[Asterisk-Users] Licensed G.729 (from digium)
Hello
everyone,
Can someone tell me which annex the G.729 codec
from digium is.
Asterisk seems to thing it's Annex B (with a
warning in trasnlate.c)
[codec_g729b.so] = (Annex B (floating
point) G.729/PCM16 Codec Translator) == Detected 10 licensed G.729
transcodersWARNING[8192]: File translate.c, Line 218 (calc_cost): Translator
'g729tolinb' does not produce sample frames. == Registered translator
'g729tolinb' from format 8 to 6, cost 9 == Registered translator
'lintog729b' from format 6 to 8, cost 20But the channels like IAX only
work when you put in allow=G729 (without the B)
When having the G729 code in the h323.conf and it's
building a connection with the H.323 channel i get:
2:51.058
ThreadID=0x00020011
h323caps.cxx(1626) H323 Added capability: G.729A{n/a}
1 2:51.059
ThreadID=0x00020011
h323caps.cxx(1687) H323 Found capability: G.729A{n/a}
1
I think this may be the source of the problems we
have with incomming H.323 call Audio only working one way...
(outgoing calls do fine though)
Is there just some inconsistencywhich needs
to be fixed, or is the codec an all G.729 codec which can doboth A B
? Or do i just have my H.323 allow=G729 wrong ?
Thanks in advance,
Tjardick van der Kraan
[Asterisk-Users] The same SIP problems...SORRY!
Hi eveybody again! I don't want to be annoying, but if nobody can help me with this, I'll have to desist of working with SIP.I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones (an analog one and a DECT one) conected to it.Also, I have two Dlink dg102s with four phones conected to them. The main problems are two. Calls between the phones conected to the SIP GW and the ones conected to the MGCP GW goes OK ONLY if I call from the MGCP to the SIP. Phones at MGCP can call without problems to the PSTN (voice quality isn't very good, with silence times, but it can be supported!). But phones at SIP can't do any call! The problem is that when I pick up the callee phone, I don't hear nothing and the call goes off inbetween 4 or 5 seconds. And the caller (SIP) doesn't realise I have picked up, because It's still hearing the calling tone.When the call goes off, the caller hear the congestion tone. I don't know what is the problem The other problem is that I can't achive to transfer calls. When I dial #, it doesn't happen anything!! And the callerID doesn't work either... My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = default transfer = yes threewaycalling = yes usecallerid = yes hidecallerid = no [sip] type=friend callerid=sip username=sip host=188.208.12.37 accountcode=sip My extensions.conf exten = ,1,dial,SIP/[EMAIL PROTECTED]|60|rTt exten = ,2,Hangup I also have done a SIP debug and I'm sneding an extract of what I have found. I can't understand why the out of SIP messages go to an IP so strange!!! (229...) I can't find this IP anywhre in my system...Any ideas? Hope someone can help!! Thanks in advance! michelle PD:188.208.12.237 is the asterisk IP (...) s=session c=IN IP4 188.208.12.237 t=0 0 =audio 13532 RTP/AVP 0 a=rtpmap:0 PCMU/8000 to 229.159.241.112:5060 Retransmitting #5 (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 229.159.241.112:5060 ;branch=z9hG4bK-3a5246f7- 8c6b606-10eb From: ;tag=0-13c4-3a5246f7-8c6b604-c3a To: ;tag=as52ed0a6a Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Contact: Content-Type: application/sdp Content-Length: 135 v=0 o=root 11673 11673 IN IP4 188.208.12.237 s=session c=IN IP4 188.208.12.237 t=0 0 =audio 13532 RTP/AVP 0 a=rtpmap:0 PCMU/8000 to 229.159.241.112:5060 -- Hungup 'IAX2[test]/1' == Spawn extension (default, , 1) exited non-zero on 'SIP/229.159.241.112:5 060' set_destination: Parsing for address/port to send t o set_destination: set destination to 188.208.12.37, port 5060 Reliably Transmitting: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 188.208.12.237:5060;branch=z9hG4bK6723148d From: ;tag=as52ed0a6a To: ;tag=0-13c4-3a5246f7-8c6b604-c3a Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 188.208.12.37:5060 Sip read: SIP/2.0 200 OK From: To: ;tag=0-13c4-3a5246f7-8c6b604-c3a Call-ID: [EMAIL PROTECTED] CSeq: 102 BYE Via: SIP/2.0/UDP 188.208.12.237:5060 ;received=188.208.12.237 ;branch=z9hG4bK67231 48d Content-Length:0 7 headers, 0 lines Message is BYE - Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Installing the wcfxs driver
Hi All when I modprobe the wcfxs drive and do a cat /proc/pci, it is sharing irq with my AGP and USB, I think this is causing the card to stop working, it would work for a couple of days or a couple of hours but then stop, I'm a complete linux newbie, how can I force the wxfxs driver onto another IRQ in case it is this causing the problem Thanks for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] The same SIP problems...SORRY!
Hi!
I thought it was the SIP device too, but I have looked for avery litle comand
of this device and I can't find this Ip address, and I see that its Ip is Ok,
and I have configurated the REGISTRAR section too... I don't know what's
happening, and I don't understand that, if the IP is wrong, why can I hear
the callee phone ringing and the call only goes off when I pick it up?
it's so strange...I think!
Michelle
gt;On Mon, 16 Jun 2003, michelle matis litio wrote: gt;gt; to
229.159.241.112:5060 gt;gt; Retransmitting #5 (no NAT): gt;gt; SIP/2.0
200 OK gt;gt; Via: SIP/2.0/UDP 229.159.241.112:5060 ;branch=z9hG4bK-
3a5246f7- gt;gt; 8c6b606-10eb gt;gt; From: ;tag=0-13c4-3a5246f7-
8c6b604-c3a gt;gt; To: ;tag=as52ed0a6a gt;gt; Call-ID: A
href=javascript:sendMsg('f93b00-0-13c4-3a5246f7-8c6b602-
[EMAIL PROTECTED]');f93b00-0-13c4-3a5246f7-8c6b602-
[EMAIL PROTECTED]/A gt;gt; CSeq: 1 INVITE gt;gt; User-Agent:
Asterisk PBX gt;gt; Contact: gt;gt; Content-Type: application/sdp
gt;gt; Content-Length: 135 gt;gt; gt;gt; v=0 gt;gt; o=root 11673
11673 IN IP4 188.208.12.237 gt;gt; s=session gt;gt; c=IN IP4
188.208.12.237 gt;gt; t=0 0 gt;gt; =audio 13532 RTP/AVP 0 gt;gt;
a=rtpmap:0 PCMU/8000 gt;Hi, gt;Its being sent to that IP address,
because that is that the gt;originating SIP device put in its Via header.
