RE: [asterisk-users] Asterisk 1.4 - no PRI and no Zap?
But apart from that: have you tried at least building that driver with 1.4.0 ? Yep. The build process seems to work just fine. The ztcfg and zttool stuff all acts normal. I copied tor2.c and tor2-hw.h from the custom 1.4.0-beta1 drivers (that work just fine with asterisk 1.4.0-beta1) and recompiled zaptel, libpri, asterisk and asterisk-addons in that order. My concern is that the custom drivers might have one or more lines changed in zaptel.c or something else. I tried a diff but there was way too much there so I bailed. I've asked the OEM to let me know when the 'official' 1.4 drivers are ready. Thanks, MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 and unicall
Guys, anybody knows if 1.4 has support for unicall or if/which version of unicall will compile on it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Question about MWI in Asterisk 1.4.0
Sounds great. What's the mechanism by which Asterisk servers communicate the mwi status between them? -Original Message- From: Jean-Yves Avenard [mailto:[EMAIL PROTECTED] Sent: Monday, December 25, 2006 11:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about MWI in Asterisk 1.4.0 Hi On 12/26/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: No, Asterisk 1.4 does not include any functionality for multi-server MWI. The SIP functionality improvements are just better support for the 'pull' model of SIP MWI, in addition to the 'push' model Asterisk has used in the past. If I adapt the patch for multi-server WMI for Asterisk 1.4, is there any chances it would be committed to trunk? Would be ace if it became a standard feature... JY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] flight and the agi
Hello, I am working with a php/agi example now and really don't like the way flight sounds...I am just using it like below. Is there a better voice app to use? Also, I am wanting the agi to hit a webservice so it will return an array, is it possible to have asterisk read the array and allow the user to go to next element, skip back, etc? Thanks! exten = 711,5,Flite(At the beep enter the zip code.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cdr_addon_mysql.so did not register itself during load
I've loaded Asterisk 1.4 with the addons 1.4, libpri 1.4 and Zaptel 1.4 as well. I can place calls but I noticed the MySQL was writing out to the database. When doing an Asterisk load with asterisk - I saw the following: [Dec 26 11:02:08] WARNING[10029]: loader.c:375 load_dynamic_module: Module 'cdr_addon_mysql.so' did not register its [Dec 26 11:02:08] WARNING[10029]: loader.c:607 load_resource: Module 'cdr_addon_mysql.so' could not be loaded. I've searched this out on Google and got no responses back that match. I checked modules.conf and both the cdr_addons_mysql.so and res_config_mysql.so are set to preload. Any ideas what is wrong here? Something change in the 1.4 that requires changes somewhere else? Thanks Kevin Savoy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cdr_addon_mysql.so did not register itself during load
Savoy, Kevin - Williston, ND wrote: I’ve loaded Asterisk 1.4 with the addons 1.4, libpri 1.4 and Zaptel 1.4 as well. I can place calls but I noticed the MySQL was writing out to the database. When doing an Asterisk load with asterisk – I saw the following: [Dec 26 11:02:08] WARNING[10029]: loader.c:375 load_dynamic_module: Module 'cdr_addon_mysql.so' did not register its [Dec 26 11:02:08] WARNING[10029]: loader.c:607 load_resource: Module 'cdr_addon_mysql.so' could not be loaded. The module that is being loaded is not a 1.4 module. It is using the old way of module loading. You should make sure that you are using 1.4 addons and that they are installed. -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] flight and the agi
blackwater dev wrote: Hello, I am working with a php/agi example now and really don't like the way flight sounds...I am just using it like below. Is there a better voice app to use? Also, I am wanting the agi to hit a webservice so it will return an array, is it possible to have asterisk read the array and allow the user to go to next element, skip back, etc? Thanks! | exten = 711,5,Flite(At the beep enter the zip code.) People say Cepstral has better voices, but you can still tell it's machine produced. Having the prompt pre-recorded would probably be best sounding. As for an array, you can just use dialplan logic and dialplan variables. -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Subscription Bug?
