Re: [Asterisk-Users] Re: dial '0' for outside line and get a dialtone...
On Fri, 17 Sep 2004 13:53:11 +0200, Evert Meulie [EMAIL PROTECTED] wrote: Maurizio Marini wrote: Thanks! That works like a charm! The only thing I'd like to do now is NOT having to press 'Dial' on my Cisco 7960 between the '0' and the rest of the number. Any options for that...? Regards, Evert Have you considered using dialplan.xml on the Cisco phones? That's what I do. I programmed the Australian dial plan in (you would setup the phone to suit your country's dial plan). The phones can produce a different dialtone upon dialing '0' by adding a comma, eg: TEMPLATE MATCH=0,02 Timeout=0 User=Phone/ -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call center application
hi... what features are you looking for to be included in such application? SJ Do you have any? Is web based? Please send me the one you have with features, price and demo (if any) on private email Bye John Fach ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] First time asterisk installation problem
Hi all, I am trying to install asterisk on my system, the compiplation and installation process all seem to work fine (make ; make install ; make samples). But astersik fails to start. Is the sample configs not supposed to work out of the box? Even more confusing, it seems to fail at different points every time I start it, but this is probobly because of threads starting differently or something? I can't really figure out exactly what it is that makes it fail, if anyone can give me a clue I would appreciate it. Startup log follows below. /Ola [EMAIL PROTECTED] asterisk-1.0-RC2]# asterisk -c == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.0-RC2, Copyright (C) 1999-2004 Digium. Written by Mark Spencer [EMAIL PROTECTED] = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action MailboxStatus == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [SetVar] == Registered application 'SetVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] = (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so = (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) == Registered channel type 'Modem' (Generic Voice Modem Channel Driver) [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found Sep 18 18:04:13 WARNING[1024]: res_musiconhold.c:543 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [res_adsi.so] = (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Found [res_features.so] = (Call Parking Resource) == Parsing '/etc/asterisk/features.conf': Found -- Registered extension context 'parkedcalls' -- Added extension '700' priority 1 to parkedcalls == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_crypto.so] = (Cryptographic Digital Signatures) Beginning asterisk shutdown Warning, flexible rate not heavily tested! Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Asterisk cleanly ending (2). [EMAIL PROTECTED] asterisk-1.0-RC2]# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Medium volume 100% SIP/IAX PBX.
I know of several that are working fine. We've got our small business on it too, but we're doing BV via a DSL; we have much lower call volume requirements. You say you use BroadVoice? How are you dealing with the voicemail issue? How about multiple simultaneous calls, are you paying for multiple plans or do they allow that on their business plan? We don't have to deal with those issues as our call volumes are so low it's not a problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Canreinvite=???
The KEY thing you are missing is that IAX does NOT use RTP for audio. IAX uses IAX for audio and IAX for signaling. You CANNOT reinvite between a SIP/RTP endpoint and an IAX endpoint. Since SIP, H323, MGCP, and SCCP/Skinny all use RTP for audio, in theory you could do RTP reinvites between these protocols. I have no idea if Asterisk supports this or not. On Sat, 2004-09-18 at 00:52, Carlos Arnt wrote: Looking at this explanation : When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the media streams directly to each other. So if i really understand this using this option i can make the RTP packets flow from one device to another when they connect leaving only the SIP to asterisk . So for example if then I put my Grandstream with a real ip address and use * with a real ip address i can make my calls from nufone flow direct to my grandstream leaving my * bandwidth free . Like this : Grandstream begin call SIP--- Asterisk | | - Nufone. Open RTP Channel Grandstream Real IP -- Nufone IP Right ?? If i'm right , i try this and with tcpdump see the even with everyone using real ip's, the RTP still going over asterisk using my bandwidth . (Note, I force grandstream to use the same codec then Nufone, G729) Can someone give-me some light ?? ;) Can i make this ??? Use asterisk only to begin the call and let the RTP flow over the client and nufone network ?? Thanks alot ! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with DTMF
On Sat, 2004-09-18 at 01:42, Gunnar Andersson wrote: I live in Sweden and can not get CallerID to work on analog incoming lines. I m trying to find out if DTMF style CallerID works on a FXO card (X100). I`v seen one solution with a modem attached in parallel with the X100 just to provide the ID on its serial port. It must be much better if this can be implemented in to the X100 driver. Any info about this would be highly appreciated. http://www.google.com/search?hl=enlr=ie=UTF-8q=site%3Alists.digium.com+Sweden+CallerID+dtmfbtnG=Search -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quintum A800 and asterisk
I just upgrade quintum A800 with new SIP firmware -- Product Name: Tenor Analog A800 Multipath Switch - 8 ports (Rev. B) Gatekeeper Status: Mini GK Calls Allowed: 8 Feature Bit Status: -PS/+RB/-ER Languages allowed: 1 Serial Number: A002-00308F Ethernet Address: 00-30-E1-00-30-8F IP Address: 10.101.0.10 Subnet Mask: 255.255.255.0 Default Gateway: 10.101.0.1 System Software Version: P5-2-1(LEC) (1678285/0xFF74) Boot Software Version: P4-1-3 (180592/0xE814) Database Version: 2.08 09-13-2000 (278376) -- I try to config for register to asteris but don't work. got error : SIPSTK : 19866575:[sess]:bbba04 RegisterSession::ProcessTimerExpiration() called SIPSTK : 19866575:[ua]:Received Sip Iuca Message: 0x3b SIPSTK : 19866575:[sess]:TransactionAbort - in RegSession SIPSTK : 19866576:[ua]:UA: Registration failed - no action taken now.. Can someone help me ? DOme C -- This mail sent through Msarn Mail Solution - Full Spam-Vitus Protect - For more information Please contact [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as an outbound call machine?
Hi All... I have a need to phone a large number of people and collect information from them. I know Asterisk has a nice IVR system, but can it be used to initiate a call to people listed in a database or text file? Don't worry, this is not an annoying marketing thing. Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as an outbound call machine?
Jim,
What you are probably looking for is a superdialer mechanism, as it is
tricky to get Asterisk to do predictive dialing.
A superdialer (if you don't know what it is) is basically a forward call
succession plan. What happens is that you connect to one phone number
after another in successive order, with your agent sitting and listening
on one end.
Accomplishing this in Asterisk is fairly easy, and I approached a
similar problem with an AGI script (written in PHP to interface with our
database).
All you would need to do is have an agent connect to a specific
extension and then launch the AGI script. Here's something off the top
of my head:
extensions.conf
exten = 5000,1,Wait(1)
exten = 5000,2,Answer
exten = 5000,3,AGI(superdialer.php,${INSERT_SOME_PARAM_HERE})
exten = 5000,4,Hangup
superdialer.php (using PHP AGI)
?php
require_once('phpagi/phpagi.php');
require_once('AwesomePhoneNumberSelectCode.inc.php');
$agi = new AGI();
$numbers = AwesomePhoneNumberSelectCode_Execute();
// assume it returns an array of phone numbers
foreach ($numbers as $number) {
$agi-conlog(SuperDialing: $number);
// dial with a 30 second timeout (approx. 5 or 6 rings)
$result = $agi-agi_exec(EXEC Dial
IAX2/[EMAIL PROTECTED]/$number|30);
if ($result['code'] != 200) {
// error here
}
$result = $agi-agi_exec('channel status');
if (!is_array($result) || $result['code'] != 200) {
// asterisk terminated on us, so exit out
break;
}
}
?
Something like the above should allow you to accomplish what you're
looking to accomplish.
-Ken Shaw...
On Sat, 2004-09-18 at 10:56, Jim Archer wrote:
Hi All...
I have a need to phone a large number of people and collect information
from them. I know Asterisk has a nice IVR system, but can it be used to
initiate a call to people listed in a database or text file?
