[Asterisk-Users] sound file installation problem
I downloaded asterisk-sounds-1.2.0-beta1, superused, then typed "make install". The installation stopped with the following error: No description for sounds/access-code.gsmmake: *** [datafiles] Error 1 Does anyone have any useful tips? I'm running Debian 3.0. Thanks, WILL ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_zap vs. Panasonic DTMF integration
The Panasonic KX-TA624 series PBXes (and similar models) support a DTMF integration feature that can be enabled for dedicated voice mail ports. What I want to do is connect an X100P FXO port to a jack on the Panasonic and make use of the Panasonic's DTMF call progress tones that it provides in DTMF integration mode. The integration works well when a Panasonic extension is forwarding into one of the VM ports and Asterisk picks it up; in this mode, after Asterisk picks up the call, the Panasonic sends #6XXX, where XXX is the voice mailbox. When a user presses a message waiting button, the Panasonic will send #6*XXX, and that also works fine. (It is necessary to delay answering by one second if the Panasonic is set to use the double-ring cadence.) Problems start when Asterisk dials an extension into the Panasonic. When the FXO port picks up and dials an extension on the Panasonic station port, the Panasonic will return the following indications as DTMF digits: 1 Ringback 2 Busy 3 Reorder 4 DND 5 Answer (recipient has answered) 6 Forwarded to other VM port 7 Forwarded to other VM port, but VM busy 8 Forwarded to non-VM extension 9 Confirmation (for turning message waiting lights on or off) #9 Recipient has disconnected This seems to be something that would best be implemented within chan_zap.c, but there's a problem. When Asterisk has finished dialing, the Panasonic returns 1-4 or 6-9 before Asterisk can detect the tone. I'd say it takes 200 ms or less for the Panasonic to return the DTMF digit. Is there any other approach that would work that wouldn't require diving into the internals? In this example, we're dialing Panasonic extension 104, and the Panasonic sends a DTMF 1 to indicate ringing, but the 1 is never received by chan_zap. It doesn't matter whether or not I have echo cancellation enabled. Here's the debug output, where an IAX extension is calling out on a Zap channel into the Panasonic: Oct 1 23:21:02 DEBUG[22774] chan_zap.c: Dialing '104' Oct 1 23:21:02 DEBUG[22774] chan_zap.c: Deferring dialing... Oct 1 23:21:02 DEBUG[22774] channel.c: Set channel Zap/1-1 to read format slin Oct 1 23:21:02 DEBUG[22774] channel.c: Set channel IAX2/103-1 to write format slin Oct 1 23:21:02 DEBUG[22774] channel.c: Set channel IAX2/103-1 to read format slin Oct 1 23:21:02 DEBUG[22774] channel.c: Set channel Zap/1-1 to write format slin Oct 1 23:21:02 DEBUG[22774] app_queue.c: Device 'IAX2/103' changed to state '2' (In use) Oct 1 23:21:02 DEBUG[22774] app_queue.c: Device 'Zap/1' changed to state '2' (In use) Oct 1 23:21:02 DEBUG[22774] app_queue.c: Device 'Zap/1' changed to state '2' (In use) Oct 1 23:21:02 DEBUG[22774] chan_iax2.c: Ooh, voice format changed to 2 Oct 1 23:21:02 DEBUG[22774] channel.c: Set channel IAX2/103-1 to read format slin Oct 1 23:21:02 DEBUG[22774] chan_zap.c: Dropping frame since I'm still dialing on Zap/1-1... [ repeated many times... ] Oct 1 23:21:02 DEBUG[22774] chan_zap.c: Exception on 20, channel 1 Oct 1 23:21:02 DEBUG[22774] chan_zap.c: Got event Hook Transition Complete(12) on channel 1 (index 0) Oct 1 23:21:02 DEBUG[22774] chan_zap.c: Dropping frame since I'm still dialing on Zap/1-1... [ repeated many times... ] Somewhere in here, the DTMF 1 sent by the Panasonic is lost. Oct 1 23:21:03 DEBUG[22774] chan_zap.c: Exception on 20, channel 1 Oct 1 23:21:03 DEBUG[22774] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0) Oct 1 23:21:03 DEBUG[22774] chan_zap.c: Enabled echo cancellation on channel 1 Oct 1 23:21:03 DEBUG[22774] channel.c: Set channel IAX2/103-1 to read format slin Oct 1 23:21:03 DEBUG[22774] channel.c: Set channel Zap/1-1 to write format slin Oct 1 23:21:03 DEBUG[22774] channel.c: Set channel Zap/1-1 to read format slin Oct 1 23:21:03 DEBUG[22774] channel.c: Set channel IAX2/103-1 to write format slin Oct 1 23:21:03 DEBUG[22774] chan_iax2.c: Answering IAX2 call Oct 1 23:21:03 DEBUG[22774] app_queue.c: Device 'Zap/1' changed to state '2' (In use) Oct 1 23:21:03 DEBUG[22774] app_queue.c: Device 'IAX2/103' changed to state '2' (In use) Oct 1 23:21:04 DEBUG[22774] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 1 [ repeated... ] Here, the destination extension answers, and the Panasonic sends a 5: Oct 1 23:21:07 DEBUG[22774] chan_zap.c: DTMF digit: 5 on Zap/1-1 Oct 1 23:21:07 DEBUG[22774] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 1 [ repeated during conversation... ] Now, the extension has disconnected, and the Panasonic sends #9: Oct 1 23:21:12 DEBUG[22774] chan_zap.c: DTMF digit: # on Zap/1-1 Oct 1 23:21:12 DEBUG[22774] chan_zap.c: DTMF digit: 9 on Zap/1-1 The Zap channel doesn't disconnect until the Panasonic sends reorder tone, which triggers Asterisk's busy detection. Russ ___ --Bandwidt
RE: [Asterisk-Users] Best way to create IVR/voicemail system
Angus, This might get you started. As an IVR developer, these examples seem pretty complex for a very simple action. I am also fairly new to *, so maybe I am wrong and will figure it out as I learn more. http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Record http://www.voip-info.org/wiki-Asterisk+tips+ivr+menu Shawn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Angus Comber Sent: Friday, September 30, 2005 1:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Best way to create IVR/voicemail system Hello I want to setup a system where people can dial a number and then a system will ask them questions for which they will leave answers. Eg something like this: Answer Playback(whatisyournamemsg) Record(yourname:gsm) Playback(whatisyourheight) Record(yourheight:gsm) Playback(thankyou) Hangup Is this the best way to do this sort of thing? Do users then just access the responses by eg *98 - or does this work a little differently to voicemail? How do we retrieve the responses? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Updated presentation of Asterisk 1.2
Olle E. Johansson a écrit : Friends, I have updated my Asterisk 1.2 presentation with the latest information. It is still available in the same place as before: http://www.astricon.net/asterisk1-2/ Just wanted to thank you for making this available. -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 483 527 / GSM: +689 797 527 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP 400 Bad Request from Cisco 7960/7940
We've been experiencing an odd issue lately. I'm not sure when it started because it's not happening on most calls--it seems confined to a couple of our queues. It's consistent though. Here's the CLI output: -- Got SIP response 400 "Bad Request" back from 192.168.249.94 -- SIP/502-9a58 is circuit-busy I've tried a few different Asterisk versions CVS-HEAD, stable, even 1.2 beta. I've also bounced between SIP firmware 7.4 and 7.5 on the 7960/7940 phones. Anyone else seeing anything like this? -Corey * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sourcing Eicon Diva V-4BRI/QuadBRI cards in Australia
Hey, If anyone supplies or knows where I can get either an Eicon Diva V-4BRI or a Junghanns QuadBRI card in Australia, please could you contact me offlist? Also, if anyone has any second-hand ones they need to get rid of, I'd be interested in making an offer! I've been checking eBay, but there don't seem to be any of these available at the moment. Thanks, Avi -- National Manager - Special Projects < Sydney . Melbourne . Canberra . Hobart . London /> Walter Turnbull Bldg T: +61 (0) 2 6233 0607 44 Sydney Ave, F: +61 (0) 2 6233 0696 Forrest, W: http://www.squiz.net/ ACT 2603 .>> Open Source - Own it - Squiz.net ./> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with VM Distribution Groups
Hi, We're experiencing a problem with basic VM distribution groups where messages won't be delivered. VM is called with a command like: Voicemail(20&30&39&... CLI Shows: Oct 1 20:54:33 NOTICE[26943]: app_voicemail.c:1990 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Oct 1 20:54:38 WARNING[26943]: app.c:1125 ast_lock_path: Failed to lock path '': File exists for each attempt and nothing is delivered. I've read a number of threads where the VM distribution list exceeds 256 characters and this breaks things, but that isn't the case here. Delivery fails with even a small number of mailboxes. Voicemail works normally otherwise. Can anyone advise? Thanks, -Corey * This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Digium T-1 and FXO cards for sale
Tracy R Reed wrote: > If anyone is interested in some used digium hardware for their projects: > T-1 card: > http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5101873998 > 4 port FXO cards: > http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5102115738 > http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5102264062 All of this stuff is still going pretty cheap with just 24 hours left. -- Tracy R Reed ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Can't compile zaptel (CVS Head) on Debian
On Sun, Oct 02, 2005 at 12:14:10AM +0100, Dogers wrote: > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Philipp von Klitzing > > > with both CVS HEAD and 1.2beta1 I don't succeed with either > > "make" or "make linux26". I checked more than once to make > > sure the required packages are in place - any suggestions? > > I had this recently - theres a problem with the 2.6.8-2 kernel, the package > maintainer knows about it. That's a different issue: zaptel.h has moved when m-a support was added and the package wasn't properly adapted. This is specific to building zaptel from the package. Not that I have this issue with building from my debs[1]. > > Install the linux-image-2.6.12 package (debian is changing from kernel-image > to linux-image, as they're changing the way theyre packaging it, or > something?!) and it'll compile fine. kernel-image/linux-image is just the binary . For building you naturally need the kernel-headers/linux-headers package. http://rapid.dotsrc.org/unstable/ http://rapid.dotsrc.org/experimental/ -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Can't compile zaptel (CVS Head) on Debian
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Philipp von Klitzing > with both CVS HEAD and 1.2beta1 I don't succeed with either > "make" or "make linux26". I checked more than once to make > sure the required packages are in place - any suggestions? I had this recently - theres a problem with the 2.6.8-2 kernel, the package maintainer knows about it. Install the linux-image-2.6.12 package (debian is changing from kernel-image to linux-image, as they're changing the way theyre packaging it, or something?!) and it'll compile fine. Andrew ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup half a call?
