - G.726 and G.722 now implemented. Is there a better way to learn about this software features without disturbing anyone ? Can we help somehow ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value inAsterisk
On Wednesday 08 August 2007 11:41:38 am Mike wrote:
But what if I wanted to write my own custom application for one specific
purpose, I can't set a return value? It's not possible at all?
Not possible, to my knowledge.
Let me put it this way then, if I needed to have some processing all done
in the same Asterisk priority (in my case, I want to use the hint
priority but I need to find the value of a variable and use it in the same
line).
Create a function and Dial(SIP/${MY_FUNKY_NEW_FUNC(ooga)})
Exten = 12345,hint,Dial(SIP/${A}) ; I need to know ${A} first, but I can't
know before this line is called (it's very DB driven).
Give the function method a try; that's about the only way I can think of doing
something like that... Note that if it's a very DB driven system, you can
use func_odbc to do what you want by declaring an SQL statement as a
function.
-A.
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Re: [asterisk-users] How to write a function with a return value in Asterisk
On Wednesday August 08 2007 12:10 pm, Mike wrote:
I can be a bit slow sometimes, but you said that it's not possible, and on
the other hand told me to write my own function (which appears to
contradict the first statement).
Your example of the use of a function is exactly what I need (Create a
function and Dial(SIP/${MY_FUNKY_NEW_FUNC(ooga)})) , what I don't know is
how to actually write the function with a return value (and Googling this
doesn't get me any relevant result, apparently).
I'd be most thankful for some link to a page that shows how to write such a
function in Asterisk.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, August 08, 2007 11:59
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to write a function with a return
valueinAsterisk
On Wednesday 08 August 2007 11:41:38 am Mike wrote:
But what if I wanted to write my own custom application for one
specific purpose, I can't set a return value? It's not possible at all?
Not possible, to my knowledge.
Let me put it this way then, if I needed to have some processing all
done in the same Asterisk priority (in my case, I want to use the hint
priority but I need to find the value of a variable and use it in the
same line).
Create a function and Dial(SIP/${MY_FUNKY_NEW_FUNC(ooga)})
Exten = 12345,hint,Dial(SIP/${A}) ; I need to know ${A} first, but I
can't know before this line is called (it's very DB driven).
Give the function method a try; that's about the only way I can think of
doing something like that... Note that if it's a very DB driven system,
you can use func_odbc to do what you want by declaring an SQL statement as
a function.
-A.
Asterisk will listen on stdin if you have your agi code write the var and
value out to stdout asterisk will then be able touse that var in the dial
plan. this is how I do this in a C++ app that i use often:
fprintf(stdout,EXEC SETVAR RESERVED=1 \n);
then in the dial plan I look at the value of ${RESERVED} and use a gotif to do
what needs to be done based on that value.
JohnM
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[asterisk-users] VoicePulse Connect
Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
On 8/8/07, Mike [EMAIL PROTECTED] wrote: I'd be most thankful for some link to a page that shows how to write such a function in Asterisk. There is a test application in the source tree (not built by default I believe), but it doesn't look like anyone has made an equivalent sample function. However, many of the functions in 1.4 are pretty simple, and would be a good jumping off point. Take func_sha1.c for example: 83 lines in the file, 4 functions and one macro. You could copy that and do the proper renaming to make MY_FUNKY_NEW_FUNC that does exactly what func_sha1 does (or alternatively, nothing by getting rid of the bulk of the sha1() function therein. How big it gets as you add whatever magic that function should perform is up to you of course. -- 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] How to write a function with a return value in Asterisk
On Wednesday 08 August 2007 12:10:47 pm Mike wrote: I can be a bit slow sometimes, but you said that it's not possible, and on the other hand told me to write my own function (which appears to contradict the first statement). That's because I'm a little slow today... I thought you were asking about writing an application that returned a value. Functions by their very nature return values. As for examples... you've got the source, choose one of the simpler functions and see what you can do. I apologize; I thought you were talking about applications returning values. -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
Re: [asterisk-users] How to write a function with a return value in Asterisk
You are looking for the AGI: http://www.voip-info.org/wiki-Asterisk+AGI Anthony James FitzGibbon wrote: On 8/8/07, *Mike* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I'd be most thankful for some link to a page that shows how to write such a function in Asterisk. There is a test application in the source tree (not built by default I believe), but it doesn't look like anyone has made an equivalent sample function. However, many of the functions in 1.4 are pretty simple, and would be a good jumping off point. Take func_sha1.c for example: 83 lines in the file, 4 functions and one macro. You could copy that and do the proper renaming to make MY_FUNKY_NEW_FUNC that does exactly what func_sha1 does (or alternatively, nothing by getting rid of the bulk of the sha1() function therein. How big it gets as you add whatever magic that function should perform is up to you of course. -- 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 ___ --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] FW: OT - Callto:// tags inside web pages
Olivier, I think you are getting confused. Call me on 212-203-4357 and I'll answer your questions but basically I think you are doing this the wrong way. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Wednesday, 8 August 2007 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT - Callto:// tags inside web pages Dean, It's for tens of single user : a couple of users at a time on as many locations I can get ! I've got a contact with an ISV with sells directories (with coporate charting capabilities). Today, its software is mainly used to edit and display charts and directories. In directory use, it displays extensions and phone numbers with convenient browsing facilities. I now have the opportunity to ask the ISV to extend its software to include click2call facilities. But given ISV background, soft will remain independant from phone infrastructure as attendant console usage is not the widest spread usage. So bottom line is if infrastructure provides click2call, that's fine and if doesn't, it doesn't really matter. So I'm looking for something as non-intrusive and general as possible : for an attendant welcoming calls all day long, click2call feature is a must and you can pick whatever browser is best for that task. But the operator uses this software from time to time, browser compliance is mandatory. Using a softphone or not for answering call, is not my main concern today (nor having this software installed or downloaded is not my focus) as I know it can somehow be done, at least with an hardphone. But what really matters for me now, is I could trigger click2dial feature. Skype Internet Exporer plugin detects phone numbers and replace them with active callto: tag enabled buttons Shall I aim the same with Asterisk instead of Skype ? Your suggestion to both use sip and callto seems very smart to me. How shall I then link those tags with my click2call or softphone software ? It looks like a Configuration Panel option, but I'm not sure. Your opinion ? 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
Re: [asterisk-users] VoicePulse Connect
John, Voicepulse Connect has been great to me. I've been using it for over a year now, and do not have any major complaints, except that there are no printable receipts for credit card transactions. SIP is also the preferable protocol, as IAX seems to have some issues. Customer service is usually pretty good, and there have been very few (although a couple) problems with service outages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 12:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoicePulse Connect Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --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] VoicePulse Connect
Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. -John From: Wes Baehr [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Date: Wed, 8 Aug 2007 12:55:29 -0400 John, Voicepulse Connect has been great to me. I've been using it for over a year now, and do not have any major complaints, except that there are no printable receipts for credit card transactions. SIP is also the preferable protocol, as IAX seems to have some issues. Customer service is usually pretty good, and there have been very few (although a couple) problems with service outages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 12:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoicePulse Connect Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --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 _ Find a local pizza place, movie theater, and more .then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theatercp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=950607encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
On Wed, Aug 08, 2007 at 11:34:49AM -0400, Andrew Kohlsmith wrote: On Wednesday 08 August 2007 11:24:45 am Mike wrote: Is it possible to write a function in Asterisk, that returns a value? Sort of like any programming language allows? Digium has taken the stance that it's better to set arbitrary variable names to arbitrary values rather than allow what many would consider the perfectly accepted method of using a $? type of return code in addition to any application-specific variables. Digium has taken the stance that Structured Programming is a Bad Idea? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
On Wednesday 08 August 2007 1:17:24 pm Jay R. Ashworth wrote: Digium has taken the stance that it's better to set arbitrary variable names to arbitrary values rather than allow what many would consider the perfectly accepted method of using a $? type of return code in addition to any application-specific variables. Digium has taken the stance that Structured Programming is a Bad Idea? I don't think it's fair to paint it quite so broadly. M opinion on it is that I have simply failed to show them how clear things become when I can check ONE variable for the status of the last-run application, whether that be a dial, system or agi application call. Look at the Asterisk source; it's not a mass of spaghetti code. Saying that Digium thinks that structured programming is a bad idea is an exaggeration. -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
Re: [asterisk-users] VoicePulse Connect
On Wed, 2007-08-08 at 17:08 +, John Meksavan wrote: Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. My experience with Voicepulse has been good and quality is usually very good. Most of the time when calls get distorted the problems can be traced to my ISP. Unfortunately you will never be able to get 100% reliability when using the Internet. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Howto generate a Manager Event from the Dialplan?
