[Asterisk-Users] Again Asterisk and VMWare - it works now!
Hi, I have succeed using Asterisk on VMWare on an [EMAIL PROTECTED] with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem is only when I try to call local services, like echo test or Digium Demo. Then, the sound of the informative message for the Digium Demo is choppy, but the sound from the Digium server (after connection) is very good. So.. the problem is only to play local files when in virtual machine (menus, informative messages, etc.). Why? It is clear that this is not a computer performance issue and/or a timing problem during the codec conversion. More, the inband DTMF works like a charm under the virtual machine. Even the known problem with double digits for Cisco phones dissapear. BR, Dan P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
As I understand it, the playback applications need to have native access to a piece of digium hardware to perform well... Under the virtual machine, that won't happen. Matt Hardeman PaperSoft - Original Message - From: David Boreham [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 2:01 AM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. Your e-mail made me chuckle. When I worked at Octel/Lucent in the mid-90's we were constantly sniped at for trying to make a voicemail system which ran on general purpose computers, operating systems, and message stores. It was hard work to get it to run smoothly back then even though we were the only application on the box. And today there's a guy who's trying to do the same thing in a VIRTUAL MACHINE !?!?!? sigh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
Hi, From the message archive I have seen that only for MOH you neet some Digium hardware or some software timers/emulators like ztdummy. You need for playback too? Then why it works on a standard linux machine without any such 'emulators'? BR, Dan - Original Message - From: Matthew Hardeman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 10:13 AM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! As I understand it, the playback applications need to have native access to a piece of digium hardware to perform well... Under the virtual machine, that won't happen. Matt Hardeman PaperSoft - Original Message - From: David Boreham [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 2:01 AM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. Your e-mail made me chuckle. When I worked at Octel/Lucent in the mid-90's we were constantly sniped at for trying to make a voicemail system which ran on general purpose computers, operating systems, and message stores. It was hard work to get it to run smoothly back then even though we were the only application on the box. And today there's a guy who's trying to do the same thing in a VIRTUAL MACHINE !?!?!? sigh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
On Sat, 2003-07-19 at 02:17, Dan wrote: Hi, From the message archive I have seen that only for MOH you neet some Digium hardware or some software timers/emulators like ztdummy. You need for playback too? Then why it works on a standard linux machine without any such 'emulators'? Install the ztdummy and see if it fixes your choppiness. If it doesn't, then you do have a performance problem from being in vmware. You will then have to just accept it. - Original Message - From: Matthew Hardeman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 10:13 AM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! As I understand it, the playback applications need to have native access to a piece of digium hardware to perform well... Under the virtual machine, that won't happen. Matt Hardeman PaperSoft - Original Message - From: David Boreham [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 2:01 AM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. Your e-mail made me chuckle. When I worked at Octel/Lucent in the mid-90's we were constantly sniped at for trying to make a voicemail system which ran on general purpose computers, operating systems, and message stores. It was hard work to get it to run smoothly back then even though we were the only application on the box. And today there's a guy who's trying to do the same thing in a VIRTUAL MACHINE !?!?!? sigh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
David Boreham wrote: P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. Your e-mail made me chuckle. When I worked at Octel/Lucent in the mid-90's we were constantly sniped at for trying to make a voicemail system which ran on general purpose computers, operating systems, and message stores. It was hard work to get it to run smoothly back then even though we were the only application on the box. And today there's a guy who's trying to do the same thing in a VIRTUAL MACHINE !?!?!? In the mid 90s much of the world's voicemail used Dialogic cards in PC servers, and ran without problems. Voicemail is easy, as the latency requires are very lax. VoIP is much harder in systems not tailored for hard real time use. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
