Re: [Asterisk-Users] Asterisk Live! CF
On Thursday 09 Jun 2005 23:45, Andrew Kohlsmith wrote: On Thursday 09 June 2005 13:15, Bob Goddard wrote: The Via processors emulate the i686 just fine. The problem has always been with GCC. Got some proof of that? It's generally regarded as common knowlege in these circles that the via processors claim 686 compatibility but lack some 686-specific instructions (CMPXCHG among them), and this is what causes the trouble. GCC says 686 instructions, ok. and the Via throws a fit (SIGILL) when seeing the ones it doesn't support. The Via C3 processors lack the CMPXCHG8B (CMOV) instructions and I assume others which are listed in the Intel documents as being optional. GCC assumes that they are always there. Look at http://radagast.bglug.ca/epia/epia_howto/x1098.html, section 13.2. This has been well documented. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is it possible to have a remote Phone work behind Nat without a VPN?
Hi Everyone, Is it possible to have a SIP Phone work remotely if it's behind a Router performing NAT without connecting the Router to a VPN? The Asterisk Box will be in the DMZ. Thanks Dan CYTEXONE Dan Levine [EMAIL PROTECTED] CYTEXONE | Your Technology Specialists 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization
I would be willing to Pay $500 for a good Asterisk / Exchange Integration Dan Levine [EMAIL PROTECTED] CYTEXONE | Your Technology Specialists R 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson Sent: Friday, June 10, 2005 12:53 AM To: 'George Pajari '; 'Asterisk Users Mailing List - Non-Commercial Discussion ' Subject: RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization (b) Anyone interested if we post a bounty? Post it and I'll see what can be done. I've been thinking about this and a watcher on the Exchange server, as Race suggests, is probably the way to go. As to deleting the voicemail, probably scp or something like that would work fine. I have good experience with MAPI and CDO; I've coded an Outlook Web Access replacement for my company that works fine. Make sure you are specific in the requirements, as there are probably a couple of dozen ways this can be implemented. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP-INFO
James H. Thompson wrote: Voip-info is back up -- in-spite of Murphy's law. This was phase I (install latest version of O/S) of an upgrade to improve performance and functionality. Hopefully with Phase II we will see much better performance and new functions. For those that asked, the primary voip-info-org sponsor: www.commpartners.us http://www.commpartners.us provides a dedicated server, bandwidth and hosting in their Las Vegas data center. Its slow not for any lack of resoruces, but because the software used is rather resource intensive. I would like to use this moment to say a big THANK YOU from the community to you and Commpartners for providing this resource to the community... /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap Clocking - Frame Slips - tdm400p wcfxo zttest cpu spikes spandsp
I've made some modifications to zttest in order to use
it as a frame clock accuracy tester / slip detector.
I'm not certain if that was it's original purpose, but it
seems that a lot of folks try to use it that way.
The result is something that I'm calling ztclock for now
to help avoid confusion.
I'm including the source at the end of this post. You
can compile it by placing the source in your zaptel
source directory as ztclock.c and building with:
cc -I. -O3 -g -Wall ztclock.c -o ztclock
Here is a sample run of the test:
-- snip --
./ztclock
ztclock - clock source accuracy test (3 passes)
Flushing input buffer...
Flush Complete.
Test is approximately 3 minutes. Please wait...
483328 samples in 60.410900 sec. (483288 sample intervals) 99.991722%
483328 samples in 60.410901 sec. (483288 sample intervals) 99.991722%
483328 samples in 60.410899 sec. (483288 sample intervals) 99.991722%
Estimate 8 frame slips every 12.083200 seconds.
-- snip --
Background:
During routine codec/dimensioning testing, I observed a strange,
recurring cpu spike occuring appoximately every 12 seconds on a
completely idle system with only the zaptel drivers loaded. I used
'vmstat 1' to monitor this. I was using the wcfxo driver (wildcard
fxo) as a timing source. After switching to the ztdummy driver and
using the usb controller as a clock source, I observed that the frequency
of the CPU spikes changed to approximately every 5.5 seconds. Hmm...
Also, as additional channels were added to the system, the CPU spikes
increased in intensity (ie: from 15% utilization to 50% each spike).
Further research indicated that many folks using various flavors of
FXO cards were experiencing similar observations as well as problems
with data applications such as spandsp. Some of them are observing
similar CPU spikes using the TDM400P hardware. I tried to use zttest
to put a finger on the problem, but could not seem to get the resulting
math to work out. I believe that ztclock may provide some insight into
what is happening with those spikes. The test attempts to first determine
the accuracy of your clock source compared to the results that could be
expected from a true 8khz clock source. Once this is accomplished, it
uses the results of the third pass in an attempt to calculate the time
frequency that you should expect to see 8 frame slips. I selected 8
frame slips for the calculation instead of 1 because it appears that the
zaptel driver moves 8 samples / interrupt / channel. It also appears
that the data is clocked across the PCI bus in a similar manner.
My calculated frame slip result seems to correspond directly with the
frequency of the CPU spikes that I observed, suggesting a possible
relationship.
I'd like to hear from anyone who may be tracking this or similar issues.
I'd also like to hear some feedback on the ztclock program itself in terms
of how it seems to work against various clock sources that you may have
available.
Thank you. Source Follows...
--snip--
#include stdio.h
#include stdlib.h
#include unistd.h
#include errno.h
#include string.h
#include fcntl.h
#include sys/time.h
#include sys/signal.h
#include math.h
int main(int argc, char *argv[])
{
int fd;
int res;
int count=0;
int pass=1;
int lastcount;
char buf[1024];
float score;
float t_usec;
float t_sec;
float t_intervals;
float sf;
struct timeval start, now;
fd = open(/dev/zap/pseudo, O_RDWR);
if (fd 0) {
fprintf(stderr, Unable to open zap interface: %s\n,
strerror(errno));
exit(1);
}
printf(\n\nztclock - clock source accuracy test (3 passes)\n);
/* Flush input buffer */
printf(\nFlushing input buffer...\n);
gettimeofday(start, NULL);
for (count = 0;count 64; count++)
res = read(fd, buf, 1024);
gettimeofday(now, NULL);
count = 0;
start = now;
printf(Flush Complete.\n\nTest is approximately 3 minutes.
Please wait...\n);
for(;;) {
res = read(fd, buf, 1024);
if (res 0) {
fprintf(stderr, Failed to read from pseudo
interface: %s\n, strerror(errno));
exit(1);
}
count += res;
if (count = 483328) {
gettimeofday(now, NULL);
t_usec = ((float)now.tv_sec - (float)start.tv_sec)
* 100;
t_usec += ((float)now.tv_usec -
(float)start.tv_usec);
t_sec = t_usec / 100;
t_intervals = ceil(t_usec / 125);
start = now;
printf(\n%d samples in %f sec. (%d sample
intervals) , count, t_sec, (int)t_intervals);
score = 100.0 - 100.0 * fabs((float)count -
t_intervals) / (float)count;
printf(%f%%
Re: [Asterisk-Users] IAX2 Max Retries dropped calls Firefly
There's an update to Firefly on Virbiage http://www.virbiage.com/firefly/download/firefly-thirdparty.exe lots of bug fixes - see if that helps -Adam Paul Redstone wrote: Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using Firefly 1.9.8 build 3945. However I cannot work out what this message means. There is some suggestion in when it occurs that it might be an IP connection issue from the softphone to the asterisk box. Connection is in one office via 100 M switches, very simple direct path. Firefly running Windows XP SP2. We're planning to try another softphone but quite like Firefly. Can anyone advise on this? Thanks Paul === Log extract -- Hungup 'Zap/1-1' == Spawn extension (bodiam, NN, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/ 10' -- Hungup 'IAX2/[EMAIL PROTECTED]/10' -- Registered '355' (AUTHENTICATED) at -- Registered '354' (AUTHENTICATED) at -- Accepting AUTHENTICATED call from requested format = 1024 , actual format = 1024 -- Executing Macro(IAX2/[EMAIL PROTECTED]/11, bodiam-iaxsip|352|IAX2/352) in new s tack -- Executing Dial(IAX2/[EMAIL PROTECTED]/11, IAX2/352|20|tT) in new stack -- Called 352 -- Call accepted by (format ilbc) -- Format for call is ilbc -- IAX2/352/15 is ringing -- IAX2/352/15 answered IAX2/[EMAIL PROTECTED]/11 -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/11 and IAX2/352/15 May 17 11:47:56 WARNING[2763]: chan_iax2.c:1480 attempt_transmit: Max retries ex ceeded to host on IAX2/[EMAIL PROTECTED]/7 (type = 6, subclass = 2, ts=3800 76, seqno=66) May 17 11:47:56 WARNING[2763]: chan_iax2.c:1480 attempt_transmit: Max retries ex ceeded to host on IAX2/[EMAIL PROTECTED]/7 (type = 6, subclass = 11, ts=380 079, seqno=67) -- Hungup 'Zap/2-1' == Spawn extension (bodiam, NN, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/ 7' -- Hungup 'IAX2/[EMAIL PROTECTED]/7' -- Hungup 'IAX2/352/15' == Spawn extension (macro-bodiam-iaxsip, s, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/11' in macro 'bodiam-iaxsip' == Spawn extension (bodiam, 352, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/11' -- Hungup 'IAX2/[EMAIL PROTECTED]/11' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PHPAGI Swift Escape Digits
I am trying to use swift in PHP/AGI. function swift($text, $escape_digits='', $frequency=8000, $voice=NULL, $fnameIn='') During swift speaking some text I want the caller to be able to press 1, 2 or 3 to do thing 1, thing 2 or thing 3. How are these digit defines and then caught? Thanks, Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Play MP3 during Record
Hallo, You are nearly right. We are working with some artists and they have many funny ideas with Asterisk. Regarding my question, the fact is that we can do this technically with any PC: you play a music file with RealPlayer and at the same time another music file with Winamp...So theoretically, it is possible to do so with asterisk. Regards, El jue, 09-06-2005 a las 00:48, Phuong Nguyen escribió: 1. Play a low background music when the user record his/her voice You Want a Karaoke? lol Regards, -- Ing CIP Alejandro Celi Mariátegui [EMAIL PROTECTED] -- Geschenkt: 3 Monate GMX ProMail gratis + 3 Ausgaben stern gratis ++ Jetzt anmelden testen ++ http://www.gmx.net/de/go/promail ++ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ENUM NL dead ?