I'd like to be able to generate a Manager Event from the dialplan but can't seem to find a way to do it. Alternatively, trigger and Event when a record in AstDB gets changed. Can anyone point me in the right direction? Thanks. By way of explanation, I've a app that connects to astmanproxy and I'd like it to know when a call group gets put into Nightservice. Putting the call group into Nightservice is done in the dialplan and sets a record in AstDB. It would be infriendly to poll AstDB; hence the requirement for the dialplan to trigger an Event. The call group could also be put into Nightservice by setting the appropriate record directly in AstDB; hence the Event triggered on a AstDB record change. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
Thanks for all the replies, after some thinking AGI seems like the way to go
(writing a function in C would certainly work, but I want to avoid anything
that makes upgrading to newer version of Asterisk a potential pain. Let's
say using C is plan B).
So, I wrote (well, plagarized directly from the Web) a simple Perl program
that prints Hello World. I call it using this:
exten = 12345,1,AGI(agi-helloworld.agi)
Seems to work (I'm not expecting anything, really, just no Asterisk error).
When I try to use it as part Noop like this:
exten = 12345,1,Noop(${AGI(agi-helloworld.agi)})
In the hope of getting to see Noop(Hello World) in my CLI, I get the
following Asterisk error:
Aug 8 13:40:48 ERROR[5771]: pbx.c:1402 ast_func_read: Function AGI not
registered
AGI certainly seems registered as it worked in the first case. Again,
something obvious I missed?
Thank you,
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: Wednesday, August 08, 2007 12:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to write a function with a return value in
Asterisk
You are looking for the AGI: http://www.voip-info.org/wiki-Asterisk+AGI
Anthony
James FitzGibbon wrote:
On 8/8/07, Mike [EMAIL PROTECTED] wrote:
I'd be most thankful for some link to a page that shows how to write such a
function in Asterisk.
There is a test application in the source tree (not built by default I
believe), but it doesn't look like anyone has made an equivalent sample
function.
However, many of the functions in 1.4 are pretty simple, and would be a good
jumping off point. Take func_sha1.c for example: 83 lines in the file, 4
functions and one macro. You could copy that and do the proper renaming to
make MY_FUNKY_NEW_FUNC that does exactly what func_sha1 does (or
alternatively, nothing by getting rid of the bulk of the sha1() function
therein.
How big it gets as you add whatever magic that function should perform is up
to you of course.
--
j.
_
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Re: [asterisk-users] FSK callerid
On Wed, 2007-08-08 at 23:55 +0900, Balgansuren Batsukh wrote: Hello, I installed Asterisk on Dell Precision workstation and configured with sample configuration. I have two TDM400 board and one with 4xFXO and second one 4xFXS module installed. I made call to telephone line connected to FXO port and never seen callerid on those lines. I tested cidsignalling and cidstart types and all doesn't work. Balgaa-- What country are you in? The CID conventions vary considerably from country to country! Your tonezone indicate US/France... Here in the US, the defaults in zapata.conf work fine for me: usecallerid=yes ;cidsignalling=bell ;cidstart=ring Are you sure your lines from the CO are providing CID? Out here where I'm at, the phone company charges extra for the privilege of CID. murf /var/log/messages: Aug 3 13:42:51 towerpbx kernel: Module 0: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 1: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 2: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 3: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: input: ImPS/2 Logitech Wheel Mouse as /class/input/input2 Aug 3 13:42:51 towerpbx kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Aug 3 13:42:51 towerpbx kernel: ACPI: PCI Interrupt :02:08.0[A] - GSI 17 (level, low) - IRQ 185 Aug 3 13:42:51 towerpbx kernel: Freshmaker version: 71 Aug 3 13:42:51 towerpbx kernel: Freshmaker passed register test Aug 3 13:42:51 towerpbx kernel: Module 0: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 1: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 2: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 3: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Aug 3 13:42:51 towerpbx kernel: ACPI: PCI Interrupt :02:09.0[A] - GSI 18 (level, low) - IRQ 177 Aug 3 13:42:51 towerpbx kernel: FALC version: 0005 Aug 3 13:42:51 towerpbx kernel: TE110P: Setting up global serial parameters for T1 FALC V2.2 Aug 3 13:42:51 towerpbx kernel: TE110P: Successfully initialized serial bus for card Aug 3 13:42:51 towerpbx kernel: Found a Wildcard: Digium Wildcard TE110P T1/E1 Aug 3 13:42:51 towerpbx kernel: Registered tone zone 0 (United States / North America) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 2 (France) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 0 (United States / North America) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 2 (France) zapata.conf usecallerid=yes callerid=asreceived cidsignalling=dtmf cidstart=ring sendcalleridafter=1 Is there any to support FSK callerid? How can I to debug callerid detection process on asterisk? Regards, Balgaa ___ --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 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] Howto generate a Manager Event from the Dialplan?
Have you checked out UserEvent: http://www.voip-info.org/wiki/view/Asterisk+cmd+UserEvent Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russell Brown Sent: Wednesday, August 08, 2007 1:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Howto generate a Manager Event from the Dialplan? I'd like to be able to generate a Manager Event from the dialplan but can't seem to find a way to do it. Alternatively, trigger and Event when a record in AstDB gets changed. Can anyone point me in the right direction? Thanks. By way of explanation, I've a app that connects to astmanproxy and I'd like it to know when a call group gets put into Nightservice. Putting the call group into Nightservice is done in the dialplan and sets a record in AstDB. It would be infriendly to poll AstDB; hence the requirement for the dialplan to trigger an Event. The call group could also be put into Nightservice by setting the appropriate record directly in AstDB; hence the Event triggered on a AstDB record change. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] RoundRobin Holding Memory?
I have a queue setup to 'roundrobin' (NOT roundrobin with memory). I have three agents. We'll call them 101, 102, and 103. When a call comes in.. I want it to always try 101 if no answer try 102.. if no answer try 103, etc. However, what it is doing is... it will ring 101... if 101 answers, next time a call comes in it will go to 102. This isn't at all what I want. Any ideas why it might be doing this? [551] wrapuptime=0 timeout=25 strategy=roundrobin retry=1 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format= member=Local/[EMAIL PROTECTED],0 member=Local/[EMAIL PROTECTED],0 member=Local/[EMAIL PROTECTED],0 maxlen=0 leavewhenempty=no joinempty=Yes context= announce-holdtime=no announce-frequency=60 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
On Wednesday 08 August 2007 1:39:34 pm Mike wrote:
exten = 12345,1,AGI(agi-helloworld.agi)
AGI is an application, and you've called it.
exten = 12345,1,Noop(${AGI(agi-helloworld.agi)})
AGI is not a function. You cannot nest applications like that. The NoOp
application cannot call another application.
-A.
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[asterisk-users] Paging Application - Polycom 601
Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a Page. From what I understand, the Page is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller to the only one who can talk. The problem I am having is about 1 out of 25 pages will crash the Polycom 601 (receptionist) and the phone will reboot. This leaves all the extensions in the conference room and each party must hit end call on their phone to get out of the conference. However, the receptionist can't do that because that phone restarts. Once it has rebooted, it does not show to be connected to the conference room. However, I feel like it is still in the conference - with no way out. After one of these crashes, the 601 phone will start having one way audio (can't hear caller), various other weirdness (side car status wrong) and the only way to completely correct the problems are to restart asterisk - which I assume kills the rogue page application. 1) Has anyone ever seen this problem? 2) Is there a way from the CLI to show and kill a page? 3) Any suggestions? Thanks Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
On 8/8/07, Mike [EMAIL PROTECTED] wrote:
So, I wrote (well, plagarized directly from the Web) a simple Perl program
that prints Hello World. I call it using this:
exten = 12345,1,AGI(agi-helloworld.agi)
Seems to work (I'm not expecting anything, really, just no Asterisk
error).