Scenario is as follows. Caller comes in over ZAP channel connects to handset on another ZAP channel. Call is bridged. I'd like the callee to be able to hangup on the caller and then be presented with a agi application. Basically the agent that answered the call has to enter a few responses to questions asterisk asks. On some ACD phone systems this is called a "wrap code". Lets you build basic call statistics. IE the agent enters a 1 on a sale and a 2 on no sale kinda thing. You run through your log file or sql db and produce a couple of basic counts. I'm using the new features.conf applicationmap to startup whatever I want on the dialplan. Instead of having the callee just hangup his phone, press *3 and launch on an extension in the dialplan. How do I hangup only the caller and let the callee continue? Another way I though to handle it is to introduce an H extension in the dialplan. h handles one side of the hangup, let H handle the other side Can anyone think of a way to handle this? I'd prefer not to introduce new features into asterisk at this time since they won;t be considered for Asterisk 1.2, and it looks like Asterisk 1.2 release release date has slipped to infinity. The patches will end up in bug database hell. -bill [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can't compile zaptel (CVS Head) on Debian
Hi there, with both CVS HEAD and 1.2beta1 I don't succeed with either "make" or "make linux26". I checked more than once to make sure the required packages are in place - any suggestions? Philipp /lib/modules/2.6.8-2-686/build make -C /lib/modules/2.6.8-2-686/build SUBDIRS=/usr/src/zaptel modules make[1]: Gehe in Verzeichnis »/usr/src/kernel-headers-2.6.8-2-686« CC [M] /usr/src/zaptel/zaptel.o In file included from include/asm/thread_info.h:16, from include/linux/thread_info.h:21, from include/linux/spinlock.h:12, from include/linux/capability.h:45, from include/linux/sched.h:7, from include/linux/module.h:10, from usr/src/zaptel/zaptel.c:44: include/asm/processor.h:87: error: array type has incomplete element type /usr/src/zaptel/zaptel.c: In function '__zt_receive_chunk': /usr/src/zaptel/zaptel.c:6162: warning: pointer targets in assignment differ in signedness make[2]: *** [/usr/src/zaptel/zaptel.o] Fehler 1 make[1]: *** [_module_/usr/src/zaptel] Fehler 2 make[1]: Verlasse Verzeichnis »/usr/src/kernel-headers-2.6.8-2-686« make: *** [linux26] Fehler 2 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Swap between callers
This just works with many phones with a single line registration, such as the Polycom IP501. - Original Message - From: "Angus Comber" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, October 01, 2005 10:09 AM Subject: [Asterisk-Users] Swap between callers > Hello > > On business phones it is often possible to have call waiting (think that is > the feature) whereby if you are talking to a caller you can see another > caller has called and you can swap between callers. For example, to say > hello, I am on call with someone else now can I call you back. > > How can this be implemented using SIP IP phones. Do you need to setup two > or more lines? How is it done? > > Angus > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > No virus found in this incoming message. > Checked by AVG Anti-Virus. > Version: 7.0.344 / Virus Database: 267.11.9/116 - Release Date: 9/30/05 > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Linksys register hangs Asterisk!
Here is a update with the solution.. Reinstallation of Debian! I think it was an update of Debian Unstable that made things stop working. Now I installed Debian stable with the same config and it works great now. Even that noone replied to my post thanks for reading it anyway! =) ~Johannes > Hey, > > I'w got a problem (bug maybe?). > > I have recently got my Asterisk to work perfect and I'm not trying to > setup some dial routes and get the system working as I wan't it to. > > Yesterday I was installing Festival and also did a "aptitude upgrade" on > my Debian Unstable installation. > After that the problem started. > > After some serious testing yesterday night and today I have tracked down > the problem to that it it is my Linksys WRTG54GP2 (Router with ATA) that > causes asterisk to stop working. > > Everytime it tries to register asterisk stops working normally. It don't > register any more information with sip debug activated. No incoming calls > is displayed and asterisk seems just to be seeing nothing that is going > on. > > I tried to restart asterisk and then make a incoming call directly, that > goes well. Asterisk answers and posts the normal route with voice answers. > Then I can see that the Linksys router is trying to register and after > that everything stops working. > > If I disable the linksys router to register itself everything works well, > asterisk answers and gived me the options to choose extension. > > So the problem is caused by the registration of Linksys. > This is the debug log from the registration until asterisk stops (moved to > the bottom of this mail) > > One interesting line is that the "Call-ID:" line after the @ contains the > IP number to the Linksys router WITHOUT THE LAST NUMBER in the address! > How can that be? The other lines containg the IP number is correct (in the > log replaced by ). > Can this be the cause for the problem ? > If not can there be anything else in this log that indicates what the > problem is? > > Hope someone got an answer because this is driving me crazy since I got it > all working this weekend after 2 weeks of trouble. > > Regards, > ~Johannes > > -- START SIP DEBUG LOG --- > Sip read: > REGISTER sip: SIP/2.0 > Via: SIP/2.0/UDP :5060;branch=z9hG4bK-d54de2e6 > From: >;tag=4c7b1b149bb4b329o0 > To: > > Call-ID: 66a3a900-ec8d9e2d@ > CSeq: 1 REGISTER > Max-Forwards: 70 > Contact: :5060>;expires=3600 > User-Agent: Linksys/RT31P2-3.1.3(LI) > Content-Length: 0 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura > > > 12 headers, 0 lines > Using latest request as basis request > Sending to : 5060 (non-NAT) > Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP :5060;branch=z9hG4bK-d54de2e6 > From: >;tag=4c7b1b149bb4b329o0 > To: > > Call-ID: 66a3a900-ec8d9e2d@ > CSeq: 1 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > > Content-Length: 0 > > > to :5060 > Transmitting (no NAT): > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP :5060;branch=z9hG4bK-d54de2e6 > From: >;tag=4c7b1b149bb4b329o0 > To: >;tag=as7ba88dca > Call-ID: 66a3a900-ec8d9e2d@ > CSeq: 1 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > > WWW-Authenticate: Digest realm="asterisk", nonce="7b426d2d" > Content-Length: 0 > > > to :5060 > Scheduling destruction of call '66a3a900-ec8d9e2d@ DIGIT IN NUMBER>' in 15000 ms > debian*CLI> > > Sip read: > REGISTER sip: SIP/2.0 > Via: SIP/2.0/UDP :5060;branch=z9hG4bK-2d99db8a > From: >;tag=4c7b1b149bb4b329o0 > To: > > Call-ID: 66a3a900-ec8d9e2d@ > CSeq: 2 REGISTER > Max-Forwards: 70 > Authorization: Digest > username="100",realm="asterisk",nonce="7b426d2d",uri="sip:",algorithm=MD5,response="b904 > 95eaf088d8696ac0cc5ebad9f990" > Contact: :5060>;expires=3600 > User-Agent: Linksys/RT31P2-3.1.3(LI) > Content-Length: 0 > Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER > Supported: x-sipura > > 13 headers, 0 lines > Using latest request as basis request > Sending to : 5060 (non-NAT) > Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP :5060;branch=z9hG4bK-2d99db8a > From: >;tag=4c7b1b149bb4b329o0 > To: > > Call-ID: 66a3a900-ec8d9e2d@ > CSeq: 2 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > > Content-Length: 0 > > > to :5060 > -- STOP --- > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update o
RE: [Asterisk-Users] Diva
Nope. At least I tried and never could get it working. It's a semiactive. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano GrandisSent: Friday, September 30, 2005 6:59 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Diva Hi all, just a question: can i use this kind of diva for asterisk? 00:14.0 Network controller: Eicon Networks Corporation Diva ISDN Pro 3.0 PCI Thanks all Giordano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Revieving some fax problems
I had some trouble going from bri>zaptel analog, but once I got the gain settings right, I would say it has worked well. Don't have stats, but any faxes I do send to it seem to go through. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Friday, September 30, 2005 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Revieving some fax problems Try Hylafax, with external fax/modem, it works 99.999% It you try to route it via Asterisk (with NVFaxDetect) your success will be about 95% -- #Joseph On Fri, 2005-09-30 at 15:57 -0400, Alexandre Leclerc wrote: > Hi, > > We are recieving some faxes, but I would say that about 50% of them do > not work. We don't know why... is it something with the faxes speed, > volume, etc? Should we use a real fax machine? > > Using a TDM13B with a rxgain of about 5.0... > > Thank you for any help. > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk + 99.999s was (Re: [Asterisk-Users] Asterisk on windows)
Patrick wrote: On Sat, 2005-10-01 at 08:31 -0400, Julio Arruda wrote: [snip] One thing interesting, coming from data background, seeing the requirements in carrier voice networks. Is a quite distinct ball-game. Devices that require 'hot-software-upgrades', still not that often seen in data. How is this being handled with Asterisk + other solutions ? Example, having a trunk gateway with a OC3 worth of TDM, is 'acceptable' that a sw upgrade will cut established calls ? Iirc Motorola has a solution that allows in-operation linux kernel upgrades. No idea how they pulled that magic off (and if it actually works). At VON IBM was going to demo a blade based Asterisk solution that has auto-failover of calls so maybe that could also be used to upgrade software. Don't have more info about this IBM solution. If you have a DS3 or OC3 worth of TDM calls then it probably makes sense to use a carrier-class box. Weird as it seems, not sure if the softswitch itself is the problem. Example, you could have a media gateway where the established calls are not torn down during a software upgrade of the Media Gateway controller 'entity'. The hardest part is the media gateway failover, I'm only familiar with Nortel (I work in Nortel) MG, and they in some cases would do these with APS and 1:1 sparing of the cards, where the sw migration is a 'hitless process', I assume others have similar options, but again, is not exactly 'in the asterisk' only, is in more than that, is in the 'solution'. Regards, Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Updated presentation of Asterisk 1.2
Please search for the word "collission" and replace with collision.. Damn! This 1.2 sure takes things to a different level. Good luck. - Original Message - From: "Olle E. Johansson" <[EMAIL PROTECTED]> To: "Users Asterisk" Sent: Saturday, October 01, 2005 4:32 PM Subject: [Asterisk-Users] Updated presentation of Asterisk 1.2 Friends, I have updated my Asterisk 1.2 presentation with the latest information. It is still available in the same place as before: http://www.astricon.net/asterisk1-2/ Please continue to test the beta of Asterisk 1.2, available at ftp.digium.com. We need all the feedback we can get. If you are a developer and have some time for community work, please check in with the bug tracker and help us resolve the unresolved bugs. See you at Astricon in Anaheim! /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP make outside call
David,
Shouldn't the [outgoing] be exten => 9.,1,Dial(ZAP/3 ... etc
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David H
Sent: 30 September 2005 17:52
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP make outside call
Hi,
I am can make local extension to and from SIP X-Lite
softphone, but I can't dial out using X-Lite but local
analog works just fine. Here are my conf files any
idea?
