Re: [Asterisk-Users] Telemarketer Torture
Very good. I made a much smaller one that just goes in an endless loop, no way out but to hang up. I figured telemarketer are too stupid to notice the same prompts over and over. I might use yours. Did you put the .gsm recording some place we can get them? My brother has the BEST solution for sales people. He makes an appointment with them to come out and gives an address across the street. It really wastes a real estate salesman or house painter's time to drive out to a dead end. Keeps em off the phone too. --- Steve Murphy [EMAIL PROTECTED] wrote: Hello-- I submitted of extensions.conf that contains my telemarketer torture menus, last week sometime to the mailing list. I got back a note from the mailing list machinery, stating that it was too big, and would be subject to approval. No such approval came, I guess. Either I missed it, or it didn't rate, or the moderator just plain hasn't gotten around to it yet. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GrandStream Budgetone * Error part 2
I took a closer look and tcpdump shows that the grandstream sending back to * an icmp port unreachable whenever * sends an rtp packet to the grandstream. The grandstream's rtp port is set to 5004 and * is sending to that port. I get the same results with 2 GS phones. Changing the rtp port to 5000 on the GS has the same effect Can't see why the GS phones are saying unreachable port tcpdump: 21:32:15.478488 192.168.1.101 host3.phx2.com: icmp: 192.168.1.101 udp port 5004 unreachable (DF) 21:32:15.497853 host3.phx2.com.15288 192.168.1.101.5004: udp 172 (DF) [tos 0x10] 21:32:15.498484 192.168.1.101 host3.phx2.com: icmp: 192.168.1.101 udp port 5004 unreachable (DF) 21:32:15.517851 host3.phx2.com.15288 192.168.1.101.5004: udp 172 (DF) [tos 0x10] 21:32:15.518485 192.168.1.101 host3.phx2.com: icmp: 192.168.1.101 udp port 5004 unreachable (DF) 21:32:15.537855 host3.phx2.com.15288 192.168.1.101.5004: udp 172 (DF) [tos 0x10] 21:32:15.538492 192.168.1.101 host3.phx2.com: icmp: 192.168.1.101 udp port 5004 unreachable (DF) Just started putting my first * together with a tdm400p and x100p. Analog phones, xlite and diax I've got working. Just got Grandstream budgetone-100. The budgetone registers with * just fine. * accepts the dtmf and dials the number. The remote phone rings. From there things go south. The CLI reports this: -- Executing StripMSD(SIP/jrb-683a, 1) in new stack -- Executing Dial(SIP/jrb-683a, Zap/1/9384074) in new stack -- Called 1/9384074 -- Zap/1-1 answered SIP/jrb-683a WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 52852 (Response) -- Hungup 'Zap/1-1' == Spawn extension (home, 9384074, 2) exited non-zero on 'SIP/jrb-683a' The budgetone is using a fixed ip, dtmf signaling, firmware version is 1.0.4,17 My sip.conf for the budgetone is: [jrb] type=friend host=dynamic username=jrb secret=x dtmfmode=rfc2833 context=home reinvite=no canreinvite=no qualify=1000 I can't find a solution to in the archives and I've looked at all the documentation I can find setting up the budgetone on *. Any pointers would be appreciated. Thanx in advance. John Breeden Plum Hall, Inc. Kamuela Hawaii ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sendmail not on localhost
You would have to look at the code in the VM app. and see if the hostname for the mail server is configurable. Likely it is simply hard coded to localhost which would send the mail to port 25 on the * sever. In theory the VM application _could_ use a remote mail server but it would have to be written that way. I'd prefer to run a local sendmail. Ths means you have a local queue and the mail gets handed off quikly even if your other server is down or slow. --- Ralf Illing [EMAIL PROTECTED] wrote: Hi . I already set-up sendmail on another network server thus it would be nice to use that one or is sendmail on * server required!? I had a look in the archive but couldn't find any information where to set the mail server from localhost to my network server . Cheers Ralf = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.9.6 now available- some fixes included
Dan, I just tried running diax 0.9.6 in a Remote Desktop (Win2k term services administrative mode) and received these errors: No hay dispositivo de sonido disponible! No hay dispositivo de sonido disponible! The second box was entitled diax, had a red X, and had the error message twice, just as I typed it. Now that I think about it, I don't know why anyone would do this on Terminal Services... But I was just trying to test the disconnection issues we've been having... I have tested myself, as a curiosity, in TS mode and see this behaviour. I think it can be solved, but do not worth do it right now. This is not a priority for me. you can't run DIAX in TS mode, as under the Win2k Terminal Server normally there is no sound card available. http://www.winnetmag.com/Article/ArticleID/7493/7493.html -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Telemarketer Torture