IIRC you were given URLs for all sorts of cheapo PCs. Perhaps you've got an old P90 lying around? Or perhaps someone else has? Use that! Not vmware! If you're to use vmware, do it the other way around - linux host with vmware windoze guest. This works fine for me on my PC. On Sat, 2003-07-19 at 08:49, Dan wrote: Hi, I have succeed using Asterisk on VMWare on an [EMAIL PROTECTED] with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem is only when I try to call local services, like echo test or Digium Demo. Then, the sound of the informative message for the Digium Demo is choppy, but the sound from the Digium server (after connection) is very good. So.. the problem is only to play local files when in virtual machine (menus, informative messages, etc.). Why? It is clear that this is not a computer performance issue and/or a timing problem during the codec conversion. More, the inband DTMF works like a charm under the virtual machine. Even the known problem with double digits for Cisco phones dissapear. BR, Dan P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
Hi Steven, Install the ztdummy and see if it fixes your choppiness. If it doesn't, then you do have a performance problem from being in vmware. You will then have to just accept it. I have this installed, but when Asterisk is started, I still get the following warning: [res_musiconhold.so] = (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found WARNING[1074416352]: File res_musiconhold.c, Line 462 (moh_register): Unable to open pseudo channel for timing... Sound may be choppy. WARNING[1074416352]: File res_musiconhold.c, Line 462 (moh_register): Unable to open pseudo channel for timing... Sound may be choppy. WARNING[1074416352]: File res_musiconhold.c, Line 462 (moh_register): Unable to open pseudo channel for timing... Sound may be choppy. and for IAX: [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) WARNING[1074416352]: File chan_iax2.c, Line 5561 (load_module): Unable to open IAX timing interface: No such device == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found WARNING[1074416352]: File chan_iax2.c, Line 5044 (set_config): Ignoring port for now == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) The strange thing is that IAX works well. ztdummy is loaded: [EMAIL PROTECTED] asterisk]# modprobe -l | grep ztdummy /lib/modules/2.4.20-8/misc/ztdummy.o Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX can be used on a different UDP port?
Hi,
I'm back with my question, maybe someone can help me:
I want to use IAX on another UDP port (not the default 5036), because I have
2 Asterisks behind the same NAT.
Changing the default port in iax.conf file from 5036 to 5038 and then
calling using the syntax:
exten = _8XXX,1,Dial(IAX/user:[EMAIL PROTECTED]:5038/${EXTEN:1})
I get the follwing error in the Asterisk console:
-- Executing Dial(SIP/351-24f4, IAX/user:[EMAIL PROTECTED]:5038/500)
in new stack
WARNING[1200825920]: File chan_iax.c, Line 1550 (create_addr): No such host:
195.3.32.191:5038
NOTICE[1200825920]: File app_dial.c, Line 488 (dial_exec): Unable to create
channel of type 'IAX'
== Everyone is busy at this time
It is something wrong?
It works like a charm with the default port, so it is not a
user/authentication problem.
Thanks,
Dan
___
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Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
Roy, Please do not give me such a solution. I know how and where to buy or how to build a very cheap PC (I work in this field). I now that this is a cheaper option (to buy or build a new pc), but I don't want another computer running 24/7 in my house. It is so difficult to understand that? I have a small flat with two rooms. I want to be able to sleep too in the same house. My wife for sure will not accept another one... I feel that it can fully work on my config (allmost it does it now). It is more challenging to make it work under those circumstances...;-) Why to choose everytime the easiest solution available? I want to do it for my ..soul...;-) Best regards, Dan P.S. I have several PCs available for this, but.. I DON'T WANT TO USE THEM! - Original Message - From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 12:33 PM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! IIRC you were given URLs for all sorts of cheapo PCs. Perhaps you've got an old P90 lying around? Or perhaps someone else has? Use that! Not vmware! If you're to use vmware, do it the other way around - linux host with vmware windoze guest. This works fine for me on my PC. On Sat, 2003-07-19 at 08:49, Dan wrote: Hi, I have succeed using Asterisk on VMWare on an [EMAIL PROTECTED] with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem is only when I try to call local services, like echo test or Digium Demo. Then, the sound of the informative message for the Digium Demo is choppy, but the sound from the Digium server (after connection) is very good. So.. the problem is only to play local files when in virtual machine (menus, informative messages, etc.). Why? It is clear that this is not a computer performance issue and/or a timing problem during the codec conversion. More, the inband DTMF works like a charm under the virtual machine. Even the known problem with double digits for Cisco phones dissapear. BR, Dan P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Transfer Anouncement