Thanks,
David
my sip.conf
[general]
bindport=5060 ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind
to (0.0.0.0 binds to all)
allow=all
[3000]
type=friend
allow=all
username=3000
secret=my_passwd
host=dynamic
context=sip
dtmfmode=rfc2833
my extension.conf
[globals]
davidHand=>Zap/1
davidVoicemail=>[EMAIL PROTECTED]
johnHand=>Zap/2
johnVoicemail=>[EMAIL PROTECTED]
davidout=>Zap/3
johnout=>Zap/4
[internal]
exten => 1000,1,Dial(${davidHand},10,r)
exten => 1000,n,Voicemail(u${davidVoicemail})
exten =>
1000,n,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten => 1000,n,Wait(1)
exten => 1000,n,Hangup()
exten => 1000,102,Voicemail(b${davidVoicemail})
exten => 1000,103,Hangup()
exten => 2000,1,Dial(${johnHand},10,r)
exten => 2000,n,Voicemail(u${johnVoicemail})
exten =>
2000,n,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten => 2000,n,Wait(1)
exten => 2000,n,Hangup()
exten => 2000,102,Voicemail(b${johnVoicemail})
exten => 2000,103,Hangup()
exten => 3000,1,Dial(SIP/3000,20,tr)
exten => 3000,n, Bye()
exten =>
i,1,Playback(/var/lib/asterisk/sounds/invalid)
exten => i,2,Goto(incoming,s,2)
exten =>
t,1,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten => t,2,Hangup()
[outgoing]
ignorepat => 9
exten => 9,1,Dial(Zap/3)
exten => 9,n,Congestion()
exten => 9,n,Hangup()
[voicemail]
exten => 2828,1,VoiceMailMain()
exten => 2828,n,Hangup()
[incoming]
exten => s,1,Answer()
exten =>
s,2,Background(/var/lib/asterisk/sounds/vm-enter-num-to-call)
include => internal
[sip]
include => internal
[default]
include => internal
include => outgoing
include => sip
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RE: [Asterisk-Users] Calls between SIP and IAX
Thank You for answer. As I try, the problem occurs when the call come to IAX channel in unknow format of codec. When the calls come in IAX channel with correct codec format (ulaw in my case) calls are O.K. Is it possible to set generally, that i’m using in all devices ulaw format (calls from H.323 trunk doesn’t set it correct). Thanks, Bob. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Saturday, October 01, 2005 7:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Calls between SIP and IAX asterisk console output and details about config files and networking are welcome, and i think, desirable. best regards On 10/1/05, Bohuslav Coufal <[EMAIL PROTECTED]> wrote: Hi all, I have a trouble when I try to configure asterisk to make calls between IAX and SIP. IAX I'm using to connect between asterisks a on SIP I have phones. The calls come from higher asterisk to my on IAX, SIP phone is ringing and when I hang up then dial command ends and connection is loss. When I'll make connection between asterisks on SIP then all work fine. Does anybody has any suggestions? Bob. P.S . - I'm using asterisk 1.0.9 on FC3. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org" ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] C Manager Interface Client
On Sat, Oct 01, 2005 at 12:17:38PM -0500, Tressler, Joshua A wrote: > When we pull up a telnet session beside this code, the telnet sessions > shows everything in blocks together. We insert lines between socket > reads, therefore we see > > Event: * > > Privilege: ** > > Instead of > > Event: * > Privilege: ** > > Below is the code that we have. We are getting ready to run a sniffer > and see if/why asterisk is doing the writes separately instead of in one > chunk. > > Joshua > > === What is that code you attached? Certainly not something that compiles. Not even a main() in there. Try adding: #include #include #include #include #include #include #include #include And then put the code in main() or otherwise in functions. Next step is to make it pass cc -Wall . After I did that, your code worked fine here. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 subject)
Jonathan k. Creasy wrote: 0930155701|cfg |3|00|0004f2022609.cfg could not be downloaded, getting next file. Any ideas? I attached the config files, I got them from somewhere else. The phone isn't finding the config file as the above log entry shows. The config file consists of the mac address of the phone with a .cfg appended. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 and RTP streams
Sherwood, I have never known the RTP audio to be on only one port in sip. I believe it's always on 2. The one way audio is always a nat/firewall problem in sip. Sherwood McGowan wrote: Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could. A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer will get one-way audio. I've read that this is commonly caused by the outgoing RTP port not being the same as the incoming RTP port. A lot of other devices (I found info on forcing Xten to do it) can be forced to use the same port for both, but these devices don't have an option (that I've been able to find, even in the provisioning configs) to do this. So, my question is two-fold: 1. Can Asterisk be told to send the RTP stream for incoming and outgoing always on the same set of ports? 2. Does anyone know something that I'm missing for the above mentioned devices? They're all the 2 line version of the ATA and/or router configs (wireless and wired) Thank you all in advance for your thoughts and comments. I apologize in advance if I missed something that was publicly available. Sherwood McGowan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adding Cepstral to Asterisk
I downloaded Cepstral to my Asterisk Box. I did the install and let it install to /opt/swift. I brought down a new CVS-HEAD as of today 10/1. I added APPS+=app_cepstral.so into the Makefile in /usr/src/asterisk/apps/Makefile Like: # Obsolete things... # #APPS+=app_sql_postgres.so #APPS+=app_sql_odbc.so APPS+=app_cepstral.so # I did this piece but wasn't sure exactly what part of the Makefile I was to add it in so I added it in here: Towards the top of the file where it talks obsolete programs are commented out. And then after the section that compiles voicemail add: app_cepstral.so: app_cepstral.c $(CC) -D_GNU_SOURCE -shared -Xlinker -x -o $@ $< -lz -lm -lswift -lceplex_us -lceplang_en -lz -ldl -L/opt/swift/lib -I/opt/swift/include Make sure the $(CC) line starts with a tab, not spaces. I didn't see a lot about voicemail: app_sql_odbc.so: app_sql_odbc.o $(CC) $(SOLINK) -o $@ $< -lodbc app_cepstral.so: app_cepstral.c $(CC) -D_GNU_SOURCE -shared -Xlinker -x -o $@ $< -lz -lm -lswift -lceplex_us -lceplang_en -lz -ldl -L/opt/swift/lib -I/opt/swift/include look: look.c $(CC) -pipe -O6 -g look.c -o look -lncurses I checked the /etc/ld.so.conf file for a line like: opt/swift/lib in the file. It wasn't there so I added it: include ld.so.conf.d/*.conf /opt/swift/lib I ran ldconfig when I was done. I can't see that Cepstral was added into Asterisk and I was wondering what I have done wrong that it doesn't work. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 subject)
My Polycom IP301 hangs on "Processing Cfg..." Here is the boot log: 0930155446|so |4|00|-- Initial log entry -- 0930155446|so |4|00|+++ Note that bootrom log times are in GMT +++ 0930155446|wdog |4|00|Initial log entry 0930155446|cfg |4|00|Initial log entry 0930155446|copy |4|00|Initial log entry 0930155446|cdp |4|00|Initial log entry 0930155446|cdp |5|00|CDP is DISABLED. 0930155446|cdp |5|00|802.1Q/VLAN tagging is DISABLED. 0930155446|so |3|00|Platform: Model=SoundPoint IP 301, Assembly=2345-11300-010 Rev=A 0930155446|so |3|00|Platform: Board=2345-11300-010 A 0930155446|so |3|00|Platform: MAC=0004f2022609, IP=Unknown, Subnet Mask=Unknown 0930155446|so |3|00|Platform: BootBlock=2.5.0 (11300_010) 06-Nov-04 08:07 0930155446|so |3|00|Application, main: Label=BOOT, Version=2.6.1.0003 04-Dec-04 14:38 0930155446|so |3|00|Application, main: P/N=3150-11069-261 0930155446|app1 |4|00|Initial log entry. 0930155447|so |3|00|Link status is Net up, PC down. 0930155455|app1 |3|00|Using resolver server 192.168.222.4 and domain local. 0930155455|app1 |3|00|DHCP returned result 0x287 from server 192.168.222.4. 0930155455|app1 |3|00| Phone IP address is 192.168.222.202. 0930155455|app1 |3|00| Subnet mask is 255.255.255.0. 0930155455|app1 |3|00| Gateway address is 192.168.222.1. 0930155455|app1 |3|00| DNS server is 192.168.222.4. 0930155455|app1 |3|00| DNS domain is local. 0930155455|app1 |3|00|Bootline: eim(0,0)bootHost:flash 0930155455|e=192.168.222.202:ff00:1c20:433d5fcf h=216.135.65.62 0930155455|g=192.168.222.1 u=jonathan pw=tenn1982 0930155455|app1 |3|00|Bootline: f=0x40 tn=CircaIP 0930155700|app1 |3|00|Time has been set from pool.ntp.org(193.170.141.4). 0930155700|cfg |3|00|Image bootrom.ld has not changed. 0930155701|cfg |3|00|0004f2022609.cfg could not be downloaded, getting next file. 0930155704|cfg |3|00|Image sip.ld has not changed. 0930155734|app1 |4|00|Loaded application sip.ld successfully, errors 0x0. 0930155734|app1 |6|00|Uploading boot log, time is FRI SEP 30 15:57:34 0930155734|2005 Any ideas? I attached the config files, I got them from somewhere else. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP make outside call
I'm going to dip my toe in the water to help out here, although I'm just a newbie... It looks to me as though your x-lite is coming in and being assigned the "sip" context, which includes just the right to call internal destinations. (Bizarrely?) Your default context seems to allow everything (and double-includes internal!) You could change extensions.conf to this, but it's still somewhat odd... [sip] include => internal include => outgoing [default] include => internal include => outgoing Best of luck, Alex ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] OOH323C
HI I got the following error during compiling ooh323. In function `h323_set_rtp_peer':chan_h323.c:2745: error: structure has no member named `tech_pvt'chan_h323.c: At top level:chan_h323.c:68: error: storage size of `h323_tech' isn't knownmake[2]: *** [chan_h323.lo] Error 1 make[2]: Leaving directory `/root/asterisk-ooh323c-0.2/src'make[1]: *** [all-recursive] Error 1make[1]: Leaving directory `/root/asterisk-ooh323c-0.2'make: *** [all] Error 2 What is the reason ? Mario On 9/30/05, Dan Austin <[EMAIL PROTECTED]> wrote: Asking which H323 channel is the best turns out to be a deeplypersonal issue, at least noting the responses in the past. I've tried and used all three. Here are my thoughts-Chan_h323 (the original)-Did not work in our environment. Known issues with Cisco'sCall Manager. Other than the requirements for OpenH323 and PWLib, it was easy to setup and configure.Chan_oh323Worked fine for us. Has the same dependencies as chan_h323,also easy to setup and configure.Chan_h323 (ooh323c based)This one has been a winner for us. No dependencies on OpenH323 or PWLib, which while not terrible to build/setup, is extra effortand can be tricky to match known working versions.Setup and configuration has been very simple. If you have configuredthe other channels, this one should seem familiar. A seperate note in favor of the new chan_h323 is the developer support.I found a couple little bugs that related to our use of Cisco CallManager, and expected little or no interest in getting them resolved. I had a test version made available to me in just over a day andcomplete resolution a few hours later.Dan-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of KanishkaSomaratneSent: Thursday, September 29, 2005 7:28 AM To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] OOH323Chihas any one used OOH323C i tried this it is installed but do not knowhow to configure has any one used this, what is the best h323 addon to use withasterisk___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best way to create IVR/voicemail system
Hello I want to setup a system where people can dial a number and then a system will ask them questions for which they will leave answers. Eg something like this: Answer Playback(whatisyournamemsg) Record(yourname:gsm) Playback(whatisyourheight) Record(yourheight:gsm) Playback(thankyou) Hangup Is this the best way to do this sort of thing? Do users then just access the responses by eg *98 - or does this work a little differently to voicemail? How do we retrieve the responses? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Any way to not overwrite sound files on compile?