Well, this is weird. After receiving a sip subscribe message from peer 2529266, here's what Asterisk responds with: -- (14 headers 0 lines)--- Found user '2529266' Looking for 2943110 in bell_CallStart (domain ua2.ipt.xxx.com) Dec 26 10:19:34 NOTICE[27345]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'bell_CallStart' Transmitting (no NAT) to xxx.yyy.142.139:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP xxx.yyy.142.139;branch=z9hG4bKd22096a5A22CE654;received=xxx.yyy.142.139 From: Foo Law sip:[EMAIL PROTECTED];tag=1AB6AFEA-D777BDB3 To: sip:[EMAIL PROTECTED];tag=as6ac26084 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 This is mighty strange, given this: hermes*CLI sip show peer 2529266 hermes*CLI * Name : 2529266 Secret : Set MD5Secret: Not set Context : bell_CallStart Subscr.Cont. : bell_WatchBLF Language : en Accountcode : 2529266 Asterisk is saying that bell_CallStart doesn't exist (which it doesn't), but because of that decides to not accept the SIP subscription. The two are not realated to one another. I'm wondering what Asterisk has been smoking over the last few days while I was away... Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP Subscription Bug?
Well there's ya problem. If 2943110 doesn't have a match in the dialplan anywhere, Asterisk pukes. What's up with that? I don't see why that is necessary. Doug. -Original Message- From: Douglas Garstang Sent: Tuesday, December 26, 2006 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP Subscription Bug? Well, this is weird. After receiving a sip subscribe message from peer 2529266, here's what Asterisk responds with: -- (14 headers 0 lines)--- Found user '2529266' Looking for 2943110 in bell_CallStart (domain ua2.ipt.xxx.com) Dec 26 10:19:34 NOTICE[27345]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'bell_CallStart' Transmitting (no NAT) to xxx.yyy.142.139:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP xxx.yyy.142.139;branch=z9hG4bKd22096a5A22CE654;received=xxx.yyy.142.139 From: Foo Law sip:[EMAIL PROTECTED];tag=1AB6AFEA-D777BDB3 To: sip:[EMAIL PROTECTED];tag=as6ac26084 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 2 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 This is mighty strange, given this: hermes*CLI sip show peer 2529266 hermes*CLI * Name : 2529266 Secret : Set MD5Secret: Not set Context : bell_CallStart Subscr.Cont. : bell_WatchBLF Language : en Accountcode : 2529266 Asterisk is saying that bell_CallStart doesn't exist (which it doesn't), but because of that decides to not accept the SIP subscription. The two are not realated to one another. I'm wondering what Asterisk has been smoking over the last few days while I was away... Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] controlled playback for MP3
Is there anything available for contolled playback of mp3 files that offers the same functionality(rewind,skip,pause) as the controlplayback command does for gsm and wav? Thanks, Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Subscription Bug?
Douglas Garstang wrote: Well there's ya problem. If 2943110 doesn't have a match in the dialplan anywhere, Asterisk pukes. What's up with that? I don't see why that is necessary. Doug. I'm slightly confused by what you mean... can you elaborate more? -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to limit the duration of the MeetMe conversation?
How to limit the duration of the MeetMe conversation? -- thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.0 (release) and G.729
Is anybody else having problems with Asterisk 1.4.0 with the G.729 codec? I am running on a Dual PIII-866 and have tried the i386, i586, i686 and pentium3m (not in that order) versions of the module. Whenever a call comes in that needs to be transcoded, there is no audio for a couple of seconds then the call is dropped. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP Subscription Bug?
To put it generically, if user A subscribes to the status of user B, and there is no dialplan match for user B, then Asterisk will return 404 Not Found to user A. -Original Message- From: Joshua Colp [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 26, 2006 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Subscription Bug? Douglas Garstang wrote: Well there's ya problem. If 2943110 doesn't have a match in the dialplan anywhere, Asterisk pukes. What's up with that? I don't see why that is necessary. Doug. I'm slightly confused by what you mean... can you elaborate more? -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 missing sound in Spanish
Today I installed Asterisk 1.4 in my office and I noticed that it is still missing the vm-youhaveno sound for voicemail. Without this sound voicemail will not work! This was reported since the first beta and is still not included. Be careful if you are upgrading. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chàvez Prats Director de Tecnologìa +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 missing sound in Spanish
Carlos Chavez wrote: Today I installed Asterisk 1.4 in my office and I noticed that it is still missing the vm-youhaveno sound for voicemail. Without this sound voicemail will not work! This was reported since the first beta and is still not included. Be careful if you are upgrading. Where did you report it? -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Subscription Bug?