Hi Michiel, -Original Message- Since you already have done something on this, can you tell us what your plan was? Complex :) ENUM was a part of a larger setup concerning roll-out of voip technology over wireless networks. Do you already have some docs about what to do and why, or do we have to setup something like this ? Motivations can be numerous, everyone needs to decide for themselves. A tool with guidelines (very rudimentary) would be usefull. Maybe it's a good idea to talk about this face to face (or in a conference call with some interested ppl) I have web/mail/dns/sip/iax2 services I can make available for this. Think so. I have made contact with dgtp again, hopefully something will come out of that in the next few days. Let's take this discussion off-list. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sirrix NT mode
Good day all Is there someone who's got a sirrix 4 port working in NT mode I got one working good in TE mode. Apparently I must add 8 jumpers in make the cross cable a straight cable But what about the sirrix.conf? Do I just change the mode from TE to NT? Please Help or advice? Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP2000 and hint LED's
That is the entire package as it was submitted to us from Grandstream. Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Peter Svensson [mailto:[EMAIL PROTECTED] Sent: Friday, June 10, 2005 1:46 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] GXP2000 and hint LED's On Thu, 9 Jun 2005, The VoIP Connection wrote: This is supposed to be the final version: http://www.thevoipconnection.com/Downloads/GXP2000_1.0.1.9/GXP2000_Rel ease_1 .0.1.9.zip Have you received an updated tftp config template as well? We asked for and received one with a 1.0.1.9 early beta version. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GXP2000 and hint LED's
On Fri, 10 Jun 2005, The VoIP Connection wrote: Have you received an updated tftp config template as well? We asked for and received one with a 1.0.1.9 early beta version. That is the entire package as it was submitted to us from Grandstream. We requested and received the template separate from the firmware release. Without the template the phones can not be mass-deployed easily. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM04B
I did that once on a cheap linejack card. Took a week to get the smell out of the office, and the bright orange from inside the server was quite interesting :) Only took 1 second to start a small flame going, but fortunately I cought it quick. I wonder if the zaptel cards have any kind of protection from this sort of thing... Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Thursday, June 09, 2005 6:37 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] TDM04B On Thursday 09 June 2005 08:21, Ariel Batista wrote: This board is FXO which you plug incoming phone lines into it. So plugging in a handset unless it's a butt set it will not give you any dial tone. In fact you damage the port doing this to it. A butt set will not give you dialtone either. And plugging telephones into FXO ports will *not* damage anything, since both the phone and the card are expecting the other side to source battery and ring. It's plugging POTS lines into FXS ports that causes nastiness. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATTN: Keith
On Thu, 2005-06-09 at 16:00 -0400, list wrote: according to RFC's your required to have reverse lookups on ur mail server, so blocking based on this is perfectly legitimate. My ISP has the option of reverse lookups, I still get blocked by some other ISPs :( -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATTN: Keith - Seriously OT
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Friday, 10 June 2005 5:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ATTN: Keith On Thu, 2005-06-09 at 16:00 -0400, list wrote: according to RFC's your required to have reverse lookups on ur mail server, so blocking based on this is perfectly legitimate. My ISP has the option of reverse lookups, I still get blocked by some other ISPs :( What are the Reject messages that you are getting. There are many reasons for having email blocked, and rDNS is not the primary one (by a long shot) Taking a look at the block lists...: 81.56.129.44 is listed in dynablock.njabl.org. It seems that your IP is part of a dial-up pool? (guessing) Your rDNS seems ok, but your MTA is greeting the recipient MX with a forged HELO or EHLO Received: from source ([81.56.129.44]) by exprod5mx8.postini.com ([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT Your MTA claimed it was called SOURCE but rDNS tells the recipient MX that it is called: mail.linuxautrement.com If you fix this, then perhaps your problems deliverg email will go away? Regards, T ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729AB codec support
Hello, Does Asterisk support G.729AB and does anyone know how to enable G.729AB codec? s it free? Thanks for your interest. Erdem HAKI [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Request OPTION and 404 Sjphone Xlite
Hi, I have install asterisk and it works fine. But when I use Sjphone and I use Ethereal a Client send Request:OPTIONS sip:obelix.foo and Server answer Status: 404 Not found. But i can talk with two client and asterisk. When I use Xlite i don't have this request it's clean. I don't understand?? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] lost g729 lic
Good day all We installed a box a long time ago and they bought g729a licenses Now we want to upgrade and reinstall,whats going to happen with the codec,if I give the box the same ip as always will it work? Please Help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729AB codec support
See this: http://lists.digium.com/pipermail/asterisk-users/2005-June/110524.html Free for non-commercial use. - Original Message - From: Erdem HAK [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, June 10, 2005 11:53 AM Subject: [Asterisk-Users] G.729AB codec support Hello, Does Asterisk support G.729AB and does anyone know how to enable G.729AB codec? s it free? Thanks for your interest. Erdem HAKI - [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help for conference
hi all
i have * box with 4 ext using sj phone, i wanna try to
make a conference. i am using ztdummy and look fine
when i install it because there is no erros message. I
checked with lsmod the zaptel and usb-uhci using
ztdummy. but why i still get error says no
application meetme ...
here is my ext.conf
exten = _66XX,1,NoOP(call for ${EXTEN})
exten = _66XX,2,Dial(SIP/${EXTEN},60,tr)
exten = _66XX,3,congestion
exten = 6690,1,meetme,1234
exten = 6691,1,meetme,2345
my meetme.conf
[room]
conf = 1234
conf = 2345,111
please advice
thks
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Re: [Asterisk-Users] G.729AB codec support
That is completely wrong, the intel code might be free for non commercial use, but you will still need a license to operate the g729, whoever wrote the code. The cost for 1 channel is 10$, and you can buy the only legal codec from digium (www.digium.com). Zoa. Soner Tari wrote: See this: http://lists.digium.com/pipermail/asterisk-users/2005-June/110524.html Free for non-commercial use. - Original Message - From: Erdem HAK [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, June 10, 2005 11:53 AM Subject: [Asterisk-Users] G.729AB codec support Hello, Does Asterisk support G.729AB and does anyone know how to enable G.729AB codec? s it free? Thanks for your interest. Erdem HAKI - [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729AB codec support
I'm saying free for non-commercial use, you're saying Intel is free for non-commercial use. And I point to the Intel code. And there is no fee for the licence for non-commercial use. So what is completely wrong about my post? - Original Message - From: Zoa [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 10, 2005 12:31 PM Subject: Re: [Asterisk-Users] G.729AB codec support ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729AB codec support
Im saying that the code is only an implementation of g729. The intel sources clearly states that you need a license for g729, not from intel but from the g729 patent holder. Zoa. signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] lost g729 lic
On Friday 10 June 2005 05:09, altus wrote: We installed a box a long time ago and they bought g729a licenses Now we want to upgrade and reinstall,whats going to happen with the codec,if I give the box the same ip as always will it work? Please do a modicum of research, hell even contact the people you got the licenses from (i.e. [EMAIL PROTECTED]). This kind of question is insulting to this entire list because it shows a total lack of resepect for everyone on it. We enjoy helping others, but at the same time there is a basic level of research which is requested in this social contract, and which you haven't displayed. -A. (the early-morning list nazi, apparently) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to Cisco Unity
I understand what you're saying, but I am not the one who makes the decisions. That decision is made already, so since I am actually getting your point and I agree with that, the only thing I can try to do right now, is try to avoid having Cisco Unity in the other 3 offices. I would love to implement Asterisk in these ones, but if it cannot be connected to Cisco this won't be an option at all, they won't consider it. So, back to the question, is it possible to connect Asterisk to Cisco and have all the functionality expected, and is it hard? Thanks, have a nice day Simone William Boehlke wrote: By the time you install the Asterisk server you have more features than Cisco delivers with Unity, for half the cost and without those annoying viruses. So instead of thinking about connecting Asterisk, consider disconnecting Unity. They make excellent landfill. Regards, William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simone Sent: Thursday, June 09, 2005 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity Hi, just wondering if my question is just unusual or if it is a quite stupid one. Thought there would be someone having this kind of scenario, but maybe I'm wrong. btw, have a nice day Simone Simone wrote: Hi all, first post. My company's office in the UK is soon going to get a Cisco VoIP solution system. What I am interested in, and couldn't find googling, is if it is possible to connect an Asterisk solution to the Cisco system and have all the nice advantages of it (mainly calling the extensions and directly reach the other office). Thanks, have a nice day Simone ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.6 - Release Date: 6/8/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP-INFO
On Friday 10 June 2005 02:28, Olle E. Johansson wrote: I would like to use this moment to say a big THANK YOU from the community to you and Commpartners for providing this resource to the community... I agree; while I personally dislike wikis I can't deny (as is evidenced by all the posts here in this thread) that voip-info.org is a very important resource for this community, and I'm sure that it is a mostly thankless job to boot. Thank you, James, for the blood sweat and tears, not to mention money, that you pour into voip-info.org. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATTN: Keith - Seriously OT
On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote: Received: from source ([81.56.129.44]) by exprod5mx8.postini.com ([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT Your MTA claimed it was called SOURCE but rDNS tells the recipient MX that it is called: mail.linuxautrement.com I too will block emails with a non-FQDN HELO or EHLO. I feel, however, that reverse should not have to match forward lookups for mail exchangers. It's an assinine requirement (my box does web, mail, dns and a host of other services, why should I need it to be called 'mail' for both forward and reverse lookups just to get mail flowing? Assinine. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail and MS Exchange Synchronization
On Friday 10 June 2005 02:15, Dan Levine wrote: I would be willing to Pay $500 for a good Asterisk / Exchange Integration What do you consider good Asterisk and Exchange integration? More than a handful of words, please. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call inband progress indication and zaphfc
Hello all, I've a little clue with zaphfc used to connect to a BRI linethat probably can be a configuration issue (really I hope so) Here, telcos (expecially mobile operators) use to substitute the dialtone with some vocal indication without answer the line. (Indications like The customer is not reachable or wait because the customer is on the phone ecc..) For asterisk this condition is a normal dial tone and the message from the telco and it's not possible to listen theese indications. As I'm using zaphfc and with X100p and a normal analog line I can listen these indications, my question is Have you tryed with PRI cards? as I don't know if this is an issue of asterisk, zaphfc or my configuration. Thank you in advance Diego ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Live! CF