Couldn't you do something like say that the files localized for your particular installation are in another language? Just do something like language=enmycompanyname, copy all the standard asterisk files to your local installation's equivalent of /var/lib/asterisk/sounds/enmycompnayname, et voila! No more overwriting of your files, and you get fallback behaviour to the standard asterisk sounds as well when new, nonlocalized sounds are included in the distribution. Just replacing the original files in their original locations seems incorrect; there's no reason to support that behaviour in the standard installation. - James Moore ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] chan_zap.so ?
On reboot I must run: modprobe wctdm _twice_ (the first time provides error about not finding hardware) before ztcfg -vv can run properly, setting up the zap hardware right where asterisk wants it.. On you system the command of course would be: modprobe wcfxo I hope this is your problem :) So try: modprobe wcfxo;modprobe wcfxo && ztcfg -vv && asterisk;asterisk -r Andrew Kohlsmith wrote: On Friday 30 September 2005 10:26, cyril SIMON wrote: I've a little problem with my asterisk server. Yes, it is a little problem. Today, I wanted to restart it and when I did it, my asterisk server didn't want to start again. It's telling you the problem pretty damn clearly: Sep 30 16:51:53 WARNING[2343]: chan_zap.c:816 zt_open: Unable to specify channel 1: No such device or address It can't find the zaptel hardware. There are a few causes for this: 1) You took the hardware out 2) You changed the configuration of the hardware 3) You didn't load the drivers for the hardware I really do not know that can be the problem. Please review the potential causes above and report back. I'm curious as to what the actual problem was. We can only improve our ability to help if you provide good feedback. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo <[EMAIL PROTECTED]> Office Manger, Horan & Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Any way to not overwrite sound files on compile?
Couldn't you do something like say that the files localized for your particular installation are in another language? Just do something like language=enmycompanyname, copy all the standard asterisk files to your local installation's equivalent of /var/lib/asterisk/sounds/enmycompnayname, et voila! No more overwriting of your files, and you get fallback behaviour to the standard asterisk sounds as well when new, nonlocalized sounds are included in the distribution. Just replacing the original files in their original locations seems incorrect to me. You're essentially creating a new branch without building any infrastructure to support that branch. - James Moore (Apologies if you see this twice; tried to mail from another account and it didn't seem to go to the list) __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Required hardware
Hi all! We have to setup 2 *servers. Now I am interested in possible capacity. Server 1. Should be used for getting traffic from our Telco using IAX and send it out using SIP. No transcoding, ulaw both ways. What is possible capacity on 1 server using required hardware? Server 2 will pick uo traffic from 1 Tollfree number sent to us by IAX with ulaw codec. Must be transcoded to G729 and sent out as SIP. What is possible capacity on 1 server using required hardware? Regards Anders Svensson CTO BoBas Communication Glimminge 2045 S-280 60 Broby Sweden Phone: +46 (0)40 608 22 50 Cell: +46 (0)703 17 13 06 MSN: [EMAIL PROTECTED] Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 and RTP streams (just bumping)
Sherwood McGowan wrote: > Bumping, just in case it got lost in the shuffle today... I think this is an > important thing to be able to do. > > Subject: [Asterisk-Users] Asterisk and RTP streams > > Guys, I've been poking around trying to find a good answer for this via > voip-info, google, etc... Haven't found anything that helps, so maybe you > mates could. > > A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using > Sipura SPA-2002s. Every once in a while, the customer will get one-way > audio. I've read that this is commonly caused by the outgoing RTP port not > being the same as the incoming RTP port. A lot of other devices (I found > info on forcing Xten to do it) can be forced to use the same port for both, > but these devices don't have an option (that I've been able to find, even in > the provisioning configs) to do this. So, my question is two-fold: > > 1. Can Asterisk be told to send the RTP stream for incoming and outgoing > always on the same set of ports? > 2. Does anyone know something that I'm missing for the above mentioned > devices? They're all the 2 line version of the ATA and/or router configs > (wireless and wired) > If you turn on nat=yes this will affect both SIP and RTP. Asterisk will then send to the same address as we receive RTP from, this is called symmetric RTP. THere's no way we can affect the address port range that the device tell us to send to, but we can ignore that in the case there is a NAT in between and send to whatever address the device sends audio from. The RTP port address we receive RTP on *from* the device is settable in rtp.conf. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Callcenter and Softphone hanging
Hi,
I run a small inbound callcenter with 3 agents doing techsupport. The
agents are logged in via softphone, using agentcallback login. Some
times the agents PC running softphone hangs, and they reboot the PC. But
* is not aware of this and tries to send calls to the PC, which gets
rejected.
-- outgoing agentcall, to agent '1009', on 'Local/[EMAIL PROTECTED],1'
-- Executing Dial("Local/[EMAIL PROTECTED],2", "SIP/1002") in new
stack
Oct 1 23:16:51 NOTICE[16907]: app_dial.c:777 dial_exec: Unable to
create channel of type 'SIP'
== Everyone is busy/congested at this time
-- Called Agent/1009
-- Timeout on Local/[EMAIL PROTECTED],2
== CDR updated on Local/[EMAIL PROTECTED],2
-- Executing Hangup("Local/[EMAIL PROTECTED],2", "") in new stack
Is there any way to logoff an agent from the queue in such cases from
the * prompt? Any better way to handle this issue?
raj
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[Asterisk-Users] Remote call pick-up
Hi, Does anyone have remote call pick-up working on * (either via SIP or otherwise)? If so then can you post your features.conf, sip.conf and/or zapata.conf? We can't seem to get this (seemingly simple) function to work. Cheers, Damian. -- FFF Managed Technology Ltd 60 Cook St P.O. 6368 Wellesley St Auckland t +64 9 356 2911 f +64 9 358 9070 m +64 21 415 297 w www.fff.co.nz ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] C Manager Interface Client
When we pull up a telnet session beside this code, the telnet sessions
shows everything in blocks together. We insert lines between socket
reads, therefore we see
Event: *
Privilege: **
Instead of
Event: *
Privilege: **
Below is the code that we have. We are getting ready to run a sniffer
and see if/why asterisk is doing the writes separately instead of in one
chunk.
Joshua
===
char * echo = "localhost";
int port = 5038;
int readCode = 0;
struct sockaddr_in serverAddr, cliAddr;
struct hostent *server;
socketHandle = socket(AF_INET,SOCK_STREAM,0);
if(socketHandle < 0)
{
printf("ERROR OPENING SOCKET\n");
exit(0);
}
server = gethostbyname(echo);
if(server == NULL)
{
printf("ERROR NO SUCH HOST\n");
exit(0);
}
bzero((char*) &serverAddr, sizeof(serverAddr));
serverAddr.sin_family = AF_INET;
bcopy((char*)server->h_addr, (char*)&serverAddr.sin_addr.s_addr,
server->h_length);
serverAddr.sin_port = htons(port);
if(connect(socketHandle,(struct sockaddr
*)&serverAddr,sizeof(serverAddr)) < 0)
{
printf("ERROR CONNECTING\n");
exit(0);
}
readCode = read(socketHandle,buffer,sizeof(buffer));
if(readCode < 0)
{
printf("ERROR READING FROM SOCKET\n");
exit(0);
}
bzero(buffer,sizeof(buffer));
strcpy(buffer,"Action: Login\r\nUsername: username\r\nSecret:
secret\r\nEvents: on\r\n\r\n");
readCode = write(socketHandle,buffer,strlen(buffer));
if(readCode < 0)
{
printf("ERROR WRITING TO SOCKET\n");
exit(0);
}
while(1)
{
bzero(buffer,sizeof(buffer));
readCode = read(socketHandle,buffer,sizeof(buffer));
if(readCode < 0)
{
printf("ERROR READING FROM SOCKET\n");
exit(0);
}
printf("%s",buffer);
}
===
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Saturday, October 01, 2005 10:19 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] C Manager Interface Client
On Fri, Sep 30, 2005 at 11:51:48AM -0500, Tressler, Joshua A wrote:
> List:
>
> This is my first manager client that I've written so please bear with
me:
>
> I am trying to write a C manager interface client to interface with
our CRM software. I am having an issue while reading the data from the
manager interface.
>
> I am writing this in C and I have the following code:
> while(1)
> {
> bzero(buffer,sizeof(buffer));
> readCode = read(socketHandle,buffer,sizeof(buffer));
> if(readCode < 0)
> {
> printf("ERROR READING FROM SOCKET\n");
> exit(0);
> }
> printf("%s",buffer);
> }
This is just the main loop, right? There has to be a login before that.
Could you please post your full code? (that is: a minimal version of it
that you verified to still be problematic).
>
> This prints out everything just as connecting to the telnet session
would print it out (I do the logging in elsewhere, that isn't the
problem here)
>
> This code will read until * has nothing else for me to read from it
> then print it all out and wait for some more stuff. Since * seems to
> print out on 5038 in "blocks" of text read(...) will never cut off in
> the middle of a block.