Ok the scenario want to acheive is: Call between A and B is established. A wants to transfer B to C, so A presses # and then the extension of C. This puts B on hold, and a connection between A and C is established. Then, when A hangs up the call, B and C get connected. I am using H323 phones, at the moment when A dials # and the extension of C, A is immediately cut off, and the B gets connected to C.. Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Actiontec's InternetPhoneWizard (USB) and Asterisk
Hi all, Anyone succeed using * with Asterisk with full functionality (dialing, ringing, etc.). Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
Hey Dan, Get a http://www.mini-itx.com/ and disguise it as a fruit bowl. She'll never know! I can't wait until someone builds one looking like a shoe or a handbag and then I can have them all over the house and the more I have, the happier the other half will be!! W - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 11:12 AM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! Roy, Please do not give me such a solution. I know how and where to buy or how to build a very cheap PC (I work in this field). I now that this is a cheaper option (to buy or build a new pc), but I don't want another computer running 24/7 in my house. It is so difficult to understand that? I have a small flat with two rooms. I want to be able to sleep too in the same house. My wife for sure will not accept another one... I feel that it can fully work on my config (allmost it does it now). It is more challenging to make it work under those circumstances...;-) Why to choose everytime the easiest solution available? I want to do it for my ..soul...;-) Best regards, Dan P.S. I have several PCs available for this, but.. I DON'T WANT TO USE THEM! - Original Message - From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 12:33 PM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! IIRC you were given URLs for all sorts of cheapo PCs. Perhaps you've got an old P90 lying around? Or perhaps someone else has? Use that! Not vmware! If you're to use vmware, do it the other way around - linux host with vmware windoze guest. This works fine for me on my PC. On Sat, 2003-07-19 at 08:49, Dan wrote: Hi, I have succeed using Asterisk on VMWare on an [EMAIL PROTECTED] with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem is only when I try to call local services, like echo test or Digium Demo. Then, the sound of the informative message for the Digium Demo is choppy, but the sound from the Digium server (after connection) is very good. So.. the problem is only to play local files when in virtual machine (menus, informative messages, etc.). Why? It is clear that this is not a computer performance issue and/or a timing problem during the codec conversion. More, the inband DTMF works like a charm under the virtual machine. Even the known problem with double digits for Cisco phones dissapear. BR, Dan P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best VoIP provider for Asterisk?
Marcus == Marcus Adolfsson [EMAIL PROTECTED] writes: Marcus Nufone.net is the best VoIP provider for Asterisk Marcus integration. They offer IAX termination, 2.9 cents outgoing Marcus long-distance and incoming 800. We use them at our office for Marcus all phone calls. I second this. But note they are now at 2.0 cents for calls to US and Canada. They change the same per minute for incoming calls on the 800 numbers. They are responsive, competent; simply great to work with. Highly recommended. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] XS4ALL Gateway now also does FWD
Yo all, I just added FWD (http://fwd.pulver.com/) to the XS4ALL PSTN-gateway. Here's a quick update on how it works: VoIP: From IAXTel, dial 31800rest of number for Dutch toll-free numbers. FWD is not (yet) directly reachable from IAXTel. I'll talk to Mark to see if he's intrested in setting this up. From FWD, dial 42442, wait for the dial-tone and dial your number. This can be any IAXTel-number or Dutch toll-free PSTN-number (starting with 0800). PSTN: From the PSTN, dial +31 20 3987567, wait for the dial-tone and dial your number. This can be any IAXTel-number or 5-digit FWD-number. Note that I'll probably include the other numbers and prefixes on FWD shortly. For now, it only does the standard 5-digit numbers. Let me know if you have any questions or comments. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