On Friday 10 June 2005 02:07, Bob Goddard wrote: The Via C3 processors lack the CMPXCHG8B (CMOV) instructions and I assume others which are listed in the Intel documents as being optional. GCC assumes that they are always there. Look at http://radagast.bglug.ca/epia/epia_howto/x1098.html, section 13.2. This has been well documented. Thanks for the response. optional CPU instructions? What, does the CPU decide that it doesn't feel like having those instructions on occassion? :-) Again thanks. I learned something today. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] lost g729 lic
Thank up very much for the response Its appreciated and it will help me allot I hope u have a nice Monday or is it Friday? ALtus (the early-morning BOER!) On Fri, 2005-06-10 at 06:05 -0400, Andrew Kohlsmith wrote: On Friday 10 June 2005 05:09, altus wrote: We installed a box a long time ago and they bought g729a licenses Now we want to upgrade and reinstall,whats going to happen with the codec,if I give the box the same ip as always will it work? Please do a modicum of research, hell even contact the people you got the licenses from (i.e. [EMAIL PROTECTED]). This kind of question is insulting to this entire list because it shows a total lack of resepect for everyone on it. We enjoy helping others, but at the same time there is a basic level of research which is requested in this social contract, and which you haven't displayed. -A. (the early-morning list nazi, apparently) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Request OPTION and 404 Sjphone Xlite
sylvain garcia wrote: Hi, I have install asterisk and it works fine. But when I use Sjphone and I use Ethereal a Client send Request:OPTIONS sip:obelix.foo and Server answer Status: 404 Not found. But i can talk with two client and asterisk. When I use Xlite i don't have this request it's clean. I don't understand?? Well, then you are in agreement with your Obelix Asterisk server! It doesn't really matter from a signalling point of view, but sometime someone should fix our OPTIONS support. /Olle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Authentication
Title: Message Hi, I use SIP softphone that is not registered at Asterisk. When I dial some extension defined in the dial plan ([EMAIL PROTECTED])with my SIP softphone, Asterisk will not ask me for username/password (will not return response 407) as I expected. The response 407 - Authentication required will be returned if username defined in the softphone's setting matches one of the SIP peers defined in sip.conf. This means that anyone can dial extension at my Asterisk and that is not good, since that person could then dial over my ZAP line. How can I configure Asterisk to allow only peers defined in sip.conf to register and dial? Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization
I would like to support these plans for exchange/outlook integration with at least $250 as well. Please have a closer look at http://www.click-and-call.com/ . Mediastreams has developed their product e-phone, which we could test a couple of months ago. Their Outlook Integration is really great: - see missed calls in inbox - right click a contact or missed call entry to dial - starting outlook, registers the extension in the system (on asterisk-server ?!) - incoming call pops up, transfer it with one click to voicemail or other extension - Managing Call Groups within outlook - Managing voicemail - Recording of calls ... But if you also have a closer look on their prices... ;-( If the community would be able to develop such a killer-app, Asterisk could really become the leading telephone application, perhaps world-wide! Developers like Thorben Jensen did a realy good job, to get things work on the client side. Perhaps, these guys with the power to code things well, should work - more - together on an Outlook Integration. My experiences with asterisk in short are, that the server-apps are running really stable, many features are developed, tested and made there way to the stable version. But what's really missing, are GUIs that normal users can work with. They have to accept them and should love to work with them. If we can't provide users with these GUIs, the powerfull features within Asterisk are only something for techies like us. Now, this is my 2cts to this discussion. Nice weekend to all and let's make Asterisk a more powerfull application Guido Hecken gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Handytone-488 FXO - PSTN in calls to Asterisk, sip.conf? (fwd)
For some reason, this didn't go through the first time, maybe because I had JUST signed up. Hello, I'm trying to configure Asterisk and my Handytone 488 to pass incoming calls coming over PSTN through the FXO port to Asterisk, which will process the calls with voicemail, or some such service. I point the Handytone 488 FXO port configuration to 192.168.0.2 (the machine that is running Asterisk) and have configured a catchall extension to receive the call: [from-pstn] exten = s,1,Playback(outputfile,noanswer) And an SIP client: ; This is the actual phone - FXS [997] port=5061 host=192.168.0.5 type=friend context=Powermac disallow=all ;allow=ulaw ;allow=alaw allow=ilbc allow=gsm ; This is the phone line - FXO [997b] port=5062 host=192.168.0.5 type=friend context=Powermac disallow=all ;allow=ulaw ;allow=alaw allow=ilbc allow=gsm However, when I call the PSTN number, asterisk -cvvd reports nothing when the phone rings. A couple of facts: 1. The FXS port is configured fine to route through asterisk to a iax2 terminator. Its awesome! Can't wait to need more lines. :-) 2. Softphone calls to the FXO port ring OK, and can get picked up by asterisk. 3. The 488 is a Lan client of an Apple Airport router, meaning the WAN port on the 488 is connected to the Asterisk server. The LAN port on the 488 is not used. I've found the router part of the unit to be VERY finicky. Questions: 1. Does the FXO port need to register with Asterisk? 2. Does Asterisk need to register with the FSO port? (I have no experience with registering, though I did get an incoming call through IPkall through to FWD to XLite) 3. With regard to parenthetic comment - I was unable to get that incoming call (or register with FWD) to work while behind my Airport router. Could this router also be blocking the FXO port from communicating with Asterisk? Thanks, Asterisk is AMAZING! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU
Hi, I have recently found a bug when using Steve Underwood chan_unicall with Asterisk 1.0.x (including 1.0.8RC) When you place a call from a SIP phone with dtmfmode=rfc2833 or dtmfmode=inband through MFCR2 via chan_unicall all goes well until you press a dtmf key. When you do this, the other end hears a garbage sound (not the dtmf tone) and cpu goes to 99.9% rendering almost unusable the PBX. If there are more than 2 calls, audio start to get choppy, more calls renders unusable the pbx. If you hangup the calling extension, almost all the time it returns to normality, if there is a moderate load on the * server, the only way of shutting down * is by killing -9 it. I have been working this with Steve and have reported this finding today. If you have any suggestion in which things could be tweaked in chan_sip.c, chan_zap.c or chan_unicall.c in order to see if this bug could be solved, I will be happy to test it. Any additional info you may require please let me know. Regards. AM. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATTN: Keith - Seriously OT
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, 10 June 2005 8:16 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] ATTN: Keith - Seriously OT On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote: Received: from source ([81.56.129.44]) by exprod5mx8.postini.com ([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT Your MTA claimed it was called SOURCE but rDNS tells the recipient MX that it is called: mail.linuxautrement.com I too will block emails with a non-FQDN HELO or EHLO. I feel, however, that reverse should not have to match forward lookups for mail exchangers. It's an assinine requirement (my box does web, mail, dns and a host of other services, why should I need it to be called 'mail' for both forward and reverse lookups just to get mail flowing? Assinine. Your server your rules, however in this day of increasing trojan SMTP engined boxes, you should expect to get les and less deliverability. The point I was making is that the MTA was using a faked name in the HELO. That is an immediate red flag to a well configured MX. Shrug, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cell redirect
Hello, In feature list I see that asterisk supports call redirect feature(as this is basic PBX feature :)). I am trying to implement this feature on my sip phones (avaya 4602). The need is to enable some feature access code for example *40 so, that user can dial it and redirect all calls to other extention. As I understood, in SIP evironment this must be done through redirect server? Can Asterisk be that redirect server? How do you configure it? Can someone explain briefly or paste some link with explanation and example of such usage? What must be done: User at ext. 222 dials *40,then dials 333 and hangs up. Now all incoming calls through asterisk must be forwarded to ext. 333 Ahter user can dial #40 to cancel forwarding. Thanks in advance as this is actually one of the last things I must solve to be shure to migrate office PBX to Asterisk. __ Advertisement: Nokia 6610i Nokia 3220 REZERVET! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATTN: Keith - Seriously OT
On Friday 10 June 2005 07:34, Terry H. Gilsenan wrote: Your server your rules, however in this day of increasing trojan SMTP engined boxes, you should expect to get les and less deliverability. I fail to see how a reverse pointer that == forward record means a more reliable message. How many SMTP servers are compromised? I far prefer smarter methods, especially in days where people are putting as many services as possible on one IP and want a reverse record that makes some kind of sense. :-) The point I was making is that the MTA was using a faked name in the HELO. That is an immediate red flag to a well configured MX. Oh absolutely and, as I said, I do the same thing. Actually my front-line postfix rules reject a lot of mail before it ever hits the real spam/virus filters. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP-INFO
If required, I'd be more than happy and willing to let voip-info.org be hosted on my hosting server. We are currently hooked up to the net with a 6MB symetrical connection, and it should be enough for voip-info. In addition, I can perform a daily incremental back to it, in the same manner I backup all the other hosted site. voip-info is one of the most valuable tools around, and having it go down on us is a disaster to everybody. Nir S Andrew Kohlsmith wrote: On Friday 10 June 2005 02:28, Olle E. Johansson wrote: I would like to use this moment to say a big THANK YOU from the community to you and Commpartners for providing this resource to the community... I agree; while I personally dislike wikis I can't deny (as is evidenced by all the posts here in this thread) that voip-info.org is a very important resource for this community, and I'm sure that it is a mostly thankless job to boot. Thank you, James, for the blood sweat and tears, not to mention money, that you pour into voip-info.org. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P and Siemens HIPATH 3750
Title: Mensaje Hi all, I have to interconnect Asterisk with a Siemens HIPATH 3750. In siemens we can configure ECMA-QSIG Master, ISO-QSIG Master,Point to Point link withCRC4 and Point to Point link withouthCRC4): Siemens has BNC connector. I use a balun with BNC and RH45 connectro. I try with basic RJ45 cable and with crossover RJ45(1-4, 2-5) but I can only see yellow led in TE410P. I have configured siemens like Point to Point with and withouth CRC4 and Asterisk with ccs,hdb3 ( with CRC4 and withouth CRC4), with pri_net and pri_cpe and signalling=euroisdn Anyone has experience with this scenario? Regards, srsergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VOIP-INFO
It sounds like there are quite a few people willing to aid in bandwidth for voip-info. I was just wondering if it wouldn't make sense to mirror the site across several locations with a round-robin DNS for a little bit of load balancing? Any thoughts? Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Nir Simionovich |Sent: Friday, June 10, 2005 6:03 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] VOIP-INFO | |If required, I'd be more than happy and willing to let voip-info.org be |hosted on my hosting server. |We are currently hooked up to the net with a 6MB symetrical connection, |and it should be enough |for voip-info. In addition, I can perform a daily incremental back to |it, in the same manner I backup |all the other hosted site. | |voip-info is one of the most valuable tools around, and having it go |down on us is a disaster to everybody. | |Nir S | |Andrew Kohlsmith wrote: | |On Friday 10 June 2005 02:28, Olle E. Johansson wrote: | | |I would like to use this moment to say a big THANK YOU from the |community to you and Commpartners for providing this resource to the |community... | | | |I agree; while I personally dislike wikis I can't deny (as is evidenced by |all |the posts here in this thread) that voip-info.org is a very important |resource for this community, and I'm sure that it is a mostly thankless |job |to boot. | |Thank you, James, for the blood sweat and tears, not to mention money, |that |you pour into voip-info.org. | |-A. |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap Clocking - Frame Slips - tdm400p wcfxo zttest cpu spikes spandsp