Using a C-based program to debug that is not very helpful. telnet
localhost 5038 and see what happens in real-time.
A sniffer could also help.
> However, on one instance, and this is the only one we can reproduce
the
> results on, * puts out Event: ** then stops, Privilege: ,
> then stops and then prints out the rest. This really screws up my
> parsing as i normally parse using a tokenizer on \r\n\r\n and pass
each
> block off to a parsing method. I found this problem using the
following code:
> while(1)
> {
> bzero(buffer,sizeof(buffer));
> readCode = read(socketHandle,buffer,sizeof(buffer));
> if(readCode < 0)
> {
> printf("ERROR READING FROM SOCKET\n");
> exit(0);
> }
> printf("%s\n",buffer); This is the main difference \n
> }
> In this case I get output as follows:
> ...
> ...
> .
> .
> Event: Hangup
>
> Privilege: call,all
>
> Channel: SIP/1542200-543f
> Uniqueid: 1128041150.26
> Cause: 0
> Cause-txt: Unknown
> ...
> ...
> .
> .
> In this case "Event: Hangup", "Privilege: call, all", and the rest all
get passed off to my parser. Obviously a problem.
>
> Is * spitting this data out to me in three seperate chunks or is my
socket not blocking correctly?
>
> Any suggestions as to why this would happen?
>
> TIA,
>
> Joshua
> ___
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>
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--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EM
[Asterisk-Users] Swap between callers
Hello On business phones it is often possible to have call waiting (think that is the feature) whereby if you are talking to a caller you can see another caller has called and you can swap between callers. For example, to say hello, I am on call with someone else now can I call you back. How can this be implemented using SIP IP phones. Do you need to setup two or more lines? How is it done? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Calls between SIP and IAX
asterisk console output and details about config files and networking are welcome, and i think, desirable. best regardsOn 10/1/05, Bohuslav Coufal <[EMAIL PROTECTED]> wrote: Hi all,I have a trouble when I try to configure asterisk to make calls betweenIAX and SIP. IAX I'm using to connect between asterisks a on SIP I havephones. The calls come from higher asterisk to my on IAX, SIP phone is ringing and when I hang up then dial command ends and connection isloss.When I'll make connection between asterisks on SIP then all work fine.Does anybody has any suggestions?Bob.P.S . - I'm using asterisk 1.0.9 on FC3.___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org" ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 - HEAD , SpanDSP, app_rx/tx Fax..what versions do you have?
[EMAIL PROTECTED] wrote: However, asterisk complains that there is unused symbols when running /usr/sbin/asterisk -vvvgc I've been bitten by this before. You've installed a newer version of SpanDSP over an older version. Remove the spandsp libraries in the /usr/local/lib folder and re-install SpanDSP. Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Faxdetection in IAX? (Missing audio samples)
Hi, please don't bother me continuing trying to fax, even if I've got convinced, that it generally won't work! I've found a strange behaviour, when sending from IAXCLIENT1 -> asterisk -> IAXCLIENT2 or from IAXCLIENT1 -> asterisk -> SIPCLIENT2 When IAXCLIENT1 is sending an absolutely constant frequence (fax detection tones at the very beginning of every fax session), audio samples get cleared (set to 0). I dumped both, the outgoing samples from IAXCLIENT1 and incoming from IAXCLIENT2, respectively heard by myself with a SIP phone. IAXCLIENT1 sending a constant tone of exactly 3000 msec, which leads in the dump file to a very periodic pattern of hex numbers, and thus is easy to find "manually" in the dump file. Arriving at IAXCLIENT2 respectively at the SIP phone is that tone for exactly 500 msec, afterwords silence for approx 2450 msec. Is there an explanation for this behaviour? Thanks for any hints! Roger. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 - HEAD , SpanDSP, app_rx/tx Fax..what versions do you have?
> On Sat, Oct 01, 2005 at 08:32:34AM -0700, [EMAIL PROTECTED] wrote: >> I'm trying to put together a package of asterisk-head, spandsp, and >> app_rx,tx fax. >> >> I can get everything to compile: >> >> spandsp-0.0.2pre20 >> asterisk-head (cvs co -r HEAD asterisk) >> the app_rx/tx from soft-switch.org in the 1.1 folder >> >> However, asterisk complains that there is unused symbols when running >> /usr/sbin/asterisk -vvvgc > > In which module? Are you sure it is not a left-over? (check dates, or > book-keeping oof you package-management system, if you use one) it's in the module app_rxfax.so, and if i comment out that one in /etc/asterisk/modules.conf then it will compain about app_txfax.so. ARGH anyways... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Empty ACK
Ronald Voermans wrote: > Hello, > > I have asterisk connected to SER/RTPProxy which is again connected to a > IP-PSTN gateway. When calling with a UA, registered at * to a SIP phone > connected to the IP-PSTN gateway, I get 'empty ACKs': > > U 192.168.0.173:5060 -> 10.254.254.1:5060 ACK SIP/2.0. > Via: SIP/2.0/UDP 192.168.0.173:5060;branch=z9hG4bK5cb7d048. > Route: ,. > From: "0161801019" ;tag=as628d39c1. > To: ;tag=00-04094-52dc5953-7c1293c27. > Contact: . > Call-ID: [EMAIL PROTECTED] > CSeq: 103 ACK. > User-Agent: Asterisk PBX. > Content-Length: 0. > > As you can see, there is no URI after the ACK statement, and SER doesn't > know what to do with it. Is this a bug in *, or is this normal? > It certainly looks odd. But you have to give us more information. Which version of Asterisk? Please also include a full SIP debug with debug level 4, verbose level 4 of the whole transaction, from the first INVITE to this weird ack. Add that to a bug report in the bug tracker, and we'll take a look at it. We should not send packets like this. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Compiling Zaptel on EM64T machine
On Sun, Oct 02, 2005 at 01:39:58AM +1000, Broadband Solutions wrote: > Hi Guys > > Im trying to complile Asterisk on my new dual Xeon 3.0ghz EM64T machine. > > Im running Debian 3.1, and have installed the 2.6.8-11-em64t-p4-smp > kernel (and headers). The system is working fine and is detecting > both CPU's (4 actually, with hyperthreading). > > But when I try to compile Zaptel, Im getting this error: > > /lib/modules/2.6.8-11-em64t-p4-smp /build This extra "-" in the end shouldn't be there. Are you using the right kernel-headers? Is it your own kernel with a modified version string? > make -C /lib/modules/2.6.8-11-em64t-p4-smp /build SUBDIRS=/usr/src/zaptel > modules > make[1]: Entering directory `/usr/src/kernel-headers-2.6.8-11- em64t-p4-smp' > CC [M] /usr/src/zaptel/zaptel.o > cc1: error: code model `kernel' not supported in the 32 bit mode > make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 > make[1]: *** [_module_/usr/src/zaptel] Error 2 > make[1]: Leaving directory `/usr/src/kernel-headers-2.6.8-11- em64t-p4-smp' > make: *** [linux26] Error 2 > > Can anyone help me out? What does "code model `kernel' not supported in the > 32 bit mode" mean, and how could I get around it? How about: apt-get install zaptel-source m-a a-i zaptel If you really need up-to-date zaptel debs, they're in experimental (or will soon be), and also in http://rapid.dotsrc.org/experimental/ -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling Zaptel on EM64T machine
Hi GuysIm trying to complile Asterisk on my new dual Xeon 3.0ghz EM64T machine.Im running Debian 3.1, and have installed the 2.6.8-11-em64t-p4-smp kernel (and headers). The system is working fine and is detecting both CPU's (4 actually, with hyperthreading).But when I try to compile Zaptel, Im getting this error:/lib/modules/2.6.8-11-em64t-p4-smp /buildmake -C /lib/modules/2.6.8-11-em64t-p4-smp /build SUBDIRS=/usr/src/zaptel modulesmake[1]: Entering directory `/usr/src/kernel-headers-2.6.8-11- em64t-p4-smp'CC [M] /usr/src/zaptel/zaptel.occ1: error: code model `kernel' not supported in the 32 bit modemake[2]: *** [/usr/src/zaptel/zaptel.o] Error 1make[1]: *** [_module_/usr/src/zaptel] Error 2make[1]: Leaving directory `/usr/src/kernel-headers-2.6.8-11- em64t-p4-smp'make: *** [linux26] Error 2Can anyone help me out? What does "code model `kernel' not supported in the 32 bit mode" mean, and how could I get around it? Thanks, Brad ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 - HEAD , SpanDSP, app_rx/tx Fax..what versions do you have?
On Sat, Oct 01, 2005 at 08:32:34AM -0700, [EMAIL PROTECTED] wrote: > I'm trying to put together a package of asterisk-head, spandsp, and > app_rx,tx fax. > > I can get everything to compile: > > spandsp-0.0.2pre20 > asterisk-head (cvs co -r HEAD asterisk) > the app_rx/tx from soft-switch.org in the 1.1 folder > > However, asterisk complains that there is unused symbols when running > /usr/sbin/asterisk -vvvgc In which module? Are you sure it is not a left-over? (check dates, or book-keeping oof you package-management system, if you use one) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] chan_cap-cm-0.6 deflect support
On Fri, 30 Sep 2005, Louis-David Mitterrand wrote: > On Fri, Sep 30, 2005 at 01:11:19PM +0200, Armin Schindler wrote: > > > Also, is there a way to detect that a SIP phone has an active forward > > > number and capi-deflect any incoming calls to that number? > > > > If you can retrieve this information from extensions.conf, then you can use > > my example above. > > > > Anyway, I noticed that the original implementation of deflect specified in > > capi.conf does not work in all cases. > > I plan to remove that and to allow capicommand(deflect|...) only. > > It's not necessary to do that in capi.conf and using different MSNs is > > difficult too. > > My idea is provide information about 'this is a call-waiting call, no > > b-channel' to extensions.conf via a variable. And the user then can decide > > what to do with that call using all features of the dialplan. > > I plan to do this for version 0.6.1. > > Yes, that would be perfect! Looking forward to that implementation. I have added this (and removed the deflect entry from capi.conf) to CVS. In addition, to make chan_capi find a free interface if all B-channels are used by Asterisk, I added a pseudo channel to each interface. This additional channel is used if it is an incoming call and the variable BCHANNELINFO is '2' only. Now you can check for BCHANNELINFO=2 to find out this is an call-waiting call and start capicommand(deflect|), if you want activate call deflection on that call. I have not fully tested it, but it should be possible to hangup/hold another call and accept the call-waiting call. Maybe you want to test it :-) Armin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 - HEAD , SpanDSP, app_rx/tx Fax..what versions do you have?