On 26/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: To put it generically, if user A subscribes to the status of user B, and there is no dialplan match for user B, then Asterisk will return 404 Not Found to user A. Yes, because the subscribe is against an extension, which is translated to a SIP (or other technology) user via the 'Hint' entry for that extension in the dialplan. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 missing sound in Spanish
On Tue, 2006-12-26 at 14:46 -0400, Joshua Colp wrote: Carlos Chavez wrote: Today I installed Asterisk 1.4 in my office and I noticed that it is still missing the vm-youhaveno sound for voicemail. Without this sound voicemail will not work! This was reported since the first beta and is still not included. Be careful if you are upgrading. Where did you report it? I did not report it myself because someone in the list had already posted the issue. I submitted the bug today to make it official. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chàvez Prats Director de Tecnologìa +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] agi+cepstral driving me nuts
I just got cepstal working fine in the dial plan using code like:
exten = 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep
enter your zip code.)
The php script it calls is based on the nerdvittles weather one so it calls
a webpage which prints to the screen, the nerdvittles code uses system to
generate the .wav file then has the dial plan call it via:
//php script
$retcode2 = system (flite -f $tmptext -o $tmpwave) ;
//extensions
exten = 411,9,NoOp(Wave file: ${TMPWAVE})
exten = 411,10,Playback(${TMPWAVE})
Since I am using capstral, I simply changed the line to below which works
fine from the command line but when calling, I never hear it, it just hangs
up. Is it timing out? Is there a better way to do this? How can I return
just a string of Text to read so I don't have to create the .wav file then
play it?
$retcode2 = system (swift -n Diane -m text -f $tmptext -o $tmpwave) ;
Thanks!
___
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RE: [asterisk-users] Asterisk 1.4.0 (release) and G.729
Even when I move my license to the new install, I got no G729 license available on the new system. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Kenyon Sent: Tuesday, December 26, 2006 1:32 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4.0 (release) and G.729 Is anybody else having problems with Asterisk 1.4.0 with the G.729 codec? I am running on a Dual PIII-866 and have tried the i386, i586, i686 and pentium3m (not in that order) versions of the module. Whenever a call comes in that needs to be transcoded, there is no audio for a couple of seconds then the call is dropped. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP Subscription Bug?
Asterisk, imho, should still accept the subscription request from user A. -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 26, 2006 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Subscription Bug? On 26/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: To put it generically, if user A subscribes to the status of user B, and there is no dialplan match for user B, then Asterisk will return 404 Not Found to user A. Yes, because the subscribe is against an extension, which is translated to a SIP (or other technology) user via the 'Hint' entry for that extension in the dialplan. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Subscription Bug?
Why? You're saying 'please update me on the status of extension '1234'' when there's no such extension. Where's it going to get the data from? Better to get a 404, know something's wrong and correct a typo than let it succeed and just not work. Peter On 26/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: Asterisk, imho, should still accept the subscription request from user A. -Original Message- From: Peter Bowyer [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 26, 2006 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Subscription Bug? On 26/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: To put it generically, if user A subscribes to the status of user B, and there is no dialplan match for user B, then Asterisk will return 404 Not Found to user A. Yes, because the subscribe is against an extension, which is translated to a SIP (or other technology) user via the 'Hint' entry for that extension in the dialplan. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
All: I am looking to move cell phone providers. I have acquired the new cell phone and LOVE my new number but want to keep the old number as well. The new provider only will allow me to use one number or the other. They will port the old number if I want, but will keep my new number if I ask them to port the old one. Where can I go to get the old number ported from my old provider (the account is still active) and have it forwarded on to my new number (for cheap)? Sincerely, Lorell Hathcock Adaptive Data Works, LLC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Number forwarding and porting?
All: (This time with a subject line!) I am looking to move cell phone providers. I have acquired the new cell phone and LOVE my new number but want to keep the old number as well. The new provider only will allow me to use one number or the other. They will port the old number if I want, but will keep my new number if I ask them to port the old one. Where can I go to get the old number ported from my old provider (the account is still active) and have it forwarded on to my new number (for cheap)? Sincerely, Lorell Hathcock Adaptive Data Works, LLC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Number forwarding and porting?