Chris Albertson [EMAIL PROTECTED] said: My brother has the BEST solution for sales people. He makes an appointment with them to come out and gives an address across the street. It really wastes a real estate salesman or house painter's time to drive out to a dead end. Keeps em off the phone too. I once got Reader's Digest direct mail department off my back by sending them a formal offer to check their mail service - every received mail piece would be reported by me (including a 'quality report' - folded, cracked, ...) and I would invoice only some 50 dollars per mail piece for that. Sending mail would constitute acceptance of the offer - never got a single piece of mail from them again (a pity, I could've been rich ;-)). Wonder whether one could build up a similar construction (the paper one was legally quite watertight, of course) for telemarketeers... -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAX 0.9.6b - available for download - multilingual issue on WinME solved
Hi all, A modified version of DIAX, 0.9.6b is now available for downlaod at: http://www.laser.com/dante http://www.geocities.com/tdanro The multilingual issue on Windows ME (and 98 too) is now solved (I hope), been completely redesigned internaly. Tested here on ME and it works, including the Default_locales mode. Added a new language (Czech) with the help of Petr Grussmann. If support for some other languages is requested, please send me a mail directly. Please send me your feedback, especially the WIn9x/ME users of DIAX. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call-waiting caller-id
Hi., - Original Message - From: Steve Dolloff [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 09, 2003 9:00 PM Subject: [Asterisk-Users] call-waiting caller-id Are there any known issues with call-waiting caller-id for SIP? Caller-ID on the first call works fine, but when the second call comes in, I hear the interrupt tone, but the caller-id doesn't display anything. I have tried this with the Cisco ATA and the SPA-2000. I have also tried two different phones to verify that it wasn't something specific to the phone. I have two ATAs and two X100P cards in an * box. When an internal call is in progress and another one comes from the PSTN line, the callwaiting callerID is correctly displayed on the phone connected to the ATA (using SIP), the same for two internal calls. but... If I'm in a call with the PSTN line then the callwaiting callerID does not work anymore when a new call arrives on the same line. It happens like in your case. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_iax2.c Warnings
On 10/12/03 05:05, Isamar Maia wrote: I'm setting up my Iaxtel connection now and I'm getting some annoying warnings What means: WARNING[7176]: File chan_iax2.c, Line 436 (iax_error_output): Ignoring unknown information element 'Unknown IE' (31) of length 4 ? And how can I fix it? Don't worry too much about it. IAX2 has various information elements with which to set up a call. These include caller ID, destination context/extension, etc. Element 31 is defined as DATETIME. This is sent to tell the remote Asterisk server what time you think it is. This can be useful for working out timezone differences, etc. If you want to remove the message, upgrade your Asterisk to a more recent version (the DATETIME element was added a couple of months ago, IIRC). Regards, Alastair Alastair Maw MXTelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sendmail not on localhost
On 10/12/03 07:41, Chris Albertson wrote: I'd prefer to run a local sendmail. Ths means you have a local queue and the mail gets handed off quikly even if your other server is down or slow. A better solution would be an SMTP fowarding agent, such as ssmtp. I'd prefer *not* to have to patch/configure/nurse multiple sendmails in my organization unless I really need to. Regards, Alastair Alastair Maw MXTelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web Interface for CDRs
I've written that ugly one :) I'll try to see if I can find it, if that was what you meant... On Tue, 2003-12-09 at 15:55, Bruce Hedreen wrote: Does anyone know where that nice .php is that was written to access the CDRs from mysql DB? Bruce W. Hedreen Computer Technologies of Eastern Carolina, LLC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.543 / Virus Database: 337 - Release Date: 11/21/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GrandStream Budgetone * Error
Try changing the order of preferred codecs on budgetone, I've had the same problems and that fixed it for me. i have pcma,pcmu,g723,g729 and it works ok. regards, robert John Breeden wrote: Just started putting my first * together with a tdm400p and x100p. Analog phones, xlite and diax I've got working. Just got Grandstream budgetone-100. The budgetone registers with * just fine. * accepts the dtmf and dials the number. The remote phone rings. From there things go south. The CLI reports this: -- Executing StripMSD(SIP/jrb-683a, 1) in new stack -- Executing Dial(SIP/jrb-683a, Zap/1/9384074) in new stack -- Called 1/9384074 -- Zap/1-1 answered SIP/jrb-683a WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 52852 (Response) -- Hungup 'Zap/1-1' == Spawn extension (home, 9384074, 2) exited non-zero on 'SIP/jrb-683a' The budgetone is using a fixed ip, dtmf signaling, firmware version is 1.0.4,17 My sip.conf for the budgetone is: [jrb] type=friend host=dynamic username=jrb secret=x dtmfmode=rfc2833 context=home reinvite=no canreinvite=no qualify=1000 I can't find a solution to in the archives and I've looked at all the documentation I can find setting up the budgetone on *. Any pointers would be appreciated. Thanx in advance. John Breeden Plum Hall, Inc. Kamuela Hawaii ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.9.6b - available for download - multilingual issue on WinME solved
