Re: [Asterisk-Users] How to use more PCI Cards with FXO/FXS in case that TDMoE doesn't work
I suggest you go the channel bank route. On Wed, 18 Aug 2004 10:16:01 +0200 Miroslav Nachev [EMAIL PROTECTED] wrote: Hi, We have a case where we need of 16 x FXS, 12 x FXO and 1 x E1. To do this using Digium products I need of 8 PCI slots. This is not possible to be done in one computer and that's why I try to start using TDMoE. Unfortunately all my tries are without success. The network is crashed everytime. Can you give me some ideas/suggestions how to solve this case? Best Regards, Miroslav Nachev ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Semi-OT: Splitting a PRI into two PRI's?
Hi I know of a product called a Parlay which does this, but its expensive. Someone on the list said that asterisk could do this with a quad T1 card. I think that would be very nifty if asterisk could transfer the isdn calls based on CLID or DNIS before the call is actually answered. If you get this working with Asterisk, let us know! good luck Regards Clive On Tue, 10 Aug 2004 17:16:21 -0500 (CDT) Nate Carlson [EMAIL PROTECTED] wrote: On Tue, 10 Aug 2004 [EMAIL PROTECTED] wrote: The Adtran Atlas 550 or 830 may do what you're looking to do. We use it to split PRI into multiple BRI. Yeah, I've been looking into their product offerings, and it does look like they very likely do what we need. I'll try to find a reseller to tell me for sure. Thanks! | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A few questions - isdn call routing
On Wed, 4 Aug 2004 13:34:02 +0200 (CEST) Peter Svensson [EMAIL PROTECTED] wrote: On Tue, 3 Aug 2004 [EMAIL PROTECTED] wrote: There is a device called a parlay made by a crowd called voxtream which will route the ISDN calls based on the DID and/or the callerid, before the call is answered. It would be nice if this feature could be done in Asterisk as well, but at this point in time, it first answers the call. Are you sure about this? When I looked at the traces on our setup it seems that CONNECT was only sent on the incoming leg after it was received from the outgoing leg. As a graph: pstn -pri- asterisk -pri- other_device pstnasteriskother device -SETUP- dial(...) -PROCEEDING- -SETUP- -ALERTING- -PROCEEDING- -ALERTING- -CONNECT -CONNECT ACK- -CONNECT- -CONNECT ACK- Peter Peter, if thats correct, then thats great! Clive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec+packet loss concealment
Hi From what I have heard, Asterisk does not allow for iLBC to take advantage of the lost packet concealment. I understand this has something to do with the jitter processing. If lost packet concealment doesnt work with ilbc, I can assume the same applies to other codecs who claim to have this feature. Hopefully this will be fixed sometime soon, especially for us folks with less than ideal IP throughput. Regards Clive On Tue, 03 Aug 2004 10:22:20 +1000 Adam Hart [EMAIL PROTECTED] wrote: Steve Underwood wrote: Adam Hart wrote: Daniel Niasoff wrote: Is G729 more sensitive to packet loss or delays due to its higher compression. If Ive generally got the bandwidth available, am I best sticking to ulaw. G.729 has lost packet concealment, G.711 doesn't. G.711 will sound better otherwise if you can afford the bandwidth. Eh? G.729 has no particular features to allow more effective packet loss concealment. iLBC has, but at the cost of a substantially higher bit rate. In fact G.711 is a little ahead of G.729 in the regard, since packets are completely independant. The smoothing in G.729 means you need the previous packet to decode the current one properly. Regards, Steve I believe you're mistaken - G.729 works similar to iLBC and speex. iLBC works better as the packets are independent but G.729 still has a function for packet loss concealment. prehaps have a look at http://www.speex.org/comparison.html -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Herbalife Independent Distributor http://www.healthiest.co.za ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 Codec+packet loss concealment
On Tue, 3 Aug 2004 05:47:59 -0400 Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 03 August 2004 02:58, [EMAIL PROTECTED] wrote: From what I have heard, Asterisk does not allow for iLBC to take advantage of the lost packet concealment. I understand this has something to do with the jitter processing. Can you provide a source for that statement? I am not disputing it but I'd like to have it in the archives for one, but also to verify the claim too. Regards, Andrew Hi Here I am quoting Steve Davies: For IAX2, at least, Asterisk does not use the lost-packet-concealment of any codec. This is because the incoming frames clock Asterisk. For iLBC's lost packet concealment to work, Asterisk would have to start calling the decoder with a NULL at the point when the missing packet should have arrived. Can't say for sure for SIP, but I'd guess that its the same. Steve Regards Clive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Herbalife Independent Distributor http://www.healthiest.co.za ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A few questions - isdn call routing
Hi
There is a device called a parlay made by a crowd called
voxtream which will route the ISDN calls based on the DID
and/or the callerid, before the call is answered.