Don't worry, this is not an annoying marketing thing.
Thanks...
___
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[Asterisk-Users] IP Intercom's
Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a loudspeaker ? Thank you, Steve Maroney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Astricon tutorials :: Open for registration again
I am sorry for my ignorance but I don't remember choosing a tutorial/track when I signed up for Astricon. How do I sugn up for a particular tutorial track? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Monday, September 13, 2004 11:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Astricon tutorials :: Open for registration again We're now opening up registrations for the Astricon tutorials again. We've been able to move to new conference rooms within the same hotel. Register on line at http://www.astricon.net We're sorry for the inconvienience our recent closing of the tutorials may have caused you. You are welcome to contact us at [EMAIL PROTECTED] to change your reservation to include the tutorials. There are a lot of things going on now, so we may not be able to answer your phone call. Even though we now have larger rooms, we want to emphasize that the seats in the tutorials are served on a first come basis. If everyone wants to attend the same tutorial, there simply won't be enough seats available. You are free to choose any tutorial, as long as we have seats available. We are also researching the possibility to tape the tutorials to make them available on line or on dvd later on. We will be well over 300 people in Atlanta. See you there! Best regards, /Olle and Steven ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] uk caller id
dear all, i am looking to enable CALLERID on an Asterisk system comprising a X101P FXO interface connecting to BT PSTN in the uk seems this is supported by the interface but there seems to be varying information on how to enable it in zapata.conf 1. usecallerid=uk 2. ukcallerid=yes being two of the configuration statements offered TIA GT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as an outbound call machine?
Hi Ken and thanks! That's great! I had never heard of the superdialer app
for Asterisk. I'm not sure what you mean by predictive dialing. But we
can write php code to go through the database.
I need to get this thing to run unatended. Is it possible for Asterisk to
recognize busy signals and answering machines? I realize answering
machines are tricky, but can it at least detect silence?
Thank you!
Jim
--On Saturday, September 18, 2004 11:25 AM -0700 Kenneth Shaw
[EMAIL PROTECTED] wrote:
Jim,
What you are probably looking for is a superdialer mechanism, as it is
tricky to get Asterisk to do predictive dialing.
A superdialer (if you don't know what it is) is basically a forward call
succession plan. What happens is that you connect to one phone number
after another in successive order, with your agent sitting and listening
on one end.
Accomplishing this in Asterisk is fairly easy, and I approached a
similar problem with an AGI script (written in PHP to interface with our
database).
All you would need to do is have an agent connect to a specific
extension and then launch the AGI script. Here's something off the top
of my head:
extensions.conf
exten = 5000,1,Wait(1)
exten = 5000,2,Answer
exten = 5000,3,AGI(superdialer.php,${INSERT_SOME_PARAM_HERE})
exten = 5000,4,Hangup
superdialer.php (using PHP AGI)
?php
require_once('phpagi/phpagi.php');
require_once('AwesomePhoneNumberSelectCode.inc.php');
$agi = new AGI();
$numbers = AwesomePhoneNumberSelectCode_Execute();
// assume it returns an array of phone numbers
foreach ($numbers as $number) {
$agi-conlog(SuperDialing: $number);
// dial with a 30 second timeout (approx. 5 or 6 rings)
$result = $agi-agi_exec(EXEC Dial
IAX2/[EMAIL PROTECTED]/$number|30);
if ($result['code'] != 200) {
// error here
}
$result = $agi-agi_exec('channel status');
if (!is_array($result) || $result['code'] != 200) {
// asterisk terminated on us, so exit out
break;
}
}
?
Something like the above should allow you to accomplish what you're
looking to accomplish.
-Ken Shaw...
On Sat, 2004-09-18 at 10:56, Jim Archer wrote:
Hi All...
I have a need to phone a large number of people and collect information
from them. I know Asterisk has a nice IVR system, but can it be used to
initiate a call to people listed in a database or text file?
Don't worry, this is not an annoying marketing thing.
Thanks...
___
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To UNSUBSCRIBE or update options visit:
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Re: [Asterisk-Users] Connecting SPA-300 to Asterisk
On Fri, 2004-09-17 at 23:37, Kristian Kielhofner wrote: Sys. Concept Inc. wrote: How to Connect SPA-3000 to Asterisk so * will answer? After setting up Asterisk on Gentoo the extension.conf contains [demo] context; but my asterisk is not answering? In SPA-3000 in PSTN Line - tab under Dial Plan 1 I have: S0:[EMAIL PROTECTED] Default dial plan is set to 1. My box's IP where Asterisk is running has IP: 10.0.0.101 Line 1 - tab has: SIP Settings Port: 5060 Nat is disabled as both Asterisk and SPA-3000 are behind firewall. What am I missing? -- #Joseph Check this out from Voxilla: http://voxilla.com/forum-viewtopic-t-557.html I checked their forum, actually Voxilla is the place I bought the SPA-3000 unit from. But the configuration instructions are not clear to me, there is a lot of information but none is complete. The SPA-3000 unit might be good but the configuration is nightmare and Sipura manual is only good for reference nothing else in addition they do not offer any support. If I was to recommend any unit, go with Digium cards and stay away from Siupra -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting SPA-300 to Asterisk
What I did was create an extension that goes to my mainmenu, ie: exten = 7000,1,Goto(mainmenu,s,2) Then I setup the SPA-3000 to dial that extension when a call comes in according to the FAQ entry on Sipura's website: http://www.sipura.com/support/spa3000faq/Section_3.html#4 4: How can I forward all PSTN callers to a VoIP number? A: You can use specify a dial plan to be used by the default PSTN caller with a hot line syntax: (S0:voip_number) where voip_number is replaced with the actual phone number (or sip url) of the VoIP destination. So I used: (S0:7000) Works like a charm. In my experience Sipura has an excellent product, I use both the SPA- 2000 and SPA-3000, and they are both great. I've had _far_ less problems with Sipura products then I have had with Digium X100P cards. Especially when it comes to echo. Sipura's email support is also exceptional, I often get replies to emails within minutes, and they have even implemented feature requests and sent me a beta firmware within days. On Sat, 2004-09-18 at 13:31 -0600, Joseph wrote: On Fri, 2004-09-17 at 23:37, Kristian Kielhofner wrote: Sys. Concept Inc. wrote: How to Connect SPA-3000 to Asterisk so * will answer? After setting up Asterisk on Gentoo the extension.conf contains [demo] context; but my asterisk is not answering? In SPA-3000 in PSTN Line - tab under Dial Plan 1 I have: S0:[EMAIL PROTECTED] Default dial plan is set to 1. My box's IP where Asterisk is running has IP: 10.0.0.101 Line 1 - tab has: SIP Settings Port: 5060 Nat is disabled as both Asterisk and SPA-3000 are behind firewall. What am I missing? -- #Joseph Check this out from Voxilla: http://voxilla.com/forum-viewtopic-t-557.html I checked their forum, actually Voxilla is the place I bought the SPA-3000 unit from. But the configuration instructions are not clear to me, there is a lot of information but none is complete. The SPA-3000 unit might be good but the configuration is nightmare and Sipura manual is only good for reference nothing else in addition they do not offer any support. If I was to recommend any unit, go with Digium cards and stay away from Siupra -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpanDSP/RxFax anomalies...