Dan, What type of "virtual harddisk" did you create in your linux vmware virtual machine? As far as I know, the default type is a kind of compressed harddisk image, to save disk space in the host machine. Perhaps that is the answer to have bad performance only in local services, opening local files in vmware: the overhead to decompress. -Eduardo Dan escreveu: Nice! If it can work without connecting it to the power supply, then will be better..;-) When we leave the home, I must disconnect from the mains EVERYTHING (the alarm system and the HA PC are the only accepted exceptions)... :-))) Dan - Original Message - From: "Simon Woodhead" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 1:31 PM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! Hey Dan, Get a http://www.mini-itx.com/ and disguise it as a fruit bowl. She'll never know! I can't wait until someone builds one looking like a shoe or a handbag and then I can have them all over the house and the more I have, the happier the other half will be!! W - Original Message - From: "Dan" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 11:12 AM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! Roy, Please do not give me such a solution. I know how and where to buy or how to build a very cheap PC (I work in this field). I now that this is a cheaper option (to buy or build a new pc), but I don't want another computer running 24/7 in my house. It is so difficult to understand that? I have a small flat with two rooms. I want to be able to sleep too in the same house. My wife for sure will not accept another one... I feel that it can fully work on my config (allmost it does it now). It is more challenging to make it work under those circumstances...;-) Why to choose everytime the easiest solution available? I want to do it for my ..soul...;-) Best regards, Dan P.S. I have several PCs available for this, but.. I DON'T WANT TO USE THEM! - Original Message - From: "Roy Sigurd Karlsbakk" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 12:33 PM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! IIRC you were given URLs for all sorts of cheapo PCs. Perhaps you've got an old P90 lying around? Or perhaps someone else has? Use that! Not vmware! If you're to use vmware, do it the other way around - linux host with vmware windoze guest. This works fine for me on my PC. On Sat, 2003-07-19 at 08:49, Dan wrote: Hi, I have succeed using Asterisk on VMWare on an [EMAIL PROTECTED] with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem is only when I try to call local services, like echo test or Digium Demo. Then, the sound of the informative message for the Digium Demo is choppy, but the sound from the Digium server (after connection) is very good. So.. the problem is only to play local files when in virtual machine (menus, informative messages, etc.). Why? It is clear that this is not a computer performance issue and/or a timing problem during the codec conversion. More, the inband DTMF works like a charm under the virtual machine. Even the known problem with double digits for Cisco phones dissapear. BR, Dan P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP in hotels
Jonathan Young wrote: Our company can offer VoIP to premises and domestic users and bill the premises as a whole. We need something to enable the hotel owner to bill each guest in a hotel in real time. What solutions do exist presently? Asterisk can certainly do exactly what you need. (PS: Our radius (and every telephony equipment outside the hotel) does not recognise which room in the hotel initiated the international (VoIP) call, so that's the main problem - Only the hotel's PMS knows which guest phoned, so with our current setup we cannot bill the individual guest, but the hotel as a whole) First drop RADIUS like a prom dress and either create or find a real billing platform. You simply need a however many T400Ps and FXS channel banks that can interface into your existing hotel room phones. Then with a proper billing platform your system will automagicly know exactly which FXS port made that call and would make the connection to which room made the call all in real-time. This is very simple, no need to over complicate the matter with a useless technology like RADIUS. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog phone not ringing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I've got my developers kit from telappliant and got a machine up and running to become the house phone system. Most things are working now, such as incoming calls, call transfer, call parking, voicemail, etc. The one thing I can't do is make my analog phone ring! I can see the call coming in on the asterisk console and can then pick up the analog phone, but no ringing! Incoming calls come in on the X100P (channel 1), the analog phone is on the only other channel from a TDM400P The one thing that I think it could be is the connector to convert from RJ45 to BT phone socket. I'm using a mod tap that I had lying around. Not sure what the wiring is like inside it. Thanks, Darren. The console has these messages: -- Starting simple switch on 'Zap/1-1' NOTICE[344081]: File chan_zap.c, Line 4135 (ss_thread): Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, ZAP/2|tr|30) in new stack -- Called 2 -- Hungup 'Zap/2-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (default, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' NOTICE[360465]: File chan_zap.c, Line 4135 (ss_thread): Got event 2 (Ring/Answered)... -- Executing Dial(Zap/1-1, ZAP/2|tr|30) in new stack -- Called 2 -- Hungup 'Zap/2-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (default, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' extensions.conf exten = s,1,Dial(ZAP/2|tr,30) ; Only 1 line... can't be busy! exten = s,2,Hangup [channels] group=1 context=pstn signalling=fxs_ks threewaycalling=yes musiconhold=default transfer=no channel = 1 group=2 context=internal signalling=fxo_ks mailbox=666 musiconhold=default threewaycalling=yes transfer=yes callerid=Living Room 101 channel = 2 - -- Darren Poulson - Unix Admin PGP Key at: http://www.22balmoralroad.net/~daz/pgp.key Death is nature's way of telling you to slow down. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.1 (GNU/Linux) iD8DBQE/GWP+eauN7FqCAjoRArC5AJ0ZR/r6fNbr3o4xG6Lv+tfdgvu17wCgp/fz 6dQW0fFJ7z8qRGk7ZosFvYg= =bENC -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dlink dg102s and G.729
Hi all, I'm making a small voip network with asterisk and Dlink dg102s gateways (MGCP). Dlink gateways work with g.711, g731 and g729 codecs. The questions: do i need a g729 codec license for asterisk for dlink use g.729 or just configure asterisk to inform gateways to utilize then ? Anyone have a .conf(s) example(s) to send-me ? Thank's Alexandre da Rosa --- Este e-mail foi certificado contra vírus no envio. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.502 / Virus Database: 300 - Release Date: 18/7/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog phone not ringing
--On Saturday, July 19, 2003 16:30:04 +0100 Darren Poulson [EMAIL PROTECTED] wrote: The one thing that I think it could be is the connector to convert from RJ45 to BT phone socket. I'm using a mod tap that I had lying around. Not sure what the wiring is like inside it. That's a pretty good bet. Depending on the phone I've regularly had to swap wires, solder wires together or add capacitors to get phones to work with my ATA 186 and even ordinary phone lines here in darkest Sussex. There seem to be a number of wiring variations that can cause problems - you can use google to get many references to the possible solutions. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speex support
John == John Todd [EMAIL PROTECTED]: John == John Todd [EMAIL PROTECTED] writes: What is the state of speex support in asterisk? I saw the codec seems to be there. John Install the Speex library support, and re-compile Asterisk. John There's probably a pre-compiled version of Speex for your system; John look around in whatever package manager you use for your Linux John distro. I do have the libraries installed. Can speex be used on IAX2 links? Is there much work still to be done? John Yes, it can be used. No work required to get functionality. Really? Have you tried it? I have. It doesn't work -- and a quick look at chan_iax2.c shows that there is a good reason for this -- get_samples() doesn't know how to calculate the number of samples for an incoming speex format frame. This results in chopped sound and hundreds of warnings: [snip] --J. PS: bad advice is worse than no advice... John I take it that comment was directed at me. John Yes, really, Speex does work, and yes, I did try it without any John of the modifications you describe above. Feel free to ask for John help if it doesn't work, but don't assume that others haven't John made it work or that I'm giving you intentionally bad advice - John it's insulting. I apologize, then -- I must have missed something, because after looking into it it seemed that there is no way it can work. I thought you just wrote without reading my posting carefully. sorry, --J. PS: I still think the patch I attached should be applied (or one that is more correct in calculating the number of samples), it made it work for me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to set specific codec ?
I have two dlink dg-102s (MGCP) gateways. Dg-102s work with g.711, g.723 and g.729 codecs. How i can set in asterisk to gateways utilize specific codec? Ex: g.723 Alexandre Rosa --- Este e-mail foi certificado contra vírus no envio. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.502 / Virus Database: 300 - Release Date: 18/7/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to set specific codec ?
Alexander de donde eres...? Alexandre Rosa wrote: I have two dlink dg-102s (MGCP) gateways. Dg-102s work with g.711, g.723 and g.729 codecs. How i can set in asterisk to gateways utilize specific codec? Ex: g.723 Alexandre Rosa --- Este e-mail foi certificado contra vírus no envio. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.502 / Virus Database: 300 - Release Date: 18/7/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] how to set specific codec ?