It would be nice if this feature could be done in Asterisk
as well, but at this point in time, it first answers the
call.
regards
Clive
On Tue, 03 Aug 2004 06:51:33 -0500
Steven Critchfield [EMAIL PROTECTED] wrote:
In trying to follow Marks advice and be nice to
newcomers, I'll just put
URLS below.
On Tue, 2004-08-03 at 06:10, Mark wrote:
We have several C/T servers with PRI lines that are
under utilised, in
the following configuration
eISDN - PRI - C/T Server 1
eISDN - PRI - C/T Server 2
eISDN - PRI - C/T Server 3
For our C/T applications we need the Dialed Number
passing from the PRI
to the C/T server - is this possible ?
Not exactly passing, but recreating.
exten = 123456,1,Dial(g2,${EXTEN})
This will essentially connect the 2 legs together and
introduce the
number on the other side.
If we install 2 or more of the Quad port ISDN cards,
and a call came in
on the first card, but was re-directed out of a second
card, is there a
dedicated bus between the cards (as with Dialogic
cards) or would it use
the Server's PCI bus ?
No, there is no Sbus or whatever it is called on
Dialogic. All calls
pass through the PCI bus. Probably covered on the Wiki
somewhere
http://www.voip-info.org/
Do you have any idea of the extra load this would put
on the CPU ?
There is a whitepaper on Digiums site discussing that.
http://www.digium.com/images/pdf/QuadCardCPUBenchmark.pdf
We also have a Samsung DCS phone switch that connects
to 4 BRI lines, do
you have or know of any product that will work with
asterisk and allow
us to connect this to the Asterisk server ?
Asterisk - 4xBRI - DCS
http://ns1.jnetdns.de/jn/relaunch/asterisk/page17.html
--
Steven Critchfield [EMAIL PROTECTED]
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Re: [Asterisk-Users] Rodopi Billing
Rodopi is a radius system. Just build your own using freeradius. Are you using Cisco that you need radius? Cheers Clive On Fri, 30 Jul 2004 11:44:14 -0700 Darren Bentley [EMAIL PROTECTED] wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? Thanks, - Darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI help - Say Number
Hi
I am trying to read a number back using the command
SayNumber. This worked fine in older versions of
asterisk, but now I am trying CVS head and I get this
error:
Jul 27 16:10:53 WARNING[507921]: file.c:1004
ast_waitstream_full: Wait failed (Interrupted system call)
The line in code is:
$AGI-exec('SayNumber',$credit);
Any ideas?