Dan, what is the codec you use for DIAX? Is there a way to select it? - Original Message - From: Dan [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Miércoles, 10 de Diciembre de 2003 07:05 a.m. Subject: [Asterisk-Users] DIAX 0.9.6b - available for download - multilingual issue on WinME solved Hi all, A modified version of DIAX, 0.9.6b is now available for downlaod at: http://www.laser.com/dante http://www.geocities.com/tdanro The multilingual issue on Windows ME (and 98 too) is now solved (I hope), been completely redesigned internaly. Tested here on ME and it works, including the Default_locales mode. Added a new language (Czech) with the help of Petr Grussmann. If support for some other languages is requested, please send me a mail directly. Please send me your feedback, especially the WIn9x/ME users of DIAX. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX 0.9.6b - available for download - multilingual issue on WinME solved
Hi, - Original Message - From: Hector Q.-datafull [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 10, 2003 2:42 PM Subject: Re: [Asterisk-Users] DIAX 0.9.6b - available for download - multilingual issue on WinME solved Dan, what is the codec you use for DIAX? Is there a way to select it? GSM only for the moment. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] On Hold - Talked about before
I should have stated this. Is there any Analog phones that can do this. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Posted At: Tuesday, December 09, 2003 11:57 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] On Hold - Talked about before Subject: Re: [Asterisk-Users] On Hold - Talked about before At 08:45 PM 12/9/2003, you wrote: Ok - Here is where I am at. I know this topic has been discussed before, but never a solid answer was set in place. Is anyone aware of any phones that can put a caller on hold and the caller hear MOH by the user pressing the hold button. I understand most phones are only muting the speaker and handset. The SNOM phones can do this, and are also excellent phones generally. Install the 1.6x software build for now; the 2.x build changes their behavior a bit and breaks MOH with asterisk. This is being worked on. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfert with IAX
Hi, I try to use Libiax in order to put un transfert button inmy iax softphone. Is there a way to make a call transfert ? Best regards rattana
[Asterisk-Users] Re: FXO cards
On Wednesday, December 10, 2003, at 01:21 AM, [EMAIL PROTECTED] wrote: DIalogic 4 and 12 port cards are expensive. The 12 port prohibitively so, so I think I would be better off getting 2 of the 4 port cards. I have found them for 900$ apiece on ebay. That compairs as: Case 1 Channel bank Asterisk box: 500$ (provably less than this, as it is a minimal machine) T100P card: 500$ Channel Bank 600$ (actually, from what I can find, provably more like 1000$ before I am done) Phones: I am going to go with the Mitel Networks 5055, they are about 400$ apiece. Total: 1600$ for the setup (to 2000$) + 2000K for 5 phones. Case 2: FXO cards Asterisk box 500$ Dialogic 4 port card 900$ Phones: same as the above Total 1400$ plus phones, 2K$ for 5 phones. Don't forget that you will then have to purchase the DS0 licenses too. err, whats a DS0 license? Michael Rowley MD FP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfert with IAX
Hi, I try to use Libiax in order to put un transfert button in my iax softphone. Is there a way to make a call transfert ? Use '#' for transfer. BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DIAX to DIAX call and disconnecting after 50-60 sec.
- Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 08, 2003 9:14 AM Subject: Re: [Asterisk-Users] DIAX to DIAX call and disconnecting after 50-60 sec. Hi, - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 08, 2003 3:30 PM Subject: Re: [Asterisk-Users] DIAX to DIAX call and disconnecting after 50-60 sec. - Original Message - From: Dan [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Monday, December 08, 2003 3:49 AM Subject: [Asterisk-Users] DIAX to DIAX call and disconnecting after 50-60 sec. Hi, There is any other user of DIAX with this problem? Thanks, Dan Yes, my calls that are DIAX to asterisk to DIAX disconnect after about a minute. These are Extension to Extension calls. HOWEVER, I tested with the IAXComm/IAXClient and had the same results. So, I don't believe it's in your app, I think it's something in the library. Can you check with IAX(1) mode too? I want to know if this is related to LIBIAX2 only.. Can you tell me the CVS version used for this test? Dan, Sorry it's taken so long, but I just ran these two test calls... Version info from Asterisk: Asterisk CVS-12/04/03-22:04:29, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer [EMAIL PROTECTED] = Connected to Asterisk CVS-12/04/03-22:04:29 currently running on bebop (pid = 5145) DIAX extension 714 is 0.9.6b on Win2k Dell desktop DIAX extension 715 is 0.9.6 on Win2k Gateway laptop For fun, (and an attempt at re-aquanting myself) I'm running both of them in Spanish. This is from the first call, IAX to IAX via Asterisk: -- Registered '715' (AUTHENTICATED) at 192.168.3.100:4569 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Accepting AUTHENTICATED call from 24.199.177.150, requested format = 2, a ctual format = 2 -- Executing Macro([EMAIL PROTECTED]/1602, ext|715|IAX/715) in