gt;Also, your SIP device didn't put any From or To in its INVITE.
gt;Perhaps you could send a sip debug from the start of a SIP call
gt;attempt. gt;But I'm sure that the trouble is with your SIP Gateway
device's gt;setup. gt;Steve
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[Asterisk-Users] Queue App
I think I solved the errors I was getting with my patch, sort of anyway. Brief over view: Tell all the callers their position in the queue. When they move, tell them their new position. I was receiving Thread xxx already blocked by xxx. I found that if I only tell caller 4 and above (Which becomes caller 3) that their position changed, I do not receive the errors. http://pbx.usedontmiss.com/queue_patch So this seems to work as of now. I will be using it today, and will let everyone know. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P creating a short-circuit on line
On Sun, 15 Jun 2003, John Laur wrote: I do not think it is necessarily a hardware issue, as the line-in-use lights do not light until the wcfxo kernel module is loaded. It would be very nice for asterisk to be able to share these lines via the PBX.. That is very interesting. I have assumed that it is hardware related. Our X100P is currently plugged into a POTS port of an ISDN TA and that seems to have overcome the line test problem. However, we really would liek it to be on a PSTN line. Regards, Kwong Li [EMAIL PROTECTED] Laser Business Systems Ltd. http://www.laser.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP REGISTER behavior change: specific domains possible in REGISTER
[EMAIL PROTECTED] (John Todd) writes: Mark has fixed the REGISTER issues to be more RFC compliant. I've created a new thread so that those of you who got bored with the old thread might read this new one. The feature that has just been added was added a while ago, but now it actually seems to _work_. :-) [snip] Thanks for this. I'll try it out later. Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 Licencing..
Hi, Does the G.729 module support adding more licences?? From what I understand the module generates a code that unlocks it for a given number of licences.. I would probably want to buy 2 or 3 licences to test with and then later as I needed more add then on as needed one or two at a time.. Is this possible?? Thanks -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 Licencing..
Hiya, Yes it does. The only thing to be careful of, as we learnt to our mistake, was that a single purchase gives you a single key for all and thus you cannot buy 10 licenses intending to use some on one server and some on another. I guess this would be possible by special request though. Simon - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 1:44 PM Subject: [Asterisk-Users] G.729 Licencing.. Hi, Does the G.729 module support adding more licences?? From what I understand the module generates a code that unlocks it for a given number of licences.. I would probably want to buy 2 or 3 licences to test with and then later as I needed more add then on as needed one or two at a time.. Is this possible?? Thanks -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] Installing the wcfxs driver
Hi My Motherboard cannot disable the IRQ sharing, can I specify on with modprobe, the IRQ to be used with a particular module? Robb Quoting Emanuele Pucciarelli [EMAIL PROTECTED]: On Mon, Jun 16, 2003 at 12:05:34PM +0100, Robert Boardman wrote: for a couple of days or a couple of hours but then stop, I'm a complete linux newbie, how can I force the wxfxs driver onto another IRQ in case it is this causing the problem You usually can, you should check your motherboard's documentation. I have an Asus MB and I can effectively disable IRQ sharing for the board in the setup area reachable at boot. Bye, -- Emanueel ___ 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] Voicemail Notification
Are you using voicemail2 or voicemail? Can you confirm that /var/spool/asterisk/vm/403/INBOX has messages and/or /var/spool/asterisk/voicemail/default/403/INBOX has messages? Mark I am using voicemail2, and I can confirm that I have messages in my inbox. -Derek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queue application
Hi, Im working on a call center application where callers input some information and get transferred to an attendant, or waits in a queue until one is available. The operator is using a PC-based system that needs to have access to the information previously input by the caller. I was thinking about making * write some control info somewhere and then make the application get it through samba/file sharing. Any other insights? Also, how to make this work if the call is queued? Best regards, PHM
Re: [Asterisk-Users] G.729 Licencing..
No, you can reinstall up to 3 times I believe. - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 2:11 PM Subject: Re: [Asterisk-Users] G.729 Licencing.. What if you change the hardware? The licenses are lost? Thanks, Dan - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 3:44 PM Subject: [Asterisk-Users] G.729 Licencing.. Hi, Does the G.729 module support adding more licences?? From what I understand the module generates a code that unlocks it for a given number of licences.. I would probably want to buy 2 or 3 licences to test with and then later as I needed more add then on as needed one or two at a time.. Is this possible?? Thanks -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] Re: Applications, dialplan not loading
Just a brief progress report on the the applications and dialplan not loading: If I don't load chan_alsa.so, by using noload=chan_alsa.so in modules.conf, I do get the dialplan, apps, and etc. (I received a hint offlist from someone who had problems who'd tried a different version of this solution.) I suspect that the problem is a conflict between the libasound2 libraries in the Debian package and the libasound provided by the latest version of ALSA. I am working the issue. Problems to solve: * Resolving library issues * Determining why asterisk does not issue sufficiently complaints about chan_alsa.so (or whatever it is that's blocking loading the dialplan.) -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 Licencing..