Thanks
Clive
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Re: [Asterisk-Users] IP phone recommendation
Hi Out of interest, (this may be not possible) but I think it would be an excellent idea to modify firmware to handle the IAX2 protocol. Especially since its a linux based phone. Thoughts? Regards Clive On Mon, 19 Jul 2004 21:54:59 + Joshua Colp [EMAIL PROTECTED] wrote: Hello Yiannis, I have an ipDialog SipTone II sitting right beside me. Overall it is an excellent phone but lacks codecs. It only has ulaw, alaw, and g729. The speakerphone is adequate for most things, call transferring works, holding, volume controller, conferencing, 2 lines, it pretty much all works. The interesting thing about the phone though is that it runs Linux. Thanks to ipDialog sending me the firmware I have been able to modify it slightly to get a telnet prompt available. I can't release the firmware though, who knows what trouble I could get into... but below is a snippet of info. Oh, be on the watch... I may end up selling the phone when my Ciscos come. - Joshua Colp. /proc cat version Linux version 2.4.10-uc2 ([EMAIL PROTECTED]) (gcc version 2.95.3 20010315 (release)) #1 Fri Mar 21 12:39:17 PST 2003 /proc cat cpuinfo Processor : STMicro STLC1502 rev 0 (v3l) BogoMIPS : 6.55 Hardware : STMicro STLC1502 Revision : Serial : On Monday 19 July 2004 12:04 pm, Yiannis Costopoulos wrote: Hi, I am looking for some affordable IP Phones. Any experiences with the SipToneII by ipDialog? What about soft phones? Any recommendations there (for Windoze and Linux)? Thanks, Yiannis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller based routing
Hi Just create a new context, and use ex girlfreind logic. cheers Clive On Wed, 21 Jul 2004 14:58:17 +0200 GIBERT Frédéric [EMAIL PROTECTED] wrote: Hello, Can someone explain me how to do caller based routing. Here is my example. I have an asterisk between a PBX and the PSTN. The second company get the same, and so, I can interconnect them by VoIP. Classic architecture. My problem is when I want to place fax. The calls between the 2 sites are in gsm codec. So the fax doesn?t work! Is there any possibilities to do caller based routing in asterisk, in order that when a fax try to send a fax, the call is automatically routed through the PSTN and not through the VoIP. Thanks. GIBERT Frédéric Mobile: +33 6 72 08 35 16 Fax : +33 1 30 71 39 33 Mail : [EMAIL PROTECTED] _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x100p and two hfc isdn cards
hi I had the saem trouble, so I just took my x100p card out and the problem went away:) I know its not the ultimate solution, but I decided to use an ATA with my analgue phone instead. I would suggest trying to put the analogue lines as channel 7 and the isdn lines as channels 1-6 Good luck regards Clive On Thu, 08 Jul 2004 11:52:23 +0200 Tomaz [EMAIL PROTECTED] wrote: hello, i have a problem starting asterisk with one x100p digium and two hfc chipset isdn cards with bri-stuff.0.0.2. ztcfg -vv shows me a this info: Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: Individual Clear channel (Default) (Slaves: 03) Channel 04: D-channel (Default) (Slaves: 04) Channel 05: Individual Clear channel (Default) (Slaves: 05) Channel 06: Individual Clear channel (Default) (Slaves: 06) Channel 07: D-channel (Default) (Slaves: 07) 7 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) - cat /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1 loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=2-3,5-6 dchan=4,7 and # cat /etc/asterisk/zapata.conf [channels] switchtype = euroisdn ; p2p TE mode signalling = bri_cpe ; prilocaldialplan=national pridialplan = unknown ; echocancel=yes group = 1 context=isdn channel = 2-3,5-6 group = 2 context=gsm signalling=fxs_ks channel = 1 - but when i start asterisk i got this errors: Parsing '/etc/asterisk/zapata.conf': Found Jul 8 13:53:58 WARNING[16384]: chan_zap.c:682 zt_open: Unable to specify channel 2: No such device or address Jul 8 13:53:58 ERROR[16384]: chan_zap.c:5397 mkintf: Unable to open channel 2: No such device or address here = 0, tmp-channel = 2, channel = 2 Jul 8 13:53:58 ERROR[16384]: chan_zap.c:7668 setup_zap: Unable to register channel '2-3' Jul 8 13:53:58 WARNING[16384]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 Jul 8 13:53:58 WARNING[16384]: loader.c:408 load_modules: Loading module chan_zap.so failed! Segmentation fault what to do? i have latest CVS asterisk .. thank you, Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HFC- Colongne TE Mode