Hi Rob, I used RxFax some time ago with success but the last CVS of asterisk crashes when asterisk is going to start receiving data. I'm trying with tiff-3.5.7 spandsp-0.0.1k and last asterisk CVS... What versions do you use? On Fri, 2004-09-10 at 14:09 -0500, Rob Fugina wrote: I've recently started playing with the RxFax application on my Asterisk box. I've had success, mostly, but I've had some failures, too... The most recent failure is specific to receiving from a particular fax machine -- a Canon Laser Class 9000S. The TIF images received are readable, but the aspect ratio is stretched horizonatlly (or squished vertically). Is this a problem anyone else has seen before? Is there a workaround? Thanks, Rob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you in Advance, Alexander ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NEWBIE - No Audio on ISDN BRI (Teles PCI)
Hello, folks! This is my first post here. I installed Asterisk from scratch and after reading a lot of information on voip-info and this mailing list I was able to get started. I can make sip-to-sip calls (just on a basic extentions structure, let's say for beginners) but now I'm trying to make this system works with my Teles ISDN BRI PCI card. I can make and receive calls through X-Ten Pro software. Calls are routed through Asterisk to/from X-Ten client but here's my real problem: NO AUDIO AT ALL! I can't hear anything, nor can the other party. Does anyone here now anything about it? Am I doing anything wrong? I'm new to ISDN under Linux, but I think it's ok since the calls are corectly routed and I can see info on isdnlog. I also tested my card on minicom, as suggested on voip-info. One more question: can I route SIP calls to a ISDN (digital) phone connected on the same bus? How do I set the TEI on which the call should be answered, in case it's possible? By the way, I never used this ISDN card to place/receive calls or connect to an ISP on Linux, but I already received/transmited faxes on Windows 2000 SP4 before installing Asterisk. Thank you so much in advance and I appologize if these questions have already been answered somewhere. I just need some guidance. Take care, [ Dhennys Pestana ] [ +55(21)9339-3737 ] Linux: SuSE 8.2 2.4.20-4GB Asterisk: 1.0 RC1 (from tarball) ISDN BRI: lspci shows Tiger Jet Network Inc. Intel 537, but I know for sure it's actually a Teles PCI cause I bought it on a sealed box ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting SPA-300 to Asterisk
Thanks Mike, it did work after so much playing around. I guess none of those devices are coming with perfect installation and/or configuration instructions. #Joseph On Sat, 2004-09-18 at 13:45, Mike Benoit wrote: What I did was create an extension that goes to my mainmenu, ie: exten = 7000,1,Goto(mainmenu,s,2) Then I setup the SPA-3000 to dial that extension when a call comes in according to the FAQ entry on Sipura's website: http://www.sipura.com/support/spa3000faq/Section_3.html#4 4: How can I forward all PSTN callers to a VoIP number? A: You can use specify a dial plan to be used by the default PSTN caller with a hot line syntax: (S0:voip_number) where voip_number is replaced with the actual phone number (or sip url) of the VoIP destination. So I used: (S0:7000) Works like a charm. In my experience Sipura has an excellent product, I use both the SPA- 2000 and SPA-3000, and they are both great. I've had _far_ less problems with Sipura products then I have had with Digium X100P cards. Especially when it comes to echo. Sipura's email support is also exceptional, I often get replies to emails within minutes, and they have even implemented feature requests and sent me a beta firmware within days. On Sat, 2004-09-18 at 13:31 -0600, Joseph wrote: On Fri, 2004-09-17 at 23:37, Kristian Kielhofner wrote: Sys. Concept Inc. wrote: How to Connect SPA-3000 to Asterisk so * will answer? After setting up Asterisk on Gentoo the extension.conf contains [demo] context; but my asterisk is not answering? In SPA-3000 in PSTN Line - tab under Dial Plan 1 I have: S0:[EMAIL PROTECTED] Default dial plan is set to 1. My box's IP where Asterisk is running has IP: 10.0.0.101 Line 1 - tab has: SIP Settings Port: 5060 Nat is disabled as both Asterisk and SPA-3000 are behind firewall. What am I missing? -- #Joseph Check this out from Voxilla: http://voxilla.com/forum-viewtopic-t-557.html I checked their forum, actually Voxilla is the place I bought the SPA-3000 unit from. But the configuration instructions are not clear to me, there is a lot of information but none is complete. The SPA-3000 unit might be good but the configuration is nightmare and Sipura manual is only good for reference nothing else in addition they do not offer any support. If I was to recommend any unit, go with Digium cards and stay away from Siupra -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 13 sec. delay what is causing it?
I've setup SPA-3000 and when the calls come through my phone is rining almost instantly but the [demo] doesn't answer till after about 13 seconds. So I have about 13 seconds delay and I don't know what setting is causing it; here is a part of my settings from extension.conf. [from_pstn] exten = 1000,1,Goto(demo,s,1) [demo] exten = s,1,Answer ; Answer the line exten = s,2,BackGround(demo-congrats) ; Play a congratulatory message exten = s,3,BackGround(demo-instruct) ; Play some instructions What setting is causing the 13-15sec. delay? #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Intercom's
There are several ways to approach this: * modify an existing SIP phone with Auto-answer (Grandstream for example) to interface with a loud speaker * use a SIP client (Asterisk for example) on a small PC and interface the sound card to a loudspeaker * use a traditional overhead paging/intercom hardware and interface to it via the sound card or via an FXS port. * use an analog auto answer door phone with an FXS interface Check these wiki pages for starting points: http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom http://www.voip-info.org/wiki-Asterisk+phone+door Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Steve Maroney [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 18, 2004 8:41 AM Subject: [Asterisk-Users] IP Intercom's Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a loudspeaker ? Thank you, Steve Maroney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Intercom's
On Sat, 18 Sep 2004 13:41:54 -0500 (CDT), Steve Maroney [EMAIL PROTECTED] wrote: Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a loudspeaker ? Thank you, Steve Maroney The easiest way is to hook up loud speakers to the sound card of the Asterisk box. I use this and it works pretty well. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 13 sec. delay what is causing it?
Perfectly normal. On analog lines, the caller id is set between the 1st and 2nd rings. So Asterisk has to wait for the caller id and depending on the speed of the computer that hosts Asterisk, 13 seconds is exactly right. A normal ring cycle is 2 secs ring on 4 seconds of silence, so the 2nd ring is 12 seconds into the call. I just put in a nice Asus motherboard with a 500 mhz front side bus, 2.4 gig AMD processor 512 meg ram for the pbx here and I get the first ring on the extensions at the same time as the second ring on the incoming ring. I was testing and trialing on a celeron 1.4ghz machine with 256 meg ram and the video borrowed some of the system ram. The analog extensions were not ringing until the third incoming ring on that slow machine. Lyle - Original Message - From: Joseph [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 18, 2004 3:54 PM Subject: [Asterisk-Users] 13 sec. delay what is causing it? I've setup SPA-3000 and when the calls come through my phone is rining almost instantly but the [demo] doesn't answer till after about 13 seconds. So I have about 13 seconds delay and I don't know what setting is causing it; here is a part of my settings from extension.conf. [from_pstn] exten = 1000,1,Goto(demo,s,1) [demo] exten = s,1,Answer ; Answer the line exten = s,2,BackGround(demo-congrats) ; Play a congratulatory message exten = s,3,BackGround(demo-instruct) ; Play some instructions What setting is causing the 13-15sec. delay? #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Intercom's
On Sat, 2004-09-18 at 16:32, James H. Thompson wrote: There are several ways to approach this: * modify an existing SIP phone with Auto-answer (Grandstream for example) to interface with a loud speaker * use a SIP client (Asterisk for example) on a small PC and interface the sound card to a loudspeaker * use a traditional overhead paging/intercom hardware and interface to it via the sound card or via an FXS port. * use an analog auto answer door phone with an FXS interface Check these wiki pages for starting points: http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom http://www.voip-info.org/wiki-Asterisk+phone+door Or you could plug an amp and some overhead speakers into the sound card on the box running Asterisk and use chan_oss or chan_alsa with auto-answer enabled. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 13 sec. delay what is causing it?