I've made some modifications to zttest in order to use it as a frame clock accuracy tester / slip detector. I'm not certain if that was it's original purpose, but it seems that a lot of folks try to use it that way. The result is something that I'm calling ztclock for now to help avoid confusion. snip Background: During routine codec/dimensioning testing, I observed a strange, recurring cpu spike occuring appoximately every 12 seconds on a completely idle system with only the zaptel drivers loaded. I used 'vmstat 1' to monitor this. I was using the wcfxo driver (wildcard fxo) as a timing source. So, chicken egg: which comes first... the cpu spiking causing missed data, or, missed data causing cpu spiking, or, none of the above? Compiled and ran on a cvs-head box with a TDM04b (4 fxo's) Rev H card, fedora 3, 3ghz celery: [EMAIL PROTECTED] zaptel]# ./ztclock ztclock - clock source accuracy test (3 passes) Flushing input buffer... Flush Complete. Test is approximately 3 minutes. Please wait... 483328 samples in 60.413670 sec. (483310 sample intervals) 99.996277% 483328 samples in 60.413665 sec. (483310 sample intervals) 99.996277% 483328 samples in 60.413670 sec. (483310 sample intervals) 99.996277% Estimate 8 frame slips every 26.851555 seconds. I see the above appears to be slightly better then the numbers posted in your example. Running spandsp fails on the above system with nothing else running on this system (no calls, no nothing). Can we draw any conclusions or limit assumptions given the output? Rich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VOIP-INFO
On Fri, 2005-06-10 at 05:35 -0700, Chris Coulthurst wrote: It sounds like there are quite a few people willing to aid in bandwidth for voip-info. I was just wondering if it wouldn't make sense to mirror the site across several locations with a round-robin DNS for a little bit of load balancing? Any thoughts? Chris Coulthurst [EMAIL PROTECTED] I would like to start by saying wikis are against my religion, however voip-info is one of the best that I have seen, fairly well maintained current info, and its *accurate* at least most of the time for most things. Prejudices asiude, round robin seems to have a bit of a problem with a wiki on its surface, and must have some stuff done to make sure that its good. Namely the database backend needs to update all servers or everything is out of sync and people get confused or problems arise when comments are properly propagated. WHy not do it for free. Start the 'VoIP documentation project' on sourceforge. It provides bandwidth, filesystem for images and all, php, and all. afaik it does not provide database connectivity so that may be limiting, but it might. If the wiki software can support filesystem access instead of say mysql then there is no problem. That would allow for a solution to everything that should be required, and provide a nice connection at zero cost to anyone (other than time to set it up, maintain it, and all that). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 support
Hello again, I relaized that older version of Asterisk supports g729 ( Pass-thru only unless g729 license obtained - in any case I want). Do you know that latest [EMAIL PROTECTED] or CVS version provide us g729 pass-thru options? Thanks for your interest Erdem HAKI [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Evening in Melbourne (again!) next Thursday
Darn, and here I was thinking small town Melbourne, FL, USA =( From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jurgenSent: Thursday, June 09, 2005 11:16 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion; Commercial and Business-Oriented Asterisk DiscussionSubject: [Asterisk-Users] Asterisk Evening in Melbourne (again!) next Thursday Hi all,If you're in Melbourne Australia and interested in Asterisk, you're invited to join us for the second in an irregularly scheduled casual evening to talk about Asterisk, VOIP, networks, and just generally get geeky about IP phone stuff. About a dozen of us got together a couple of months ago, and had a good time chatting about all things Asterisk. Beverages were also consumed.Anyone with an interest is welcome; from Asterisk Gods to newbies who have recently downloaded it, from people administering several hundred seats to people playing with it at home and annoying their families.When: Next Thursday evening, the 16th, at 7pm.Where: Niagara Hotel, 383 Lonsdale Street (between Queen and Elizabeth) in the city. The Niagara's a relaxed, comfortable place, people seemed to like it last time. Also, like last time, I'll get an old phone and put it on the table, so those of us who haven't met will be able to recognise each other.Any questions, you can reach me on 0415 276 127, or email [EMAIL PROTECTED].Hope to see you there!...jurgen-- [EMAIL PROTECTED] is jurgen's gmail address.Visit http://jurgen.ca/ for more yummy goodness. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is it possible to have a remote Phone work behind Nat without a VPN?
If you get this working, please let me know -- I'm testing out the same situation, using [EMAIL PROTECTED], and have 3 SIP phones -- one softphone on a Samsung i700, one Avaya IP Phone and one softphone on a PC. The latter two are behind NAT and the i700 softphone is not, but I can't originate an inbound call from the i700 or call any extension from any other! I must be missing something.On 6/10/05, Dan Levine [EMAIL PROTECTED] wrote: Hi Everyone, Is it possible to have a SIP Phone work remotely if it's behind a Router performing NAT without connecting the Router to a VPN? The Asterisk Box will be in the DMZ. Thanks Dan CYTEXONE Dan Levine [EMAIL PROTECTED] CYTEXONE | Your Technology Specialists 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --Give a man a match, and he'll be warm for a minute.But set a man on fire, and he'll be warm for the rest of his life. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: PHPAGI Swift Escape Digits
Michael...
I don't believe that PHPAGI supports this currently. What you are looking
for is a combination of 2 functions: get_data() and swift().
PHPAGI code is very easy to follow so build your own function to do what you
want and add it to your copy of PHPAGI.php. Ain't OSS wonderful?
I did the following... (my apologies to PHPAGI and PHP gurus for my
inelegant code... but it works for me).
snip
/**
* Use Cepstral Swift to read text and get dtmf
*/
function swift_get_data($text, $frequency=8000, $voice=NULL,
$addl_params='', $timeout=NULL, $max_digits=NULL)
{
$text = trim($text);
if($text == '') return true;
if(!is_null($voice))
$voice = -n $voice;
elseif(isset($this-config['cepstral']['voice']))
$voice = -n {$this-config['cepstral']['voice']};
if($addl_params != '')
$addl_params = ,$addl_params;
// create the wave file
$fname = $this-config['phpagi']['tempdir'] . DIRECTORY_SEPARATOR;
$fname .= str_replace('.', '_', 'swift_' .
$this-request['agi_uniqueid']);
$p = popen({$this-config['cepstral']['swift']} -p
audio/channels=1,audio/sampling-rate=$frequency$addl_params $voice -o
$fname.wav -f -, 'w');
fputs($p, $text);
pclose($p);
// stream it
$ret = $this-get_data($fname, $timeout, $max_digits);
// destroy it
if(file_exists($fname . '.wav'))
unlink($fname . '.wav');
return $ret;
}
/snip
usage:
snip
$result = $agi-swift_get_data('To do thing 1 press 1. 'To do thing 2
press 2. 'To do thing 3 press 3. ',8000,'David','',2500,1);
$keys = $result['result'];
if ($keys == 1) {
// * do thing 1
} elseif ($keys == 2) {
// * do thing 2
} elseif ($keys == 3) {
// * do thing 3
}
/snip
Clarke Kawakami
Open Telephony Labs LLC
http://www.optellabs.com
I am trying to use swift in PHP/AGI.
function swift($text, $escape_digits='', $frequency=8000, $voice=NULL,
$fnameIn='')
During swift speaking some text I want the caller to be able to press
1, 2 or 3 to do thing 1, thing 2 or thing 3.
How are these digit defines and then caught?
Thanks,
Michael
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RE: [Asterisk-Users] Asterisk to Cisco Unity
We have Cisco Callmangler V4 in one office and several * servers in others, we use a SIP trunk out of the Cisco and it works perfectly. Peter -Original Message- From: Simone [mailto:[EMAIL PROTECTED] Sent: 10 June 2005 10:15 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity I understand what you're saying, but I am not the one who makes the decisions. That decision is made already, so since I am actually getting your point and I agree with that, the only thing I can try to do right now, is try to avoid having Cisco Unity in the other 3 offices. I would love to implement Asterisk in these ones, but if it cannot be connected to Cisco this won't be an option at all, they won't consider it. So, back to the question, is it possible to connect Asterisk to Cisco and have all the functionality expected, and is it hard? Thanks, have a nice day Simone William Boehlke wrote: By the time you install the Asterisk server you have more features than Cisco delivers with Unity, for half the cost and without those annoying viruses. So instead of thinking about connecting Asterisk, consider disconnecting Unity. They make excellent landfill. Regards, William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simone Sent: Thursday, June 09, 2005 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to Cisco Unity Hi, just wondering if my question is just unusual or if it is a quite stupid one. Thought there would be someone having this kind of scenario, but maybe I'm wrong. btw, have a nice day Simone Simone wrote: Hi all, first post. My company's office in the UK is soon going to get a Cisco VoIP solution system. What I am interested in, and couldn't find googling, is if it is possible to connect an Asterisk solution to the Cisco system and have all the nice advantages of it (mainly calling the extensions and directly reach the other office). Thanks, have a nice day Simone ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.6 - Release Date: 6/8/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ __ This email has been scanned for all viruses by the Star Internet Virus Screen. The service is provided in partnership with MessageLabs, the email security company. For more information on a higher level of virus protection visit www.star.net.uk __ __ This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP-INFO
WHy not do it for free. Start the 'VoIP documentation project' on sourceforge. It provides bandwidth, filesystem for images and all, php, Erk! My vote is against Sourceforge, definately -- although it's free, you get what you pay for. Clumsy interface and *shockingly* slow load times. I don't know what they're running for lines, but DAMN it takes a long time to get anywhere on there. That and their penchant for going down more often than a thermometer in Canada, I'd rather not have the resource there. :) (they're getting better, mind you, but it's still not great) Nathan -- - Nathan E. Pralle Give the director a serpent deflector. www.nathanpralle.com - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Banks
I have many old channel banks around that I would like to use to generate analog extensions. Will most channel banks work with Asterisk? Dave ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SpanDSP wownt compile
I am trying to patch the latest release version of Asterisk (1.0.7) with SpanDSP(0.0.2pre18). It seems that the Makefile for Asterisk was revamped since SpanDSP was released and the patch file that comes with SpanDSP for adding rxfax.c and txfax.c no longer work. I am not familiar with how make works and so i have no idea how to fix this. Does anyone know of any remedy for this? The SpanDSP patch file can be found here: ftp://ftp.soft-switch.org/pub/spandsp/spandsp-0.0.2pre18/ Thanks, -Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 404 not found
I use client Sjphone which work fine but i have Sniff a traffic.. - Sjphone send packet with OPTIONS to Asterisk - Asterisk ask with 404 not found This sequence come back often in my log. I don't understand why Sjphone Sens OPTION, and 404 not found.. Thanks for your help ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SoftPhone - Solaris