I'm trying to put together a package of asterisk-head, spandsp, and app_rx,tx fax. I can get everything to compile: spandsp-0.0.2pre20 asterisk-head (cvs co -r HEAD asterisk) the app_rx/tx from soft-switch.org in the 1.1 folder However, asterisk complains that there is unused symbols when running /usr/sbin/asterisk -vvvgc ARGH.. Does someone have a package with files that I could try? I would greatly appreciate it. Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_zap.c: Ring/Off-hook in strange state 6 on channel 1
Asterisk version: 1.2.0-beta1 -OR- CVS HEAD Hardware: Generic X100P clone connected to Panasonic KX-TA624-5 extension port The problem happens IF AND ONLY IF: - the Panasonic is set to use double rings - the X100P is set to answer immediately (I'm using DISA here) It does not happen consistently; sometimes Asterisk behaves normally. When the "strange state" happens, DISA usually won't respond to dialing. I've tried it with busy and call progress detection on and off, and it makes no difference. I have two workarounds, either of which will work: - Insert a one-second wait before answering with DISA - Set the Panasonic to send single rings instead of double It seems that what's going on here is that the FXO port picks up the line after the first half of the double ring, just as the second half comes in, and Asterisk gets confused because the ring hasn't been fully tripped. I don't yet understand enough about the guts of chan_zap.c to propose a fix, though. On an unrelated note, I've found that the clone boards need to have both Tx and Rx gain set to -1.5 to prevent echo problems on a Panasonic extension port. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] C Manager Interface Client
On Fri, Sep 30, 2005 at 11:51:48AM -0500, Tressler, Joshua A wrote:
> List:
>
> This is my first manager client that I've written so please bear with me:
>
> I am trying to write a C manager interface client to interface with our CRM
> software. I am having an issue while reading the data from the manager
> interface.
>
> I am writing this in C and I have the following code:
> while(1)
> {
> bzero(buffer,sizeof(buffer));
> readCode = read(socketHandle,buffer,sizeof(buffer));
> if(readCode < 0)
> {
> printf("ERROR READING FROM SOCKET\n");
> exit(0);
> }
> printf("%s",buffer);
> }
This is just the main loop, right? There has to be a login before that.
Could you please post your full code? (that is: a minimal version of it
that you verified to still be problematic).
>
> This prints out everything just as connecting to the telnet session would
> print it out (I do the logging in elsewhere, that isn't the problem here)
>
> This code will read until * has nothing else for me to read from it
> then print it all out and wait for some more stuff. Since * seems to
> print out on 5038 in "blocks" of text read(...) will never cut off in
> the middle of a block.
Using a C-based program to debug that is not very helpful. telnet
localhost 5038 and see what happens in real-time.
A sniffer could also help.
> However, on one instance, and this is the only one we can reproduce the
> results on, * puts out Event: ** then stops, Privilege: ,
> then stops and then prints out the rest. This really screws up my
> parsing as i normally parse using a tokenizer on \r\n\r\n and pass each
> block off to a parsing method. I found this problem using the following code:
> while(1)
> {
> bzero(buffer,sizeof(buffer));
> readCode = read(socketHandle,buffer,sizeof(buffer));
> if(readCode < 0)
> {
> printf("ERROR READING FROM SOCKET\n");
> exit(0);
> }
> printf("%s\n",buffer); This is the main difference \n
> }
> In this case I get output as follows:
> ...
> ...
> .
> .
> Event: Hangup
>
> Privilege: call,all
>
> Channel: SIP/1542200-543f
> Uniqueid: 1128041150.26
> Cause: 0
> Cause-txt: Unknown
> ...
> ...
> .
> .
> In this case "Event: Hangup", "Privilege: call, all", and the rest all get
> passed off to my parser. Obviously a problem.
>
> Is * spitting this data out to me in three seperate chunks or is my socket
> not blocking correctly?
>
> Any suggestions as to why this would happen?
>
> TIA,
>
> Joshua
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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RE: [Asterisk-Users] C Manager Interface Client
I just had a thought. Is this something
that I should post to the Asterisk-Dev list? I didn’t want to cross post,
but I’m not sure to which list I should have originally posted it.
Thanks,
Joshua
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tressler, Joshua A
Sent: Friday, September 30, 2005
11:52 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] C
Manager Interface Client
List:
This is my first manager client that I've
written so please bear with me:
I am trying to write a C manager interface client to
interface with our CRM software. I am having an issue while reading the data
from the manager interface.
I am writing this in C and I have the
following code:
while(1)
{
bzero(buffer,sizeof(buffer));
readCode = read(socketHandle,buffer,sizeof(buffer));
if(readCode < 0)
{
printf("ERROR READING FROM SOCKET\n");
exit(0);
}
printf("%s",buffer);
}
This prints out everything just as connecting to the telnet session would print
it out (I do the logging in elsewhere, that isn't the problem here)
This code will read until * has nothing else for me to read from it then print
it all out and wait for some more stuff. Since * seems to print out on
5038 in "blocks" of text read(...) will never cut off in the middle
of a block.
However, on one instance, and this is the
only one we can reproduce the results on, * puts out Event: ** then stops,
Privilege: , then stops and then prints out the rest. This really
screws up my parsing as i normally parse using a tokenizer on \r\n\r\n and pass
each block off to a parsing method. I found this problem using the
following code:
while(1)
{
bzero(buffer,sizeof(buffer));
readCode = read(socketHandle,buffer,sizeof(buffer));
if(readCode < 0)
{
printf("ERROR READING FROM SOCKET\n");
exit(0);
}
printf("%s\n",buffer);
This is the main difference \n
}
In this case I get output as follows:
...
...
.
.
Event: Hangup
Privilege: call,all
Channel: SIP/1542200-543f
Uniqueid: 1128041150.26
Cause: 0
Cause-txt: Unknown
...
...
.
.
In this case "Event: Hangup",
"Privilege: call, all", and the rest all get passed off to my
parser. Obviously a problem.
Is * spitting this data out to me in three seperate chunks or is my socket not
blocking correctly?
Any suggestions as to why this would happen?
TIA,
Joshua
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Re: [Asterisk-Users] error on loading zaptel module
On Sat, Oct 01, 2005 at 05:06:13PM +0330, Paradise Dove wrote: > i get this error on dmesg: > > zaptel: Unknown symbol __stack_smash_handler > zaptel: Unknown symbol __guard Seems you built zaptel with different kernel headers/config than the one you're currently running. Care to give more details? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] cisco phones problems
Whatever you have the voice vlan set it is what they operate on. You cannot provision that on the phone manually. If they are small switches (35xx, etc), then you need to configure without .1q trunking as those switch imply it automatically. For the larger switches 1.q trunking in the config is required for phones to properly operate on dhcp and the pcs attached to function properly. > On 14:51, Fri 30 Sep 05, Edwin Lam wrote: > > after much struggles. i've found out that if i ping the phone unit > > from another computer constantly (couple pings every 5-10 sec) > > the phone will operate fine. once i stopped the pings, the UNREACHABLE > > message started to pop up and the drop calls problems starts. seems > > like it's the firmware issue. does anyone uses Cisco SIP 7.3 (or 6.0, > > i've tried downgraded it at some point) and have similar problems? > > > > p.s. another piece of info: the phone units are set to a non default > > vlan manually since we share the physical lan for both data & voice. > > > Hi, > > I had the same problem with only 1 Cisco 7905 every once in > a while. All problems were solved as soon as I reverted the > phones to SCCP and started using chan_sccp.so > There's no lag anymore between the phones and asterisk. > > So maybe this is an extra reason to suspect the firmware ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] cisco phones problems
This sounds like you have CDP disable. Cisco CDP is taken care of the ability to see the phones. You cannot disable that if you use a Cisco switch, but you need to stop CDP on ports that going outside your network. It works like keepalive. Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Friday, September 30, 2005 6:35 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] cisco phones problems On 14:51, Fri 30 Sep 05, Edwin Lam wrote: > after much struggles. i've found out that if i ping the phone unit > from another computer constantly (couple pings every 5-10 sec) > the phone will operate fine. once i stopped the pings, the UNREACHABLE > message started to pop up and the drop calls problems starts. seems > like it's the firmware issue. does anyone uses Cisco SIP 7.3 (or 6.0, > i've tried downgraded it at some point) and have similar problems? > > p.s. another piece of info: the phone units are set to a non default > vlan manually since we share the physical lan for both data & voice. Hi, I had the same problem with only 1 Cisco 7905 every once in a while. All problems were solved as soon as I reverted the phones to SCCP and started using chan_sccp.so There's no lag anymore between the phones and asterisk. So maybe this is an extra reason to suspect the firmware -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?" ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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-oh323-0.6.7
there is a problem with oh323 and incomming calls . the problem is at https://skylab.inaccessnetworks.com/mantis/view.php?id=15 there is a patch to solve this issue, has any one used the patch with oh323-0.6.7 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 can I tranfer a call form one SIP phone to other during the call (unattended transfer)
Hi all. I have both t and T options in dial command. SIP phones configured with canreinvite=no and when I press #1 (as I have in features.conf) during call there is nothing to happened. Thanks for any suggestions. Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 and RTP streams (just bumping)