When I try to use it as part Noop like this:
exten = 12345,1,Noop(${AGI(agi-helloworld.agi)})
In the hope of getting to see Noop(Hello World) in my CLI, I get the
following Asterisk error:
Aug 8 13:40:48 ERROR[5771]: pbx.c:1402 ast_func_read: Function AGI not
registered
AGI certainly seems registered as it worked in the first case. Again,
something obvious I missed?
This is calling the AGI application:
exten = something,priority,AGI(program|args)
This is an attempt to call a function called AGI (which doesn't exist) and
pass the results of said non-existent function to the NoOp application:
exten = something,priority,NoOp(${AGI(program|args)})
look at 'core show applications' and 'core show functions' to see what you
can call in each case. Applications and functions aren't interchangeable.
If you want to use an AGI script to set a variable you can later use as an
arg to Dial(), then you want to call the AGI application from your dialplan,
then from inside the AGI script do your calculations and issue the AGI
command SET VARIABLE name value.
So if you have a very basic AGI script that just does this:
echo SET VARIABLE foo bar
then your dialplan could look something like this:
exten = foo,1,AGI(foo.agi)
exten = foo,n,NoOp(${foo})
And you'd expect to see NoOp(bar) on your console when you called that
extension.
Of course, you'd want to use one of the available AGI frameworks to do the
heavy lifting of parsing the input that Asterisk gives an AGI script and
take care of the error handing when you issue a command back to Asterisk
from the AGI script.
--
j.
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Re: [asterisk-users] Learn some terminalogy before mountingthistask.
SLA is not BLF. The only thing you need to configure to have BLF is adding hint priority to your dial plan. On 8/8/07, James Collier [EMAIL PROTECTED] wrote: Flash Operator Panel would do it. Also the Aastra 55i phones with the expansion module, which has 36 lines on it should work, but you will need to cofigure your Asterisk for Shared Line Appearances (also called Bridged Line Appearance) for the Busy Lamp Field (BLF) to work. The Aastra 55i would show you if they are talking or not. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de James R. Stevens Enviado el: lunes, 06 de agosto de 2007 5:39 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Learn some terminalogy before mountingthistask. All, In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, July 02, 2007 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Learn some terminalogy before mounting thistask. On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote: All, It's been some time since this thread was alive but we are now seeing some progress in this project. Which I will document. We have ordered a T1 for the new building which we are moving (We are getting 14 channels of the T1.) and have a Sangoma A101 card for a 3U rack server. The T1 will have B8ZF decoding and ESF framing which the sangoma card should handle. They asked me if we want NI1 or NI2 ?? Is this a reference to the PRI ? Yes. You want NI2. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Marceau Sent: Tuesday, April 10, 2007 11:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Learn some terminalogy before mounting this task. James, I'm sorry that I can't add anything but just wanted you to know that I am watching this thread with great interest and suspect that many others will too. Thanks in advance for posting lots of details as you go thru the process. Pierre [EMAIL PROTECTED] 4/10/2007 10:41:36 PM Hi James, Admittedly, the terminological and conceptual barrier may present some impediments to the completeness and specificity of answers, so we might have to work at this a bit, but let's see how we can help: On Tue, 10 Apr 2007, James R. Stevens said something to this effect: We have a T1 coming into the building(FYI-Our Voice and Data are on separate T's) terminating at the Smart Jack. Are you implying that there are two T1 circuits -- one voice, and one data? Or do you mean that the T1 is channelised and some of the channels are used for voice and some for data? That's kind of what it sounds like. Sounds like you can do 7 calls on voice channels and the rest are provisioned as a clear-channel data pipe. That would mean that you have some equipment for breaking them out on your premises. The channel bank would break out the voice lines as FXO analogue lines (if you set it to) and those probably feed into your PBX. The rest of the channels used for data would probably be signaled out on another T1 interface, but with some subrate DS0 channels missing. That's ust a guess. But what you say below suggests that my theory is wrong, so perhaps it is the case that you have separate voice and data T1s after all, even though you refer to it in the singular. Do be aware that under no circumstances does anyone generally refer to a T1 as a T. :) I can tell you our current phone system can handle 7 phone calls at a time: Does this mean the T only has 7 channels provisioned out of the 24 possible? This is possible. Do you happen to know what kind of signaling is used on it? Is it an ISDN PRI, or an EM trunk? Does a channel (In terms of the T1) = a port? A port on what? The channel bank? Channel banks generally do break the DS0s (subrate 64 kbps channels, of which there are 24 on a T1) out, but some more sophisticated ones have the capability to do other things as well. If so, the answer is yes. How many phone calls can one TDM400 support concurrently? (four ??) If it has four FXO ports and four FXO modules, yes. They come in different combinations. Some come with 2 FXO (outside POTS lines to CO) and 2 FXS (plain analogue POTS
Re: [asterisk-users] Siemens Openstage Asterisk ?
Hi Olivier, Hi, I don't have this answer but would be curious to know its price for reseller. Any clue ? no, I'm sorry. We're only responsible for the configuration of the devices. Our client will buy all the necessary hardware. I will ask him about the prices, but these will be end user prices and it may take 4-6 weeks until I can send you the details. Bye, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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] Help : problem in SLA (Shared Line Apperence
Clarify this, what you are trying to achieve?
To see if handsets are being used or not?
Or to see if any trunk is being used or not and share it?
These are 2 different concepts, first is BLF you can have your asterisk to
provide that information with hint priority, and the second one is SLA.
On 8/8/07, raviprakash sunkara [EMAIL PROTECTED] wrote:
On 8/7/07, raviprakash sunkara [EMAIL PROTECTED]
wrote:
Hello Russell,
Nice To meet U and Good Morning. I got u r mail-Id from
http://www.asterisk.org/node/48325
Recently i started the SLA configuration. But i didn't understand the
Flow of its Functionality
One of the My Client Ask to have do deploySLA feature
He Using the Aastra 55i, when users is busy , Aastra 55i will blink
lamps
in SLA.conf
slatest]
type=trunk
device=SIP/1001
autocontext=slatest
[slatest1]
type=trunk
device=SIP/1003
autocontext=slatest1
[slateststation]
type=station
device=SIP/1002
autocontext=slateststation
trunk=slatest
trunk=slatest1
sip.conf
[1001]
type=friend
username=1001
secret=1001
host=dynamic
;context=slatest
context=slatest
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1002]
type=friend
username=1002
secret=1002
host=dynamic
;context=default1
context=slateststation
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1003]
type=friend
username=1003
secret=1003
host=dynamic
;context=default1
context=slatest1
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
Dialplan
[testing]
exten = _100X,1,Dial(SIP/${EXTEN}/${EXTEN})
exten = 101,1,Goto(slateststation|102|1)
exten = 102,1,Goto(slatest|1|1)
exten = 103,1,Goto(slatest1|1|1)
exten = h,1,Hangup()
[slatest]
exten = 1,1,SLATrunk(slatest)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slatest1]
exten = 1,1,SLATrunk(slatest1)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slateststation]
exten = 102,1,SLAStation(slateststation)
Thanks Regards
Ravi Prakash Sunkara
India
--
Thanks Regards
Ravi Prakash Sunkara
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Re: [asterisk-users] How to write a function with a return value inAsterisk
AH! Thanks, I've been thinking that apps and functions were interchangeable,
hoping that I could return values with functions. Now that this is very
clear in my mind (at least I think it is) I'll go and write a function.
Might as well ask this before I go out, not find my answer and come back to
ask the question: What are the best practice when it comes to building these
functions, and making them useable from Asterisk, without needing to build
Asterisk every single time...? Can I compile/build them separately, and
somehow register them into Asterisk?
I`m much better at writing C code then actually integrating that code in
larger project...unfortunately.