Hi, I am looking for a softphone (sip or iax) that works in Solaris/SPARC with sunray100 terminals. I found iaxcomm but it doesn't work. Also I am trying sip-communicator but I have several errors from JMF/RTP. Does anyone have a softphone working over this platform? which one? I don't care if it is a commercial product, I can buy it if works fine. thanks in advance. Sebas -- Sebastian Silva G R U P O G A U S S Depto. Sistemas Av. Libertador 6250 4 piso Tl.: 4 706- (int. 121) [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ATTN: Keith - Seriously OT
On Friday, June 10, 2005 3:16 AM, Andrew Kohlsmith [SMTP:[EMAIL PROTECTED] wrote: On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote: Received: from source ([81.56.129.44]) by exprod5mx8.postini.com ([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT Your MTA claimed it was called SOURCE but rDNS tells the recipient MX that it is called: mail.linuxautrement.com I too will block emails with a non-FQDN HELO or EHLO. I feel, however, that reverse should not have to match forward lookups for mail exchangers. It's an assinine requirement (my box does web, mail, dns and a host of other services, why should I need it to be called 'mail' for both forward and reverse lookups just to get mail flowing? Assinine. -A. Your server does not have to be called 'mail' for DNS and rDNS to work properly for mail delivery. All that is required is that a reverse lookup returns whatever the actual name of the server is and the server needs to use that same name when it issues HELO. My server at home is called 'fs-1' and the one at work is 'troutdale'. Both systems work properly just because I set up the DNS and rDNS records to match the names of the servers. There are a lot of broken rDNS records on the internet, and that's not likely to change anytime soon. I only have control of a very tiny portion of DNS and rDNS space, but I still feel obligated to make my part work properly. It's what makes the internet work. Would you feel OK driving around in your car, knowing that some large percentage of the street signs were not correct? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM04B
-Original Message- From: Gregory Wiktor - ADCom Corp. [mailto:[EMAIL PROTECTED] I did that once on a cheap linejack card. Took a week to get the smell out of the office, and the bright orange from inside the server was quite interesting :) Only took 1 second to start a small flame going, but fortunately I cought it quick. Reminds me of when I smoked cheap a sound card connecting it to the tape output of a tube amp. The sound card apparently had no DC blocking capacitor on its input, and the tube amp had some DC on its output... I wonder if the zaptel cards have any kind of protection from this sort of thing... No, they don't. Someone mentioned damaging one this way just a couple weeks ago. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail and MS Exchange
-Original Message- From: magnus [mailto:[EMAIL PROTECTED] From my perspective, not sure I would want Exchange (Which is difficult enough to manage) to be cluttered up with potentially large voicemail files, That's a concern, especially since bugs in current Asterisk versions require you to use uncompressed WAV files to get acceptable volume levels. However, this *is* a common configuration for other products. We used to have a CallXpress system that used Exchange as a message store. It stored voice messages in people's Exchange mailboxes, and could even read email messages over the phone via text-to-speech. The interface with Exchange was kind of kludgy, though, and not entirely reliable. It actually used a copy of Outlook on the voicemail server to talk to Exchange. I would have thought that most Exchange clients are most likely to be Outlook based, who could use pst Imap (Or pop3 if asterisk could auto forward and then delete voice mail) to retrieve voicemail via email without having to worry about central Exchange issues. IMAP is no good. Outlook, at least in older versions, cannot handle both an IMAP account and an Exchange account at the same time. (They can do POP3 and Exchange together, though.) A voicemail app that used an IMAP server as its message store would still be a nice feature, though. It might even work with Exchange, which can act as an IMAP server. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP-INFO
On Fri, 2005-06-10 at 08:12 -0500, Nathan Pralle wrote: WHy not do it for free. Start the 'VoIP documentation project' on sourceforge. It provides bandwidth, filesystem for images and all, php, Erk! My vote is against Sourceforge, definately -- although it's free, you get what you pay for. Clumsy interface and *shockingly* slow load times. I don't know what they're running for lines, but DAMN it takes a long time to get anywhere on there. That and their penchant for going down more often than a thermometer in Canada, I'd rather not have the resource there. :) (they're getting better, mind you, but it's still not great) I havent had that problem, maybe its more of a peering problem rather than them ... A network connection between you and them is slow, overloaded and goes down. I have *never* had a problem getting to sf.net nor slow load times (although some pages load slow because the underlying php is inherently slow, most are really fast). They have a really good raid system (a php stats program gives info off the machine it runs on and does run on sf.net) so it shouldnt be a FS problem. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronizatio n
-Original Message- From: Iassen Hristov [mailto:[EMAIL PROTECTED] Dumb, hacky idea...but just so crazy it might work: Have Asterisk include a read receipt request when sending the voice mail message. Write a script, triggered from a sendmail alias or .forward file, that will parse the incoming receipts and handle the message deletion. Bonus points: When someone listens to the message on the voicemail server, send an Outlook message retraction request. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help for conference
check /etc/asterisk/modules.conf and make sure that you have load = meetme.so
best regards
On 6/10/05, craz sead [EMAIL PROTECTED] wrote:
hi all
i have * box with 4 ext using sj phone, i wanna try to
make a conference. i am using ztdummy and look fine
when i install it because there is no erros message. I
checked with lsmod the zaptel and usb-uhci using
ztdummy. but why i still get error says no
application meetme ...
here is my ext.conf
exten = _66XX,1,NoOP(call for ${EXTEN})
exten = _66XX,2,Dial(SIP/${EXTEN},60,tr)
exten = _66XX,3,congestion
exten = 6690,1,meetme,1234
exten = 6691,1,meetme,2345
my meetme.conf
[room]
conf = 1234
conf = 2345,111
please advice
thks
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Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Seems things have just got worse. Just got reports that 800 numbers are not terminating. For example, can not dial: 800-888-9358 or 800-922-4684 Had to pull voipjet out of our routes until this gets fixed. On 6/9/05, Moody [EMAIL PROTECTED] wrote: We have been having serious quality problems using the westcoast server - been using the East coast server with increased success but seeing some issues related to going cross continent. Voipjet, you listening? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] D-Link DVG-1402S
Hi friends, Has anybody used a D-Link DVG-1402S VoIP gateway with * ?Please. Can send me any information to configurate thisgateway? Many thanks in advance. Luis ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P and Siemens HIPATH 3750
Hello, Sergio Serrano wrote: I have to interconnect Asterisk with a Siemens HIPATH 3750. I have configured siemens like Point to Point with and withouth CRC4 and Asterisk with ccs,hdb3 ( with CRC4 and withouth CRC4), with pri_net and pri_cpe and signalling=euroisdn Anyone has experience with this scenario? If you have a yellow LED, it is likely, that your wiring is still wrong. We have connecected an Asterisk Server between the PSTN and a Siemens Hipath 3750 with a TMS2M. The Asterisk has two TE110P cards. One TE110P is configured pri_cpe (the one at the PSTN), the other is pri_net (at the HiPath). The TMS2M is configured to Euro-Amt PP (with CRC4). Our zaptel.conf === span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 loadzone=fr defaultzone=fr Our zapata.conf === [channels] context=remote pridialplan=unknown prilocaldialplan=unknown usecallingpres=yes busydetect=no callprogress=no callwaitingcallerid=yes echotraining=no echocancel=yes echocancelwhenbridged=no overlapdial=yes immediate=no callerid=asreceived language=de musiconhold=default rxgain=0.0 txgain=0.0 switchtype=euroisdn signalling=pri_cpe group=1 channel = 1-15,17-31 signalling=pri_net group=2 pridialplan=local prilocaldialplan=local channel = 32-46,48-62 Regards, Henry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is it possible to have a remote Phone work behind Nat without a VPN?
Hi Everyone, Is it possible to have a SIP Phone work remotely if it's behind a Router performing NAT without connecting the Router to a VPN? The Asterisk Box will be in the DMZ. Thanks Dan CYTEXONE Dan Levine [EMAIL PROTECTED] CYTEXONE | Your Technology Specialists 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is it possible to have a remote Phone work behindNat without a VPN?
nat=yes in sip.conf From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan LevineSent: Friday, June 10, 2005 10:27 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Is it possible to have a remote Phone work behindNat without a VPN? Hi Everyone, Is it possible to have a SIP Phone work remotely if it's behind a Router performing NAT without connecting the Router to a VPN? The Asterisk Box will be in the DMZ. Thanks Dan CYTEXONE Dan Levine [EMAIL PROTECTED] CYTEXONE | Your Technology Specialists 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 69
Hi All i used cangoma card, connected with E1, using unicall. asterisk 1.1.x. when i dial to asterisk server. asterisk show error as belows: -- Unicall/9 extension '9' in context 'from-pstn' from '71811242' does not exist. RejectingcallJun 10 16:47:59 WARNING[28159]: chan_unicall.c:2655 handle_uc_event: Unicall/9 event Far end disconnectedJun 10 16:47:59 WARNING[28159]: chan_unicall.c:2941 handle_uc_event: CRN 32769 - far disconnected cause=Normal Clearing [16]Jun 10 16:47:59 WARNING[28159]: chan_unicall.c:2655 handle_uc_event: Unicall/9 event Drop callJun 10 16:47:59 WARNING[28159]: chan_unicall.c:2655 handle_uc_event: Unicall/9 event Release call -- Unicall/9 released my caller id: 071811242 dialed number: 119 what is error ?. how to modify extensions.conf my setting extensions.conf [from-pstn]exten = s,1,wait(1)exten = s,2,Answerexten = s,3,Background(custom/aa_main) Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and mpg123 lock up
I have had a number of occasions where asterisk stopped working. (1.0.7) When this occured I tried to issue an asterisk -rx "stop now" and nothing happened. I then killall -9 asterisk, and it stops - but mpg123 is still hung. I then killall -9 mpg123 and it stops. I then restart asterisk and everything is fine again. Anyone else having this problem and what to do about it? Thanks, Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Banks
most yes On 6/10/05, David Sampson [EMAIL PROTECTED] wrote: I have many old channel banks around that I would like to use to generate analog extensions. Will most channel banks work with Asterisk? Dave ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] config problem
Hi, I am brand new with asterisk Just finished to install it Have some problems to configure it 1st case: IPphone LAN-- asterisk server LANFW--internetdiax software 2nd case: GSMtelephone lineasterisk serverLAN--FWinternetdiax software I would to have communication between diax software and IPphone or GSM Any pointers, tips, config files ? Thanks G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clicks in audio with TE100P PRI
It seems that configuring span=1,1,0,ccs,hdb3 and changing jitterbuffer=16 resolves or masks the issue. What I will do now is reduce again jitterbuffer to default to see what happens. To answer some of the questions I don't see hard disk activity when the clicks appear, also the hard disk has very low usage. The clicks I listened were continuous and periodic. If the other party stays in silence I also listen the click every half second. Also to check, I run zttest and gives me Best=100%, average=99.989%. Once tested again I'll write the results to see what happen. Thanks to all Alejandro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] config problem
its a good idea to read all the comments in the configuration files in /etc/asterisk/ in special asterisk.conf, extensions.conf, sip.conf, iax.conf and zapata.conf best regards On 6/10/05, Georges Henroteaux [EMAIL PROTECTED] wrote: Hi, I am brand new with asterisk Just finished to install it Have some problems to configure it 1st case: IPphone LAN-- asterisk server LANFW--internetdiax software 2nd case: GSMtelephone lineasterisk serverLAN--FWinternetdiax software I would to have communication between diax software and IPphone or GSM Any pointers, tips, config files ? Thanks G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Best BootRom SIP Code for Poly600?