On Sat, 2004-09-18 at 17:21, Lyle Giese wrote: Perfectly normal. On analog lines, the caller id is set between the 1st and 2nd rings. So Asterisk has to wait for the caller id and depending on the speed of the computer that hosts Asterisk, 13 seconds is exactly right. A normal ring cycle is 2 secs ring on 4 seconds of silence, so the 2nd ring is 12 seconds into the call. I just put in a nice Asus motherboard with a 500 mhz front side bus, 2.4 gig AMD processor 512 meg ram for the pbx here and I get the first ring on the extensions at the same time as the second ring on the incoming ring. I was testing and trialing on a celeron 1.4ghz machine with 256 meg ram and the video borrowed some of the system ram. The analog extensions were not ringing until the third incoming ring on that slow machine. System speed has VERY little to do with this. If Asterisk expects to get Caller*ID and the PSTN line does not have Caller*ID service on the line. Asterisk has to wait until the beginning of the second ring before giving up on getting any Caller*ID. If your PSTN line doesn't have Caller*ID service then tell Asterisk not to expect Caller*ID then the delay will be MUCH less. This is covered over and over and over again in the mailing list archives. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp
I think the port.h in this distribution may have been created from tiffv3.5.7 while you have tiffv3.6.0 - (or maybe something else), anyway I had this problem, and installing tiffv3.5.7 and copying the port.h from that distribution to /usr/local/include fixed it HTH Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maurizio Marini Sent: 17 September 2004 11:31 To: [EMAIL PROTECTED] Cc: administrator tootai Subject: Re: [Asterisk-Users] spandsp On Thursday 19 August 2004 23:29, administrator tootai wrote: I made one. Can be found at http://ftp2.tootai.net/spandsp-0.0.1k-whole.tar.gz The 3 headers files are included, made a short readme file for installation and modify the Makefile.patch (remove the dtmftotext). Comments welcome. debian sid with littiff3-dev libtiff4-dev installed; compiling spandsp i get this error: gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c t4.c -MT t4.lo -MD -MP -MF .deps/t4.TPlo -fPIC -DPIC -o .libs/t4.lo In file included from /usr/include/tiffiop.h:45, from t4.c:38: /usr/include/tif_dir.h:240: error: conflicting types for `TIFFFieldInfo' /usr/include/tiffio.h:448: error: previous declaration of `TIFFFieldInfo' make[2]: *** [t4.lo] Error 1 `TIFFFieldInfo' is defined in tif_dir.h and in my tiffio.h: /usr/include# grep TIFFFieldInfo * tif_dir.h:} TIFFFieldInfo; tif_dir.h:externvoid _TIFFMergeFieldInfo(TIFF*, const TIFFFieldInfo[], int); tif_dir.h:externconst TIFFFieldInfo* _TIFFFindFieldInfo(TIFF*, ttag_t, TIFFDataType); tif_dir.h:externconst TIFFFieldInfo* _TIFFFieldWithTag(TIFF*, ttag_t); tiffio.h:} TIFFFieldInfo; tiffio.h:const TIFFFieldInfo *info; tiffio.h:extern void TIFFMergeFieldInfo(TIFF*, const TIFFFieldInfo[], int); tiffio.h:extern const TIFFFieldInfo* TIFFFindFieldInfo(TIFF*, ttag_t, TIFFDataType); tiffio.h:extern const TIFFFieldInfo* TIFFFieldWithTag(TIFF*, ttag_t); tiffiop.h: TIFFFieldInfo** tif_fieldinfo; /* sorted table of registered tags */ what do u suggest me? -- Maurizio Marini ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Intercom's
As far as I understand it is possible to implement an Intercom with Aastras 390 / 480 ADSI compatible phones. One of the ADSI commands sent to the phone like caller-Id, takes the 390/480 off hook. This way it would be possible to implement a * server controlled auto answer function. I do not have any documentation about the exact command sequence though. Alfred R. Nurnberger Eric Wieling wrote: On Sat, 2004-09-18 at 16:32, James H. Thompson wrote: There are several ways to approach this: * modify an existing SIP phone with Auto-answer (Grandstream for example) to interface with a loud speaker * use a SIP client (Asterisk for example) on a small PC and interface the sound card to a loudspeaker * use a traditional overhead paging/intercom hardware and interface to it via the sound card or via an FXS port. * use an analog auto answer door phone with an FXS interface Check these wiki pages for starting points: http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom http://www.voip-info.org/wiki-Asterisk+phone+door Or you could plug an amp and some overhead speakers into the sound card on the box running Asterisk and use chan_oss or chan_alsa with auto-answer enabled. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 13 sec. delay what is causing it?
Eric, I am very aware of what you are talking about. However I am comparing two different computers connected to the exact same PSTN line in the US, using the exact same Digium TDM card. Huge difference in cpu's and bus speeds. * still needs to get the data from the TDM or X100P card and parse it internally before ringing the pbx extensions. Lyle - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, September 18, 2004 5:44 PM Subject: Re: [Asterisk-Users] 13 sec. delay what is causing it? On Sat, 2004-09-18 at 17:21, Lyle Giese wrote: Perfectly normal. On analog lines, the caller id is set between the 1st and 2nd rings. So Asterisk has to wait for the caller id and depending on the speed of the computer that hosts Asterisk, 13 seconds is exactly right. A normal ring cycle is 2 secs ring on 4 seconds of silence, so the 2nd ring is 12 seconds into the call. I just put in a nice Asus motherboard with a 500 mhz front side bus, 2.4 gig AMD processor 512 meg ram for the pbx here and I get the first ring on the extensions at the same time as the second ring on the incoming ring. I was testing and trialing on a celeron 1.4ghz machine with 256 meg ram and the video borrowed some of the system ram. The analog extensions were not ringing until the third incoming ring on that slow machine. System speed has VERY little to do with this. If Asterisk expects to get Caller*ID and the PSTN line does not have Caller*ID service on the line. Asterisk has to wait until the beginning of the second ring before giving up on getting any Caller*ID. If your PSTN line doesn't have Caller*ID service then tell Asterisk not to expect Caller*ID then the delay will be MUCH less. This is covered over and over and over again in the mailing list archives. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEWBIE - No Audio on ISDN BRI (Teles PCI)
use chan_capi from www.junghanns.net for the fritzcard. Or even better, get rid of the fritz card and get a hfc-pci card and use it with bri-stuff from the same address On Sat, 18 Sep 2004 17:23:30 -0300, Pestana Networks, Inc. [EMAIL PROTECTED] wrote: Hello, folks! This is my first post here. I installed Asterisk from scratch and after reading a lot of information on voip-info and this mailing list I was able to get started. I can make sip-to-sip calls (just on a basic extentions structure, let's say for beginners) but now I'm trying to make this system works with my Teles ISDN BRI PCI card. I can make and receive calls through X-Ten Pro software. Calls are routed through Asterisk to/from X-Ten client but here's my real problem: NO AUDIO AT ALL! I can't hear anything, nor can the other party. Does anyone here now anything about it? Am I doing anything wrong? I'm new to ISDN under Linux, but I think it's ok since the calls are corectly routed and I can see info on isdnlog. I also tested my card on minicom, as suggested on voip-info. One more question: can I route SIP calls to a ISDN (digital) phone connected on the same bus? How do I set the TEI on which the call should be answered, in case it's possible? By the way, I never used this ISDN card to place/receive calls or connect to an ISP on Linux, but I already received/transmited faxes on Windows 2000 SP4 before installing Asterisk. Thank you so much in advance and I appologize if these questions have already been answered somewhere. I just need some guidance. Take care, [ Dhennys Pestana ] [ +55(21)9339-3737 ] Linux: SuSE 8.2 2.4.20-4GB Asterisk: 1.0 RC1 (from tarball) ISDN BRI: lspci shows Tiger Jet Network Inc. Intel 537, but I know for sure it's actually a Teles PCI cause I bought it on a sealed box ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk presence utility
Hello, On Sat, 18 Sep 2004 07:54:15 +0100, Bill Seddon [EMAIL PROTECTED] wrote: I've spent a couple of evenings writing a presence utility in C# so that a window, listing the currently registered SIP phones, can be displayed and a user can see who is on the phone and who is not. It uses the Manager API and works well, updating the display as API events are received. But what I want to be able to do is add to the list of displayed users when a phone registers and remove them when a phone unregisters. However the Manager API does not seem to generate event messages for these events. Is this correct or have a I missed an option somewhere? Certainly the register and unregister event is displayed on the Asterisk command line. It is possible to have the utility run the sip show peers command periodically and update its list based on the results of the command. However that means polling the server periodically, comsuming resources, instead of being notified just once. Latest CVS does include manager events for register / unregister / reachable / unreachable / lagged events for sip and iax peers. You will still need to poll for initial status. -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] English vs American voice files
OK Folks, I've spent the afternoon recording all the files for the English speaking VM etc. I've parked the file here http://www.g7ltt.com/VoIP/vmukmale.tgz I did it with Audacity at 44.1KHz x 16bit and thenused sox to raise the levels to -3db and then again to down sample them into 8KHz GSM files. The few that I've listened to sound fine. How are you getting on with yours Bill? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No sound from IVR scripts, yet calls placed without any problem.