new stack -- Executing Dial([EMAIL PROTECTED]/1602, IAX/715|90|tT) in new stack -- Calling using options 'exten=s;callerid=Andrew Thompson 714;language= en;formats=2;capability=2;version=1;adsicpe=0' -- Called 715 -- Call accepted by 192.168.3.100 (format GSM) -- Format for call is GSM -- IAX[715]/1603 is ringing -- IAX[715]/1603 answered [EMAIL PROTECTED]/1602 -- Attempting native bridge of [EMAIL PROTECTED]/1602 and IAX[715]/1603 -- Channel 'IAX[715]/1603' unable to transfer -- Channel '[EMAIL PROTECTED]/1602' unable to transfer -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:5036 -- Hungup 'IAX[715]/1603' == Spawn extension (macro-ext, s, 1) exited non-zero on '[EMAIL PROTECTED]/1602' in macro 'ext' == Spawn extension (trusted, 715, 1) exited non-zero on '[EMAIL PROTECTED]/1602' -- Hungup '[EMAIL PROTECTED]/1602' -- Registered '715' (AUTHENTICATED) at 192.168.3.100:4569 -- Registered '714' (AUTHENTICATED) at 24.199.177.150:4569 This call lasted about 10 minutes before I hung it up. This is the second call, IAX2 to IAX2, via Asterisk: -- Accepting AUTHENTICATED call from 24.199.177.150, requested format = 2, actual format = 2 -- Executing Macro([EMAIL PROTECTED]/2, ext|715|IAX2/715) in new stack -- Executing Dial([EMAIL PROTECTED]/2, IAX2/715|90|tT) in new stack -- Called 715 -- Call accepted by 192.168.3.100 (format GSM) -- Format for call is GSM -- IAX2[715]/6 is ringing -- IAX2[715]/6 answered [EMAIL PROTECTED]/2 -- Attempting native bridge of [EMAIL PROTECTED]/2 and IAX2[715]/6 -- Hungup 'IAX2[715]/6' == Spawn extension (macro-ext, s, 1) exited non-zero on '[EMAIL PROTECTED]/2' in macro 'ext' == Spawn extension (trusted, 715, 1) exited non-zero on '[EMAIL PROTECTED]/2' -- Hungup '[EMAIL PROTECTED]/2' -- Registered '714' (AUTHENTICATED) at 24.199.177.150:4569 -- Registered '715' (AUTHENTICATED) at 192.168.3.100:4569 Note that a small amount of time (30 seconds to a
Re: [Asterisk-Users] Re: FXO cards
On Wed, 2003-12-10 at 08:18, Michael Rowley wrote: On Wednesday, December 10, 2003, at 01:21 AM, [EMAIL PROTECTED] wrote: DIalogic 4 and 12 port cards are expensive. The 12 port prohibitively so, so I think I would be better off getting 2 of the 4 port cards. I have found them for 900$ apiece on ebay. That compairs as: Case 1 Channel bank Asterisk box: 500$ (provably less than this, as it is a minimal machine) T100P card: 500$ Channel Bank 600$ (actually, from what I can find, provably more like 1000$ before I am done) Phones: I am going to go with the Mitel Networks 5055, they are about 400$ apiece. Total: 1600$ for the setup (to 2000$) + 2000K for 5 phones. Case 2: FXO cards Asterisk box 500$ Dialogic 4 port card 900$ Phones: same as the above Total 1400$ plus phones, 2K$ for 5 phones. Don't forget that you will then have to purchase the DS0 licenses too. err, whats a DS0 license? Because the drivers for the Dialogic cards are not open source, you have to pay Digium for each phone line that will be in use. I think the cost was $15, but I may be wrong. So for your 6 line install, you are looking at the cost of the cards plus the cost of the license at $90 or so. None of us are pushing you to a channel bank because we will make money on this. We are trying to help you out. If you go the channel bank route, replacement parts are easier to come by and expansion is much easier. It is a lot easier to justify an expenditure that has growth ability at essentially the same cost as an option that has less. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trouble with AGI and SAY DIGITS and WAIT FOR DIGIT using PHP
Hi there,
AGI is partially not working for me with SAY DIGITS, SAY NUMBER, WAIT FOR
DIGITS etc. It appears that result is always one for any function that
looks for user input, regardless of which key was pressed. Playing sound
is only possible using EXEC. This applies to two * servers with RH 7.2
and RH 7.3 and very recent CVS. Any suggestion or tips about where I
goofed?
Thanks, Philipp
-- [CLI output] --
demo.agi: write: ANSWER
demo.agi: read: 200 result=1
demo.agi: write: SAY DIGITS 68 7
-- Playing 'digits/6' (language 'en') -- no sound 6 only
demo.agi: read: 200 result=1
(null): write: EXEC SayNumber 34 -- but this works fine
-- AGI Script Executing Application: (SayNumber) Options: (34)
-- Playing 'digits/30' (language 'en')
-- Playing 'digits/4' (language 'en')
(null): read: 200 result=1
(null): write: WAIT FOR DIGIT 5000
(null): read: 200 result=1-- user pressed 4, but we get result=1
-- [extensions.conf] --
exten = _104[013-9],1,Answer
exten = _104[013-9],2,Playback(beep)
exten = _104[013-9],3,AGI(demo.agi)
exten = _104[013-9],4,Playback(beep)
exten = _104[013-9],5,Hangup
-- [demo.agi] --
#!/usr/bin/php -q
?php
ob_implicit_flush(true);
set_time_limit(0);
$in = fopen(php://stdin,r);
...
function __read__() {
global $in, $debug;
$input = str_replace(\n, , fgets($in, 4096));
return $input;
}
function __write__($line) {
print $line.\n;
}
...
__write__(ANSWER);
__read__();
__write__(SAY DIGITS 68 \7\);
__read__();
__write__(EXEC SayNumber 34);
__read__();
__write__(WAIT FOR DIGIT 5000);
__read__();
___
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[Asterisk-Users] IAX and PDAs
Hi all, Does anyone know of any work in progress on IAX based telephony for PDAs? Putting iaxcomm on a Zaurus or iPAQ, for example. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP PUBLISH and SUBSCRIBE extensions?