Hey all, Just getting started playing around with my Polycom 600. According to the wiki, it looks like it's recommended to run BootRom 2.6.1 and SIP 1.4.1. Is that info still current, or is it safe to upgrade to 3.0.1 and 1.5.2? Justin -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AAH 1.1 cannot call between extensions (xten lite softphones)
Hello all, I've installed AAH 1.1 on my VIA C3 powered mini PC. I've made the necessary changes to the * makefile, so the compilation went well. The first thing I did was configuring two extensions from AMP, namely 200 and 201. Then I installed X-lite on two PC's and configured them with one of the extensions: System settings - SIP proxy - Default: Username: 200 Authorisation user: 200 Password: Domain/Realm: babbelbox SIP Proxy: babbelbox Babbelbox is the hostname of my * server and DNS is working properly. Now here's my problem, I can't call from one extension to the other. I tried both ways, but after about 5 seconds of silence, I get the voicemail (which works by the way). Also, I can make outbound calls (after configuring a SIP trunk to my ITSP), but I cannot receive calls through this trunk. Something makes me believe there is something wrong with the configuration of my X-lite softphones... Here's the logfile output: Jun 10 11:47:38 DEBUG[18503]: Expression is '0' Jun 10 11:47:38 VERBOSE[18503]: -- Executing GotoIf(SIP/201-1fe7, 0?4:3) in new stack Jun 10 11:47:38 VERBOSE[18503]: -- Goto (macro-dial,s,3) Jun 10 11:47:38 VERBOSE[18503]: -- Executing SetCIDName(SIP/201-1fe7, Test) in new stack Jun 10 11:47:38 VERBOSE[18503]: -- Executing AGI(SIP/201-1fe7, dialparties.agi) in new stack Jun 10 11:47:38 VERBOSE[18503]: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: request = dialparties.agi Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: priority = 4 Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: extension = s Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: language = en Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: accountcode = Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: uniqueid = 1118418458.50 Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: channel = SIP/201-1fe7 Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: callerid = Test 201 Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: context = macro-dial Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: type = SIP Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: rdnis = unknown Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: enhanced = 0.0 Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: dnid = 200 Jun 10 11:47:38 VERBOSE[18503]: dialparties.agi: Caller ID name is 'Test' number is '201' Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: Added extension 200 to extension map Jun 10 11:47:38 DEBUG[18503]: Unable to find key '200' in family 'CF' Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: Extension 200 cf is disabled Jun 10 11:47:38 DEBUG[18503]: Unable to find key '200' in family 'DND' Jun 10 11:47:38 VERBOSE[18503]: -- dialparties.agi: Extension 200 do not disturb is disabled Jun 10 11:47:49 VERBOSE[18503]: -- AGI Script dialparties.agi completed, returning 0 Jun 10 11:47:49 VERBOSE[18503]: -- Executing NoOp(SIP/201-1fe7, Returned from dialparties with no extensions to call) in new stack Jun 10 11:47:49 VERBOSE[18503]: -- Executing SetVar(SIP/201-1fe7, DIALSTATUS=BUSY) in new stack Jun 10 11:47:49 DEBUG[18503]: Expression is '0' Jun 10 11:47:49 VERBOSE[18503]: -- Executing GotoIf(SIP/201-1fe7, 0?s-BUSY|1) in new stack Jun 10 11:47:49 DEBUG[18503]: Not taking any branch Jun 10 11:47:49 DEBUG[18503]: Expression is '0' Jun 10 11:47:49 VERBOSE[18503]: -- Executing GotoIf(SIP/201-1fe7, 0?s-BUSY|1) in new stack Jun 10 11:47:49 DEBUG[18503]: Not taking any branch Jun 10 11:47:49 VERBOSE[18503]: -- Executing NoOp(SIP/201-1fe7, Sending to Voicemail box [EMAIL PROTECTED]) in new stack Jun 10 11:47:49 VERBOSE[18503]: -- Executing Macro(SIP/201-1fe7, vm|[EMAIL PROTECTED]|BUSY) in new stack Jun 10 11:47:49 VERBOSE[18503]: -- Executing Goto(SIP/201-1fe7, s-BUSY|1) in new stack Jun 10 11:47:49 VERBOSE[18503]: -- Goto (macro-vm,s-BUSY,1) Jun 10 11:47:49 VERBOSE[18503]: -- Executing VoiceMail(SIP/201-1fe7, [EMAIL PROTECTED]) in new stack Jun 10 11:47:49 DEBUG[18503]: voicemail/default/200/busy doesn't exist, doing what we can Jun 10 11:47:49 DEBUG[18503]: Ooh, format changed from unknown to ulaw Jun 10 11:47:49 DEBUG[18503]: Scheduling timer at 160 sample intervals Jun 10 11:47:49 VERBOSE[18503]: -- Playing 'vm-theperson' (language 'en') Jun 10 11:47:49 DEBUG[1720]: Stopping retransmission on '[EMAIL PROTECTED]' of Response 52707: Found Jun 10 11:47:51 DEBUG[18503]: Scheduling timer at 0 sample intervals Jun 10 11:47:51 DEBUG[18503]: Scheduling timer at 0 sample intervals Jun 10 11:47:51 DEBUG[18503]: Scheduling timer at 160 sample intervals Jun 10 11:47:51 VERBOSE[18503]: -- Playing 'digits/2' (language 'en') Jun 10 11:47:51 DEBUG[1720]: Auto destroying call '[EMAIL PROTECTED]' Jun 10 11:47:51 DEBUG[18503]: Scheduling timer at 0 sample intervals Jun 10 11:47:51 DEBUG[18503]: Scheduling
Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Why would you even be routing 800 numbers out voipjet? They CHARGE you! On 6/10/05, Pedro [EMAIL PROTECTED] wrote: Seems things have just got worse. Just got reports that 800 numbers are not terminating. For example, can not dial: 800-888-9358 or 800-922-4684 Had to pull voipjet out of our routes until this gets fixed. On 6/9/05, Moody [EMAIL PROTECTED] wrote: We have been having serious quality problems using the westcoast server - been using the East coast server with increased success but seeing some issues related to going cross continent. Voipjet, you listening? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
I'm using the east coast server and am not experiencing any issues either US based or international. On 6/10/05, Pedro [EMAIL PROTECTED] wrote: Seems things have just got worse. Just got reports that 800 numbers are not terminating. For example, can not dial: 800-888-9358 or 800-922-4684 Had to pull voipjet out of our routes until this gets fixed. On 6/9/05, Moody [EMAIL PROTECTED] wrote: We have been having serious quality problems using the westcoast server - been using the East coast server with increased success but seeing some issues related to going cross continent. Voipjet, you listening? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G711 ( alaw or ulaw ) pass-thru
Hi, Its possible to make a pass-trhu conection with alaw or ulaw? Thanks -- Este mensaje ha sido analizado por C4I S.A. Mail Server en busca de virus y otros contenidos peligrosos, y se considera que está limpio. MailScanner agradece a transtec Computers por su apoyo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Clicks in audio with TE100P PRI
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Alejandro G Sent: Friday, June 10, 2005 8:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Clicks in audio with TE100P PRI It seems that configuring span=1,1,0,ccs,hdb3 and changing jitterbuffer=16 resolves or masks the issue. What I will do now is reduce again jitterbuffer to default to see what happens. Your issue is very likely the size of the zaptel jitterbuffers setting. If the zaptel driver is not immediately available to accept a frame of data it places it in an internal queue of pending writes. If that queue is full then the write is refused by the zaptel layer and then silently discarded by chan_zap causing a gap in the audio once it is played out of the zaptel card. If you crank up the debug level you will probably see 'Write returned -1...' (aka. EAGAIN) debugs that mostly correlate to the pops and clicks. Note that the zaptel driver legitimatly (if perhaps not appropriately) also refuses data when the channel is muted, such as during DTMF generation and at other times, so not _all_ EAGAIN debugs are a sign of problems. There is more background on my experience with the T100P popclick issue in http://lists.digium.com/pipermail/asterisk-dev/2005-May/012432.html Hope that helps. Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Best BootRom SIP Code for Poly600?