Porto Alegre, Brazil. -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nome de Fernando Zuluaga Enviada em: sábado, 19 de julho de 2003 16:02 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] how to set specific codec ? Alexander de donde eres...? Alexandre Rosa wrote: I have two dlink dg-102s (MGCP) gateways. Dg-102s work with g.711, g.723 and g.729 codecs. How i can set in asterisk to gateways utilize specific codec? Ex: g.723 Alexandre Rosa --- Este e-mail foi certificado contra vírus no envio. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.502 / Virus Database: 300 - Release Date: 18/7/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.502 / Virus Database: 300 - Release Date: 18/7/2003 --- Este e-mail foi certificado contra vírus no envio. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.502 / Virus Database: 300 - Release Date: 18/7/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog phone not ringing
On Sat, 19 Jul 2003, Darren Poulson wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I've got my developers kit from telappliant and got a machine up and running to become the house phone system. Most things are working now, such as incoming calls, call transfer, call parking, voicemail, etc. The one thing I can't do is make my analog phone ring! I can see the call coming in on the asterisk console and can then pick up the analog phone, but no ringing! Incoming calls come in on the X100P (channel 1), the analog phone is on the only other channel from a TDM400P The one thing that I think it could be is the connector to convert from RJ45 to BT phone socket. I'm using a mod tap that I had lying around. Not sure what the wiring is like inside it. In the UK, phones use a three-wire connection with a separate line for the ringer. This is supposed to stop phone bells tinkling when another phone is dialled. Anyway - you need the RJ11 to BT adapter that includes the ringing capacitor. Maplin sells the adapters both with and without the cap. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] file repository
Hey All, I asked around on irc about the existence of a file repository for various asterisk modules, patches, etc - that are not included in the asterisk cvs tree. I did not come across anything so feel free to utilize what I've setup at ftp.packetwest.com /pub/asterisk will hopefully fill with lots of goodies. If you have something to post, please dump it to /incoming and I will regularly transfer it over to /pub/asterisk after verifying the integrity of any compressed/packaged files. If I've already duplicated someone else's effort please let me know! I will be happy to provide a mirror of a preexisting repository that is active. Thanks, Steve Bourg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk crashes when trying to load G.729 module.
Before few days i bought few g.729 licenses. When i try to load the codec, asterisk crahses. I tried with and without oh323 module, same result: -- Jul 20 07:06:49 WARNING[589851]: File codec_g729b.c, Line 413 (load_module): Unable to initialize va stuff: -1 -- Here the ldd result: -- [EMAIL PROTECTED]:~$ ldd -v /usr/lib/asterisk/modules/codec_g729b.so libc.so.6 = /lib/libc.so.6 (0x40039000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x8000) Version information: /usr/lib/asterisk/modules/codec_g729b.so: libc.so.6 (GLIBC_2.1.3) = /lib/libc.so.6 libc.so.6 (GLIBC_2.2) = /lib/libc.so.6 libc.so.6 (GLIBC_2.1) = /lib/libc.so.6 libc.so.6 (GLIBC_2.0) = /lib/libc.so.6 /lib/libc.so.6: ld-linux.so.2 (GLIBC_2.1) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_2.0) = /lib/ld-linux.so.2 ld-linux.so.2 (GLIBC_PRIVATE) = /lib/ld-linux.so.2 --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Techfone VOIP phone
On Sat, Jul 19, 2003 at 01:01:52AM +0800, Steve Underwood wrote: That looks a bit like this one: http://www.planet.com.tw/product/product_intro.php?menu_id=3 rather expensive to me. These things have less DSP and compute to do than an ADSL modem, and should cost no more. That would make a reasonable price about US$50-60. I suspect it won't be long before we Market volume means a lot to prices... So far, the VoIP market is way behind the DSL market. mattf wrote: I ran across this company today and wondered if anyone was using their phone: http://www.techfone.com/ it's only $189 and has pretty good features for a VOIP phone although it's only got H323 (no SIP support). -- Scott LambertKC5MLE Unix SysAdmin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Again Asterisk and VMWare - it works now!