As I understand, the key you get depend on the software hardware installation you have. If you change Asterisk to another computer (different hardware), then you still can use that codec? I have installed Asterisk on a Compaq Armada 1700 notebook (celeron/300MHz) and it works like a charm with 6 IP phones and 2 analog phones through a Cisco ATA186. I need now to add a FXO interface and for this purpose I need a system with a PCI bus. I can try the codec now on this installation (notebook) and then move it to the new system when it will be available and still keep working? Thanks, Dan - Original Message - From: Simon Woodhead [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 4:38 PM Subject: Re: [Asterisk-Users] G.729 Licencing.. No, you can reinstall up to 3 times I believe. - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 2:11 PM Subject: Re: [Asterisk-Users] G.729 Licencing.. What if you change the hardware? The licenses are lost? Thanks, Dan - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 3:44 PM Subject: [Asterisk-Users] G.729 Licencing.. Hi, Does the G.729 module support adding more licences?? From what I understand the module generates a code that unlocks it for a given number of licences.. I would probably want to buy 2 or 3 licences to test with and then later as I needed more add then on as needed one or two at a time.. Is this possible?? Thanks -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] [OF] Cable Pinouts
ethernet ? this is E1, so you need a balun you should find a 406 balun at www.patton.com or on ebay that will translate coax to rj48C On Tuesday 27 May 2003 3:45 pm, Roger Schreiter wrote: Eduardo Goncalves schrieb: ... Digium's E400P has RJ45 conector and my E1 link has BNC concetor. Could someone tell me the cable pinouts to make this conection? ... Hi, you will need a hub or a switch to connect. You can't connect your components using only passive components, since the electronic specs for twisted pair connected ethernet and coaxial connected ethernet aren't compatible at all. Roger. ___ 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] chan_capi and hanging channels
hi using chan_capi, I get _lots_ of hanging channels after a while. This was first beleived to be SIP related, but I doubt it. below, 'roy' is on MGCP, and 'fax' is just a bridged dial if someone dials in, it's re-routed to another external number roy asterisk1*CLI show channels Channel (ContextExtensionPri ) State Appl. Data CAPI[contr2/22545060] (pronto 22545060 1 )Down (None) (None) CAPI[contr1/22545070] (roy22545070 1 )Down (None) (None) CAPI[contr1/22545070] (roy22545070 1 )Down (None) (None) CAPI[contr2/22545069] (anette 22545069 1 )Down (None) (None) CAPI[contr2/22545069] (anette 22545069 1 )Down (None) (None) CAPI[contr1/22545060] (pronto 22545060 1 )Down (None) (None) CAPI[contr1/22545060] (pronto 22545060 1 )Down (None) (None) CAPI[contr2/22545060] (pronto 22545060 1 )Down (None) (None) CAPI[contr2/22545061] (fax22545061 1 )Down (None) (None) CAPI[contr2/22545061] (fax22545061 1 )Down (None) (None) CAPI[contr2/22545073] (gorm 22545073 1 )Down (None) (None) CAPI[contr2/22545073] (gorm 22545073 1 )Down (None) (None) CAPI[contr2/22545079] (ola22545079 1 )Down (None) (None) CAPI[contr2/22545079] (ola22545079 1 )Down (None) (None) CAPI[contr2/22545066] (torgeir22545066 1 )Down (None) (None) CAPI[contr2/22545066] (torgeir22545066 1 )Down (None) (None) CAPI[contr2/22545070] (roy22545070 1 )Down (None) (None) CAPI[contr2/22545070] (roy22545070 1 )Down (None) (None) 18 active channel(s) -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the wcfxs driver
No , the bios sets the irq. You can try to force an irq to the slot via the bios setup menu (ie from bios setup you can set irq 5 to pci slot 2, for ex.), or move the card to a different pci slot. In general, pci slot #1 shares with pci slot #5, #2 with #6, 3,4 with onboard facilities (agp,eth and so on). But that could not be always true. Matteo. Il lun, 2003-06-16 alle 15:22, Robert Boardman ha scritto: Hi My Motherboard cannot disable the IRQ sharing, can I specify on with modprobe, the IRQ to be used with a particular module? Robb Quoting Emanuele Pucciarelli [EMAIL PROTECTED]: On Mon, Jun 16, 2003 at 12:05:34PM +0100, Robert Boardman wrote: for a couple of days or a couple of hours but then stop, I'm a complete linux newbie, how can I force the wxfxs driver onto another IRQ in case it is this causing the problem You usually can, you should check your motherboard's documentation. I have an Asus MB and I can effectively disable IRQ sharing for the board in the setup area reachable at boot. Bye, -- Emanueel ___ 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] Whoooaaa!!! Feaky - but in a good way
If this is through your Telco, they may have turned on the Callerid-Name field along with your number. I had mine turn on the Callerid-Name field for us. -Original Message- From: Andy Powell [mailto:[EMAIL PROTECTED] Sent: Sunday, June 15, 2003 3:25 PM To: [EMAIL PROTECTED] Subject:[Asterisk-Users] Whoooaaa!!! Feaky - but in a good way Ok, this has really freaked me out, but in a good way - sort of.. I've made no changes at all to my system, save messing with ADSI. However this has nothing to do with ADSI. The thing is all of a sudden my DECT phones have started reporting caller id, and not just the number, the name too! They have never done this before in the couple of months that I've had * running. I'm pleased that they have decided to work, but I am confused and concerned as to how and why it suddenly started ... anyone got any ideas? Andy ___ 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] G.729 Licencing..
We've just moved servers and it went fine. - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 2:57 PM Subject: Re: [Asterisk-Users] G.729 Licencing.. As I understand, the key you get depend on the software hardware installation you have. If you change Asterisk to another computer (different hardware), then you still can use that codec? I have installed Asterisk on a Compaq Armada 1700 notebook (celeron/300MHz) and it works like a charm with 6 IP phones and 2 analog phones through a Cisco ATA186. I need now to add a FXO interface and for this purpose I need a system with a PCI bus. I can try the codec now on this installation (notebook) and then move it to the new system when it will be available and still keep working? Thanks, Dan - Original Message - From: Simon Woodhead [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 4:38 PM Subject: Re: [Asterisk-Users] G.729 Licencing.. No, you can reinstall up to 3 times I believe. - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 2:11 PM Subject: Re: [Asterisk-Users] G.729 Licencing.. What if you change the hardware? The licenses are lost? Thanks, Dan - Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 3:44 PM Subject: [Asterisk-Users] G.729 Licencing.. Hi, Does the G.729 module support adding more licences?? From what I understand the module generates a code that unlocks it for a given number of licences.. I would probably want to buy 2 or 3 licences to test with and then later as I needed more add then on as needed one or two at a time.. Is this possible?? Thanks -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error chan_oh323.so