Hi Justin - Just getting started playing around with my Polycom 600. According to the wiki, it looks like it's recommended to run BootRom 2.6.1 and SIP 1.4.1. Is that info still current, or is it safe to upgrade to 3.0.1 and 1.5.2? I've been testing 1.5.2 for a few weeks now, and I'd have to say that I much prefer it over all previous versions. Everything works well. The dialplan and administration are much easier, and the soft buttons on the phone are more logically organized (especially forwarding calls). I'm still using bootrom version 2.6.1, though, in case I do ever need to go back to an earlier sip version. - Noah___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G711 ( alaw or ulaw ) pass-thru
Hi, Both of those are fully uncompressed codecs and free to use. Regards, Sahil Gupta VoiceValley On Fri, 10 Jun 2005, Edgardo Bermejo wrote: Hi, Its possible to make a pass-trhu conection with alaw or ulaw? Thanks -- Este mensaje ha sido analizado por C4I S.A. Mail Server en busca de virus y otros contenidos peligrosos, y se considera que está limpio. MailScanner agradece a transtec Computers por su apoyo. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
We are a VoIP provider and need to push out 100,000 - 200,000 minutes per month (ie. need a carrier-level package - not a Vonage, etc.). To date I have not found a wholesale SIP/IAX VoIP provider provide 800 termination for free. However, if you have one, please provide the information and I will definately check them out. On 6/10/05, Pedro [EMAIL PROTECTED] wrote: Please provide the SIP or IAX provider you are using that allows you to terminate to 800 numbers for free. On 6/10/05, Matt [EMAIL PROTECTED] wrote: Why would you even be routing 800 numbers out voipjet? They CHARGE you! On 6/10/05, Pedro [EMAIL PROTECTED] wrote: Seems things have just got worse. Just got reports that 800 numbers are not terminating. For example, can not dial: 800-888-9358 or 800-922-4684 Had to pull voipjet out of our routes until this gets fixed. On 6/9/05, Moody [EMAIL PROTECTED] wrote: We have been having serious quality problems using the westcoast server - been using the East coast server with increased success but seeing some issues related to going cross continent. Voipjet, you listening? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC what has been changed
Darren Wiebe wrote: The new version has an update database button. Install over your old version and then press the update-database button in 'configure'. This worked for me but... I think the default is not to use pins but it is very easy to set yourself. Unfortunately my case is not that easy!!! My motherboard of the machine, where Asterisk and ASTCC was installed is broken. I had copied (fortunately) the database to a database server, but that is all!! I do not have the config files as they have been on the old machine. I do not know what the config files should be. How can I create the config files and make sure that I don't loose the database? When I use just save and than go to ASTCC cgi. than I can see the routes, the brand names. However, if I go to the cards, and try to list the cards, than I come to http://cgi-bin/ ... which is translated automatically in my browser to: http://www.unhcr.ch/cgi-bin/texis/vtx/home I cannot find where it is set to my web domainname Also with save not all parameters are saved (mostlikely there is my problem) I do not use the SIP/IAXfriends. It created only one config file with save: cat /var/lib/astcc/astcc-config.conf ; ; Automatically created by astcc-admin.cgi. ; friendsdb = NO dbuser = user dbhost = 192.168.20.133 dbname = astcc2005 cardlength = 12 ; Automatically created by astcc-admin.cgi. = startingdigit = 1000 dbpass = passwd emailadd = [EMAIL PROTECTED] mailprog = sendmail email = YES ; = BTW, the Makefile still misses the astcc-user.cgi to move to the web. Any detail hints welcome ;-) (like a working config file, ...) bye Ronald Darren Wiebe [EMAIL PROTECTED] Ronald Wiplinger wrote: Darren Wiebe wrote: I'm not sure, check the bug tracker. However, regarding the PIN question In the configure sheet there is an option that says Require Pins (Yes/NO). Set that at NO if you do not wish to use pins. They will be generated but will not be required. Ok, I try the question with different words ;-) : How to upgrade from the previous version? Is the default NO for pins? How to change the database from old to new? (I guess only the require pin has changed) bye Ronald Darren Wiebe Ronald Wiplinger wrote: What has been changed at the ASTCC from the previous head to the current one? How to use the PIN? Can I avoid it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] blindtransfers with IAX
Hello,
I use the ${BLINDTARNSFER} variable for transfers from SIP accounts, but
this variable seems to be unavailable for IAX channels. Is this supposed
to be this way, is there another variable???
Many thanks for your help,
Marc
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[Asterisk-Users] Dropping Frame of G729
Here is the setup: Phone -SIP G729- AsteriskA -IAX G729- AsteriskB -SIP G729- Carrier The call completes but AsteriskA prints on the screen a ton of those Dropping Frame of G729 messages starting about 5 seconds into the call: Jun 10 11:17:14 NOTICE[14277]: frame.c:135 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Jun 10 11:17:14 NOTICE[14277]: frame.c:135 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Jun 10 11:17:14 NOTICE[14277]: frame.c:135 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Jun 10 11:17:14 NOTICE[14277]: frame.c:135 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end I'm guessing the Carrier is causing this? But I don't see any messages like this on AsteriskB. Any ideas? Thanks, -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G711 ( alaw or ulaw ) pass-thru
Actually, they are compressed, but they are free to use :-) Steve Sahil Gupta wrote: Hi, Both of those are fully uncompressed codecs and free to use. Regards, Sahil Gupta VoiceValley On Fri, 10 Jun 2005, Edgardo Bermejo wrote: Hi, Its possible to make a pass-trhu conection with alaw or ulaw? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cell redirect
http://www.voip-info.org/tiki-index.php?page=Asterisk+call+forwarding -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of [EMAIL PROTECTED]Sent: Friday, June 10, 2005 3:02 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Cell redirectHello, In feature list I see that asterisk supports call redirect feature(as this is basic PBX feature :)). I am trying to implement this feature on my sip phones (avaya 4602). The need is to enable some feature access code for example "*40" so, that user can dial it and redirect all calls to other extention. As I understood, in SIP evironment this must be done through redirect server? Can Asterisk be that redirect server? How do you configure it? Can someone explain briefly or paste some link with explanation and example of such usage? What must be done: User at ext. 222 dials *40,then dials 333 and hangs up. Now all incoming calls through asterisk must be forwarded to ext. 333 Ahter user can dial #40 to cancel forwarding. Thanks in advance as this is actually one of the last things I must solve to be shure to migrate office PBX to Asterisk. __Advertisement: Nokia 6610i Nokia 3220 REZERVET! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: PHPAGI Swift Escape Digits
On 6/10/05, Clarke Kawakami [EMAIL PROTECTED] wrote: Michael... I don't believe that PHPAGI supports this currently. What you are looking for is a combination of 2 functions: get_data() and swift(). That's what I was beginning to think but kept getting thrown off by the escape digits parameter in swift. I see that is there to just break out of that particular stream but no record the digits. PHPAGI code is very easy to follow so build your own function to do what you want and add it to your copy of PHPAGI.php. Ain't OSS wonderful? Of course! I did the following... (my apologies to PHPAGI and PHP gurus for my inelegant code... but it works for me). Thanks! Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Clicks in audio with TE100P PRI
I tested all again. No matter if span=1,1,0 or span=1,0,0 if I configure jitterbufer=4 I have glitches that I'm almost sure that are holes in audio. If I raise jitterbufer=16 the problem disappear (or becames impercetible). Anyway I am interested in understand what is happening. Your issue is very likely the size of the zaptel jitterbuffers setting. If the zaptel driver is not immediately available to accept a frame of data it places it in an internal queue of pending writes. If that queue is full then the write is refused by the zaptel layer and then silently discarded by chan_zap causing a gap in the audio once it is played out of the zaptel card. If you crank up the debug level you will probably see 'Write returned -1...' (aka. EAGAIN) debugs that mostly correlate to the pops and clicks. Note that the zaptel driver legitimatly (if perhaps not appropriately) also refuses data when the channel is muted, such as during DTMF generation and at other times, so not _all_ EAGAIN debugs are a sign of problems. This makes perfect sense but again some issues of the problem do not match. I set debug at level 9 and there is no message of errors. Another thing I do not understand is why the same configuration: PAP2 - LAN - Asterisk - TE100P works perfect, and instead of LAN using internet generates the problem. Shouldn't it be the same for both configs? The only difference I see is that the rtp packets came from another Ethernet card, but if I call to terminate calls with another carrier using that eth works fine. What is clear is that jitterbuffer=16 corrects the problem. One more thing: no matter what codec I use, G729 or G711 the sound clicks are almost the same. Is anyway I could debug at RTP level in asterisk to see what is happening and check if there is packet loose? Thanks Alejandro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization
Good things are happening. Another aside from having done this before: If configuration requires the user to do anything or the user to load a piece of software it won't work. Everything must be configured from an admin consol or it won't work. You will go crazy trying to keep all the laptops and desktops up to date with DLLs. Keep everything on the back end. All the user needs to do is ask for it. There should be an asterisk voicemail.conf that turns it on for all, or for each individual mailbox. Also, sound files. 1. Come up with a sound file format that your users can us on their PC's without having to load more stuff. 2. Conversion of sound files is about 2:1. That is it takes the CPU about 2 seconds to convert 1 second of sound in batch processing. Sure it might seem like it could be done faster but moving, converting, dispatching of files takes time and energy. 3. Noise! It was hilarious the first time we let a large room full of users get their voicemail in their emails. We had tested it using Executives in private offices. Once they liked it the Cubarium was given permission to use voicemail playback in their emails. Everyone was given a pair of speakers (they were not allowed to have headphones or CD-ROM drives as that might cause them to listen to music when they were suppose to be working) for their desk. Then a week later we turned on the voicemail in email for them Imagine 150 cubes all listening to their voicemails on their desktop speakers. Absolute Pandemonium ensued. The best part was the voicemails left by spouses telling them to get home from work or else. Then there were the inter-office love affair voicemails broadcast at high volume. The Executives removed the speakers and issued headphones the next week. My point is that you have to realize the human factors and costs in doing this. Race the tyrant Vanderdecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization
Good Idea, but not practical as it breaks the second commandment of IT user management. 1. Thou shall not require any brain cells on the part of the end-user. 2. Thou shall not require any settings to be set on the users equipment. ... More rules to follow... Race the tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Brodbeck Sent: Friday, June 10, 2005 10:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Re: Voicemail and MS Exchange Synchronization -Original Message- From: Iassen Hristov [mailto:[EMAIL PROTECTED] Dumb, hacky idea...but just so crazy it might work: Have Asterisk include a read receipt request when sending the voice mail message. Write a script, triggered from a sendmail alias or .forward file, that will parse the incoming receipts and handle the message deletion. Bonus points: When someone listens to the message on the voicemail server, send an Outlook message retraction request. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Clicks in audio with TE100P PRI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
Alejandro G
Sent: Friday, June 10, 2005 9:57 AM
To: Asterisk
Subject: [Asterisk-Users] Clicks in audio with TE100P PRI
I tested all again. No matter if span=1,1,0 or span=1,0,0 if
I configure jitterbufer=4 I have glitches that I'm almost sure that are
holes in
audio.
FYI, changing the sync source is a non-trivial thing. You need to be sure
you're also changing the sync source on the opposite end. If the T100P is
interfaced to a Telco then you really ought not to be changing this - it _will_
break things if set wrong.
{clip}
This makes perfect sense but again some issues of the problem
do not match. I set debug at level 9 and there is no message of errors.
The now-canceled patch in http://bugs.digium.com/view.php?id=4107 shows where
the data is being discarded - the reporting of this may have changed in the
current code base and/or zaptel may be accepting the data and discarding it
internally now. Asterisk is a fast moving target, take all behavioural
comparisons with a grain of salt.
Another thing I do not understand is why the same configuration:
PAP2 - LAN - Asterisk - TE100P works perfect, and
instead of LAN using internet generates the problem. Shouldn't it be the
same for both configs?
Not neccessarially; if we assume that there is no congestion on the LAN then
packets arrive off the wire, say, every 20ms, regular as clockwork +/- a few
microseconds. On the Internet there _is_ congestion, hence buffering, hence
packets might arrive in a 'squirt' as quickly as the nic can deliver them,
followed by a 'large' gap, followed by another squirt etc. Asterisk doesn't
re-marshal the packets as they pass through the core, thus the bunching up that
occurs at chan_zap, which mandates evenly spaced, well marshaled blocks of
data.