I haven't actually seen a 100 or 105, but my understanding is that they do not have the soft keys with LEDs like the SNOM 200 and the really nice SNOM 220 that is supposed to be out next year with 30 something soft keys. Apparently, from what I've read, the Cisco extension monitoring LEDs don't work with SIP and the skinny drivers don't yet support it for asterisk. Surely, someone has had need of this feature for the attendant or secretary or something... I just can't find anything about it. ADSI phones were deliberately crippled to be without this feature (actually, the specification was) so as not to compete with commercial PBX/phone offerings. Finally, this PUBLISH method's pre-RFC draft was just released less than two months ago. On Tuesday, 09 December, 2003 23:33, Juan J. Sierralta P. wrote: On Tue, 2003-12-09 at 23:06, Ulexus wrote: After having received my brand new SNOM 200 phones and trying to get the remote extension monitoring to work, if seems that they use the SUBSCRIBE and PUBLISH SIP methods to do this. Does Snom 100/105 remote extension monitoring also ? I think that feature isnt in current * implementation, since it means patches on the whole code instead of a patch only in chann_sip.c. Anyway its a really nice feature ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: was FXO cards
I am just afraid of the channel bank. I just don't know anything about them. If I buy the wrong crap, it gets really expensive fast, plus adds another layer of complexity. Michael, Don't be afraid of the channel bank. I got one (Adtran) on e-bay for $99.00 and was a complete neophyte at the time. I was pleasantly surprised at how easy it was to set up, providing that you read the documentation carefully, and there are lots of surplus cards (FXO and FXS) floating around in the $100 - $200 price range. The 600 and the 750 are both completely compatible with asterisk. For the 600, just make sure that you get the L2 series FXO cards if you want caller ID. I don't know about the 750. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] was FXO cards
Case 1 and 2 are ties in my eyes, except the channel bank would provably be cheaper to upgrade to 8 lines. I am just afraid of the channel bank. I just don't know anything about them. If I buy the wrong crap, it gets really expensive fast, plus adds another layer of complexity. You could also talk with your local phone company and other CLECs to find out pricing on fractional voice T1s/partial PRI. Depending on the locality, the breakeven point is usually 6-8 lines. You might even be able to get a deal on a hybrid data/voice circuit. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unknown RTP codec 19
Hi Folks, I had success with making SIP to H323 g/w calls using chan_h323 and G.729. Calls go through well though it takes incredibly long time to get connected which I thing due to fast start being disabled at the remote end, will ask remote side to fix it. There is a Notice message which appears on my * console which bothers me. While in a conversation, I get the following; File rtp.c line 418 (ast_rtp_read): Unknown rtp codec 19 received. I read in the mail archive that rtp codec 19 is comfort noise. Should I ignore this notice ? Or is there a better way to tell * asterisk to handle comfort Noise ? Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue bug with call transfer
Kevin, I've looked at the source of app_queue.c and can see if the logic for the * key, but nothing related to # in the code. Am I missing something? Thanks, Jonathan Kevin Bockman wrote: --- Jonathan Tew [EMAIL PROTECTED] wrote: We've got the app_queue configured to supposedly allow for a call to be transferred. When the call comes in and an agent answers it (using X-Lite Pro) and then decides to transfer the call (using the SIP phone interface) they get disconnected from their call and after left logged in to the queue system. Obviously we're doing something wrong to transfer the call. We hit * to hangup the call. Is there some other way to transfer the call? I've looked through the source and didn't see it. Thanks, Jonathan ___ Wow, surprised no one answered. Maybe your subject scared everyone off. * is for hangup. # is for transfer. I'm not sure if there is native sip transfer in x-lite or not or if it works or not. Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WAV file volume
We are using Voicemail 2 and use the e-mail with WAV file attachment option. When we record a message from an internal extension, the WAV file sounds fine. However, if a caller from outside leaves a message, the volume on the WAV file is so low it is almost useless. When we access the voicemail through an extension, however, the volume sounds fine. Has anyone had this problem or any idea of a way to fix it? - Glenn Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Errors after re-plugging T1
Hi, After temporarily pulling the T1 cable out of our Asterisk box, we ended up getting a strange error messages even after the cable was plugged back in. [...] Dec 10 09:01:11 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 Dec 10 09:01:21 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 Dec 10 09:03:42 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 Dec 10 09:03:52 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 [...] So I stopped asterisk, unloaded the kernel modules and restarted everything, but still: Dec 10 09:06:16 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: !! No channel map, no channel, and no ds1? What am I supposed to identify? Dec 10 09:06:16 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: !! Unable to add IE 'Channel Identification' Dec 10 09:06:20 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 62 failed: Unknown error 500 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 1 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 2 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 3 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 4 [...] Dec 10 09:18:07 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 62 failed: Unknown error 500 Dec 10 09:18:07 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 1 Dec 10 09:18:07 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 2 I tried this several times, to no avail. Only rebooting the box helped. The question now is: is there a way to avoid rebooting in a situation like this and still get everything to work again? Rebooting can be a huge pain. Thanks. Regards, Markus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] next stable release?