Hi all, I want to install h.323 support for *, but when I launch * from shell command asterisk -vvvc I have the next error screen: [chan_oh323.so]WARNING[1024]: File loader.c, Line 226 (ast_load_resource): liboh323wrap.so: cannot open shared object file: No such file or directory WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_oh323.so failed! It can't loading chan_oh323.so, I have this module in the /usr/lib/asterisk/modules directory, but it does not recognize this library, and at the same time does not recognize liboh323wrap.so Someone has installed and using with success this oh323 package from inaccess networks ??? thanks in advance, Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error chan_oh323.so
Hi all, I want to install h.323 support for *, but when I launch * from shell command asterisk -vvvc I have the next error screen: [chan_oh323.so]WARNING[1024]: File loader.c, Line 226 (ast_load_resource): liboh323wrap.so: cannot open shared object file: No such file or directory WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_oh323.so failed! It can't loading chan_oh323.so, I have this module in the /usr/lib/asterisk/modules directory, but it does not recognize this library, and at the same time does not recognize liboh323wrap.so Someone has installed and using with success this oh323 package from inaccess networks ??? thanks in advance, Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP REGISTER
Hi! I have a new problem with my SIP device.I have done some changes and now I receive continuosly a SIP message: 501 Not impelmented back from the SIP Gateway. I can see that it doesn't register to Asterisk. I have in the SIP device: Registrar 1:UnRegisteredto: registrar: 188.208.12.237 5060expires: 2000 name: gateway passwd: 123 My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = default transfer = yes threewaycalling = yes usecallerid = yes hidecallerid = no register = gateway:[EMAIL PROTECTED]/ [gateway] type=friend callerid=sip username=gateway host=188.208.12.37 secret=123 My extensions.conf exten = ,1,dial,SIP/[EMAIL PROTECTED]|60|rTt exten = ,2,Hangup I'm going crazy with this...I think that I'm not doing well the registration but I can't find why!! 188.208.12.237 is the IP of the asterisk and 188.208.12.37 is the IP of the SIP gateway. is one of the phones of the SIP Gateway...Anyone can helpPlease! Thanks very very much Michelle - Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Local PBX
I am running Asterisk. I want to make my local PBX. I have Cisco ATA 186-I1. i want to connect two analog telephone connected to ATA 186 and make them extention to dial each other. how i can make it. Imme -- __ Sign-up for your own FREE Personalized E-mail at Mail.com http://www.mail.com/?sr=signup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Whoooaaa!!! Feaky - but in a good way
On 16/06/2003 at 10:26 DUSTIN WILDES wrote: If this is through your Telco, they may have turned on the Callerid-Name field along with your number. I had mine turn on the Callerid-Name field for us. No, not from my teleco, this is from * via the TDM card to the DECT phones that's why it spooks me... I don't have caller id on my pstn line, since it's a chargable option here in NL and I have no idea if KPN's callerid works with the Digium card. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie: isdn4linux and BRI (FRANCE)
hi i would like samples examples to configure with isdn4linux i have hisax card : gazel and an ISDN(BRI) line (2 channels B and 1D) In fist time i'll use sjphone only Perhaps there is french people on this list who can help me to do first steps with Asterisk thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:GASTMAN AUTH QUESTION
Edit /etc/asterisk/manager.conf Hi, Any of you know where to define the user and password for gastman.???PLEAS HELP ME!Alvaro Parres
Re: [Asterisk-Users] Installing the wcfxs driver
Il lun, 2003-06-16 alle 15:22, Robert Boardman ha scritto: My Motherboard cannot disable the IRQ sharing, can I specify on with modprobe, the IRQ to be used with a particular module? I do not know to what extent you can play with the kernel code in order to change how IRQ's are handled. Possibly none, but even if it is possible, I have no idea myself how to do it. Surely, though, that cannot be done with modprobe. Until now, my best solution to your problem has been moving the S400P board to a computer with a different motherboard. :( -- E. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dual T400P, SMP, performance issues
Title: RE: [Asterisk-Users] Dual T400P, SMP, performance issues Mark, As far as pings - we have cases when we could ping the box on both interfaces and there are cases when we could not (we tried 3-4 sets of NICs and drivers). All telnets, X, ssh etc. are definitely dead. No coredumps (asterisk was started with -g option), no kernel panics. Black console, Alt-SysRq combinations don't work. Pretty much no options but rebooting the box. As far as SMP and single T400P - we'll try and report the results but the idea was to go with as high density as possible ... What do you think of using hyperthreading - should we enable or disable it for the box running asterisk? What about -DCONFIG_ZAPTEL_WATCHDOG ? Can it help and how to use it? Thank you. Alex Zarubin -Original Message- From: Mark Spencer [mailto:[EMAIL PROTECTED]] Sent: Saturday, June 14, 2003 10:23 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dual T400P, SMP, performance issues When you say stops responding do you mean no more pings, telnet dead, etc? Or do you mean asterisk stops responding? Is there a segfault or kernel panic, or any other failure diagnostic? Mark On Thu, 12 Jun 2003, Alex Zarubin wrote: Zaptel was compiled with -D__SMP__ We've installed irqbalance and the picture improved a lot (thanks to Jared Smith). Do you still see problems in our /proc/interrupts? The big issue for us now is that after 24+ hours of the test load PRI-SIP our Dell PE2650, dual 2.6 GHz Xeon, 2 Gb RAM, 2 T400P, 2.4.20-18.7smp #1 SMP stops responding to anything. So the questions are: - are there known issues with PE2650 and ways to fix them? - can someone recommend the 'stable' 2.4 SMP kernel for this kind of load? - any expertise in this area will be appreciated CPU0 CPU1 CPU2 CPU3 0: 230710 30030 50050 0 IO-APIC-edge timer 1: 5 0 0 233 IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 5: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0 IO-APIC-edge rtc 14: 27 0 2 0 IO-APIC-edge ide0 20: 2085442 400221 0 230232 IO-APIC-level tor2 24: 293848 1841658 10010 570568 IO-APIC-level tor2 28: 5 25643 0 0 IO-APIC-level eth0 29: 5 0 5165040 0 IO-APIC-level eth1 30: 43720 35467 1291 3296 IO-APIC-level aacraid NMI: 0 0 0 0 LOC: 310618 310616 310616 310616 ERR: 0 MIS: 0 Thank you. Alex Zarubin -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED]] Sent: Tuesday, June 10, 2003 9:48 AM To: '[EMAIL PROTECTED]' Subject: Re: [Asterisk-Users] Dual T400P, SMP, performance issues Are you sure that you compiled zaptel for __SMP__ ? Edit your zaptel/Makefile. 