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
FitzGibbon
Sent: Wednesday, August 08, 2007 14:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to write a function with a return value
inAsterisk
On 8/8/07, Mike [EMAIL PROTECTED] wrote:
So, I wrote (well, plagarized directly from the Web) a simple Perl program
that prints Hello World. I call it using this:
exten = 12345,1,AGI(agi-helloworld.agi)
Seems to work (I'm not expecting anything, really, just no Asterisk error).
When I try to use it as part Noop like this:
exten = 12345,1,Noop(${AGI(agi-helloworld.agi)})
In the hope of getting to see Noop(Hello World) in my CLI, I get the
following Asterisk error:
Aug 8 13:40:48 ERROR[5771]: pbx.c:1402 ast_func_read: Function AGI not
registered
AGI certainly seems registered as it worked in the first case. Again,
something obvious I missed?
This is calling the AGI application:
exten = something,priority,AGI(program|args)
This is an attempt to call a function called AGI (which doesn't exist) and
pass the results of said non-existent function to the NoOp application:
exten = something,priority,NoOp(${AGI(program|args)})
look at 'core show applications' and 'core show functions' to see what you
can call in each case. Applications and functions aren't interchangeable.
If you want to use an AGI script to set a variable you can later use as an
arg to Dial(), then you want to call the AGI application from your dialplan,
then from inside the AGI script do your calculations and issue the AGI
command SET VARIABLE name value.
So if you have a very basic AGI script that just does this:
echo SET VARIABLE foo bar
then your dialplan could look something like this:
exten = foo,1,AGI(foo.agi)
exten = foo,n,NoOp(${foo})
And you'd expect to see NoOp(bar) on your console when you called that
extension.
Of course, you'd want to use one of the available AGI frameworks to do the
heavy lifting of parsing the input that Asterisk gives an AGI script and
take care of the error handing when you issue a command back to Asterisk
from the AGI script.
--
j.
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Re: [asterisk-users] Polycom 320 - Can it actually be configured?
Ideally you specify the ftp server via DHCP. Then you don't have to touch the phone. If your files are right on the server, you just plug in the phone and go. That's how we do it and it's a breeze. Scott Doug wrote: At 21:02 8/1/2007, Doug, wrote: At 16:49 8/1/2007, Douglas Garstang wrote: Don't know about the 320, but we provisioned the 301's. They're provisioning is basically the same as the 501's and 601's. What problems are you having? Have no problems with 501s or 601s or 430s. I punch in the provisioning server IP, but the phone doesn't save it. Usually, a phone will prompt to save the config, but this one doesn't. Aa! The key is the left arrow button. By pressing it, you can back out of where you are and get to the save config option. Sure would be nice if they made it user-friendly, instead of user-hostile. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Doug Sent: Wednesday, August 01, 2007 2:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom 320 - Can it actually be configured? Just got one of these. Horrible to program. Trying to key in the FTP server. Won't even remember the info after rebooting. Anybody know the proper way to beat on this stupid beast so it will work? ___ --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
[asterisk-users] FW: The trixbox Revolution Continues! Sign up for the Webinar.
Hmm beginning of the end of free trixbox by the sounds of it. It was good while it lasted but time to download the latest iso while it's still available by the sounds of it. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). From: trixbox [mailto:[EMAIL PROTECTED] Sent: Wednesday, 8 August 2007 2:00 PM To: Dean Collins Subject: The trixbox Revolution Continues! Sign up for the Webinar. trixbox http://echo4.bluehornet.com/ct/ct.php?t=1839203c=1686662876m=mtype=1 h=24C5C6D927F8E6A426872B034CFC94F4 The Official Unveiling You've been hearing about it for a while. And now it's finally here. Introducing the next evolution of the trixbox product family: trixbox Pro. What exactly is trixbox Pro, you ask? We'll tell you...next Monday. Sign up for the Webinar where we'll show you trixbox Pro in action and all the possibilities it brings you. We can only accommodate the first 1000 trixboxers on the webinar so sign up today and call in early! August 13 @ 9:00 AM PDT (16:00 UTC/GMT) http://echo4.bluehornet.com/ct/ct.php?t=1839204c=1686662876m=mtype=1 h=24C5C6D927F8E6A426872B034CFC94F4 Fonality, Inc. Fonality | 200 Corporate Pointe Suite 350 | Los Angeles, CA 90230 | www.trixbox.org http://www.trixbox.org/ This message was intended for: [EMAIL PROTECTED] You were added to the system May 3, 2007. Click here for more information http://echo4.bluehornet.com/subscribe/source.htm?c=bhKnjO3S0caW.email= [EMAIL PROTECTED]cid=105ab23bdcfebb496b833bf4db11024f | Unsubscribe http://echo4.bluehornet.com/phase2/survey1/survey.htm?CID=ypniytaction =update[EMAIL PROTECTED]_mh=8316fb751136f25ce5ff1f7461654597 http://echo4.bluehornet.com/imagelibrary/N-1686662876-979C4976A0579E088 A711D4CD3EC3723.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] Paging Application - Polycom 601
I'm using Page application with Polycom 501 and 601 and have not seen these issue, i would check firmware on 601 and play with couple different firmware. are you checking if the chanavail before sending the Page? On 8/8/07, Bill Andersen [EMAIL PROTECTED] wrote: Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a Page. From what I understand, the Page is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller to the only one who can talk. The problem I am having is about 1 out of 25 pages will crash the Polycom 601 (receptionist) and the phone will reboot. This leaves all the extensions in the conference room and each party must hit end call on their phone to get out of the conference. However, the receptionist can't do that because that phone restarts. Once it has rebooted, it does not show to be connected to the conference room. However, I feel like it is still in the conference - with no way out. After one of these crashes, the 601 phone will start having one way audio (can't hear caller), various other weirdness (side car status wrong) and the only way to completely correct the problems are to restart asterisk - which I assume kills the rogue page application. 1) Has anyone ever seen this problem? 2) Is there a way from the CLI to show and kill a page? 3) Any suggestions? Thanks Bill ___ --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] FSK callerid
Hello, I am from Mongolia and when I use telephone set with callerid it display callerid. Yes, our phone company charge for callerid service. Balgaa - Original Message - From: Steve Murphy [EMAIL PROTECTED] To: Balgansuren Batsukh [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 09, 2007 2:40 AM Subject: Re: [asterisk-users] FSK callerid On Wed, 2007-08-08 at 23:55 +0900, Balgansuren Batsukh wrote: Hello, I installed Asterisk on Dell Precision workstation and configured with sample configuration. I have two TDM400 board and one with 4xFXO and second one 4xFXS module installed. I made call to telephone line connected to FXO port and never seen callerid on those lines. I tested cidsignalling and cidstart types and all doesn't work. Balgaa-- What country are you in? The CID conventions vary considerably from country to country! Your tonezone indicate US/France... Here in the US, the defaults in zapata.conf work fine for me: usecallerid=yes ;cidsignalling=bell ;cidstart=ring Are you sure your lines from the CO are providing CID? Out here where I'm at, the phone company charges extra for the privilege of CID. murf /var/log/messages: Aug 3 13:42:51 towerpbx kernel: Module 0: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 1: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 2: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 3: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: input: ImPS/2 Logitech Wheel Mouse as /class/input/input2 Aug 3 13:42:51 towerpbx kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Aug 3 13:42:51 towerpbx kernel: ACPI: PCI Interrupt :02:08.0[A] - GSI 17 (level, low) - IRQ 185 Aug 3 13:42:51 towerpbx kernel: Freshmaker version: 71 Aug 3 13:42:51 towerpbx kernel: Freshmaker passed register test Aug 3 13:42:51 towerpbx kernel: Module 0: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 1: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 2: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 3: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Aug 3 13:42:51 towerpbx kernel: ACPI: PCI Interrupt :02:09.0[A] - GSI 18 (level, low) - IRQ 177 Aug 3 13:42:51 towerpbx kernel: FALC version: 0005 Aug 3 13:42:51 towerpbx kernel: TE110P: Setting up global serial parameters for T1 FALC V2.2 Aug 3 13:42:51 towerpbx kernel: TE110P: Successfully initialized serial bus for card Aug 3 13:42:51 towerpbx kernel: Found a Wildcard: Digium Wildcard TE110P T1/E1 Aug 3 13:42:51 towerpbx kernel: Registered tone zone 0 (United States / North America) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 2 (France) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 0 (United States / North America) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 2 (France) zapata.conf usecallerid=yes callerid=asreceived cidsignalling=dtmf cidstart=ring sendcalleridafter=1 Is there any to support FSK callerid? How can I to debug callerid detection process on asterisk? Regards, Balgaa ___ --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] How to write a function with a return value in Asterisk