you should not fear cvs. many of us are using current (or semi-current) cvs version in production systems without issues. if you're in a test environment, you won't have problem. Also many of latest cvs additions are bug fixes, nothing really new, apart of cdr_odbc. See asterisk-cvs list more more details on that. matteo Il mer, 2003-12-10 alle 18:47, john lawler ha scritto: Hi guys, I've been running 0.5.0, which is dated sometime in September of this year and I've noticed a couple of new features in more recent code that I'd like to use, but am hesitant to go w/ CVS code. My system is not exactly a production system, it's mostly test, but I'm still leery of the fresh code. I'm wondering when the next stable release might come out, and how those work in general. Thanks, jl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registration refused on DIAX096b
Anybobdy can helpme with this error? IAX.CONF [tito2] type=user username=tito2 secret=mysecret host=dynamic context=demo ;sendani=no ;host=asterisk.linux-support.net ;port=5036 ;mask=255.255.255.255 qualify=yes ; Make sure this peer is alive On the server: NOTICE[1142135600]: File chan_iax2.c, Line 2832 (register_verify): No registration for peer 'tito2' (from myipaddress) Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 1ms SCall: 1 DCall: 02606 [myipaddress:56439] CAUSE : Registration Refused thanks anybody. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX and PDAs
Hi Steve, - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 10, 2003 5:47 PM Subject: [Asterisk-Users] IAX and PDAs Hi all, Does anyone know of any work in progress on IAX based telephony for PDAs? Putting iaxcomm on a Zaurus or iPAQ, for example. Regards, Steve I have in my plans a pocketpc version of DIAX too. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Computing horsepower needed
I have been reading asterisks and everything I can get my hands on for the past week. I want to know what class processor is the bare minimum I need for a four port Asterisk installation? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX and PDAs
On Wed, 2003-12-10 at 10:47, Steve Underwood wrote: Hi all, Does anyone know of any work in progress on IAX based telephony for PDAs? Putting iaxcomm on a Zaurus or iPAQ, for example. If someone where doing this, I would be the most ecstatic, happy person in the entire whole big wide world. Seriously... someone PLEASE make an IAX client for my Zaurus :) (I would if I had programming knowledge... doh!) -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com PS: I really want an IAX client for my Zaurus! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX and PDAs
On Wed, 2003-12-10 at 13:37, Dan wrote: I have in my plans a pocketpc version of DIAX too. The only problem I have with that is that there are already a couple of PocketPC versions available. Many people have a Zaurus, or iPaq running a Linux based OS, and we don't really have any options (there's tkcPhone, but I've heard it's fairly bloated, and doesn't really work all that well). DIAX on my Zaurus would be a dream come true. -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] next stable release?
I think the word here is that stable release is a do it yourself project. You need to set up a test environment where you can check out new CVS updates, apply patches you want and see that it all works. Only then would you but the code on a productin system. --- Brancaleoni Matteo [EMAIL PROTECTED] wrote: you should not fear cvs. many of us are using current (or semi-current) cvs version in production systems without issues. if you're in a test environment, you won't have problem. Also many of latest cvs additions are bug fixes, nothing really new, apart of cdr_odbc. See asterisk-cvs list more more details on that. matteo Il mer, 2003-12-10 alle 18:47, john lawler ha scritto: Hi guys, I've been running 0.5.0, which is dated sometime in September of this year and I've noticed a couple of new features in more recent code that I'd like to use, but am hesitant to go w/ CVS code. My system is not exactly a production system, it's mostly test, but I'm still leery of the fresh code. I'm wondering when the next stable release might come out, and how those work in general. Thanks, jl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
On Wed, 2003-12-10 at 12:47, Trench Shoring wrote: I have been reading asterisks and everything I can get my hands on for the past week. I want to know what class processor is the bare minimum I need for a four port Asterisk installation? This type of question comes up quite often. I don't know whether it is the frequency that annoys me or the seemingly implied I don't want to use a modern machine, will this POS that I was about to throw in the dump going to make this work. Basically the answer as usual is, it depends on what you want to do. We run production machines on P3 and Celeron(coppermine) machines. Just remember that once you deploy, it is harder to add CPU power in a non disruptive way. Of course if you are a bit over on the CPU power, you eventually will grow into it. So stepping away from the bare minimum part, you should shoot for something in the P3 range, and decent memory. It is much easier to debug any problems you have when you don't have to wonder if it is the hardware or not. Also a decent P3 machine shouldn't be too difficult to come by for little cost. The machine I had at home was a 1400 Duron that cost $106 for the CPU and motherboard. On this Duron I was running a T100P card. So again, it doesn't have to be an expensive machine, but please save your self time and aggravation and get a somewhat modern machine. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] next stable release?
I've been running 0.5.0, which is dated sometime in September of this year and I've noticed a couple of new features in more recent code that I'd like to use, but am hesitant to go w/ CVS code. My system is not exactly a production system, it's mostly test, but I'm still leery of the fresh code. I'd like to resolve all MAJOR and CRASH and BLOCK issues from the bug tracker for the next release and then do a feature freeze with the goal of having a 1.0 release as soon as possible. We have CVS now on track for making releases, thanks to Thorston. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Starting an AGI app from cli
Hello Maybe it sound pretty easy , but.. How can I start an application (let's say Dial) from the CLI ? Actually , I want to make a little script , who connects to asterisk once per hour and make a short call.. Anybody can help me ? Thanks Alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