0: 75283844 75241320 75286285 75247088 IO-APIC-edge timer 1: 1 0 1 1 IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 3: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0 IO-APIC-edge rtc 15: 1 0 0 1 IO-APIC-edge ide1 16: 22134870 22120997 22135905 22122829 IO-APIC-level eth0 25: 4670 4548 4614 4518 IO-APIC-level tor2 All the four CPU's should have IRQ's like in the example above. Martin On Mon, 9 Jun 2003, Alex Zarubin wrote: Hi, We are trying to validate Asterisk as a media gateway PRI - SIP with two T400P (8 T1s) per box. The first experience with BOX1 (Compaq, 2.53 GHz, 1 Gb RAM) and just one T400P was encouraging - on the load test with 3 T1s worth of calls we had on average 75% idle CPU. Not so with BOX2 (Dell, single 2.6 GHz Xeon, 1 Gb RAM, 2 T400P) and BOX3 (Dell, dual 2.6 GHz Xeon, 2 Gb RAM, 2 T400P, asterisk/zaptel is built with SMP support). On the similar load test (as with the BOX1) BOX2 was showing 0% idle CPU 70% of the time. Just 3 T1s out of 8. On the load test with just 2 T1s BOX3 was very close to 0% idle on CPU0, CPU1 was at 95% idle. The process ksoftirqd_CPU0 was close to the top of the 'top', with /proc/interrupts showing tor2 related numbers growing very fast. We had 2 T1s plugged into the first T400P board, with nothing going into the second, but the number of interrupts for the both boards was growing at the same pace. Here are the interrupts (after the box reboot, so they are not that big as they were) - do they look OK? CPU0 CPU1 CPU2 CPU3 0: 122556 0 0 0 IO-APIC-edge timer 1: 4 0 0 0 IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 5: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0 IO-APIC-edge rtc 12: 20 0 0 0 IO-APIC-edge PS/2 Mouse 14: 23 0 2 0 IO-APIC-edge ide0 20: 516930 0 0 0 IO-APIC-level tor2 24: 516524 0 0 0 IO-APIC-level tor2 28: 10600 0 0 0 IO-APIC-level eth0 29: 4837 0 0 0 IO-APIC-level eth1 30: 24831 0 0 0 IO-APIC-level aacraid NMI: 0 0 0 0 LOC: 122430 122429 122429 122428 ERR: 0 MIS: 0 Not sure what went wrong. Any suggestions on how to work with 2 T400P in a box (without hurting performance) and how to get advantage of SMP for Asterisk would be appreciated. Any known Linux kernel related issues (2.4.20-13.7smp #1 SMP for BOX3 )? Thank you. Alex Zarubin
RE: [Asterisk-Users] h323 compile error
Steven, The old releases are still on the server, they just don't provide a link on their website to access it. Here are the URLs for the openh323 code that works with chan_h323 in Asterisk. http://www.openh323.org/bin/pwlib_1.4.11.tar.gz http://www.openh323.org/bin/openh323_1.11.7.tar.gz Regards, Michael -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steven P. Donegan The openh323 code is not the 'latest cvs' but the one offered on their web site for ftp. Not sure how one would go about getting an 'old' release. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue application
Paulo Mannheimer wrote:
Hi,
Im working on a call center application where callers input some
information and get transferred to an attendant, or waits in a queue
until one is available. The operator is using a PC-based system that
needs to have access to the information previously input by the
caller. I was thinking about making * write some control info
somewhere and then make the application get it through samba/file sharing.
Any other insights? Also, how to make this work if the call is queued?
Best regards,
PHM
If your operators can use Gnophone, then all you need is to append an
URL to the Dial or Queue application.
exten =
1,1,Queue(queue,tH,http://shuttle.esoft.pf/cgi-bin/crm.pl?code=${CODE})
Then when the operator picks up the call, the URL is automatically
pushed to is Gnophone browser.
--
Jean-Denis Girard
Essential Software - Ingnierie Informatique
Solutions Linux Open Source en Polynsie franaise
http://www.esoft.pf/
Tl: (689) 54 12 95
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[Asterisk-Users] chan_h323 - pwlib 1.4.11, openh 1.11.7 comiple problems
I'm having a problem with chan_h323 compiling for Asterisk. RedHat 7.3 PWLIB 1.4.11 pwlib_1.4.11.tar.gz OpenH323 1.11.7 openh323_1.11.7.tar.gz [EMAIL PROTECTED] h323]# make clean rm -f *.o *.so core.* [EMAIL PROTECTED] h323]# make cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations chan_h323.c g++ -g -pg -c -o ast_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations ast_h323.cpp chan_h323.h:30: warning: `sockaddr_in bindaddr' defined but not used g++ -g -pg -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L/usr/src/pwlib/lib -lpt_linux_x86_r -L/usr/src/openh323/lib -lh323_linux_x86_r -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat /usr/bin/ld: cannot find -lpt_linux_x86_r collect2: ld returned 1 exit status make: *** [chan_h323.so] Error 1 [EMAIL PROTECTED] h323]# Any ideas? Thanks, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 - pwlib 1.4.11, openh 1.11.7 comipleproblems
you need to build pwlib and/or setup your environment properly. See asterisk/channels/h323/README Jeremy McNamara [EMAIL PROTECTED] wrote: I'm having a problem with chan_h323 compiling for Asterisk. RedHat 7.3 PWLIB 1.4.11 pwlib_1.4.11.tar.gz OpenH323 1.11.7 openh323_1.11.7.tar.gz [EMAIL PROTECTED] h323]# make clean rm -f *.o *.so core.* [EMAIL PROTECTED] h323]# make cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations chan_h323.c g++ -g -pg -c -o ast_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations ast_h323.cpp chan_h323.h:30: warning: `sockaddr_in bindaddr' defined but not used g++ -g -pg -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L/usr/src/pwlib/lib -lpt_linux_x86_r -L/usr/src/openh323/lib -lh323_linux_x86_r -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat /usr/bin/ld: cannot find -lpt_linux_x86_r collect2: ld returned 1 exit status make: *** [chan_h323.so] Error 1 [EMAIL PROTECTED] h323]# Any ideas? Thanks, Michael ___ 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] chan_h323 problems