At 10:17 AM 8/8/2007, you wrote: Digium has taken the stance that Structured Programming is a Bad Idea? While it seems that way in hindsight, I'd guess that no thought was put into dialplan programming when it was started and by the time someone realized it was wrong, the person in charge said, we can't break all the current dialplans, which is essentially what it would take to make any real improvement. They could have done it going to 1.4, but the longer they wait, the harder it gets. Been there, done that. Wrote the letter that said, oops, I designed this wrong and as soon as you install the new version of my library all your reports will no longer print correctly. It sucks, but the sooner you do it, the better it is in the end. Ira ___ --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] les.net losing DID's
Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging Application - Polycom 601
Bill Andersen wrote: Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a Page. From what I understand, the Page is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller to the only one who can talk. I would be curious to see how you set up the phones to accept paging, just to make sure there isn't something iffy with your phone configuration. The problem I am having is about 1 out of 25 pages will crash the Polycom 601 (receptionist) and the phone will reboot. Is the 601 calling the page, or receiving a page from another phone? This leaves all the extensions in the conference room and each party must hit end call on their phone to get out of the conference. However, the receptionist can't do that because that phone restarts. Once it has rebooted, it does not show to be connected to the conference room. However, I feel like it is still in the conference - with no way out. You feel like it? Do you know for sure? If the phone does not show an active call, it's not connected to anything. I don't see how it would be in a conference after a reboot. Your problems below are probably caused by something else. The spontaneous reboot is telling. After one of these crashes, the 601 phone will start having one way audio (can't hear caller), various other weirdness (side car status wrong) and the only way to completely correct the problems are to restart asterisk - which I assume kills the rogue page application. The 601s with sidecars have been problematic. What Polycom firmware are you using? 1) Has anyone ever seen this problem? Other users have reported problems with 601s crashing. Check your firmware. AFAIK, the current firmware is 2.1.3. 2) Is there a way from the CLI to show and kill a page? 'show channels' will show you active calls (in 1.2; in 1.4, use 'core show channels') 'meetme kick' lets you kick channels/users from a conference. Still, I don't think that's what's happening here. -Stephen- ___ --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] les.net losing DID's
Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
On Wed, Aug 08, 2007 at 01:34:56PM -0400, Andrew Kohlsmith wrote: On Wednesday 08 August 2007 1:17:24 pm Jay R. Ashworth wrote: Digium has taken the stance that it's better to set arbitrary variable names to arbitrary values rather than allow what many would consider the perfectly accepted method of using a $? type of return code in addition to any application-specific variables. Digium has taken the stance that Structured Programming is a Bad Idea? I don't think it's fair to paint it quite so broadly. M opinion on it is that I have simply failed to show them how clear things become when I can check ONE variable for the status of the last-run application, whether that be a dial, system or agi application call. Look at the Asterisk source; it's not a mass of spaghetti code. Saying that Digium thinks that structured programming is a bad idea is an exaggeration. The original responder unclearly implied that functions couldn't return parameters except as globals; it's been cleared up. My apologies. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --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] les.net losing DID's
Yes they are co-operating to port DID to another provider and they have given time till august 23 so DID will continue to work till then but they are not providing any substitute DID though ( i dont expect that ) but atleast they should partially refund amount for remaining days ( i dont expect that either :P ) . On 09/08/2007, Stephen Bosch [EMAIL PROTECTED] wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk wait for traling digits
This should be configured in phone system instead of asterisk :) . On 08/08/2007, Michael Rice [EMAIL PROTECTED] wrote: This is part f the phones dial plan. Our aastra phones do the same thing. Most phones allow you to configure the dial plan on them. satish patel wrote: i have only one single 16XX dialplan for reached to avaya system then why i have to wait for more digit satish patel */Don Pobanz [EMAIL PROTECTED]/* wrote: satish patel said I have asterisk setup now what happend when i dial 4 digit number my asterisk wait for few digit why when i press # key it is dialing fast but without # wait for few number is there any configuration for dialplan This part of the dial plan looks like it should dial without the wait. Could there be another part of your dial plan that starts with '16'? If not have you reloaded extenions.conf either by restarting asterisk or doing an 'extensions reload'? Don Pobanz ___ --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 Pinpoint customers http://portal.mxlogic.com/redir/?atTQSjhOUqenT3qtXTvhvp7ndw0SWt53ySAWRVvfcPeoujvLw1g0tfSdyqKNa_ek2f5J9RHO-r5rablxiIvgF-NIj5j9EVU8AGD1cojjjsqIGIs1Z9RGRqpAUgmy30RGxM7qECsd3rh0V-VK_nLt6WtQXTdTdXivNBgGnrFYq5O5mUm-wafBitegAhASHOVJNdwQsCQknD7TAm1P1JZAS2_id41FrSA_zaxkKTjUQdbFEwSA_zaxkQg6dBcQgeRyq89NQ-k29EwgAhBexKvxYYmfSk3q9J4SDtBxBwQszDC3vZCceKlB who are looking for what you sell. ___ --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 -- Michael Rice Systems Administrator Office: 210-366-2500 Ext. : 231 Direct: 210-293-6231 McClelland and Hine, Inc. ___ --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] les.net losing DID's
Quoting Mail list [EMAIL PROTECTED]: In general how painful has this sort of thing been to people so far ? I am pretty hesitant to put any sort of number like that on letterhead, website etc., when there might be doubt about having it long term when its provided by a small company. It seems you do voip to save money but to have long term stability you have to get a number from a large company and the savings disappear, or the terms are restrictive to use, or some other negative, and then you end up doing nothing. Yes they are co-operating to port DID to another provider and they have given time till august 23 so DID will continue to work till then but they are not providing any substitute DID though ( i dont expect that ) but atleast they should partially refund amount for remaining days ( i dont expect that either :P ) . On 09/08/2007, Stephen Bosch [EMAIL PROTECTED] wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
Why not use
1-Ruby RAGI
2-http://adhearsion.com/
or similar tools which overcome Asterisk dial plan limitations?
-E
On 8/8/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Wednesday 08 August 2007 1:39:34 pm Mike wrote:
exten = 12345,1,AGI(agi-helloworld.agi)
AGI is an application, and you've called it.
exten = 12345,1,Noop(${AGI(agi-helloworld.agi)})
AGI is not a function. You cannot nest applications like that. The
NoOp
application cannot call another application.
-A.