On 17 Sep 2004 at 19:35, Nick Barnes wrote: On Behalf Of R Wong: how about the sound file location? i've got an experience IF you are not using the english, you need to let asterisk know the file location. I should, perhaps, have been more clear. I Ghosted my Asterisk box which is fully working. There were no changes to anything other than the extra HFC card (I think this is a bit of a red herring, but I mention it anyway). This box was also fully working in my office before I took it to his. Nick Barnes This is really unlikely but is it possible he has an internal firewall or something and the Asterisk box is in the DMZ? Maybe a NAT problem? I.E. rtp being blocked? What protocol are the calls? I would say the problem could be: 1) Power is different (Extremely unlikely and PC wouldn't work) 2) LAN is different (maybe left over setting from ghost i.e bindaddr etc) 3) VOIP accounts are different Cheers, Matt Riddell http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk stopped answering the calls
Asterisk stopped answering the calls. I'm just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0:[EMAIL PROTECTED] my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid=PSTN GW 3000 deny=0.0.0.0 permit=10.0.0.154 ;SPA-3000 IP address dtmfmode=rfc2833 canreinvite=no cantransfer=yes My extension.conf [globals] PSTN_GW=10.0.0.154:5062 [from_pstn] exten = 1000,1,Goto(demo,s,1) [demo] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,BackGround(demo-congrats When I type show channels I get 0 active channel(s) Why isn't asterisk answering the call? -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEWBIE - No Audio on ISDN BRI (Teles PCI)
I'd love to do that, but there's no such card for sale in Brazil. I'd pay expensive taxes (+60%) to import one, although I'd consider it if I had any idea how much would it cost me. Can you tell me? By the way, does chan_capi from different cards work between each other? How am I supposed to get a capi driver for a Teles PCI? Thanx, -Dhennys - Original Message - From: Michael Bielicki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, September 18, 2004 20:46 Subject: Re: [Asterisk-Users] NEWBIE - No Audio on ISDN BRI (Teles PCI) use chan_capi from www.junghanns.net for the fritzcard. Or even better, get rid of the fritz card and get a hfc-pci card and use it with bri-stuff from the same address On Sat, 18 Sep 2004 17:23:30 -0300, Pestana Networks, Inc. [EMAIL PROTECTED] wrote: (...) I can make and receive calls through X-Ten Pro software. Calls are routed through Asterisk to/from X-Ten client but here's my real problem: NO AUDIO AT ALL! I can't hear anything, nor can the other party. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New MOH stream for each queue member?
Hi there, Is it possible to start a new MOH stream for each queue member? My queue MOH is a message rather than music and I want customers to enter the queue and hear it from the start rather than from some random point. The first person entering the queue always seems to hear it from the start. I know could (ab)use the position announcement settings (which interrupts the MOH), changing the message and disabling the actual position announcement, but my understanding of how this works is that it would block queue members from exiting the queue whilst the announcement was playing (which I don't want). Any tips on getting this working - would it require changes to the code? I'm not a programmer. Thanks in advance for any help. regards, Julien ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Timing source on SMP system
I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. Im sure the DL360 G2 has a OHBI rather than UHBI controller. That said, I got this message during modprobe ztdummy: [EMAIL PROTECTED] zaptel]# modprobe ztdummy /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed I then moved on to zaprtc. However, I was told that this solution will not work with SMP systems. My DL360G2 is a dual proc machine. Im running out of options hereplease advise. Thanks, Chad M. Brown Infrastructure Architect identity mine, inc. - http://www.identitymine.com [EMAIL PROTECTED] 253.927.7737- Office 866.4ID.MINE (866.443.6463)- Toll free 253.405.6726 - Cellular 253.444.5170 - Fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk stopped answering the calls
Just an FYI localmask is deprecated... As of latest CVS: localnet=ip/mask bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joseph Sent: Saturday, September 18, 2004 9:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk stopped answering the calls Asterisk stopped answering the calls. I'm just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0:[EMAIL PROTECTED] my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid=PSTN GW 3000 deny=0.0.0.0 permit=10.0.0.154 ;SPA-3000 IP address dtmfmode=rfc2833 canreinvite=no cantransfer=yes My extension.conf [globals] PSTN_GW=10.0.0.154:5062 [from_pstn] exten = 1000,1,Goto(demo,s,1) [demo] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,BackGround(demo-congrats When I type show channels I get 0 active channel(s) Why isn't asterisk answering the call? -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk stopped answering the calls
Is it possible to put command line in debug mode, for example display line by line the command that is being executed to see where I'm getting stuck. My phone rings but when I answer it there silence, the voice is not going through. It rings few times, stops and rings again #Joseph On Sat, 2004-09-18 at 21:22, Brian West wrote: Just an FYI localmask is deprecated... As of latest CVS: localnet=ip/mask bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joseph Sent: Saturday, September 18, 2004 9:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk stopped answering the calls Asterisk stopped answering the calls. I'm just experimenting with asterisk, upon setup there is a [demo] context. I have SPA-3000 with PSTN line: Dial plan 2: S0:[EMAIL PROTECTED] my sip.conf localnet = 10.0.0.101 localmask = 255.255.255.0 [3000] type=friend host=dynamic username=3000 secret=my_secret mailbox=3000 context=from_pstn callerid=PSTN GW 3000 deny=0.0.0.0 permit=10.0.0.154 ;SPA-3000 IP address dtmfmode=rfc2833 canreinvite=no cantransfer=yes My extension.conf [globals] PSTN_GW=10.0.0.154:5062 [from_pstn] exten = 1000,1,Goto(demo,s,1) [demo] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,BackGround(demo-congrats When I type show channels I get 0 active channel(s) Why isn't asterisk answering the call? -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Timing source on SMP system
try zaprtc from www.junghanns.net. Works fine in my SMP systems - Original Message - From: Chad Brown [EMAIL PROTECTED] Date: Sat, 18 Sep 2004 20:23:54 -0700 Subject: [Asterisk-Users] Timing source on SMP system To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. I'm sure the DL360 G2 has a OHBI rather than UHBI controller. That said, I got this message during modprobe ztdummy: [EMAIL PROTECTED] zaptel]# modprobe ztdummy /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed I then moved on to zaprtc. However, I was told that this solution will not work with SMP systems. My DL360G2 is a dual proc machine. I'm running out of options hereplease advise. Thanks, Chad M. Brown Infrastructure Architect identity mine, inc. - http://www.identitymine.com [EMAIL PROTECTED] 253.927.7737 - Office 866.4ID.MINE (866.443.6463) - Toll free 253.405.6726 - Cellular 253.444.5170 - Fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial 0 to outbound
Hi Folks. I see that can put 0 to call out using a x101p (zaptel) or even a pstn service. Thats great, but when press the 0 i just dial then the numbers to call out. There is any way after hit 0 (ear) the line sound ?? I know it's just a style way put some users, really like it !! So after hit 0 to call for example a pstn the user will ear the line sound to dial out. I read lot's of doc's but can't find nothing explaining this method. Thanks alot ! Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Timing source on SMP system
Michael, What is the trick to getting this solution installed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Bielicki Sent: Saturday, September 18, 2004 9:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Timing source on SMP system try zaprtc from www.junghanns.net. Works fine in my SMP systems - Original Message - From: Chad Brown [EMAIL PROTECTED] Date: Sat, 18 Sep 2004 20:23:54 -0700 Subject: [Asterisk-Users] Timing source on SMP system To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. I'm sure the DL360 G2 has a OHBI rather than UHBI controller. That said, I got this message during modprobe ztdummy: [EMAIL PROTECTED] zaptel]# modprobe ztdummy /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o failed /lib/modules/2.4.20-6smp/kernel/drivers/usb/usb-uhci.o: insmod ztdummy failed I then moved on to zaprtc. However, I was told that this solution will not work with SMP systems. My DL360G2 is a dual proc machine. I'm running out of options here...please advise. Thanks, Chad M. Brown Infrastructure Architect identity mine, inc. - http://www.identitymine.com [EMAIL PROTECTED] 253.927.7737 - Office 866.4ID.MINE (866.443.6463) - Toll free 253.405.6726 - Cellular 253.444.5170 - Fax ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Background() command