Hi I got the billion card for hfc-s to work, but not with the rh9 kernel, I downloaded a new kernel 2.4.26 The trick then is to make sure you have the symbolic links correct, then it compiles and works like a dream! hope this helps. Regards Clive On Wed, 7 Jul 2004 13:07:16 +0200 Thomas Niesel [EMAIL PROTECTED] wrote: Hallo Junaid Saeed Uppal On Wed, 7 Jul 2004 15:41:49 +0500 you wrote: Hello There, I am trying to get Asterisk to work with Billion ISDN Adaptor, But i couldnt get isdn4linux to work. I am pretty new with isdn card but this is the only available option here right now , I've looked at this post and found that the author has been successful in installation of the same card. Can someone please help me out by giving the author's email address , so i can actually talk to him directly. ? or with configuration of this adaptor. I am using redhat 9 , default install. http://lists.digium.com/pipermail/asterisk-users/2004-February/037641.html i4l is the wrong way for that card! Have deep look in here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20zaphfc This will help -- Tho/\/\as ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HFC- Colongne TE Mode
Sorry, I forgot to mention, you need to use bristuff 0.0.2 thats the zaphfc driver cheers Clive n Wed, 7 Jul 2004 15:41:49 +0500 Junaid Saeed Uppal [EMAIL PROTECTED] wrote: Hello There, I am trying to get Asterisk to work with Billion ISDN Adaptor, But i couldnt get isdn4linux to work. I am pretty new with isdn card but this is the only available option here right now , I've looked at this post and found that the author has been successful in installation of the same card. Can someone please help me out by giving the author's email address , so i can actually talk to him directly. ? or with configuration of this adaptor. I am using redhat 9 , default install. http://lists.digium.com/pipermail/asterisk-users/2004-February/037641.html regards ~uppal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN Bristuff + analogue
Hi does anyone know if its possible to run Bristuff together with a tdm card in the same computer. I get an error when trying to start asterisk in chan_zap.c My zaptel.conf looks like this: loadzone=us defaultzone=us fxsks=1 fxoks=2 fxoks=3 span=1,1,3,ccs,ami bchan=4-5 dchan=6 Thanks Clive _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware Transcoder
The most economical way is just multiple asterisk boxes, even though it may use more space. On Thu, 3 Jun 2004 20:13:03 -0600 brian k. west [EMAIL PROTECTED] wrote: Go spec some hardware dsp chips and boards that can do 100 channels... I think you will fall out of your chair. bkw - Original Message - From: Isaac McDonald [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 03, 2004 2:45 PM Subject: [Asterisk-Users] Hardware Transcoder Does anyone know of a hardware transcoder? Or a software transcoder for that matter. I would consider using asterisk but it seems that Asterisk per the WIKI can only support at most 100 channels transcoding from g.711 to g.729. I would be transcoding from g.711 to g.723.1 or g.729. Thanks, IAM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware Transcoder
I have done a quick search and there are some nice looking dsp-pci cards out there. (Dunno abt prices). It may take some coding to get them working with Asterisk , and one would not require a super-power quad xeon processor if it had a huge dsp card. May be an interesting way to scale asterisk for a large install. Most of us use asterisk for smaller applications, so this is not a major concern. Good luck! On Thu, 3 Jun 2004 21:48:43 -0400 Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Thursday 03 June 2004 16:45, Isaac McDonald wrote: Does anyone know of a hardware transcoder? Or a software transcoder for that matter. I would consider using asterisk but it seems that Asterisk per the WIKI can only support at most 100 channels transcoding from g.711 to g.729. I would be transcoding from g.711 to g.723.1 or g.729. So use multiple boxes! -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bri stuff Issues
Hi all I am attempting to install bristuff, and have not had much success. I have my kernel sources installed (RH9), and am following the instructions step by step. Things seems to fall off the rails when I try make make clean all in the zaphfc directory, which is part of the install.sh script. The error messages I have included below. I am wondering if the fact that I have fxo and fxs modules installed in my machine as well is an issue. Any advice or pointers will be appreciated. Thanks and regards Clive [EMAIL PROTECTED] bri-stuff.0.0.2]# cd zaphfc [EMAIL PROTECTED] zaphfc]# make clean all rm -f zaphfc.o *.ko *.mod.c *.mod.o .