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Re: [asterisk-users] les.net losing DID's
One of the things that we've done is get a standard PSTN line in place that rings down to the VoIP lines. In smaller shops there's a single copper line, in larger shops they might have a T1/PRI. It's obviously more expensive than pure VoIP lines, but the stability of the number is solid; You know that number isn't going away. If you have to change the number that your inbound rings down to because your VoIP DID just disappeared, then so be it. Pay the fee to your telco and make the change if that happens. But at least you know that one number you have that you've published everywhere isn't going away anytime soon. We originally found our incumbent very resistant to this type of ring strategy (they didn't want to let the call roll over to a number that wasn't theirs), so we moved to using CLECs. Recently we've found that the incumbent has allowed us to do this on certain line types too... So much the better from a stability perspective. AR On 8/8/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Mail list [EMAIL PROTECTED]: In general how painful has this sort of thing been to people so far ? I am pretty hesitant to put any sort of number like that on letterhead, website etc., when there might be doubt about having it long term when its provided by a small company. It seems you do voip to save money but to have long term stability you have to get a number from a large company and the savings disappear, or the terms are restrictive to use, or some other negative, and then you end up doing nothing. Yes they are co-operating to port DID to another provider and they have given time till august 23 so DID will continue to work till then but they are not providing any substitute DID though ( i dont expect that ) but atleast they should partially refund amount for remaining days ( i dont expect that either :P ) . On 09/08/2007, Stephen Bosch [EMAIL PROTECTED] wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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 Robar [EMAIL PROTECTED] ___ --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] les.net losing DID's
Quoting Alex Robar [EMAIL PROTECTED]: surely you wouldn't do this where you are getting voip numbers so you can have local numbers in other areas. Having an analog or other rung through like that would be impossible in most cases and hugely expensive where actually possible. One of the things that we've done is get a standard PSTN line in place that rings down to the VoIP lines. In smaller shops there's a single copper line, in larger shops they might have a T1/PRI. It's obviously more expensive than pure VoIP lines, but the stability of the number is solid; You know that number isn't going away. If you have to change the number that your inbound rings down to because your VoIP DID just disappeared, then so be it. Pay the fee to your telco and make the change if that happens. But at least you know that one number you have that you've published everywhere isn't going away anytime soon. We originally found our incumbent very resistant to this type of ring strategy (they didn't want to let the call roll over to a number that wasn't theirs), so we moved to using CLECs. Recently we've found that the incumbent has allowed us to do this on certain line types too... So much the better from a stability perspective. AR On 8/8/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Mail list [EMAIL PROTECTED]: In general how painful has this sort of thing been to people so far ? I am pretty hesitant to put any sort of number like that on letterhead, website etc., when there might be doubt about having it long term when its provided by a small company. It seems you do voip to save money but to have long term stability you have to get a number from a large company and the savings disappear, or the terms are restrictive to use, or some other negative, and then you end up doing nothing. Yes they are co-operating to port DID to another provider and they have given time till august 23 so DID will continue to work till then but they are not providing any substitute DID though ( i dont expect that ) but atleast they should partially refund amount for remaining days ( i dont expect that either :P ) . On 09/08/2007, Stephen Bosch [EMAIL PROTECTED] wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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 Robar [EMAIL PROTECTED] Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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] VoicePulse Connect
If you cannot afford any dropped calls or poor audio quality, you need a PRI or POTS connection. It doesn't matter how great the carrier is, the Internet is an unreliable medium. 2-3 times VoicePulse has had issues with incomings calls ringing busy. Once incoming calls were all garbled on my end, although the customer could hear me fine. Generally, the outbound service is reliable. However, you should have a backup carrier anyway. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 1:09 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. -John From: Wes Baehr [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Date: Wed, 8 Aug 2007 12:55:29 -0400 John, Voicepulse Connect has been great to me. I've been using it for over a year now, and do not have any major complaints, except that there are no printable receipts for credit card transactions. SIP is also the preferable protocol, as IAX seems to have some issues. Customer service is usually pretty good, and there have been very few (although a couple) problems with service outages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 12:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoicePulse Connect Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --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 _ Find a local pizza place, movie theater, and more.then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theater cp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=95060 7encType=1FORM=MGAC01 ___ --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] les.net losing DID's
Jon, No, not at all - Sorry, that's not what I meant. Indeed, a local extension would be quite prohibitively expensive. What we tend to do with people who require out-of-area calling ability is grab a toll free DID from a bit of a bigger or more stable provider. Here in Ontario, Canada, we've had great success with Unlimitel for providing toll free DIDs. AR On 8/8/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Alex Robar [EMAIL PROTECTED]: surely you wouldn't do this where you are getting voip numbers so you can have local numbers in other areas. Having an analog or other rung through like that would be impossible in most cases and hugely expensive where actually possible. One of the things that we've done is get a standard PSTN line in place that rings down to the VoIP lines. In smaller shops there's a single copper line, in larger shops they might have a T1/PRI. It's obviously more expensive than pure VoIP lines, but the stability of the number is solid; You know that number isn't going away. If you have to change the number that your inbound rings down to because your VoIP DID just disappeared, then so be it. Pay the fee to your telco and make the change if that happens. But at least you know that one number you have that you've published everywhere isn't going away anytime soon. We originally found our incumbent very resistant to this type of ring strategy (they didn't want to let the call roll over to a number that wasn't theirs), so we moved to using CLECs. Recently we've found that the incumbent has allowed us to do this on certain line types too... So much the better from a stability perspective. AR On 8/8/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Mail list [EMAIL PROTECTED]: In general how painful has this sort of thing been to people so far ? I am pretty hesitant to put any sort of number like that on letterhead, website etc., when there might be doubt about having it long term when its provided by a small company. It seems you do voip to save money but to have long term stability you have to get a number from a large company and the savings disappear, or the terms are restrictive to use, or some other negative, and then you end up doing nothing. Yes they are co-operating to port DID to another provider and they have given time till august 23 so DID will continue to work till then but they are not providing any substitute DID though ( i dont expect that ) but atleast they should partially refund amount for remaining days ( i dont expect that either :P ) . On 09/08/2007, Stephen Bosch [EMAIL PROTECTED] wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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 Robar [EMAIL PROTECTED] Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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 Robar [EMAIL
Re: [asterisk-users] les.net losing DID's
Quoting Alex Robar [EMAIL PROTECTED]: Jon, No, not at all - Sorry, that's not what I meant. Indeed, a local extension would be quite prohibitively expensive. What we tend to do with people who require out-of-area calling ability is grab a toll free DID from a bit of a bigger or more stable provider. Here in Ontario, Canada, we've had great success with Unlimitel for providing toll free DIDs. I have run across that name before as well - anyone else have any experience wth them ? (I am in ontario as well) AR On 8/8/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Alex Robar [EMAIL PROTECTED]: surely you wouldn't do this where you are getting voip numbers so you can have local numbers in other areas. Having an analog or other rung through like that would be impossible in most cases and hugely expensive where actually possible. One of the things that we've done is get a standard PSTN line in place that rings down to the VoIP lines. In smaller shops there's a single copper line, in larger shops they might have a T1/PRI. It's obviously more expensive than pure VoIP lines, but the stability of the number is solid; You know that number isn't going away. If you have to change the number that your inbound rings down to because your VoIP DID just disappeared, then so be it. Pay the fee to your telco and make the change if that happens. But at least you know that one number you have that you've published everywhere isn't going away anytime soon. We originally found our incumbent very resistant to this type of ring strategy (they didn't want to let the call roll over to a number that wasn't theirs), so we moved to using CLECs. Recently we've found that the incumbent has allowed us to do this on certain line types too... So much the better from a stability perspective. AR On 8/8/07, Jon Pounder [EMAIL PROTECTED] wrote: Quoting Mail list [EMAIL PROTECTED]: In general how painful has this sort of thing been to people so far ? I am pretty hesitant to put any sort of number like that on letterhead, website etc., when there might be doubt about having it long term when its provided by a small company. It seems you do voip to save money but to have long term stability you have to get a number from a large company and the savings disappear, or the terms are restrictive to use, or some other negative, and then you end up doing nothing. Yes they are co-operating to port DID to another provider and they have given time till august 23 so DID will continue to work till then but they are not providing any substitute DID though ( i dont expect that ) but atleast they should partially refund amount for remaining days ( i dont expect that either :P ) . On 09/08/2007, Stephen Bosch [EMAIL PROTECTED] wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --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 Robar [EMAIL PROTECTED] Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation Provided by
Re: [asterisk-users] RoundRobin Holding Memory?
Matt wrote: I have a queue setup to 'roundrobin' (NOT roundrobin with memory). I have three agents. We'll call them 101, 102, and 103. When a call comes in.. I want it to always try 101 if no answer try 102.. if no answer try 103, etc. However, what it is doing is... it will ring 101... if 101 answers, next time a call comes in it will go to 102. This isn't at all what I want. Any ideas why it might be doing this? [551] wrapuptime=0 timeout=25 strategy=roundrobin retry=1 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting music=default monitor-join=yes monitor-format= member=Local/[EMAIL PROTECTED],0 member=Local/[EMAIL PROTECTED],0 member=Local/[EMAIL PROTECTED],0 maxlen=0 leavewhenempty=no joinempty=Yes context= announce-holdtime=no announce-frequency=60 ___ --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 That is actually the expected behavior, the memory part of rrmemory is it remembers if it was unable to reach a queue member so that it can try them again. To achieve actual circular call distribution you can add a penalty to each number, with the penalty being higher for each later rang agent, so in your queues.conf you would place the agents as: 101,1 102,2 103,3 And so on, this why it will always start at the agent with the lowest penalty. Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to write a function with a return value in Asterisk
Mike wrote:
Thanks for all the replies, after some thinking AGI seems like the way
to go (writing a function in C would certainly work, but I want to
avoid anything that makes upgrading to newer version of Asterisk a
potential pain. Let's say using C is plan B).