On Fri, 17 Sep 2004, Paul Penrod wrote: [message] exten = s,1,Answer() exten = s,2,Wait(2) ;Pause to let the user end catch up with the connection exten = s,3,Background(demo-congrats) exten = s,4,Goto(3) Remove the goto. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID with DTMF
Hi Everyone! I live in Sweden and can not get CallerID to work on analog incoming lines. I m trying to find out if DTMF style CallerID works on a FXO card (X100). I`v seen one solution with a modem attached in parallel with the X100 just to provide the ID on its serial port. It must be much better if this can be implemented in to the X100 driver. Any info about this would be highly appreciated. rgds Gunnar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk presence utility
I've spent a couple of evenings writing a presence utility in C# so that a window, listing the currently registered SIP phones, can be displayed and a user can see who is on the phone and who is not. It uses the Manager API and works well, updating the display as API events are received. But what I want to be able to do is add to the list of displayed users when a phone registers and remove them when a phone unregisters. However the Manager API does not seem to generate event messages for these events. Is this correct or have a I missed an option somewhere? Certainly the register and unregister event is displayed on the Asterisk command line. It is possible to have the utility run the sip show peers command periodically and update its list based on the results of the command. However that means polling the server periodically, comsuming resources, instead of being notified just once. Any insight gratefully received. Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Martin Croome
Does anyone know how to get in touch with Martin Croome, author of a C# library for the Asterisk Manager API? Bill Seddon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with DTMF
On Sat, 18 Sep 2004, Gunnar Andersson wrote: I live in Sweden and can not get CallerID to work on analog incoming lines. I m trying to find out if DTMF style CallerID works on a FXO card (X100). I`v seen one solution with a modem attached in parallel with the X100 just to provide the ID on its serial port. It must be much better if this can be implemented in to the X100 driver. Any info about this would be highly appreciated. Have a look at http://bugs.digium.com/bug_view_page.php?bug_id=009. I'm not sure if Swedish style callerid works with X100 or not, but it does work with the tdm400 fxo card. In any case you need the patches from that bug. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] English vs American voice files
My wife has been recording the text published on the wiki. A couple of questions for you: 1) One of the recordings says please enter the full 10 digit number starting with the area code. Any opinions on whether this should be changed for the UK and, if so, to what? 2) The recordings seem dull on playback even though we are recording using a good quality microphone with matching impedance. Initially we recorded using 16 bit/8K sampling on the basis that this is what is required by Asterisk but that was really terrible. So we're sampling at higher rates on the basis that we can use sox to change it as necessary. Any thoughts on what we can do to make the recordings sound sharper? Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Linus Surguy Sent: September 18, 2004 8:28 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] English vs American voice files Ah, this brings up an interesting point. I've noted that BT are calling # square rather than hash. What do the other providers call it back in Blighty? 'Hash' is by far the most common used. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call center application
Hi Is there a application to manage a call center with asterisk ? Thanks Jerome Vous manquez despace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail sur http://fr.benefits.yahoo.com/ Le nouveau Yahoo! Messenger est arrivé ! Découvrez toutes les nouveautés pour dialoguer instantanément avec vos amis. A télécharger gratuitement sur http://fr.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call center application
[EMAIL PROTECTED] wrote: Hi Is there a application to manage a call center with asterisk ? hi... what features are you looking for to be included in such application? SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: cisco 7960 CTLSEP
[EMAIL PROTECTED] wrote: How to load the SIP image on these phones? create emtpy required files... that will trick it.. :) ta SJ When I create an empty CTLSEPmac.tlv files I start looking for SEPmac.xml.cnf files. When I create this file also (empty) I will ask for the CTLSEP again :( No requests for SIP.. hmm... odd we had one unit completely left lifeless because it did not load the firmware correctly but as far I can remember the empty files did the trick. what firmware version do you have loaded at the moment? SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] English vs American voice files
Hi, Bill asked: 1) One of the recordings says please enter the full 10 digit number starting with the area code. Any opinions on whether this should be changed for the UK and, if so, to what? Yes, definitely. How's about Please enter the full telephone number including the STD code. Any thoughts on what we can do to make the recordings sound sharper? I did something similar to this a while back and had all sorts of problems. I found: a - Sampling at the highest rate available let me do some post processing on the file before I resampled it down for the final application. b - The environment the sound was recorded in had a much greater effect on the sound quality than I would ever have thought. Try adding, removing or repositioning reflective and non-reflective surfaces around the person and the microphone. c - If you are using a spit/pop guard, make sure it's positioned correctly. That's my 2p anyway. Nick Barnes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] English vs American voice files
Bloody hell Bill, don't hang about will ya! I agree that the suggested sample format is pretty bad. I'd do it at 44x16. As for sounding dull. Dunno what you mean here but it sounds like either your mic doesn't have the right amount of range or that you need to add som EQ to your mic line to emphasise it. I find that I need to boost my mid range and so I twiddle the 2.5k pot on my mixer. If this isn;t avasilable to you, ship me some audio and I'll take a crack at it and see what I can do. Also, what environment are you recording it in? I built a small cubby hole out of plasterboard and 2x4's which I then hung carpet onto to reduce the hardness of the walls. The carpet absorbes andy sound that might otherwise get bounced back and picked up. Works quite well. Mark Bill Seddon said: My wife has been recording the text published on the wiki. A couple of questions for you: 1) One of the recordings says please enter the full 10 digit number starting with the area code. Any opinions on whether this should be changed for the UK and, if so, to what? 2) The recordings seem dull on playback even though we are recording using a good quality microphone with matching impedance. Initially we recorded using 16 bit/8K sampling on the basis that this is what is required by Asterisk but that was really terrible. So we're sampling at higher rates on the basis that we can use sox to change it as necessary. Any thoughts on what we can do to make the recordings sound sharper? Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Linus Surguy Sent: September 18, 2004 8:28 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] English vs American voice files Ah, this brings up an interesting point. I've noted that BT are calling # square rather than hash. What do the other providers call it back in Blighty? 'Hash' is by far the most common used. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] English vs American voice files
Agreed. Lets not get involved with dictating how many numbers someone dials here. Yes, definitely. How's about Please enter the full telephone number including the STD code. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP-Asterisk-GnuGK-Cisco 5300
UPDATE.