This is obvious (However not an Asterisk question): You need to ask your old cell provider to port the number to the new one. Also, in the meantime you need to ask to forward the number to your new number. However, since there is nothing forced to him to do that, this is going to happen if your old provider is willing to do that. There are a lot of provider that can make your life miserable when it start something like this moves. They not only hate to loose you as a customer, they also hate to loose their numbers ( This is because they believe that the did belongs to them and not to the customer that pays for it). Good Luck, Carlos Alperin Seneca Communications, LLC _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lorell Hathcock Sent: Tuesday, December 26, 2006 2:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Number forwarding and porting? All: (This time with a subject line!) I am looking to move cell phone providers. I have acquired the new cell phone and LOVE my new number but want to keep the old number as well. The new provider only will allow me to use one number or the other. They will port the old number if I want, but will keep my new number if I ask them to port the old one. Where can I go to get the old number ported from my old provider (the account is still active) and have it forwarded on to my new number (for cheap)? Sincerely, Lorell Hathcock Adaptive Data Works, LLC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4 with a nortel call server 1000 running SIP (sdp headers)
Has anyone tried to get 1.4 running with a call server 1000 and SIP? I had 1.0.X running with a call server 1000 and had to tweek the code due to multipart SDP headers. Has multipart SDP headers been enhanced in 1.4. THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Questions about 1.4
At 09:37 AM 12/25/2006, you wrote: The Asterisk Development Team is pleased to announce the first release in the Asterisk 1.4 series, Asterisk 1.4.0! Another thing I've noticed is that twice today while sitting at the CLI prompt I was throw to the command line because Asterisk had exited. Typing asterisk started it right up again and then I could get back to the CLI. I think both times it happened at the end of a call. I'd try to list more info, but I don't know where to look. And three more times today Asterisk has just stopped. Starting to really annoy the wife! If it might have left any logs indicating the problem I'd be glad to look at them. The messages file has no indications of any problems that I can see. Ira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.0 (release) and G.729
Carlos Alperin wrote: Even when I move my license to the new install, I got no G729 license available on the new system. When you say new system, are you referring to a different computer? Your license is bound to the machine (well the network cards/MAC addresses) that it is registered to. If you use them in a machine with a different network configuration (or change the network configuration of the machine asterisk is running on), you need to re-register. You can do this by running the registration application again (can only be changed once without speaking to digium). The problem I'm having, the number of licenses show up correctly but as soon as the codec is in use, there is no sound and the call drops. The codecs I'm using are from: ftp://ftp1.digium.com/pub/asterisk/g729/asterisk-1.4/32-bit/i686/codec_g729a.so etc. The module loads correctly (afaict) without reporting any errors and if you query the number of licenses, it appears to work. It's only when the codec is actually encoding/decoding a stream that things go wrong. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Kenyon Sent: Tuesday, December 26, 2006 1:32 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4.0 (release) and G.729 Is anybody else having problems with Asterisk 1.4.0 with the G.729 codec? I am running on a Dual PIII-866 and have tried the i386, i586, i686 and pentium3m (not in that order) versions of the module. Whenever a call comes in that needs to be transcoded, there is no audio for a couple of seconds then the call is dropped. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4.0 (release) and G.729