The only difference I see is that the rtp packets came from
another Ethernet card, but if I call to terminate calls with another carrier
using that eth works fine.
You mean a network-asterisk-network call path? Then, yes, that also
implies an issue at the zap layer.
One more thing: no matter what codec I use, G729 or G711 the
sound clicks are almost the same.
That implies that the issue isn't on the rtp side, rather in the core or on the
zap side as the data is transcoded to ulaw/alaw by the time it hits chan_zap.
If you were using chan_iax with the new iax jitterbuffer and enabled genericplc
you'd find the popclick didn't dissapear either.
You could also eliminate the rtp side as the source of the noise by dialing
across the network into an echo() target on the Asterisk server and then
running a stream of sound through - you might need to have a headset and play a
stereo or something into the mic. If the drop is occuring on the network or in
the core then the echo'ed audio will also contain the popclick effect.
Is anyway I could debug at RTP level in asterisk to see what
is happening and check if there is packet loose?
Depends on the channel type (sip, iax etc.) - each channel has it's own variety
of commented out very low level debugging. Use the source... ;-)
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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[Asterisk-Users] Call disconnect
When connecting from providers (I have tried 3 now) in the UK and having the calls routed to my asterisk server in the US, I am suffering a call disconnect problem. The problem occurs whenever I record a call, either using record or sending the call to the voicemail application. This however does not occur when I route calls from US providers to the same Asterisk server.The calls are being disconnected after about 30-45 seconds of recording, and appear to be terminated normally. I can however listen to messages or stay on hold for 10 minutes or more. I have tried this on several other servers and get the same problem. Is there something I am missing in the configuration? Is is possibly due to there not being any audio being sent back to the PSTN gateway in the UK during a record function? Any help would be appreciated, I am really stumped on this one after many hours. Scott England ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization
))) Please see comments inline. From my perspective, not sure I would want Exchange (Which is difficult enough to manage) to be cluttered up with potentially large voicemail files, That's a concern, especially since bugs in current Asterisk versions require you to use uncompressed WAV files to get acceptable volume levels. However, this *is* a common configuration for other products. ) I do not worry about this. It is 'only' storage space. We used to have a CallXpress system that used Exchange as a message store. It stored voice messages in people's Exchange mailboxes, and could even read email messages over the phone via text-to-speech. The interface with Exchange was kind of kludgy, though, and not entirely reliable. It actually used a copy of Outlook on the voicemail server to talk to Exchange. )) We are not now thinking about using this type of design, no client running on the Exchange server nor on the laptop. I would have thought that most Exchange clients are most likely to be Outlook based, who could use pst IMAP (or POP3 if asterisk could auto forward and then delete voice mail) to retrieve voicemail via email without having to worry about central Exchange issues. IMAP is no good. Outlook, at least in older versions, cannot handle both an IMAP account and an Exchange account at the same time. (They can do POP3 and Exchange together, though.) ))) Again, no need for IMAP client on the laptop, just ordinary Outlook connected directly to its Exchange Server or connected to its IMAP server. Yes, you're correct, old versions of Outlook didn't allow use of Exchange Server connection IMAP connection inside the same Outlook profile, but that is hopefully not a concern for too many folks if it is, argh, I don't know what to suggest, other than moving to new emailer or new version of their emailer. A voicemail app that used an IMAP server as its message store would still be a nice feature, though. It might even work with Exchange, which can act as an IMAP server. Yes, Exchange Server would be the IAMPd. We can simultaneously connect to Exchange via any protocol, as many times, per machines, per user, etc (to the machines limits). Please post your bounty to http://tinyurl.com/bf64x Please also let us keep this on the task of only voicemail-to-email integration. I know that as soon as someone sees the words Outlook, they go crazy, wanting everything under the sun, the reason I know this is that I am that person too, but we will never get all that, let's get the voicemail synchronization done, it will be huge, it will benefit everyone, it will be huge step in the right direction. The issue of tying Outlook all up ought to be a second issue as such I have created a second bounty for that. DavidB, you may or may not want to pony up for that one here: http://tinyurl.com/8wmrb ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cell redirect
not sure but this may help you http://voip-info.org/tiki-index.php?page=Asterisk%20call%20forwarding Additionally, i can tell you that im using AGI to detect redirection number. I Allow my users to set redirection from their Web based User Panel, they can check their calls, and edit their redirection number. So, Asterisk allows you to do it in many ways :-) Best Regards On 6/10/05, Ugis Racko [EMAIL PROTECTED] wrote: http://www.voip-info.org/tiki-index.php?page=Asterisk+call+forwarding -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, June 10, 2005 3:02 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cell redirect Hello, In feature list I see that asterisk supports call redirect feature(as this is basic PBX feature :)). I am trying to implement this feature on my sip phones (avaya 4602). The need is to enable some feature access code for example *40 so, that user can dial it and redirect all calls to other extention. As I understood, in SIP evironment this must be done through redirect server? Can Asterisk be that redirect server? How do you configure it? Can someone explain briefly or paste some link with explanation and example of such usage? What must be done: User at ext. 222 dials *40,then dials 333 and hangs up. Now all incoming calls through asterisk must be forwarded to ext. 333 Ahter user can dial #40 to cancel forwarding. Thanks in advance as this is actually one of the last things I must solve to be shure to migrate office PBX to Asterisk. __ Advertisement: Nokia 6610i Nokia 3220 REZERVET! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
http://www.freeworldialup.com/advanced/peering_numbers But I'm not sure if they would like you to terminate a lot of minutes over it, just check it out. Roman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pedro Sent: Freitag, 10. Juni 2005 18:15 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems? We are a VoIP provider and need to push out 100,000 - 200,000 minutes per month (ie. need a carrier-level package - not a Vonage, etc.). To date I have not found a wholesale SIP/IAX VoIP provider provide 800 termination for free. However, if you have one, please provide the information and I will definately check them out. On 6/10/05, Pedro [EMAIL PROTECTED] wrote: Please provide the SIP or IAX provider you are using that allows you to terminate to 800 numbers for free. On 6/10/05, Matt [EMAIL PROTECTED] wrote: Why would you even be routing 800 numbers out voipjet? They CHARGE you! On 6/10/05, Pedro [EMAIL PROTECTED] wrote: Seems things have just got worse. Just got reports that 800 numbers are not terminating. For example, can not dial: 800-888-9358 or 800-922-4684 Had to pull voipjet out of our routes until this gets fixed. On 6/9/05, Moody [EMAIL PROTECTED] wrote: We have been having serious quality problems using the westcoast server - been using the East coast server with increased success but seeing some issues related to going cross continent. Voipjet, you listening? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Thanks a lot to all for the input. I have now switched to the voipjet east coast back-up server and everything seems to be back to normal now. Thanks, Roman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Freitag, 10. Juni 2005 17:58 To: Pedro; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems? I'm using the east coast server and am not experiencing any issues either US based or international. On 6/10/05, Pedro [EMAIL PROTECTED] wrote: Seems things have just got worse. Just got reports that 800 numbers are not terminating. For example, can not dial: 800-888-9358 or 800-922-4684 Had to pull voipjet out of our routes until this gets fixed. On 6/9/05, Moody [EMAIL PROTECTED] wrote: We have been having serious quality problems using the westcoast server - been using the East coast server with increased success but seeing some issues related to going cross continent. Voipjet, you listening? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best BootRom SIP Code for Poly600?
Justin Ellison wrote: Hey all, Just getting started playing around with my Polycom 600. According to the wiki, it looks like it's recommended to run BootRom 2.6.1 and SIP 1.4.1. Is that info still current, or is it safe to upgrade to 3.0.1 and 1.5.2? I am still running BootRom 2.6.1 with Firmware 1.5.2 works great. I don't want to upgrade the rom due to not being able to down grade. Justin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best BootRom SIP Code for Poly600?
I'm using bootrom 2.6.1 with 1.5.2 for the same reason. I would suggest the upgrade to 1.5.2 for some non trivial enhancements such as multiple line/call appearance. Also the menu system is significantly improved. Louie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ariel Batista Sent: Friday, June 10, 2005 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Best BootRom SIP Code for Poly600? Justin Ellison wrote: Hey all, Just getting started playing around with my Polycom 600. According to the wiki, it looks like it's recommended to run BootRom 2.6.1 and SIP 1.4.1. Is that info still current, or is it safe to upgrade to 3.0.1 and 1.5.2? I am still running BootRom 2.6.1 with Firmware 1.5.2 works great. I don't want to upgrade the rom due to not being able to down grade. Justin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone noticed Voipjet voice quality problems?
Just did the same and it seems (cross fingers) to be fine now too. However, I have to wonder. What happens to the load on that East Coast box when we all switch over to it. Sure would be nice to hear from VoipJet. Considering hwo many times I have recommended them, it would make me feel better. Cheers, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roman Zhovtulya Sent: Friday, June 10, 2005 11:14 AM To: 'Matt'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Anyone noticed Voipjet voice quality problems? Thanks a lot to all for the input. I have now switched to the voipjet east coast back-up server and everything seems to be back to normal now. Thanks, Roman -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Freitag, 10. Juni 2005 17:58 To: Pedro; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone noticed Voipjet voice quality problems? I'm using the east coast server and am not experiencing any issues either US based or international. On 6/10/05, Pedro [EMAIL PROTECTED] wrote: Seems things have just got worse. Just got reports that 800 numbers are not terminating. For example, can not dial: 800-888-9358 or 800-922-4684 Had to pull voipjet out of our routes until this gets fixed. On 6/9/05, Moody [EMAIL PROTECTED] wrote: We have been having serious quality problems using the westcoast server - been using the East coast server with increased success but seeing some issues related to going cross continent. Voipjet, you listening? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Toll Free DIDs
I have several toll free numbers that get forwarded to a single local number assigned to a trunkgroup. I've asked the telco to not forward those toll free numbers but to assign them as DIDs to the trunkgroup, so that I can differentiate via DNID. They said that they can't do that. That toll free numbers must forward. I know that I could have them each forward to different local DIDs assigned to the trunkgroup, but that just doesn't seem necessary. Is the telco correct? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Toll Free DIDs
I have several toll free numbers that get forwarded to a single local number assigned to a trunkgroup. I've asked the telco to not forward those toll free numbers but to assign them as DIDs to the trunkgroup, so that I can differentiate via DNID. They said that they can't do that. That toll free numbers must forward. I know that I could have them each forward to different local DIDs assigned to the trunkgroup, but that just doesn't seem necessary. Is the telco correct? Technically they are partly correct. 800 numbers are pointed to a local number. Although, they can pass the 800-XXX- to you IF they choose to. In my experience(limited as it may be) it is easier to have them point to a specific local number assigned to the trunkgroup. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