I found the problem. A 'make opt' doesn't create the pwlib/lib directory when compiling pwlib. You have to do a 'make'. I did a 'make install' for h323 but I get a Segmentation Fault when I start Asterisk with chan_h323. A backtrace shows the following: (gdb) bt #0 0x42029241 in kill () from /lib/i686/libc.so.6 #1 0x46bfd5b4 in PAssertFunc () from /data/gnugk/pwlib/lib/libpt_linux_x86_r.so.1 #2 0x46c11e02 in PAssertFunc () from /data/gnugk/pwlib/lib/libpt_linux_x86_r.so.1 #3 0x4741d991 in H323EndPoint::SetLocalUserName () from /data/gnugk/openh323/lib/libh323_linux_x86_r.so.1 #4 0x47488aff in H323Gatekeeper::SetPassword () from /data/gnugk/openh323/lib/libh323_linux_x86_r.so.1 #5 0x4741798b in H323EndPoint::InternalCreateGatekeeper () from /data/gnugk/openh323/lib/libh323_linux_x86_r.so.1 #6 0x47417634 in H323EndPoint::SetGatekeeper () from /data/gnugk/openh323/lib/libh323_linux_x86_r.so.1 #7 0x41fb014c in h323_set_gk (gatekeeper_discover=0, gatekeeper=0x41fb8800 65.39.220.195, secret=0x41fb8880 ) at ast_h323.cpp:915 #8 0x41fab366 in load_module () at chan_h323.c:1646 #9 0x08053db6 in ast_load_resource (resource_name=0x80cbdab chan_h323.so) at loader.c:298 #10 0x080541ec in load_modules () at loader.c:393 #11 0x0807a39a in main (argc=2, argv=0xb894) at asterisk.c:1330 #12 0x42017499 in __libc_start_main () from /lib/i686/libc.so.6 (gdb) Regards, Micahel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, June 16, 2003 11:27 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_h323 - pwlib 1.4.11, openh 1.11.7 comiple problems I'm having a problem with chan_h323 compiling for Asterisk. RedHat 7.3 PWLIB 1.4.11 pwlib_1.4.11.tar.gz OpenH323 1.11.7 openh323_1.11.7.tar.gz [EMAIL PROTECTED] h323]# make clean rm -f *.o *.so core.* [EMAIL PROTECTED] h323]# make cc -g -pg -c -o chan_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations chan_h323.c g++ -g -pg -c -o ast_h323.o -pipe -Wall -fPIC -Wmissing-prototypes -Wmissing-declarations -DP_LINUX -D_REENTRANT -D_GNU_SOURCE -march=i686 -DPBYTE_ORDER=PLITTLE_ENDIAN -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPHAS_TEMPLATES -DPTRACING -DP_USE_PRAGMA -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -Wno-missing-prototypes -Wno-missing-declarations ast_h323.cpp chan_h323.h:30: warning: `sockaddr_in bindaddr' defined but not used g++ -g -pg -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L/usr/src/pwlib/lib -lpt_linux_x86_r -L/usr/src/openh323/lib -lh323_linux_x86_r -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat /usr/bin/ld: cannot find -lpt_linux_x86_r collect2: ld returned 1 exit status make: *** [chan_h323.so] Error 1 [EMAIL PROTECTED] h323]# Any ideas? Thanks, Michael ___ 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] .gsm files
On Sun, 2003-06-15 at 11:37, Dan wrote: Hi, There is any available GSM file player for Windows, compatible with the Asterisk GSM format? I receive the voicemail messages by mail as attachment and the sound is in GSM format. Sox can convert gsm to a wav file, also you may want to mail yourself the msgsm format which already has a header on it. - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 15, 2003 6:26 PM Subject: Re: [Asterisk-Users] .gsm files At 10:34 2003-06-15 -0400, you wrote: Hi guys, Being a true Linux geek, I've never been too much into sounds or sound files other than a few .mp3 songs I got. My question is pretty straightforward and simple. I see that the music format of choice for asterisk is .gsm. What can I use to listen to files in .gsm format and what is the most effective way of recording files into .gsm format? The sox program will convert wav into gsm: $ sox foo.wav foo.gsm does the trick. -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ 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] Dual T400P, SMP, performance issues
Yo, I've seen very similar Zaptel-related freezes on a wide variety of mainboards (SMP as well as non-SMP), with X100P's as well as with an E100P. At some point, almost always at the moment a call through one of those cards connects or disconnects, the machine completely stops responding and needs a reset to come back to life. A very nice way to trigger it with the E100P seems to be to put around 10-20 channels of it into a meetme-conference and then issue the stop now-command on the Asterisk-console. A high volume of connects / disconnects seems to trigger the freezes. I'm still investigating the issue and am going to try different kernels and some custom kernel-patches. One of my boxes (dual PIII-750, Intel L440GX+-board) with an X100P and a TDM40P in it hasn't frozen since I installed kernel 2.4.21-rc2 with the ACPI-patch (http://sourceforge.net/projects/acpi/). I'll probably try that on the box with the E100P first. Be sure enable Power Management support in your kernel-config, disable APM, enable ACPI and check all ACPI-options, except for CPU Enumeration Only. Note that this ACPI- patch also handles IRQ-routing and might help in cases where the BIOS assigns the same IRQ to some devices (or, as was the case for me, none at all). Grtz, Oliver On Mon, Jun 16, 2003 at 13:03:20 -0500, Alex Zarubin wrote: Mark, As far as pings - we have cases when we could ping the box on both interfaces and there are cases when we could not (we tried 3-4 sets of NICs and drivers). All telnets, X, ssh etc. are definitely dead. No coredumps (asterisk was started with -g option), no kernel panics. Black console, Alt-SysRq combinations don't work. Pretty much no options but rebooting the box. As far as SMP and single T400P - we'll try and report the results but the idea was to go with as high density as possible ... What do you think of using hyperthreading - should we enable or disable it for the box running asterisk? What about -DCONFIG_ZAPTEL_WATCHDOG ? Can it help and how to use it? Thank you. Alex Zarubin -Original Message- From: Mark Spencer [mailto:[EMAIL PROTECTED] Sent: Saturday, June 14, 2003 10:23 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Dual T400P, SMP, performance issues When you say stops responding do you mean no more pings, telnet dead, etc? Or do you mean asterisk stops responding? Is there a segfault or kernel panic, or any other failure diagnostic? Mark On Thu, 12 Jun 2003, Alex Zarubin wrote: Zaptel was compiled with -D__SMP__ We've installed irqbalance and the picture improved a lot (thanks to Jared Smith). Do you still see problems in our /proc/interrupts? The big issue for us now is that