No, I didn't said a new machine. I have a license with 23 active channels for G.729 for testing purposes. I only erase the old configuration before recompile it, since I was doing tests with the beta-2 release where the license was working. So, the only thing I didn't delete was the /licenses directory (/var/lib/asterisk/licenses). I'm using an AMD Athlon Dual 3800, so I'm using the 64 bits x86_64 code. After finishing installing everything, however I put back the codec formatxx.so files on /usr/lib/asterisk/modules And then go the console, I don't see any reference to G.729 on the list. And when I go to translation options I have no reference to any translation going to or from G729. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Kenyon Sent: Tuesday, December 26, 2006 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4.0 (release) and G.729 Carlos Alperin wrote: Even when I move my license to the new install, I got no G729 license available on the new system. When you say new system, are you referring to a different computer? Your license is bound to the machine (well the network cards/MAC addresses) that it is registered to. If you use them in a machine with a different network configuration (or change the network configuration of the machine asterisk is running on), you need to re-register. You can do this by running the registration application again (can only be changed once without speaking to digium). The problem I'm having, the number of licenses show up correctly but as soon as the codec is in use, there is no sound and the call drops. The codecs I'm using are from: ftp://ftp1.digium.com/pub/asterisk/g729/asterisk-1.4/32-bit/i686/codec_g729a .so etc. The module loads correctly (afaict) without reporting any errors and if you query the number of licenses, it appears to work. It's only when the codec is actually encoding/decoding a stream that things go wrong. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Kenyon Sent: Tuesday, December 26, 2006 1:32 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4.0 (release) and G.729 Is anybody else having problems with Asterisk 1.4.0 with the G.729 codec? I am running on a Dual PIII-866 and have tried the i386, i586, i686 and pentium3m (not in that order) versions of the module. Whenever a call comes in that needs to be transcoded, there is no audio for a couple of seconds then the call is dropped. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] I cant install zaptel drivers in suse 10.1
Hi, All How do I install Zaptel drivers on a system running Suse? Make results: grep: /lib/modules/2.6.16.13-4-smp/build/include/linux/autoconf.h: No existe el fichero o el directorio make[1]: Entering directory /usr/src/asterisk/zaptel/zaptel-1.4.0-beta2 make -C /lib/modules/2.6.16.13-4-smp/build SUBDIRS=/usr/src/asterisk/zaptel/zaptel-1.4.0-beta2 modules make[2]: Entering directory /usr/src/linux-2.6.16.13-4-obj/i386/smp make[2]: *** No hay ninguna regla para construir el objetivo modules. Altc. Make[2]: Leaving directory /usr/src/linux-2.6.16.13-4-obj/i386/smp make[1]: *** [linux26] Error 2 make[1]: Leaving directory /usr/src/asterisk//zaptel/zaptel-1.4.0-beta2 make: *** [all] Error 2 Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I cant install zaptel drivers in suse 10.1
This has been fixed in 1.4.0 - I would strongly recommend using that instead of the beta. - Original Message - From: Marco Torrez [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, December 26, 2006 5:44:41 PM GMT-0600 US/Central Subject: [asterisk-users] I cant install zaptel drivers in suse 10.1 Hi, All How do I install Zaptel drivers on a system running Suse? Make results: grep: /lib/modules/2.6.16.13-4-smp/build/include/linux/autoconf.h: No existe el fichero o el directorio make[1]: Entering directory /usr/src/asterisk/zaptel/zaptel- 1.4.0-beta2 make -C /lib/modules/2.6.16.13-4-smp/build SUBDIRS=/usr/src/asterisk/zaptel/zaptel-1.4.0-beta2 modules make[2]: Entering directory /usr/src/linux-2.6.16.13-4-obj/i386/smp make[2]: *** No hay ninguna regla para construir el objetivo modules. Altc. Make[2]: Leaving directory /usr/src/linux-2.6.16.13-4-obj/i386/smp make[1]: *** [linux26] Error 2 make[1]: Leaving directory /usr/src/asterisk//zaptel/zaptel-1.4.0-beta2 make: *** [all] Error 2 Thanks -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] I cant install zaptel drivers in suse 10.1
You need the kernel source installed to compile Zaptel. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Torrez Sent: Tuesday, December 26, 2006 4:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] I cant install zaptel drivers in suse 10.1 Hi, All How do I install Zaptel drivers on a system running Suse? Make results: grep: /lib/modules/2.6.16.13-4-smp/build/include/linux/autoconf.h: No existe el fichero o el directorio make[1]: Entering directory /usr/src/asterisk/zaptel/zaptel- 1.4.0-beta2 make -C /lib/modules/2.6.16.13-4-smp/build SUBDIRS=/usr/src/asterisk/zaptel/zaptel-1.4.0-beta2 modules make[2]: Entering directory /usr/src/linux-2.6.16.13-4-obj/i386/smp make[2]: *** No hay ninguna regla para construir el objetivo modules. Altc. Make[2]: Leaving directory /usr/src/linux-2.6.16.13-4-obj/i386/smp make[1]: *** [linux26] Error 2 make[1]: Leaving directory /usr/src/asterisk//zaptel/zaptel-1.4.0-beta2 make: *** [all] Error 2 Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] agi+cepstral driving me nuts
Why don't you try app_swift?