I've set up a test enviroment and beed trying to answer that question. I think if you are carfull NOT to do dumb things like running X11 and a browser and so on on the server you can use a pretty low power system. Just do not plug in a CRT, mouse or keyboard. Use telnet or ssh. The requirements to run a graphic interfaceare are greater then to run a low-end asterisk server. Asterisk seemed to run well on am old 400Mhz Pentium but I'm using an ADM2600+ with 128MB ram and am not taxing the system much at all. I think a 1Ghz Pantium would be well more then required. OK that said. BIG remaining question. I've got some echo problems with the FXO card. Fixing this might take a lot of CPU power to do the required DSP. I don't know yet. But it works with two calls open at about 2% of the CPU utilization. ond the ADM 2600+ Pushing 8K sample/sec data aound is a very lightload audo at 8K is a very low data rate. My goal is to reduce the heat and electic power. I may try _under_ clocking the 2600+ and see if that makes it run cool enough that I can remove a fan. --- Trench Shoring [EMAIL PROTECTED] wrote: I have been reading asterisks and everything I can get my hands on for the past week. I want to know what class processor is the bare minimum I need for a four port Asterisk installation? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue bug with call transfer
Yeah I guess you did. I'm not really a programmer so I don't know where it is. All I know is that it's there somewhere and it works. --- Jonathan Tew [EMAIL PROTECTED] wrote: Kevin, I've looked at the source of app_queue.c and can see if the logic for the * key, but nothing related to # in the code. Am I missing something? Thanks, Jonathan Kevin Bockman wrote: --- Jonathan Tew [EMAIL PROTECTED] wrote: We've got the app_queue configured to supposedly allow for a call to be transferred. When the call comes in and an agent answers it (using X-Lite Pro) and then decides to transfer the call (using the SIP phone interface) they get disconnected from their call and after left logged in to the queue system. Obviously we're doing something wrong to transfer the call. We hit * to hangup the call. Is there some other way to transfer the call? I've looked through the source and didn't see it. Thanks, Jonathan ___ Wow, surprised no one answered. Maybe your subject scared everyone off. * is for hangup. # is for transfer. I'm not sure if there is native sip transfer in x-lite or not or if it works or not. Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
This type of question comes up quite often. I don't know whether it is the frequency that annoys me or the seemingly implied I don't want to use a modern machine, will this POS that I was about to throw in the dump going to make this work. Amen to that. I am running 1FXO and 1FXS on a P200MMX but I can't use iLBC because there just isn't enough horsepower there, but then again I'm just playing around. I am a minimalist myself but c'mon people, modern systems are not expensive and unless you're doing multiple T1s stop worry about the CPU. Spring for an Athlon or P3-class machine and start playing. It's trivial to upgrade hardware and if you've already cheaped out on the motherboard/CPU then you have extra money to get something bigger should the need arise. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Starting an AGI app from cli
On Wed, 2003-12-10 at 13:11, Alexandru Coseru wrote: Hello Maybe it sound pretty easy , but.. How can I start an application (let's say Dial) from the CLI ? Actually , I want to make a little script , who connects to asterisk once per hour and make a short call.. Anybody can help me ? No need for CLI. Use sample.call dropped into /var/spool/asterisk/outgoing -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] External Email Notification
Can anyone help in pointing me in the direction to configure my asterisk box to send a voice mail message waiting notification via my external POP3 server? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sendmail not on localhost
Alastair Maw wrote: On 10/12/03 07:41, Chris Albertson wrote: I'd prefer to run a local sendmail. Ths means you have a local queue and the mail gets handed off quikly even if your other server is down or slow. A better solution would be an SMTP fowarding agent, such as ssmtp. I'd prefer *not* to have to patch/configure/nurse multiple sendmails in my organization unless I really need to. Okokok. I've contributed a patch so you can configure any mailer for vm, but with the recent changes in voicemail.c it's out of date. I'll update this and you can help me try it out if it works for you. If so, add comments in bugs.digium.com. Give me a day or two to fix this, then download from bugs.digium.com. With that patch, you can configure ssmtp, postfix or anything to send the mail. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
On Wed, 2003-12-10 at 13:15, Chris Albertson wrote: My goal is to reduce the heat and electic power. I may try _under_ clocking the 2600+ and see if that makes it run cool enough that I can remove a fan. You will probably run into stability issues. Underclocking can be just as bad for stability as overclocking. Maybe you should check into the ACPI or other power management functions to help reduce power requirements. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External Email Notification
On Wed, 2003-12-10 at 13:38, Kevin wrote: Can anyone help in pointing me in the direction to configure my asterisk box to send a voice mail message waiting notification via my external POP3 server? pop3 is not able to send messages. pop3 is just for picking up messages. You need to use sendmail or an equivalent application to send the mail via SMTP to the location you pick up mail. We covered this yesterday. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sendmail not on localhost
On Wed, 2003-12-10 at 13:46, Olle E. Johansson wrote: Alastair Maw wrote: On 10/12/03 07:41, Chris Albertson wrote: I'd prefer to run a local sendmail. Ths means you have a local queue and the mail gets handed off quikly even if your other server is down or slow. A better solution would be an SMTP fowarding agent, such as ssmtp. I'd prefer *not* to have to patch/configure/nurse multiple sendmails in my organization unless I really need to. Okokok. I've contributed a patch so you can configure any mailer for vm, but with the recent changes in voicemail.c it's out of date. I'll update this and you can help me try it out if it works for you. If so, add comments in bugs.digium.com. Give me a day or two to fix this, then download from bugs.digium.com. With that patch, you can configure ssmtp, postfix or anything to send the mail. postfix and exim should provide a sendmail link or binary that should be command line compatible as the original for sending mail. I don't know about ssmtp. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] External Email Notification -2
Perhaps I should have been more specific, it wasn't clear in the discussion on this forum yesterday. I would my voice message waiting notification to be forwarded to my external ISP POP3 account. What is involved to set this up? Is configuring sendmail what is required? If so, are there any pointers for a configuration? Not a unix expert here... Can anyone help in pointing me in the direction to configure my asterisk box to send a voice mail message waiting notification via my external POP3 server? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