*o.cmd *~ cc -c zaphfc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer -O2 -Wall -I/lib/modules/2.4.20-8/build/include -I../zaptel -Wall -DMODVERSIONS -include /lib/modules/2.4.20-8/build/include/linux/modversions.h -DCONFIG_ZAPATA_BRI_DCHANS In file included from zaphfc.c:15: /lib/modules/2.4.20-8/build/include/linux/kernel.h:60: invalid suffix on integer constant /lib/modules/2.4.20-8/build/include/linux/kernel.h:60: parse error before numeric constant /lib/modules/2.4.20-8/build/include/linux/kernel.h:61: invalid suffix on integer constant /lib/modules/2.4.20-8/build/include/linux/kernel.h:61: parse error before numeric constant /lib/modules/2.4.20-8/build/include/linux/kernel.h:62: `panic_R_ver_str' declared as function returning a function /lib/modules/2.4.20-8/build/include/linux/kernel.h:68: parse error before numeric constant /lib/modules/2.4.20-8/build/include/linux/kernel.h:68: `simple_strtoul_R_ver_str' declared as function returning a function /lib/modules/2.4.20-8/build/include/linux/kernel.h:69: invalid suffix on integer constant (a whole lot more errors)... lib/modules/2.4.20-8/build/include/linux/dcache.h:254: warning: implicit declaration of function `__out_of_line_bug_R8b0fd3c5' zaphfc.c: In function `hfc_shutdownCard': zaphfc.c:48: warning: implicit declaration of function `printk_R1b7d4074' zaphfc.c: In function `hfc_interrupt': zaphfc.c:545: warning: implicit declaration of function `sprintf_R1d26aa98' make: *** [zaphfc.o] Error 1 _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] South-Africa
My advice is just sell them. no-one I know is bothered with Icasa approval, as long as it works, its fine. That card has FCC approval, as far as I know. ALles van die beste! Regards Clive On Fri, 30 Apr 2004 15:17:13 +0100 WipeOut [EMAIL PROTECTED] wrote: Altus Snyman wrote: Good day all I'm in South-Africa,currently we are using openline4 cards for our pbx systems.Now we first need approval on the cards form icasa(a government standards) before we can use the card.The market here is very big for a system like asterisk.The only problem is to get a card approved(for a small company like us) its just about impossible. Now what I'm looking for is a company that will import an approve a card or if someone out of South-Africa now of such a card? The market is very big here Let me Know Thanks Altus Just don't tell anyone.. ;) We tried getting Modems approved in SA about 8 years ago and in the end it just wasn't worth it.. The regulators were a joke and their costs were rediculous.. It may have improved now.. Good luck.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ For super low premiums ,click here http://www.dialdirect.co.za/quote ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with IAX2?
Hi I am also having jitter trouble on IAX2, and I can vouch that the jitter buffer is busted. On Wed, 07 Apr 2004 09:56:01 -0400 Steve Kann [EMAIL PROTECTED] wrote: Andrew Kohlsmith wrote: Are there open problems/issues with iax2 and jitter (quality)? Just upgraded to today's dev cvs about an hour ago, and it seems the iax conversations are lower quality then a month or two ago. iax2 show firmware says version 13. (Test call originated from C7960 with g711.) I noticed the same thing. Jitter buffer apparently is broken, and has always been. I was advised to say jitterbuffer=no in iax.conf, but I swear it's better with it set to yes and then executing iax2 set jitter 250 in the CLI. At least it was before I cvs up'd. :-) I found a jitter buffer bug in IAX2 a short while ago. It could potentially lead to misordered frames in conversations, and does so quite often when the sender of frames is using iaxclient under win9x. I compensated for this with a change in iaxclient, but the problem could also happen in asterisk-generated frames. See : http://sourceforge.net/mailarchive/forum.php?thread_id=4096021forum_id=29380 I don't know if this is the bug people are hitting, or not, though. Jeremy (of NuFone fame) has his jitterbuffer=no on his servers and since he's my VOIP provider I tend to just try and match his setup in terms of IAX2 anyway. I dunno, I do agree with you that it seemed better a while ago. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Herbalife Independent Distributor http://www.healthiest.co.za ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial up adapter
Jason, hi Don't waste your time on old technology. This was done on the old komodo phone , sold by net2phone as a yap jack and there are some ipphones (VIC phone) comes to mind with an analogue modem built in. We even have adsl here in parts of Africa, (not that its got any bandwidth throughput) so rather go buy a nice cisco or grandstream. my 2c :) regards Clive On Mon, 1 Mar 2004 13:53:31 -0800 Jason Miller [EMAIL PROTECTED] wrote: I was wondering if anyone has used an adapter to dial up to a local internet service then used the VOIP phone instead of needing a computer. If so what product do you suggest? An idea of what I am looking for, configure the device which has a analog port to dial said ISP and authenticate Has an ethernet port to hook up to the phone Or am I just dreaming up a new product to market? Jason __ http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Jitter Buffer Configuration (typo in iax.conf)