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Swift
This one even compiles on 1.4, and has buffering, meaning that it
doesn't have to wait for the tts to generate the complete output.
http://www.loopfree.net/app_swift/
exten = s,1,AGI(getinfo.php)
exten = s,2,Swift( ${RESULT_INFORMATION} )
Julián J. M.
On 12/26/06, blackwater dev [EMAIL PROTECTED] wrote:
I just got cepstal working fine in the dial plan using code like:
exten = 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep
enter your zip code.)
The php script it calls is based on the nerdvittles weather one so it calls
a webpage which prints to the screen, the nerdvittles code uses system to
generate the .wav file then has the dial plan call it via:
//php script
$retcode2 = system (flite -f $tmptext -o $tmpwave) ;
//extensions
exten = 411,9,NoOp(Wave file: ${TMPWAVE})
exten = 411,10,Playback(${TMPWAVE})
Since I am using capstral, I simply changed the line to below which works
fine from the command line but when calling, I never hear it, it just hangs
up. Is it timing out? Is there a better way to do this? How can I return
just a string of Text to read so I don't have to create the .wav file then
play it?
$retcode2 = system (swift -n Diane -m text -f $tmptext -o $tmpwave) ;
Thanks!
___
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Re: [asterisk-users] Asterisk 1.4.0 (release) and G.729
Carlos Alperin wrote: No, I didn't said a new machine. Sorry, misunderstanding, when you said new system I read that as new machine. I have a license with 23 active channels for G.729 for testing purposes. I only erase the old configuration before recompile it, since I was doing tests with the beta-2 release where the license was working. So, the only thing I didn't delete was the /licenses directory (/var/lib/asterisk/licenses). I'm using an AMD Athlon Dual 3800, so I'm using the 64 bits x86_64 code. After finishing installing everything, however I put back the codec formatxx.so files on /usr/lib/asterisk/modules And then go the console, I don't see any reference to G.729 on the list. Which formatxx.so files did you copy back? The new installation should have created a new format_g729.so. (if not, check with make menuselect that it is in fact selected to compile). Did you see the module loading messages in /var/log/asterisk/messages? Does show modules like g729 report that they are loaded? If the above is untrue, does module load codec_g729a.so produce an error ? And when I go to translation options I have no reference to any translation going to or from G729. When codec_g729.so loads the translation paths for to and from slin, there shouild be a line like the following in /var/log/asterisk/messages. == Registered translator 'g729tolin' from format g729 to slin, cost 4 == Registered translator 'lintog729' from format slin to g729, cost 19 Loaded codec_g729a.so = (Annex A/B (floating point) G.729 Codec (optimized for x86_64)) Thanks I'm just hoping someone comes along and tells me that It's a known bug, otherwise I'll probably end up spending a day trying to track down what's wrong. Oh, and if everything I've written above reads as utter bollocks, please remember that I did write it at 1am. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Input on Dundi
The RealTime command pulls all the entire record from the database and prepends all the fields with the last argument (here is have DN_) so when the record is pulled, all the records info is available as a variable like DN_port and DN_ipaddr. This is a really cool command. Hope this helps. Wow, thanks for the examples JR. This is exactly what I needed. I was not aware of the RealTime command. That will be very useful. I also stumbled across RealTimeUpdate recently and documented it on the wiki. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 5.8gig phone MWI
Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. Uniden DSS7815 MWI works with SPA3K. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4 with a nortel call server 1000 running SIP(sdp headers)
Actually, there was recently a bug fixed regarding multipart SDP parsing in chan_sip. That should have fixed the issue with CS1000s and SIP (among other things). I haven't actually tried it yet on my CS1000, but it should work. Regards, - Brad From: [EMAIL PROTECTED] on behalf of Jerry Geis Sent: Tue 12/26/2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 1.4 with a nortel call server 1000 running SIP(sdp headers) Has anyone tried to get 1.4 running with a call server 1000 and SIP? I had 1.0.X running with a call server 1000 and had to tweek the code due to multipart SDP headers. Has multipart SDP headers been enhanced in 1.4. THanks, Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about MWI in Asterisk 1.4.0
Hi On 12/27/06, Douglas Garstang [EMAIL PROTECTED] wrote: Sounds great. What's the mechanism by which Asterisk servers communicate the mwi status between them? With new IAX commands. The client can ask the server how many messages are waiting. I've started to port the modification on 1.4, but there's been a lot of changes between 1.2 and 1.4 with the introduction of context etc.. I need to understand how 1.4 is working now which may take a while. JY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] agi+cepstral driving me nuts
Too bad Cepstral hasnt still made a decent Spanish voice, the ones they
have still sound too computer like, not like the English ones they have
which sound great!