Changed the authentication scheme in GnuGK and now Asterisk can register
succesfully
BUT still can't place calls.
I'm getting a IE: Cause - No route to destination error. I'm pasting
bellow the H323 trace for the call originated from Asterisk... and below
that... a H323 trace for a call from a Cisco ATA.
The ATA can place the call to the same destination just perfect.
I APOLOGIZE FOR THE HUGE POST.
CALL FROM ASTERISK:
2004/09/18 01:20:01.294 2 RasSrv.cxx(2224) GK Read
from 68.88.232.68:32779
2004/09/18 01:20:01.298 3 RasSrv.cxx(2237) GK
admissionRequest {
requestSeqNum = 38834
callType = pointToPoint null
endpointIdentifier = 9 characters {
0034 0035 0032 0036 005f 0065 006e 0064 4526_end
0070 p
}
destinationInfo = 1 entries {
[0]=dialedDigits 835218329287863
}
destCallSignalAddress = ipAddress {
ip = 4 octets {
42 76 ee c6Bv..
}
port = 1720
}
srcInfo = 3 entries {
[0]=dialedDigits 2000
[1]=h323_ID 8 characters {
0061 0073 0074 0065 0072 0069 0073 006b asterisk
}
[2]=dialedDigits 12812812281
}
bandWidth = 10
callReferenceValue = 16568
conferenceID = 16 octets {
20 4d 6f ea b0 07 d9 11 85 6d 00 a0 cc 5c d3 e4Mo..m...\..
}
activeMC = FALSE
answerCall = FALSE
canMapAlias = TRUE
callIdentifier = {
guid = 16 octets {
b2 4c 6f ea b0 07 d9 11 85 6d 00 a0 cc 5c d3 e4 .Lo..m...\..
}
}
gatekeeperIdentifier = 12 characters {
0074 0065 006c 0063 006f 006e 0063 0065 telconce
0070 0074 0067 006b ptgk
}
willSupplyUUIEs = TRUE
}
2004/09/18 01:20:01.300 1 RasSrv.cxx(1321) GK ARQ Received
2004/09/18 01:20:01.300 4 gkauth.cxx(319) GkAuth default
check ok
2004/09/18 01:20:01.305 3 RasSrv.cxx(1585) GK ARQ will
request bandwith of 10
2004/09/18 01:20:01.307 2 RasTbl.cxx(1885)
CallTable::Insert(CALL) Call No. 203, total sessions : 1
2004/09/18 01:20:01.308 2 RasSrv.cxx(1696)
ACF|192.168.1.3:1720|4526_endp|16568|835218329287863:dialedDigits|2000:dialedDigits=asterisk:h323_ID=12812812281:dialedDigits|false;
2004/09/18 01:20:01.309 3 RasSrv.cxx(2164) GK Send to
68.88.232.68:32779
admissionConfirm {
requestSeqNum = 38834
bandWidth = 10
callModel = gatekeeperRouted null
destCallSignalAddress = ipAddress {
ip = 4 octets {
42 76 ee c6Bv..
}
port = 1721
}
irrFrequency = 120
willRespondToIRR = FALSE
uuiesRequested = {
setup = FALSE
callProceeding = FALSE
connect = FALSE
alerting = FALSE
information = FALSE
releaseComplete = FALSE
facility = FALSE
progress = FALSE
empty = FALSE
status = FALSE
statusInquiry = FALSE
setupAcknowledge = FALSE
notify = FALSE
}
}
2004/09/18 01:20:01.309 5 RasSrv.cxx(2178) GK Sent
Successful
2004/09/18 01:20:01.378 3ProxyThread.cxx(503) ProxyL
Connected from 68.88.232.68:33474
2004/09/18 01:20:01.379 5ProxyThread.cxx(538) ProxyH(1) add a
socket, total 2
2004/09/18 01:20:01.379 4ProxyThread.cxx(659) ProxyH(1) 1
sockets selected from 2, total 1/2
2004/09/18 01:20:01.380 5ProxyThread.cxx(354) Q931s Reading
from 68.88.232.68:33470
2004/09/18 01:20:01.380 3 ProxyChannel.cxx(417) Q931s
Received: ReleaseComplete CRV=16566 from 68.88.232.68:33470
2004/09/18 01:20:01.384 4 ProxyChannel.cxx(373) Q931Received: {
q931pdu = {
protocolDiscriminator = 8
callReference = 16566
from = originator
messageType = ReleaseComplete
IE: Cause - No route to destination = {
80 83 ..
}
IE: User-User = {
25 80 06 00 08 91 4a 00 02 01 11 00 4c c2 7e 58 %.J.L.~X
b0 07 d9 11 85 6d 00 a0 cc 5c d3 e4 02 80 01 00 .m...\..
}
}
h225pdu = {
h323_uu_pdu = {
h323_message_body = releaseComplete {
protocolIdentifier = 0.0.8.2250.0.2
callIdentifier = {
guid = 16 octets {
4c c2 7e 58 b0 07 d9 11 85 6d 00 a0 cc 5c d3 e4
L.~X.m...\..
}
}
}
h245Tunneling = FALSE
}
}
}
CALL FROM ATA:
2004/09/18 01:37:38.102 2 RasSrv.cxx(2224) GK Read
from 68.88.232.68:40232
2004/09/18 01:37:38.105 3 RasSrv.cxx(2237) GK
admissionRequest {
requestSeqNum = 3247
callType = pointToPoint null
endpointIdentifier = 9 characters {
0034 0034 0038 0033 005f 0065 006e 0064 4483_end
0070 p
}
destinationInfo = 1 entries {
[0]=dialedDigits 835212812928686
}
srcInfo = 2 entries {
[0]=dialedDigits 3001
Re: [Asterisk-Users] English vs American voice files
1) One of the recordings says please enter the full 10 digit number starting with the area code. Any opinions on whether this should be changed for the UK and, if so, to what? Whilst you might be targeting the UK, it is still best to keep it generic - my suggestion would be simply 'please enter the full telephone number including the area code' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RSS Feed Added to Asterisk News Site
I've put up an RSS feed on my site so that you can browse the headlines etc without actually visiting the site. If you use FireFox, Sage is a nice RSS reader. The URL is: http://www.sineapps.com/rssfeed.php _This is not a webpage._ It is a feed of the headlines and descriptions for Asterisk news for RSS compatible clients. Have a quick search on Google if you don't already have one because they make browsing news so much easier. Cheers, Matt Riddell (New Zealand Digium Distribution/Custom Software) http://www.sineapps.com/news.php (asterisk news) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 2, Issue 163
On 17 Sep 2004 at 13:20, Angel Diaz wrote: Hi Matt, I have verified with ztmonitor the audio level and it was too low, then with this the fax machine report Not Response. I modified the audio level in zapata.conf and after that the fax machine report Commnunication Error. Do you an idea what could be ? Thanks, Angel. I'd say it's probably now too high. Lower it by one or two and try again. Matt Riddell (New Zealand Digium Distribution/Custom Software) http://www.sineapps.com/contact.php (contact us) http://www.sineapps.com/news.php (asterisk news) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How To get response of command from another socket
hi i logged on to manager API from other terminal by telnet IPADDR 5038 now logged in with username mark let's say this connection Window A now i opened another connection with Manager API with same usename lets say this window B now if i give a command like originate,Redirect through window A connection , can i able to see its response:success/failure Originate:failed/succesfully queued.. in another window B i think its not possible to see a response of command from another socket of the same user is it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No sound
Hello,
I have just set up an asterisk box (Debian unstable) and I would like
to test it with a H.323 application (gnomemeeting). When I call the
demo voice menu, I can't hear any sound. asterisk says that the
soundfile is played:
-- Executing BackGround(H323/ip$212.9.189.172:30005/29597, demo-congrats) in new
stack
-- Playing 'demo-congrats' (language 'en')
Using strace I found out that the file is actually read, but there is
no subsequent traffic on the network. Neither a firewall nor NAT is
involved. gnomemeeting says in its history that the codec gets
immediately closed after the connection is established:
12:05:56 Opened codec G.711-ALaw-64k{sw} for transmission
12:05:56 Connected with root using The NuFone Network's H.323 Channel Driver for
Asterisk 1.0.0
12:05:56 Closed codec G.711-ALaw-64k{sw} which was opened for transmission
I edited h323.conf to use different codecs but all produced the same
result. Strangely, I can send dtmf tones and they are interpreted by
asterisk.