So, I wrote (well, plagarized directly from the Web) a simple Perl
program that prints Hello World. I call it using this:
exten = 12345,1,AGI(agi-helloworld.agi)
Seems to work (I'm not expecting anything, really, just no Asterisk
error).
When I try to use it as part Noop like this:
exten = 12345,1,Noop(${AGI(agi-helloworld.agi)})
In the hope of getting to see Noop(Hello World) in my CLI, I get the
following Asterisk error:
Aug 8 13:40:48 ERROR[5771]: pbx.c:1402 ast_func_read: Function AGI
not registered
AGI certainly seems registered as it worked in the first case. Again,
something obvious I missed?
Thank you,
Mike
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
*Anthony Francis
*Sent:* Wednesday, August 08, 2007 12:39
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] How to write a function with a return
value in Asterisk
You are looking for the AGI: http://www.voip-info.org/wiki-Asterisk+AGI
Anthony
James FitzGibbon wrote:
On 8/8/07, *Mike* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
I'd be most thankful for some link to a page that shows how to
write such a
function in Asterisk.
There is a test application in the source tree (not built by default
I believe), but it doesn't look like anyone has made an equivalent
sample function.
However, many of the functions in 1.4 are pretty simple, and would be
a good jumping off point. Take func_sha1.c for example: 83 lines in
the file, 4 functions and one macro. You could copy that and do the
proper renaming to make MY_FUNKY_NEW_FUNC that does exactly what
func_sha1 does (or alternatively, nothing by getting rid of the bulk
of the sha1() function therein.
How big it gets as you add whatever magic that function should
perform is up to you of course.
--
j.
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Yeah to assign a return variable you need to consult the AGI
documentation for the perl variant of setvar. In other words, you call
the AGI the first way, have the agi set the var, then you can ref the
var in later steps.
Anthony
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Re: [asterisk-users] les.net losing DID's
What we tend to do with people who require out-of-area calling ability is grab a toll free DID from a bit of a bigger or more stable provider. Here in Ontario, Canada, we've had great success with Unlimitel for providing toll free DIDs. I have run across that name before as well - anyone else have any experience wth them ? (I am in ontario as well) We use them (Unlimitel) here in Ottawa. They are small, but they are stable and responsive. Outages occur occasionally (every few months), but they are dealt with rapidly and a detailed email usually explains what went wrong. I'm not really aware of anyone else in the area who handles Asterisk well. - 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] FSK callerid
On Thu, 2007-08-09 at 01:55 +0900, Balgansuren Batsukh wrote: Hello, I am from Mongolia and when I use telephone set with callerid it display callerid. Yes, our phone company charge for callerid service. Balgaa Balgaa-- If you have tried all combinations of cidsignalling/cidstart, then it means that you must investigate the details of CID provided by your phone company. Does it come before/after first/second ring? Or a polarity shift? Is it really FSK or is it DTMF, or what? If it is different from the rest of world, perhaps someone, somehow will be able to make mods to the zaptel drivers to make it work. But they never will be able to, if you don't know exactly what the phone company is providing. Is the callerid display you use, an off-the-shelf item for US consumption? murf - Original Message - From: Steve Murphy [EMAIL PROTECTED] To: Balgansuren Batsukh [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 09, 2007 2:40 AM Subject: Re: [asterisk-users] FSK callerid On Wed, 2007-08-08 at 23:55 +0900, Balgansuren Batsukh wrote: Hello, I installed Asterisk on Dell Precision workstation and configured with sample configuration. I have two TDM400 board and one with 4xFXO and second one 4xFXS module installed. I made call to telephone line connected to FXO port and never seen callerid on those lines. I tested cidsignalling and cidstart types and all doesn't work. Balgaa-- What country are you in? The CID conventions vary considerably from country to country! Your tonezone indicate US/France... Here in the US, the defaults in zapata.conf work fine for me: usecallerid=yes ;cidsignalling=bell ;cidstart=ring Are you sure your lines from the CO are providing CID? Out here where I'm at, the phone company charges extra for the privilege of CID. murf /var/log/messages: Aug 3 13:42:51 towerpbx kernel: Module 0: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 1: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 2: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: Module 3: Installed -- AUTO FXO (FCC mode) Aug 3 13:42:51 towerpbx kernel: input: ImPS/2 Logitech Wheel Mouse as /class/input/input2 Aug 3 13:42:51 towerpbx kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Aug 3 13:42:51 towerpbx kernel: ACPI: PCI Interrupt :02:08.0[A] - GSI 17 (level, low) - IRQ 185 Aug 3 13:42:51 towerpbx kernel: Freshmaker version: 71 Aug 3 13:42:51 towerpbx kernel: Freshmaker passed register test Aug 3 13:42:51 towerpbx kernel: Module 0: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 1: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 2: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Module 3: Installed -- AUTO FXS/DPO Aug 3 13:42:51 towerpbx kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Aug 3 13:42:51 towerpbx kernel: ACPI: PCI Interrupt :02:09.0[A] - GSI 18 (level, low) - IRQ 177 Aug 3 13:42:51 towerpbx kernel: FALC version: 0005 Aug 3 13:42:51 towerpbx kernel: TE110P: Setting up global serial parameters for T1 FALC V2.2 Aug 3 13:42:51 towerpbx kernel: TE110P: Successfully initialized serial bus for card Aug 3 13:42:51 towerpbx kernel: Found a Wildcard: Digium Wildcard TE110P T1/E1 Aug 3 13:42:51 towerpbx kernel: Registered tone zone 0 (United States / North America) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 2 (France) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 0 (United States / North America) Aug 3 13:42:51 towerpbx kernel: Registered tone zone 2 (France) zapata.conf usecallerid=yes callerid=asreceived cidsignalling=dtmf cidstart=ring sendcalleridafter=1 Is there any to support FSK callerid? How can I to debug callerid detection process on asterisk? Regards, Balgaa ___ --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 -- Steve Murphy Software Developer Digium 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] VoicePulse Connect
Wes, I'm working through some issues with IAX and Voicepulse right now. It was regarding dropped inbound calls. I was able to put my server into a different location though, and so far the issues have disappeared so hopefully it was a network problem somewhere between us.Just curious what problems you encountered as I would prefer to use IAX if possible. John, I've tried a few services, and Voicepulse was the clear winner for me. I still have two other services in my dialplan for failover, but Voicepulse will remain the primary for now. The voice quality has been very good, and their technical support has been absolutely fantastic for a no-charge service. Wes Baehr wrote: If you cannot afford any dropped calls or poor audio quality, you need a PRI or POTS connection. It doesn't matter how great the carrier is, the Internet is an unreliable medium. 2-3 times VoicePulse has had issues with incomings calls ringing busy. Once incoming calls were all garbled on my end, although the customer could hear me fine. Generally, the outbound service is reliable. However, you should have a backup carrier anyway. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 1:09 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. -John From: Wes Baehr [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Date: Wed, 8 Aug 2007 12:55:29 -0400 John, Voicepulse Connect has been great to me. I've been using it for over a year now, and do not have any major complaints, except that there are no printable receipts for credit card transactions. SIP is also the preferable protocol, as IAX seems to have some issues. Customer service is usually pretty good, and there have been very few (although a couple) problems with service outages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 12:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoicePulse Connect Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --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 _ Find a local pizza place, movie theater, and more.then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theater cp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=95060 7encType=1FORM=MGAC01 ___ --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] MoH mysteriously stopped working
Folks, I have somewhat of a serious issue here. My music on hold mysteriously stopped working. I have made no changes to my Asterisk box in the past month and up until an hour ago, MoH was working fine (as far as I know). CLI: -- Started music on hold, class 'default', on channel 'IAX2/lobby-2' -- Stopped music on hold on IAX2/lobby-2 voip*CLI moh reload voip*CLI 1 class reloaded. == Destroying musiconhold processes == Parsing '/etc/asterisk/musiconhold.conf': Found Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:422 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:504 monmp3thread: Unable to spawn mp3player musiconhold.conf: - [default] mode = quietmp3 directory = /var/lib/asterisk/mohmp3 random = yes I have had .gsm (and only .gsm) files in that directory since day one, and it's always played them just fine. The .gsm files are still in that directory, and transferring them to my computer and playing them works just fine. I have autoload set in modules.conf, and I can't figure out why my music on hold suddenly stopped working. Any thoughts? Thanks in advance, Jay ___ --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] les.net losing DID's