Also using the -p (real time option) to start asterisk, may also help. James Taylor wrote: Carlos, You may have something here. Dan - you might try to connect via the virtual network adapter to your host machine's hard drive. (just map a drive) it could be that networking from your VM to the host is faster. James Taylor -- Original Message -- From: Carlos Eduardo Cremon [EMAIL PROTECTED] Dan, What type of virtual harddisk did you create in your linux vmware virtual machine? As far as I know, the default type is a kind of compressed harddisk image, to save disk space in the host machine. Perhaps that is the answer to have bad performance only in local services, opening local files in vmware: the overhead to decompress. -Eduardo Dan escreveu: Nice! If it can work without connecting it to the power supply, then will be better..;-) When we leave the home, I must disconnect from the mains EVERYTHING (the alarm system and the HA PC are the only accepted exceptions)... :-))) Dan - Original Message - From: Simon Woodhead [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 1:31 PM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! Hey Dan, Get a http://www.mini-itx.com/ and disguise it as a fruit bowl. She'll never know! I can't wait until someone builds one looking like a shoe or a handbag and then I can have them all over the house and the more I have, the happier the other half will be!! W - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 11:12 AM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! Roy, Please do not give me such a solution. I know how and where to buy or how to build a very cheap PC (I work in this field). I now that this is a cheaper option (to buy or build a new pc), but I don't want another computer running 24/7 in my house. It is so difficult to understand that? I have a small flat with two rooms. I want to be able to sleep too in the same house. My wife for sure will not accept another one... I feel that it can fully work on my config (allmost it does it now). It is more challenging to make it work under those circumstances...;-) Why to choose everytime the easiest solution available? I want to do it for my ..soul...;-) Best regards, Dan P.S. I have several PCs available for this, but.. I DON'T WANT TO USE THEM! - Original Message - From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 19, 2003 12:33 PM Subject: Re: [Asterisk-Users] Again Asterisk and VMWare - it works now! IIRC you were given URLs for all sorts of cheapo PCs. Perhaps you've got an old P90 lying around? Or perhaps someone else has? Use that! Not vmware! If you're to use vmware, do it the other way around - linux host with vmware windoze guest. This works fine for me on my PC. On Sat, 2003-07-19 at 08:49, Dan wrote: Hi, I have succeed using Asterisk on VMWare on an [EMAIL PROTECTED] with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem is only when I try to call local services, like echo test or Digium Demo. Then, the sound of the informative message for the Digium Demo is choppy, but the sound from the Digium server (after connection) is very good. So.. the problem is only to play local files when in virtual machine (menus, informative messages, etc.). Why? It is clear that this is not a computer performance issue and/or a timing problem during the codec conversion. More, the inband DTMF works like a charm under the virtual machine. Even the known problem with double digits for Cisco phones dissapear. BR, Dan P.S. Please do not answer again that this setup cannot work. In this moment I cannot accept such an answer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL
Re: [Asterisk-Users] Techfone VOIP phone
Scott Lambert wrote: On Sat, Jul 19, 2003 at 01:01:52AM +0800, Steve Underwood wrote: That looks a bit like this one: http://www.planet.com.tw/product/product_intro.php?menu_id=3 rather expensive to me. These things have less DSP and compute to do than an ADSL modem, and should cost no more. That would make a reasonable price about US$50-60. I suspect it won't be long before we Market volume means a lot to prices... So far, the VoIP market is way behind the DSL market. That seems a rather US view of things. The practice is Asia, and particularly in Taiwan, has generally been to cut prices to the bone on new things, and try to drive market volumes from there. They made ADSL cheap when the volumes were still modest. I'm not convinced the volumes are that small. I believe Cisco alone has shipped several million lines of VoIP. That would be more than enough to get the Asian makers interested in aggressive cost reduction work. It doesn't need a lot of complex expensive and risky development. It doesn't, for example, require complex custom chips, as the DSL business did. An OMAP chip, for example, has most of what you need, except the ethernet port. Those are some of the biggest volume chips made, so prices are very keen. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