|-Original Message-
|From: [EMAIL PROTECTED] [mailto:asterisk-users-
|[EMAIL PROTECTED] On Behalf Of Julian J. M.
|Sent: Tuesday, December 26, 2006 6:26 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [asterisk-users] agi+cepstral driving me nuts
|
|Why don't you try app_swift?
|http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Swift
|
|This one even compiles on 1.4, and has buffering, meaning that it
|doesn't have to wait for the tts to generate the complete output.
|
|http://www.loopfree.net/app_swift/
|
|exten = s,1,AGI(getinfo.php)
|exten = s,2,Swift( ${RESULT_INFORMATION} )
|
|Julián J. M.
|
|On 12/26/06, blackwater dev [EMAIL PROTECTED] wrote:
| I just got cepstal working fine in the dial plan using code like:
|
| exten = 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep
| enter your zip code.)
|
|
| The php script it calls is based on the nerdvittles weather one so it
calls
| a webpage which prints to the screen, the nerdvittles code uses system to
| generate the .wav file then has the dial plan call it via:
|
| //php script
| $retcode2 = system (flite -f $tmptext -o $tmpwave) ;
|
| //extensions
| exten = 411,9,NoOp(Wave file: ${TMPWAVE})
| exten = 411,10,Playback(${TMPWAVE})
|
|
| Since I am using capstral, I simply changed the line to below which works
| fine from the command line but when calling, I never hear it, it just
hangs
| up. Is it timing out? Is there a better way to do this? How can I
return
| just a string of Text to read so I don't have to create the .wav file
then
| play it?
|
| $retcode2 = system (swift -n Diane -m text -f $tmptext -o $tmpwave) ;
|
|
| Thanks!
|
|___
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|
|asterisk-users mailing list
|To UNSUBSCRIBE or update options visit:
| http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Re: Number forwarding and porting?
Lorell Hathcock wrote: I am looking to move cell phone providers. I have acquired the new cell phone and LOVE my new number but want to keep the old number as well. The new provider only will allow me to use one number or the other. They will port the old number if I want, but will keep my new number if I ask them to port the old one. Where can I go to get the old number ported from my old provider (the account is still active) and have it forwarded on to my new number (for cheap)? Cell phones can only have one number attached to them (at least in the US). You can port any US number to any other local phone line within the same rate center (local switch area). Cellular phones can accept ANY US number. You pay a charge for LNP features on every bill so the carrier can not refuse unless it is to a land line out of the area of the number. If you want both numbers to ring to your cell phone you can get a carrier to do a call forward. CLEC's or call centers are best for this. They port the number from your old cell carrier and program your old number to forward all calls to your new cell number. Your outbound calls will have to carry the caller ID of the number assigned to your cell phone. They can not use the call forward number for the outbound caller ID. If you have local phone service check with that carrier for their call forward options. Otherwise check with a few answering service companies in your area. Allen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agent presence
Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on pause. I'm using chan_agent for the agents, so agents are logged in and out using AgentCallbackLogin (I know it's deprecated in 1.4, but it's working well for us at the moment) and the agents are put on pause using PauseQueueMember and UnpauseQueueMember. I've figured out I can show whether an agent is logged in or out by creating a dummy extension with a hint as follows:- exten = 151,1,Dial(Agent/151) exten = 151,hint,Agent/151 X-Lite quite happily shows the agent as Ready when they're logged in, unavailable when logged out and On the Phone when (funnily enough) they're taking a call. However, when the agent is on pause, they are still shown as Ready. Is this a limitation of chan_agent, Pause/UnpauseQueueMember, Asterisk 1.2's presence support, or is there something else I can do in order to get the agent shown indicated as something other than Ready when they're on pause? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