after 24+ hours of the test load PRI-SIP our Dell PE2650, dual 2.6 GHz Xeon, 2 Gb RAM, 2 T400P, 2.4.20-18.7smp #1 SMP stops responding to anything. So the questions are: - are there known issues with PE2650 and ways to fix them? - can someone recommend the 'stable' 2.4 SMP kernel for this kind of load? - any expertise in this area will be appreciated CPU0 CPU1 CPU2 CPU3 0: 230710 30030 50050 0IO-APIC-edge timer 1: 5 0 0233IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 5: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0IO-APIC-edge rtc 14: 27 0 2 0IO-APIC-edge ide0 20:2085442 400221 0 230232 IO-APIC-level tor2 24: 2938481841658 10010 570568 IO-APIC-level tor2 28: 5 25643 0 0 IO-APIC-level eth0 29: 5 05165040 0 IO-APIC-level eth1 30: 43720 35467 1291 3296 IO-APIC-level aacraid NMI: 0 0 0 0 LOC: 310618 310616 310616 310616 ERR: 0 MIS: 0 Thank you. Alex Zarubin -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 10, 2003 9:48 AM To: '[EMAIL PROTECTED]' Subject: Re: [Asterisk-Users] Dual T400P, SMP, performance issues Are you sure that you compiled zaptel for __SMP__ ? Edit your zaptel/Makefile. 0: 75283844 75241320 75286285 75247088IO-APIC-edge timer 1: 1 0 1 1IO-APIC-edge keyboard 2: 0 0 0 0 XT-PIC cascade 3: 0 0 0 0 IO-APIC-level usb-ohci 8: 1 0 0 0IO-APIC-edge rtc 15: 1 0 0 1IO-APIC-edge ide1 16: 22134870 22120997 22135905 22122829 IO-APIC-level eth0
Re: [Asterisk-Users] .gsm files
use wav49 for the format. On Mon, 2003-06-16 at 14:44, Dan wrote: Hi Steve, ..., also you may want to mail yourself the msgsm format which already has a header on it. How can I do this from Asterisk? Thanks, Dan - Original Message - From: Moshe Yudkowsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, June 15, 2003 6:26 PM Subject: Re: [Asterisk-Users] .gsm files At 10:34 2003-06-15 -0400, you wrote: Hi guys, Being a true Linux geek, I've never been too much into sounds or sound files other than a few .mp3 songs I got. My question is pretty straightforward and simple. I see that the music format of choice for asterisk is .gsm. What can I use to listen to files in .gsm format and what is the most effective way of recording files into .gsm format? The sox program will convert wav into gsm: $ sox foo.wav foo.gsm does the trick. -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ 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 ___ 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
[Asterisk-Users] 1X1 PBX
I have Asterisk and Cisco ATA 186. How i can make small PBX. let me know the step and configuration made in conf files. Imme -- __ Sign-up for your own FREE Personalized E-mail at Mail.com http://www.mail.com/?sr=signup ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP REGISTER
I'm afraid I have no idea what your goal is here. Do you have a phone somewhere in this configuration? I don't see it. Please explain what it is you are trying to do. From what I see (though much data is missing from your explanatin) anytime you place a call, it will result in a loop. While you're at it, include the following information: sip show peers sip show registry sip debug (and wait for a cycle of SIP messages to go by) JT Hi! I have a new problem with my SIP device.I have done some changes and now I receive continuosly a SIP message: 501 Not impelmented back from the SIP Gateway. I can see that it doesn't register to Asterisk. I have in the SIP device: Registrar 1:UnRegisteredto: registrar: 188.208.12.237 5060expires: 2000 name: gateway passwd: 123 My sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 context = default transfer = yes threewaycalling = yes usecallerid = yes hidecallerid = no register = gateway:[EMAIL PROTECTED]/ [gateway] type=friend callerid=sip username=gateway host=188.208.12.37 secret=123 My extensions.conf exten = ,1,dial,SIP/[EMAIL PROTECTED]|60|rTt exten = ,2,Hangup I'm going crazy with this...I think that I'm not doing well the registration but I can't find why!! 188.208.12.237 is the IP of the asterisk and 188.208.12.37 is the IP of the SIP gateway. is one of the phones of the SIP Gateway...Anyone can helpPlease! Thanks very very much Michelle - Tu cuenta de correo gratuita Mixmail con Antivirus y Antispam http://mixmail.ya.com Ya.com ADSL Home 24h, Módem + Alta + 1 mes Gratis http://acceso.ya.com/adslhome24h/ ___ 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] Queue App
This patch is still not bug-free. It was causing my server to crash without warning... One of these days I will understand Mark's full plan and outline. It is hard, since I am only looking at a few of the apps that are there, and have not spent too much time on the whole thing. John On Monday, June 16, 2003, at 04:47 PM, John Todd wrote: Haven't tried this one, but please make sure to get Mark to put it in the standard distribution if it gets approved and is bug-free. This sounds like something my customers will want once they get things settled with Asterisk, and I'd hate to have it just be a patch that gets lost in the mailing list somewhere... JT I think I solved the errors I was getting with my patch, sort of anyway. Brief over view: Tell all the callers their position in the queue. When they move, tell them their new position. I was receiving Thread xxx already blocked by xxx. I found that if I only tell caller 4 and above (Which becomes caller 3) that their position changed, I do not receive the errors. http://pbx.usedontmiss.com/queue_patch So this seems to work as of now. I will be using it today, and will let everyone know. John ___ 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] How do i make best use of Macro?
To comply with Marks request, here is a new patch against app_meetme.c
that copies data to a localbuffer. While I feel moderately comfortable
with this patch, please review my usage of strsep, malloc, and free.
When I run make from the apps/ directory I get a error message about
strsep passing incompatible argument, and under the root level make file
I get several messages that are similar to the 1 from the apps/
directory. As for the malloc and free, I would just feel better if
someone checks my sanity.
On Fri, 2003-06-13 at 11:38, Mark Spencer wrote:
data should be considered a const void * in all honesty (maybe i should
enforce it). Always copy to a local buffer before strseping it.
mark
On Wed, 11 Jun 2003, Steven Critchfield wrote:
Since there was some interest in this, here is the diff against current
cvs. Someone that is better at C should look into my use of strsep
because there is a couple of warnings. Also there is a warning on my use
of pbx_builtin_setvar_helper, but I can't see whats wrong here.
BTW, SayNumber doesn't seem to say '0'.
Usage is like this.