Thanks for the heads up. I guess BOTH devices then have to be configured to use the same port for outgoing as the incoming came in on. I know some UA's have this option, I'll just have to figure it out. Cheers, Sherwood ->-Original Message- ->From: [EMAIL PROTECTED] ->[mailto:[EMAIL PROTECTED] On Behalf Of ->Rich Adamson ->Sent: Saturday, October 01, 2005 9:21 AM ->To: Asterisk Users Mailing List - Non-Commercial Discussion ->Subject: RE: [Asterisk-Users] Asterisk and RTP streams (just bumping) -> ->Just as a FYI, the rtp port that is used for any device ->(asterisk or a sip adapter) is choosen "by the device". If a ->sip adapter as an example attempts to initiate an rtp ->session, changing asterisk won't have any impact. -> ->Or, saying this a little different, asterisk can initiate an ->rtp session using an rtp/udp port from within the range ->specified in rtp.conf, but the other end of that rtp session ->(sip adapter) gets to pick its own udp port for the return ->data. The udp port selected is often times picked from a ->range that you have control over, but in all cases that I can ->think of, you can't tell the device to always pick a single ->predetermined udp port. -> ->So, unless I've really missed the point in the discussion ->below, I don't believe the proposed port selection changes ->can be accomplished. -> -> -> ->> Bumping, just in case it got lost in the shuffle today... I ->think this ->> is an important thing to be able to do. ->> ->> Subject: [Asterisk-Users] Asterisk and RTP streams ->> ->> Guys, I've been poking around trying to find a good answer for this ->> via voip-info, google, etc... Haven't found anything that helps, so ->> maybe you mates could. ->> ->> A lot of my customers are using Linksys UAs (router/ATA ->PAP2) and some ->> using Sipura SPA-2002s. Every once in a while, the customer ->will get ->> one-way audio. I've read that this is commonly caused by ->the outgoing ->> RTP port not being the same as the incoming RTP port. A lot ->of other ->> devices (I found info on forcing Xten to do it) can be ->forced to use ->> the same port for both, but these devices don't have an ->option (that ->> I've been able to find, even in the provisioning configs) ->to do this. So, my question is two-fold: ->> ->> 1. Can Asterisk be told to send the RTP stream for incoming and ->> outgoing always on the same set of ports? ->> 2. Does anyone know something that I'm missing for the ->above mentioned ->> devices? They're all the 2 line version of the ATA and/or router ->> configs (wireless and wired) ->> ->> Thank you all in advance for your thoughts and comments. I ->apologize ->> in advance if I missed something that was publicly available. ->> ->> Sherwood McGowan ->> ->> ->> ->> ___ ->> --Bandwidth and Colocation sponsored by Easynews.com -- ->> ->> Asterisk-Users mailing list ->> Asterisk-Users@lists.digium.com ->> http://lists.digium.com/mailman/listinfo/asterisk-users ->> To UNSUBSCRIBE or update options visit: ->>http://lists.digium.com/mailman/listinfo/asterisk-users ->> -> ->---End of Original Message- -> -> ->___ ->--Bandwidth and Colocation sponsored by Easynews.com -- -> ->Asterisk-Users mailing list ->Asterisk-Users@lists.digium.com ->http://lists.digium.com/mailman/listinfo/asterisk-users ->To UNSUBSCRIBE or update options visit: -> http://lists.digium.com/mailman/listinfo/asterisk-users -> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] strange wave like noise on sip handset
On Sat, 2005-10-01 at 14:47 +0100, Angus Comber wrote: > No it happens on our asterisk and at a customers. Not that noticeable but > not crystal clear. Didn't happen on a Snom 190. [Snip] > Sipura SPA-841 - when receiving an incoming call echoy for about 2-3 seconds > at start of call then echo went away. Remote end did not hear any echo. > Also wave like hiss as per my message. Angus - does the SPA-841 have AGC? Turn it off if you can. Brett ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Transcoding
Quoting Anders Svensson <[EMAIL PROTECTED]>: this page might help. http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning > > > Hi all! > > Is it possible to have a setup with a server only dedicated for transcoding > from ulaw/alaw to G729. What is the capacity of a server like that in > simultaneous calls? > > > > > > Regards > > Anders > > > > > > This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] strange wave like noise on sip handset
We have all Cisco - and they are pricey, but work great otherwise. Both with chsn_sccp and SIP. 05 -> 70s and a few 20s -Greg On Sat, 2005-10-01 at 14:47 +0100, Angus Comber wrote: > No it happens on our asterisk and at a customers. Not that noticeable but > not crystal clear. Didn't happen on a Snom 190. > > I have been working my way through IP handsets with these results: > > Grandstream BT-100 series. OKish for the price but a bit echoy. > > Grandstream GXP-2000 - OK but if used on hands free a bit echoy. > > Snom 190. Very clear. However, on a customer site they complained that > full volume was still not load enough. But didn't extensively test. > > Sipura SPA-841 - when receiving an incoming call echoy for about 2-3 seconds > at start of call then echo went away. Remote end did not hear any echo. > Also wave like hiss as per my message. > > Next phones to try are a Polycom 300 and a CISCO 7940. > > I suppose it depends on how demanding customer is. I would hope that I can > find a phone with no echo / hiss /other problems. Perhaps I need to think > about using channel banks/FXS cards and analog phones! But would prefer IP > phones for flexibility etc. > > Anyone found a perfect IP phone? > > Angus > > > - Original Message - > From: "Leif Madsen" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Saturday, October 01, 2005 2:33 PM > Subject: Re: [Asterisk-Users] strange wave like noise on sip handset > > > On 9/30/05, Angus Comber <[EMAIL PROTECTED]> wrote: > > On a Sipura SPA-841 handset (and also at other end) you hear a sea wave > > like > > sound - it gets louder then softer and continually repeats. > > > > I don't remember hearing this when using other handsets. But what is this > > effect? How can I reduce it? > > I heard the same thing from a remote users Polycom 501 - seems it was > sitting too close to a fan in a computer. Could it be something > similar to that? > > Just a thought since this happened to me yesterday :) > > -- > Leif Madsen - http://www.leifmadsen.com > Astricon 2005, Anaheim, CA, October 12-14 > http://www.astricon.net > http://www.oreilly.com/catalog/asterisk > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system?
[EMAIL PROTECTED] wrote on 09/30/2005 01:10:34 PM: > On Fri, Sep 30, 2005 at 01:32:07PM +0100, Angus Comber wrote: > > Hello > > > > I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU. Is this > > likely to be enough power for a 8 extension system with 6 external pstn > > lines? > > Probably, yes. I use a 533MHz to handle two outside lines and 4 internal extensions with no problems. There is very little CPU usage even with nearly everything in use. However, I'm doing no transcoding and no-compression codecs. Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] strange wave like noise on sip handset
No it happens on our asterisk and at a customers. Not that noticeable but not crystal clear. Didn't happen on a Snom 190. I have been working my way through IP handsets with these results: Grandstream BT-100 series. OKish for the price but a bit echoy. Grandstream GXP-2000 - OK but if used on hands free a bit echoy. Snom 190. Very clear. However, on a customer site they complained that full volume was still not load enough. But didn't extensively test. Sipura SPA-841 - when receiving an incoming call echoy for about 2-3 seconds at start of call then echo went away. Remote end did not hear any echo. Also wave like hiss as per my message. Next phones to try are a Polycom 300 and a CISCO 7940. I suppose it depends on how demanding customer is. I would hope that I can find a phone with no echo / hiss /other problems. Perhaps I need to think about using channel banks/FXS cards and analog phones! But would prefer IP phones for flexibility etc. Anyone found a perfect IP phone? Angus - Original Message - From: "Leif Madsen" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, October 01, 2005 2:33 PM Subject: Re: [Asterisk-Users] strange wave like noise on sip handset On 9/30/05, Angus Comber <[EMAIL PROTECTED]> wrote: On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? I heard the same thing from a remote users Polycom 501 - seems it was sitting too close to a fan in a computer. Could it be something similar to that? Just a thought since this happened to me yesterday :) -- Leif Madsen - http://www.leifmadsen.com Astricon 2005, Anaheim, CA, October 12-14 http://www.astricon.net http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: RHEL / CentOS Enable APIC
Hello! I'm setting up Asterisk on a new system. In the past, all of my Asterisk boxes have either been embedded-style systems that do not supoort APIC, or multi-processor systems where APIC comes along with SMP. However, now I'm trying to install Asterisk on a single CPU (and non-HT) system that does support APIC (A P4 Northwood an Intel 845 chipset). I've used both RHEL3 and RHEL4 (and CentOS 3 as part of [EMAIL PROTECTED]). For the life of me, though, I cannot seem to get an APIC enabled kernel installed. It seems that because it's a uniprocessor system, the default is to load a uniprocessor, non-APIC kernel. On the 2.6 kernels I've tried adding "lapic" as a kernel parameter, but it does not help. I must say that I'm surprised that [EMAIL PROTECTED] doesn't do this automatically, given the benefits of an APIC-enabled kernel for Asterisk. Is there a way with either RHEL or CentOS to force it to use an APIC-enabled kernel? I've tried Googling but no success. Thank you very much for your help! Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Now can I tranfer call form one SIP phone to other during call (unattended transfer)
I have both t and T options in dial command. SIP phones configured with canreinvite=no and when I pres #1 (as I have in features.conf) during call there nothing to happened. Thank for any suggestions. Bob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error on loading zaptel module
i get this error on dmesg: zaptel: Unknown symbol __stack_smash_handler zaptel: Unknown symbol __guard paradise dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] strange wave like noise on sip handset
On 9/30/05, Angus Comber <[EMAIL PROTECTED]> wrote: > On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like > sound - it gets louder then softer and continually repeats. > > I don't remember hearing this when using other handsets. But what is this > effect? How can I reduce it? I heard the same thing from a remote users Polycom 501 - seems it was sitting too close to a fan in a computer. Could it be something similar to that? Just a thought since this happened to me yesterday :) -- Leif Madsen - http://www.leifmadsen.com Astricon 2005, Anaheim, CA, October 12-14 http://www.astricon.net http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 create IVR system using *
Hello I want to setup a system where people can dial a number and then a system will ask them questions for which they will leave answers. Eg something like this: Answer Playback(whatisyournamemsg) Record(yourname:gsm) Playback(whatisyourheight) Record(yourheight:gsm) Playback(thankyou) Hangup Is this the best way to do this sort of thing? Do users then just access the responses by eg *98 - or does this work a little differently to voicemail? How do we retrieve the responses? Or can I email the responses as WAV files? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Updated presentation of Asterisk 1.2
Friends, I have updated my Asterisk 1.2 presentation with the latest information. It is still available in the same place as before: http://www.astricon.net/asterisk1-2/ Please continue to test the beta of Asterisk 1.2, available at ftp.digium.com. We need all the feedback we can get. If you are a developer and have some time for community work, please check in with the bug tracker and help us resolve the unresolved bugs. See you at Astricon in Anaheim! /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk + 99.999s was (Re: [Asterisk-Users] Asterisk on windows)
On Sat, 2005-10-01 at 08:31 -0400, Julio Arruda wrote: [snip] > One thing interesting, coming from data background, seeing the > requirements in carrier voice networks. Is a quite distinct ball-game. > Devices that require 'hot-software-upgrades', still not that often seen > in data. How is this being handled with Asterisk + other solutions ? > Example, having a trunk gateway with a OC3 worth of TDM, is 'acceptable' > that a sw upgrade will cut established calls ? Iirc Motorola has a solution that allows in-operation linux kernel upgrades. No idea how they pulled that magic off (and if it actually works). At VON IBM was going to demo a blade based Asterisk solution that has auto-failover of calls so maybe that could also be used to upgrade software. Don't have more info about this IBM solution. If you have a DS3 or OC3 worth of TDM calls then it probably makes sense to use a carrier-class box. Regards, Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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:Any way to not overwrite sound files on compile?