That is why you need to start posting info about the providers at http://www.bochterservices.com/phpbb/ so everyone knows This is a FREE SERVICE provided by Bochter Services and it is not going away any time soon. There will be more added by your request Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Join our forum. This is where you can talk about VOIP You can overview some providers others have used. http://bochterservices.com/phpbb/ --- Stephen Bosch wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --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 Inbound (clean). Database: 000764-2, 08/08/2007 - 8/8/2007 5:31:56 PM ___ --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] MoH mysteriously stopped working
I've had MOH die probably 4-5 times in the last 2+ years and the only way to get it back is to stop * and restart it. Reloading MOH or just doing a regular reload doesn't work. I have to actually do a stop now and then asterisk to get it to work again. * restarts and MOH works fine. No clue why, but I have seen it on multiple versions of *. Jay Moore wrote: Folks, I have somewhat of a serious issue here. My music on hold mysteriously stopped working. I have made no changes to my Asterisk box in the past month and up until an hour ago, MoH was working fine (as far as I know). CLI: -- Started music on hold, class 'default', on channel 'IAX2/lobby-2' -- Stopped music on hold on IAX2/lobby-2 voip*CLI moh reload voip*CLI 1 class reloaded. == Destroying musiconhold processes == Parsing '/etc/asterisk/musiconhold.conf': Found Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:422 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:504 monmp3thread: Unable to spawn mp3player musiconhold.conf: - [default] mode = quietmp3 directory = /var/lib/asterisk/mohmp3 random = yes I have had .gsm (and only .gsm) files in that directory since day one, and it's always played them just fine. The .gsm files are still in that directory, and transferring them to my computer and playing them works just fine. I have autoload set in modules.conf, and I can't figure out why my music on hold suddenly stopped working. Any thoughts? Thanks in advance, Jay ___ --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] Using CURL
Hi,
Here is my first step (call it a proof of concept) in using the hint
priority with dynamic values.
Background - this works
exten = 12345,hint,SIP/12345-1
To make this a little dynamic, I used a web page to return to me the value
of the sip registration. In other words, http://www.somepage.com/test.html
returns the following (without quotes): SIP/12345-1
I should therefore be getting the same result by using the following:
exten = 12345,hint,${CURL(http://www.somepage.com/test.html)}
BUTno. I get the following in the Asterisk CLI when reloading the
config:
Aug 8 18:24:27 NOTICE[26765]: pbx.c:1508
pbx_substitute_variables_helper_full: Error in extension logic (missing '}')
Now, I'm paying close attention to my Asterisk code up there, and I don't
see a missing '}' .
Anybody has an explanation for me? Is there some deeper meaning to this
notice I am getting?
Mike
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Re: [asterisk-users] VoicePulse Connect
I had a lot of problems with garbled IAX calls (even when calling into just the IVR). The problem was compacted when I would bridge an incoming IAX call to an outgoing SIP call, though that may be a fault of Asterisk. Since using SIP everything has been working perfectly. I never had any real problems with dropping calls (that weren't on my end). However, I don't use IAX anymore, so I am not aware of any current issues. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Robinson Sent: Wednesday, August 08, 2007 4:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] VoicePulse Connect Wes, I'm working through some issues with IAX and Voicepulse right now. It was regarding dropped inbound calls. I was able to put my server into a different location though, and so far the issues have disappeared so hopefully it was a network problem somewhere between us.Just curious what problems you encountered as I would prefer to use IAX if possible. John, I've tried a few services, and Voicepulse was the clear winner for me. I still have two other services in my dialplan for failover, but Voicepulse will remain the primary for now. The voice quality has been very good, and their technical support has been absolutely fantastic for a no-charge service. Wes Baehr wrote: If you cannot afford any dropped calls or poor audio quality, you need a PRI or POTS connection. It doesn't matter how great the carrier is, the Internet is an unreliable medium. 2-3 times VoicePulse has had issues with incomings calls ringing busy. Once incoming calls were all garbled on my end, although the customer could hear me fine. Generally, the outbound service is reliable. However, you should have a backup carrier anyway. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 1:09 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. -John From: Wes Baehr mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com asterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' mailto:asterisk-users@lists.digium.com asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Date: Wed, 8 Aug 2007 12:55:29 -0400 John, Voicepulse Connect has been great to me. I've been using it for over a year now, and do not have any major complaints, except that there are no printable receipts for credit card transactions. SIP is also the preferable protocol, as IAX seems to have some issues. Customer service is usually pretty good, and there have been very few (although a couple) problems with service outages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 12:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoicePulse Connect Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --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 _ Find a local pizza place, movie theater, and more.then map the best route! http://maps.live.com/default.aspx?v=2 http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theate r ss=yp.bars~yp.pizza~yp.movie%20theater cp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=95060 7encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Question on the Monitor command on AMI
Hi all, Is there a way to have this command to record a mixed file instead of one for in and one for out? I have set the Mix parameter to 1, but it is still generating two files. I would prefer it to have the in and out files mixed. Thnx. ___ --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] MoH mysteriously stopped working
Peder, Unfortunately, this did not work. Any other thoughts? Jay Peder @ NetworkOblivion wrote: I've had MOH die probably 4-5 times in the last 2+ years and the only way to get it back is to stop * and restart it. Reloading MOH or just doing a regular reload doesn't work. I have to actually do a stop now and then asterisk to get it to work again. * restarts and MOH works fine. No clue why, but I have seen it on multiple versions of *. Jay Moore wrote: Folks, I have somewhat of a serious issue here. My music on hold mysteriously stopped working. I have made no changes to my Asterisk box in the past month and up until an hour ago, MoH was working fine (as far as I know). CLI: -- Started music on hold, class 'default', on channel 'IAX2/lobby-2' -- Stopped music on hold on IAX2/lobby-2 voip*CLI moh reload voip*CLI 1 class reloaded. == Destroying musiconhold processes == Parsing '/etc/asterisk/musiconhold.conf': Found Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:422 spawn_mp3: Found no files in '/var/lib/asterisk/mohmp3' Aug 8 16:27:33 WARNING[7984]: res_musiconhold.c:504 monmp3thread: Unable to spawn mp3player musiconhold.conf: - [default] mode = quietmp3 directory = /var/lib/asterisk/mohmp3 random = yes I have had .gsm (and only .gsm) files in that directory since day one, and it's always played them just fine. The .gsm files are still in that directory, and transferring them to my computer and playing them works just fine. I have autoload set in modules.conf, and I can't figure out why my music on hold suddenly stopped working. Any thoughts? Thanks in advance, Jay ___ --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] VoicePulse Connect
Wes Baehr wrote: I had a lot of problems with garbled IAX calls (even when calling into just the IVR). The problem was compacted when I would bridge an incoming IAX call to an outgoing SIP call, though that may be a fault of Asterisk. Since using SIP everything has been working perfectly. I never had any real problems with dropping calls (that weren’t on my end). However, I don’t use IAX anymore, so I am not aware of any current issues. This is interesting information -- I've had similar problems with IAX trunks on totally different carriers. Example: Callers do not hear the remote ringing, or only some of the rings, or don't hear the beep tone for voice mail. IAX is easier if you're behind a firewall :( -Stephen- ___ --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