exten = 1234,1,MeetMeCount(1234|var)
exten = 1234,2,SayNumber(${var})
exten = 1234,3,MeetMe(1234)
-
diff -U3 -r asterisk-orig/apps/app_meetme.c asterisk/apps/app_meetme.c
--- asterisk-orig/apps/app_meetme.c 2003-06-11 23:14:38.0 -0500
+++ asterisk/apps/app_meetme.c 2003-06-11 22:58:32.0 -0500
@@ -54,9 +54,10 @@
'q' -- quiet mode (don't play enter/leave sounds)\n;
static char *descrip2 =
- MeetMeCount(confno): Plays back the number of users in the specified MeetMe\n
-conference. Returns 0 on success or -1 on a hangup. A ZAPTEL INTERFACE\n
-MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY.\n;
+ MeetMeCount(confno[|var]): Plays back the number of users in the specifiedi\n
+MeetMe conference. If var is specified, playback will be skipped and the value\n
+will be returned in the variable. Returns 0 on success or -1 on a hangup.\n
+A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY.\n;
STANDARD_LOCAL_USER;
@@ -465,19 +466,29 @@
int res = 0;
struct conf *conf;
int cnt;
+ char* confnum;
+ char val[5] = 0; /* I don't think we will ever get 99,999 callers into a
single meetme */
+
if (!data || !strlen(data)) {
ast_log(LOG_WARNING, MeetMeCount requires an argument (conference
number)\n);
return -1;
}
LOCAL_USER_ADD(u);
- conf = find_conf(data, 0);
+ confnum = strsep((char*) data,|);
+ conf = find_conf(confnum, 0);
if (conf)
cnt = conf-users;
else
cnt = 0;
- if (chan-_state != AST_STATE_UP)
- ast_answer(chan);
- res = ast_say_number(chan, cnt, , chan-language);
+ if(strlen(data)){
+ /* have var so load it and exit */
+ sprintf(val,%i,cnt);
+ pbx_builtin_setvar_helper(chan,(char*) data,val);
+ }else{
+ if (chan-_state != AST_STATE_UP)
+ ast_answer(chan);
+ res = ast_say_number(chan, cnt, , chan-language);
+ }
LOCAL_USER_REMOVE(u);
return res;
}
--
Steven Critchfield [EMAIL PROTECTED]
___
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--
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--- apps/app_meetme.c 2003-06-16 16:11:53.0 -0500
+++ /home/critch/app_meetme.c 2003-06-16 16:11:18.0 -0500
@@ -54,9 +54,10 @@
'q' -- quiet mode (don't play enter/leave sounds)\n;
static char *descrip2 =
- MeetMeCount(confno): Plays back the number of users in the specified MeetMe\n
-conference. Returns 0 on success or -1 on a hangup. A ZAPTEL INTERFACE\n
-MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY.\n;
+ MeetMeCount(confno[|var]): Plays back the number of users in the specifiedi\n
+MeetMe conference. If var is specified, playback will be skipped and the value\n
+will be returned in the variable. Returns 0 on success or -1 on a hangup.\n
+A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY.\n;
STANDARD_LOCAL_USER;
@@ -465,19 +466,37 @@
int res = 0;
struct conf *conf;
int cnt;
+ char *confnum,*localdata,*mhandle;
+ char val[5] = 0; /* I don't think we will ever get 99,999 callers into a
single meetme */
+
if (!data || !strlen(data)) {
ast_log(LOG_WARNING, MeetMeCount requires an argument (conference
Re: [Asterisk-Users] 1X1 PBX
http://www.digium.com/handbook-draft.pdf matteo. Il lun, 2003-06-16 alle 22:24, Imran Muneer ha scritto: I have Asterisk and Cisco ATA 186. How i can make small PBX. let me know the step and configuration made in conf files. Imme ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P questions
On Mon, Jun 16, 2003 at 12:39:20AM -0500, Asterisk wrote: Hello! I've been following this list for several weeks now and would like to purchase some hardware for a VOIP/ voice-mail solution. This card appears strikingly similar to a modem. Is it? Is there a product to bring in more Well it has an RJ12 port and plugs into your computer and is green and has some chips and stuff on it, it is very similar looking :-) I think it is very similar in hardware and concept to a winmodem/linuxmodem/softwaremodem. than one POTS line short of a full T1? It just seems silly that the technology hasn't advanced any further than to have a single line per card. Well there are some products which may fit your bill, but will take more of your bills: www.voicetronix.com sells a 4 port PCI FXO at about AU$1000, a 6 port FXO/FXS (jumper selectable) at about AU$1500, and a 12 port PCI FXO/FXS (jumpers) at about AU$3000. (AU$1.50 =~ US$1) Also Dialogic: Dialogic D/41JCT-LS 4-port analog + voice - US$882.29 VFX/41JCT-LS 4-port analog + voice + fax - ? D/120JCT-LS 12-port analog + voice - US$1500 ebay cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Fw: [Asterisk-Users] X100P questions
Well there are some products which may fit your bill, but will take more of your bills: www.voicetronix.com sells a 4 port PCI FXO at about AU$1000, a 6 port I think their 4 port card is FXS, can it be FXO as well? -Matt I'm sorry, I pick a bad week to stop sniffing glue, you are entirely correct. Aloha, Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P questions
Is there a product to bring in more than one POTS line short of a full T1? It just seems silly that the technology hasn't advanced any further than to have a single line per card. We are working on an FXO module for the TDM400P and hope to have it ready in a couple of months for initial testing, but no firm deadline. When that's ready we will have a multiport solution. In the mean time, there is always VoiceTronix, which now has support within Asterisk contributed. Does that mean that the Voicetronix is navibly supported by Asterisk? -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_h323 problems
I've done this, with the exact versions you state, 3 times today - every one does the full , proper thing. I did: cd pwlib;make clean;make opt;make install cd ../openh323;make clean;make opt;make install cd ../asterisk/asterisk/channels/h323;make clean;make install;make samples works every time on a clean RedHat 7.2 100% install I hope something in there helps... - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 16, 2003 8:20 PM Subject: RE: [Asterisk-Users] chan_h323 problems I did RTFM. It looks like the instructions conflict each other. Here's what it says: 4. Build the debug and release versions of the PWLib library as follows: cd $PWLIBDIR make both Your README under channels/h323/README says: cd /path/to/pwlib make clean opt Which one do I follow? If I do a 'make opt' it won't build the libs in pwlib. I tried it twice, 'make opt' won't build it but 'make both' will. I'm using PWLib 1.4.11 and Openh323 1.11.7. If I've misread something, please let me know. Asterisk now loads without core dumping (chan_oh323 was installed, it's been removed now). Although, the outgoing quality of the call is very choppy. Incoming works fine, no problems. Any idea what would cause outgoing calls to have problems? I'm sending these calls to GnuGK which then sends the calls to a Quintum or Cisco H323 Gateway (both are having the same problem). Regards, Michael No.. you MUST do a make opt. RTFM http://www.openh323.org/build.html ___ 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] ProSLIC error message
I have just updated to the current CVS from CVS of 12 June and I now receive the following error message when I start *. Freshmaker version: 62 Freshmaker passed register test Module 0: Initialized Module 1: Initialized ProSLIC on module 2 failed to powerup within 510 ms Unable to do INITIAL ProSLIC powerup on module 2 Module 2: Not installed Module 3: Initialized Found a Wildcard FXS: Wildcard S400P Prototype (4 modules) Any help appreciated. Richard Scobie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