Do you have an idea on what's going on here and how to fix it?
Thanks a lot,
Hendrik
___
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Re: [Asterisk-Users] Asterisk presence utility
On Sat, 18 Sep 2004, Bill Seddon wrote: I've spent a couple of evenings writing a presence utility in C# so that a window, listing the currently registered SIP phones, can be displayed and a user can see who is on the phone and who is not. It uses the Manager API and works well, updating the display as API events are received. But what I want to be able to do is add to the list of displayed users when a phone registers and remove them when a phone unregisters. However the Manager API does not seem to generate event messages for these events. Is this correct or have a I missed an option somewhere? Certainly the register and unregister event is displayed on the Asterisk command line. It is possible to have the utility run the sip show peers command periodically and update its list based on the results of the command. However that means polling the server periodically, comsuming resources, instead of being notified just once. Any insight gratefully received. Bill Seddon This is purely conjecture on my part, but it seems to me if the console is generating SIP Un/Registration messages, that it would be trivial for the manager interface to display them as well. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] English vs American voice files
Mark Phillips [EMAIL PROTECTED] wrote: [...] Could some clever wag that deals with the language bits of * create some other languages like British, Aussie, SouthAfrican. I'd also be looking for Welsh too (anyone here speak Taff?) I don't, but I know people who do. I get the distinct impression that Plaid Cymru could put up a candidate in Birmingham and win a seat. How about Georgie (I'm kidding about that one). You may be kidding, but at least when I was calling Newcastle in 1998-ish, Digital Dot (as we affectionately call BT's spoken announcements) had a completely different accent in the 0191 exchange to the rest of the UK. It wasn't a SysX/SysY thing, but apparently special announcements just for that exchange. (I've also heard custom announcements on 01902, but thankfully not in Wulv'rumpt'n dialect.) If I were commissioning a voice, I'd probably go for an educated Scottish, Welsh or Irish accent. The London and Essex voice seems overused and also irritates me. The bonus is that a provincial voice actor should be cheaper than a London-based one. All these modes of English are more than just a dialect. My 7 or so years as an Ex-Pat in the US have taught me that American really is a valid language. Whilst most of us English speakers can cope with American we'd be a bit suprised when calling a VM system in Slough, Cooperpedy or Pretoria only to be spoken to in American. I *think* most people are aware that it's the voicemail system that's American, not the company they're calling. I'm usually more surprised to hear a British speaker in voicemail prompts :) Am I just ranting here or does someone get my point? Well, it'd definitely be nice if all our voice prompts were consistent, but as it is there's an occasionally jarring mix of the Digium lady and a bloke from our office. It works though, and callers don't get confused, which is the main thing. -- I want to know how God created this world. I am not interested in this or that phenomenon, in the spectrum of this or that element. I want to know His thoughts; the rest are details. - Albert Einstein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How To get response of command from another socket
You are correct that you will not be able to see a command response directly. However Asterisk will generate events that will be received by any manager api connection and it is these events that you will want to monitor from one or both connection windows. For example, when you originate a call, Asterisk will generate a newexten event that will contain the name of the originating device and some other stuff. If as a result of the origination, macros need to be evaluated, each macro evaluation will generate an event. Eventually there will be an event representing the dial() command from the dialplan followed by a ringing event. At the same time, an event showing that the called party device has been called will be generated. At some point the call may be answered and at this time a link event is generated showing the two parties connected in a call. Finally one or other of the parties will hangup and hangup events (one for each connected device) are generated. So although you cannot see the response generated by an action in one connection from another, you can pretty much infer the response from the events. In your case, a successful action response in one window will generate a set of events that can be seen in another window. Bill Seddon From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of vrushank Sent: September 20, 2004 12:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] How To get response of command from another socket hi i logged on to manager API from other terminal by telnet IPADDR 5038 now logged in with username mark let's say this connection Window A now i opened another connection with Manager API with same usename lets say this window B now if i give a command like originate,Redirect through window A connection , can i able to see its response:success/failure Originate:failed/succesfully queued.. in another window B i think its not possible to see a response of command from another socket of the same user is it? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] English vs American voice files
Bill Seddon [EMAIL PROTECTED] wrote: My wife has been recording the text published on the wiki. A couple of questions for you: 1) One of the recordings says please enter the full 10 digit number starting with the area code. Any opinions on whether this should be changed for the UK and, if so, to what? I'd probably go for please enter the full national number, including the area code. A play with BT's automated services on 150 (I'm thinking particularly of the Friends and Family number change section) should be fruitful as to what BT thinks is the best wording. 2) The recordings seem dull on playback even though we are recording using a good quality microphone with matching impedance. Initially we recorded using 16 bit/8K sampling on the basis that this is what is required by Asterisk but that was really terrible. So we're sampling at higher rates on the basis that we can use sox to change it as necessary. Any thoughts on what we can do to make the recordings sound sharper? How about making the recordings on a telephone? Leave yourself a voicemail containing the choice phrases, then edit them out using something like Audacity. -- I have four children which is not bad considering I'm not a Catholic. - Sir Peter Ustinov ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: FW: [Asterisk-Users] Polycom IP500
When I send the message from the phone, I use only the phones extension. Or, I think I am. I set up each phone's buddy list with the extension (only) of the other phone. When I send the message, I just picked the buddy off the list. Both the sip.conf and phone configs are pretty vanilla. [2014] type=friend username=2014 secret=secret host=dynamic context=from-internal canreinvite=no auth=md5 dtmfmode=rfc2833 pickupgroup=1 callergroup=1 qualify=yes The second phone's config is identical except for peername, username and secret. The phone configs are entirely standard, except for adding the SIP proxy info to sip.cfg, and the particular phone's information (username, display label, password, etc) to phoneXXX.cfg. I really didn't have a use for this feature, so I didn't explore it to great detail. - Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] English vs American voice files
Bill Seddon wrote: My wife has been recording the text published on the wiki. A couple of questions for you: 1) One of the recordings says please enter the full 10 digit number starting with the area code. Any opinions on whether this should be changed for the UK and, if so, to what? 2) The recordings seem dull on playback even though we are recording using a good quality microphone with matching impedance. Initially we recorded using 16 bit/8K sampling on the basis that this is what is required by Asterisk but that was really terrible. So we're sampling at higher rates on the basis that we can use sox to change it as necessary. Any thoughts on what we can do to make the recordings sound sharper? You can't have have sharp sounding voice over the telephone. Telephone calls really do sound dull. We are so used to them, we tend to blot that out of our perception. You can record at a higher sampling rate, and then convert to 8K for final use. However, once you convert to 8K it will still sound dull. You might use an equaliser to pep up the 2.5kHz to 3.5kHz area a bit. That might make the perceived quality a little better. Don't expect miracles, though. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