> > When compile only type 'make' and copy manually your module/s from asterisk > > apps directory into your asterisk modules directory. > > > > regards. > > G. > > > > > > Matt wrote: > > > > >Every time I recompile Asterisk (or upgrade to a new CVS-HEAD, > > >whatever) asterisk overwrites custom files I have made. Granted, > > >these files are named the same as the asterisk default files > > >(vm-login.gsm, etc) because we had a person here record them to > > >customize them a bit more for our application. > > > > > >Short of keeping them somewhere and copying them back every time > > >(which isn't all that often) I do a re-compile. Is there some flag or > > >something to tell Asterisk not to install sound files, or at the very > > >least not to overwrite ones already existing? > > > > Or just use "make upgrade" it just puts in place the binaries. I believe you meant to say "make update". "upgrade" is not a defined parameter. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Asterisk + 99.999s was (Re: [Asterisk-Users] Asterisk on windows)
Patrick wrote: On Wed, 2005-09-28 at 23:17 +0800, Steve Underwood wrote: [snip] An effective DOS attack on a $300,000 Alpha running NT I used to use was "wiggle the mouse" :-) I never really understood how that brought a multi-CPU machine to a standstill, but it did. Reminds me of an Internet Call Diversion pilot WorldCom did back in 2000 where Alcatel & some M$ drones brought in 2 very big Alpha servers running NT. These boxes needed to be rebooted multiple times. They were surprised WCOM felt having to reboot these boxes all the time was unacceptable in an environment requiring 5nines availability. Never laughed so hard when I saw the incredulous faces of the M$ drones. We brought in a Stratus based solution and won the project. Alcatel folks where not surprised, I'm sure ;-) One thing interesting, coming from data background, seeing the requirements in carrier voice networks. Is a quite distinct ball-game. Devices that require 'hot-software-upgrades', still not that often seen in data. How is this being handled with Asterisk + other solutions ? Example, having a trunk gateway with a OC3 worth of TDM, is 'acceptable' that a sw upgrade will cut established calls ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 and RTP streams (just bumping)
Just as a FYI, the rtp port that is used for any device (asterisk or a sip adapter) is choosen "by the device". If a sip adapter as an example attempts to initiate an rtp session, changing asterisk won't have any impact. Or, saying this a little different, asterisk can initiate an rtp session using an rtp/udp port from within the range specified in rtp.conf, but the other end of that rtp session (sip adapter) gets to pick its own udp port for the return data. The udp port selected is often times picked from a range that you have control over, but in all cases that I can think of, you can't tell the device to always pick a single predetermined udp port. So, unless I've really missed the point in the discussion below, I don't believe the proposed port selection changes can be accomplished. > Bumping, just in case it got lost in the shuffle today... I think this is an > important thing to be able to do. > > Subject: [Asterisk-Users] Asterisk and RTP streams > > Guys, I've been poking around trying to find a good answer for this via > voip-info, google, etc... Haven't found anything that helps, so maybe you > mates could. > > A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using > Sipura SPA-2002s. Every once in a while, the customer will get one-way > audio. I've read that this is commonly caused by the outgoing RTP port not > being the same as the incoming RTP port. A lot of other devices (I found > info on forcing Xten to do it) can be forced to use the same port for both, > but these devices don't have an option (that I've been able to find, even in > the provisioning configs) to do this. So, my question is two-fold: > > 1. Can Asterisk be told to send the RTP stream for incoming and outgoing > always on the same set of ports? > 2. Does anyone know something that I'm missing for the above mentioned > devices? They're all the 2 line version of the ATA and/or router configs > (wireless and wired) > > Thank you all in advance for your thoughts and comments. I apologize in > advance if I missed something that was publicly available. > > Sherwood McGowan > > > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transcoding
Hi all! Is it possible to have a setup with a server only dedicated for transcoding from ulaw/alaw to G729. What is the capacity of a server like that in simultaneous calls? Regards Anders ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 on windows
On Wed, 2005-09-28 at 23:17 +0800, Steve Underwood wrote: [snip] > An effective DOS attack on a $300,000 Alpha running NT I used to use was > "wiggle the mouse" :-) I never really understood how that brought a > multi-CPU machine to a standstill, but it did. Reminds me of an Internet Call Diversion pilot WorldCom did back in 2000 where Alcatel & some M$ drones brought in 2 very big Alpha servers running NT. These boxes needed to be rebooted multiple times. They were surprised WCOM felt having to reboot these boxes all the time was unacceptable in an environment requiring 5nines availability. Never laughed so hard when I saw the incredulous faces of the M$ drones. We brought in a Stratus based solution and won the project. Regards, Patrick ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip registration Failure
Hi List, I am very new to asterisk. I downloaded asterisk from CVS head yesterday and compiled it in Redhat linux 9. I created a sip account for testing and configured it in the Firefly. While Firefly try to connect to the asterisk server i am getting an error as below and failing the registration. Oct 1 08:00:57 NOTICE[23415]: chan_sip.c:10646 handle_request_register: Registration from '"200" ' failed for '192.168.10.200' - Not a local SIP domain My sip configuration is as below, [general] context=default ; Default context for incoming calls bindport=5060 bindaddr=0.0.0.0 [test] type=peer secret=200 username=200 host=dynamic nat=no #disallow=all allow=all context=default Please help me to find out the problem in my configuration. Thanks in advance Rgds Anil ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Developer help needed
I need help from somebody to make a Boot CD which includes a Linux distribution and Asterisk, it should install more or less automatically (like [EMAIL PROTECTED]). I need this to make a version that is optimized for use with IPManager, IPSpeedDial and IPSwitchBoard. If you think you are capable of making this please contact me off list so we can discuss your fees and timeframe. Regards Thorben http://ipsoftware.thorben.dk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calls between SIP and IAX
Hi all, I have a trouble when I try to configure asterisk to make calls between IAX and SIP. IAX I'm using to connect between asterisks a on SIP I have phones. The calls come from higher asterisk to my on IAX, SIP phone is ringing and when I hang up then dial command ends and connection is loss. When I'll make connection between asterisks on SIP then all work fine. Does anybody has any suggestions? Bob. P.S. - I'm using asterisk 1.0.9 on FC3. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Voice Encryption
Hello, > I went over the code. AES128 is the only algorithm that is suppored > today. More importantly there are some concerns on the vulnerability as > discussed in > http://lists.digium.com/pipermail/asterisk-security/2005-August/60.html. > People are using UDP VPNs to satisfy customer requirements. > http://lists.digium.com/pipermail/asterisk-users/2005-August/120293.html we are using plain ipsec here. From softphones, we just use operating system's native ipsec support. For hardphones, we have a custom device based on soekris board with vpn hardware encryption accelerator, which does the job for the phone. Juraj. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Empty ACK
Hello, I have asterisk connected to SER/RTPProxy which is again connected to a IP-PSTN gateway. When calling with a UA, registered at * to a SIP phone connected to the IP-PSTN gateway, I get 'empty ACKs': U 192.168.0.173:5060 -> 10.254.254.1:5060 ACK SIP/2.0. Via: SIP/2.0/UDP 192.168.0.173:5060;branch=z9hG4bK5cb7d048. Route: ,. From: "0161801019" ;tag=as628d39c1. To: ;tag=00-04094-52dc5953-7c1293c27. Contact: . Call-ID: [EMAIL PROTECTED] CSeq: 103 ACK. User-Agent: Asterisk PBX. Content-Length: 0. As you can see, there is no URI after the ACK statement, and SER doesn't know what to do with it. Is this a bug in *, or is this normal? Regards, Ronald ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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:Any way to not overwrite sound files on compile?
On Fri, 2005-09-30 at 10:17 -0300, Gustavo A. Gonzalez wrote: > When compile only type 'make' and copy manually your module/s from asterisk > apps directory into your asterisk modules directory. > > regards. > G. > > > Matt wrote: > > >Every time I recompile Asterisk (or upgrade to a new CVS-HEAD, > >whatever) asterisk overwrites custom files I have made. Granted, > >these files are named the same as the asterisk default files > >(vm-login.gsm, etc) because we had a person here record them to > >customize them a bit more for our application. > > > >Short of keeping them somewhere and copying them back every time > >(which isn't all that often) I do a re-compile. Is there some flag or > >something to tell Asterisk not to install sound files, or at the very > >least not to overwrite ones already existing? > Or just use "make upgrade" it just puts in place the binaries. -- Dave Cotton <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceGateway Design - Request for comments/suggestions
Hello Group, I'd like to hear your thoughts for the following design. To implement a Voice Gateway solution with atleast 2 PRI (60 channel) capacity, is it better to implement - 1 Grunty Server with a BiPRI card (TE205P, 210P) / Quad PRI (TE405P, TE410P) or - 2 Normal Servers (Dell 420/430 with 2 Gig RAM and 80 Gig SATA) with Single PRI (TE110P) or BiPRI (TE205P, 210P) and Load balance between them. My second question in similar lines is, is it better to implement - 2 single slot ISDN Card or - 1 multiple slot ISDN Card. While option 1 gives us card level redundancy, option 2 only gives us slot level redundancy. With regards to responsivenes/performance (memory usage et al) is option 1 costlier? Has anyone experienced malfunctioning of some slots an ISDN Card, but other slots in the same card functioning normally? Thanks for sharing your thoughts. -r ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 and RTP streams (just bumping)
Bumping, just in case it got lost in the shuffle today... I think this is an important thing to be able to do. Subject: [Asterisk-Users] Asterisk and RTP streams Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could. A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer will get one-way audio. I've read that this is commonly caused by the outgoing RTP port not being the same as the incoming RTP port. A lot of other devices (I found info on forcing Xten to do it) can be forced to use the same port for both, but these devices don't have an option (that I've been able to find, even in the provisioning configs) to do this. So, my question is two-fold: 1. Can Asterisk be told to send the RTP stream for incoming and outgoing always on the same set of ports? 2. Does anyone know something that I'm missing for the above mentioned devices? They're all the 2 line version of the ATA and/or router configs (wireless and wired) Thank you all in advance for your thoughts and comments. I apologize in advance if I missed something that was publicly available. Sherwood McGowan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 backup asterisk installation for upgrade
I want to backup Asterisk based on 1.07 to install 1.09 and 1.2beta on another server. Which files and folders do I have to backup, in order to restore if things don't work right? This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