Hi I havent been able to get the jitter buffer to work even with correct typing. If you have any luck, please let me know how it performs for you. Thanks and Regards Clive On Thu, 12 Feb 2004 19:56:27 + (GMT) Michael T Farnworth [EMAIL PROTECTED] wrote: I had noticed that the jitterbuffer settings under Asterisk didn't seem to work very well, then I noticed that there was a typo in my iax.conf file where I had: maxexccessbuffer=750 which should have been maxexcessbuffer=750 I have just realised that I didn't make this typo, it is actually a typo in the sample iax.conf file which is provided with Asterisk. People might want to take a look at their own settings and check if you have the same problem! Michael -- Michael T Farnworth Maxima Systems Ltd (http://www.maximasystems.com) 16 Woodbourne Sq Douglas Isle of Man IM1 4DB Tel: +44 (0)1624 665826 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 jitter buffer help
Steve hi Yup, adsl, seems to be getting slower by the day. Maybe we can configure * to change the iax to port 21 udp ? Regards Clive On Thu, 5 Feb 2004 13:21:08 +0200 (SAST) Stephen Davies [EMAIL PROTECTED] wrote: On Thu, 5 Feb 2004 [EMAIL PROTECTED] wrote: Hi I wonder if anyone has a fix or any advice for the IAX2 jitter buffer. My internet connection here in South Africa has an international ping time of 550ms +- 50 ms. According to the scientific approach I would like to add a 100ms jitter buffer. (nevermind the latency)! I have tried playing with maxjitterbuffer and maxexcessjitterbuffer settings, I also tried from the CLI IAX2 set jitter 700 with all kinds of parameters. Hi Clive, Are you on a Telkom ADSL line? I've found it unusable for VOIP over the last two weeks - simply not enough throughput. Its only a few prioritised ports (eg port 80 - web, 21 - ftp) that have any decent throughput. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [OT] South african laws - was [Asterisk-Users] iax2 jitter buffer help
Basically voip is only legal if used between branch offices of a company that are connected using leased lines. Archaic.. yes, stupid... yes, but thats the law here..:( Our telco is strangling the country so they can line their pockets. On Thu, 05 Feb 2004 11:57:57 + Chris Lee [EMAIL PROTECTED] wrote: On the subject of South Africa What are the laws regarding using the Internet to carry telephone traffic? What are the laws regarding connecting digium kit to Telkom equipment? As I recall they are quite restrictive, have they been eased up a bit? Regards Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 jitter buffer help
Steve, I still would love to know how to improve the jitter settings:) I still have managed a conversation, but its not great at all with the sound breaking up. Some sort of jitter control will definitly help. Thanks Clive On Thu, 05 Feb 2004 14:19:07 +0200 [EMAIL PROTECTED] wrote: Steve hi Yup, adsl, seems to be getting slower by the day. Maybe we can configure * to change the iax to port 21 udp ? Regards Clive On Thu, 5 Feb 2004 13:21:08 +0200 (SAST) Stephen Davies [EMAIL PROTECTED] wrote: On Thu, 5 Feb 2004 [EMAIL PROTECTED] wrote: Hi I wonder if anyone has a fix or any advice for the IAX2 jitter buffer. My internet connection here in South Africa has an international ping time of 550ms +- 50 ms. According to the scientific approach I would like to add a 100ms jitter buffer. (nevermind the latency)! I have tried playing with maxjitterbuffer and maxexcessjitterbuffer settings, I also tried from the CLI IAX2 set jitter 700 with all kinds of parameters. Hi Clive, Are you on a Telkom ADSL line? I've found it unusable for VOIP over the last two weeks - simply not enough throughput. Its only a few prioritised ports (eg port 80 - web, 21 - ftp) that have any decent throughput. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 jitter buffer help
Hi I wonder if anyone has a fix or any advice for the IAX2 jitter buffer. My internet connection here in South Africa has an international ping time of 550ms +- 50 ms. According to the scientific approach I would like to add a 100ms jitter buffer. (nevermind the latency)! I have tried playing with maxjitterbuffer and maxexcessjitterbuffer settings, I also tried from the CLI IAX2 set jitter 700 with all kinds of parameters. Any advice would be appreciated. Thanks Clive __ http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceGlo
I wonder which voice codec they use, they say one can use a 28k modem using their service which rules out ilbc. On Mon, 1 Dec 2003 17:34:41 -0500 Chris HARIGA [EMAIL PROTECTED] wrote: Hi, VoiceGlo is comercial version of Asterisk? :))) loo Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ ___ Look Good, Feel Good www.healthiest.co.za ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