I have been reading asterisks and everything I can get my hands on for the past week. I want to know what class processor is the bare minimum I need for a four port Asterisk installation? As a low-end data point (probably not cool for a reasonable production box), I purchased a eMachine T2240 with a 2.2ghz Celeron, 40 gig drive, 384 meg ram, with integrated 10/100 nic from Circuit City (new, open box, with warranty, $300). Its running asterisk with 2 x100p's, festival, sendmail, apache, mysql, MOH, X11, Gnome, etc, just fine. Asterisk has three iax trunks running, a couple of remote nat'ed 7960's, a few local sip phones, and nothing very fancy for a dial plan. The size of the box (and its architecture) is far more related to voice traffic volumes and uptime then it is anything else. (e.g, if you never place a phone call, you don't need any resources; if you never click the console mouse, gnome is not consuming any cpu resource, etc, etc.) In an idle condition (no calls being processed), top is the heaviest app. Placing a single asterisk demo call from a sip phone (forcing iax2 to Digium) causes asterisk to bump towards the top at about 0.3% cpu utilization with an occasional random peak at 2% cpu. (A single pstn call via x100p to a 7960 averages about 1.0% cpu. Both of these are eyeball inspection of top.) I don't know what you mean by port in your statement, so can't comment on machine size. If you mean four physical pstn lines, I'd have to venture a guess and say the above machine could handle four x100p's. But, as you've already seen from the list, there are probably a dozen different ways to configure * to support any given set of requirements with different cost/benefit trade offs for each. The above just happens to be one of the the cheapest around for new assembled equipment with a so called warranty. Don't think I'd install it at a hospital or police department though. ;) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sendmail not on localhost
I already set-up sendmail on another network server thus it would be nice to use that one or is sendmail on * server required!? I had a look in the archive but couldn't find any information where to set the mail server from localhost to my network server . Get nullmailer. it is the best dumb mailer I've ever run across. The system never knows it isn't running a full-blown mail server. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Article on Asterisk: German Linux magazine
Hi there, a friend just notified my of the the cover story of freeX 1'2004: Linux als Telefonanalage (engl.: Linux as PBX) http://www.cul.de/freex.html Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registration refused on DIAX096b
- Original Message - From: Hector Q.-datafull [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 10, 2003 1:21 PM Subject: [Asterisk-Users] Registration refused on DIAX096b Anybobdy can helpme with this error? IAX.CONF [tito2] type=user username=tito2 secret=mysecret host=dynamic context=demo ;sendani=no ;host=asterisk.linux-support.net ;port=5036 ;mask=255.255.255.255 qualify=yes ; Make sure this peer is alive On the server: NOTICE[1142135600]: File chan_iax2.c, Line 2832 (register_verify): No registration for peer 'tito2' (from myipaddress) Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ Timestamp: 1ms SCall: 1 DCall: 02606 [myipaddress:56439] CAUSE : Registration Refused Dan or Mark should have the definitave answer here, but I've been setting my DIAX extensions to type=friend. - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External Email Notification -2
On Wed, 2003-12-10 at 14:42, Kevin wrote: Perhaps I should have been more specific, it wasn't clear in the discussion on this forum yesterday. I would my voice message waiting notification to be forwarded to my external ISP POP3 account. What is involved to set this up? Is configuring sendmail what is required? If so, are there any pointers for a configuration? Not a unix expert here... Drop sendmail and use exim. It has a simple configurator and will ask you the questions you need to get configured. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Article on Asterisk: German Linux magazine
a friend just notified my of the the cover story of freeX 1'2004: Linux als Telefonanalage (engl.: Linux as PBX) http://www.cul.de/freex.html Notice that gnophone was on the cover. We really need to resurect it as a project! Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registration refused on DIAX096b
Hi, - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 10, 2003 11:00 PM Subject: Re: [Asterisk-Users] Registration refused on DIAX096b - Original Message - From: Hector Q.-datafull [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 10, 2003 1:21 PM Subject: [Asterisk-Users] Registration refused on DIAX096b Anybobdy can helpme with this error? IAX.CONF [tito2] type=user I do not think you can register as user. Use peer or friend. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] External Email Notification -2
I was hopeful that it was a simple config file entry to accomplish external voice mail notification. Perhaps there is a knowledgeable individual that can help for a small fee... -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 10, 2003 3:52 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] External Email Notification -2 On Wed, 2003-12-10 at 14:42, Kevin wrote: Perhaps I should have been more specific, it wasn't clear in the discussion on this forum yesterday. I would my voice message waiting notification to be forwarded to my external ISP POP3 account. What is involved to set this up? Is configuring sendmail what is required? If so, are there any pointers for a configuration? Not a unix expert here... Drop sendmail and use exim. It has a simple configurator and will ask you the questions you need to get configured. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pridump
Hi All, Can anyone tell me what are the dev1 dev2 parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip.conf and Codecs
I have been doing some testing and have found issue with certain devices and negotiating codecs in doing this Ihave noticed something that seems peculiar to me. It seems that including allow=all yields different results than having no disallow or allows in the sip.conf. Could someone please explain why that is true? Thanks
[Asterisk-Users] Native Bridging and Polycom 600 Solved
Hi,
The Polycom 600 phones do not natively bridge with Asterisk. I've solved the
problem, but I'm not sure how general it is, so I thought I'd ask this list
for advice.
It's necessary to use a recent Asterisk CVS for this, since there was a
problem with session versions in earlier CVS builds.
The problem now is the Via field. When the reinvite goes out, the branch
number does not change from its value in the previous invite. However, the
Polycom phone tracks its transactions this way - the branch numbers must be
different for new invites. So here's the change:
In chan_sip.c, in transmit_reinvite_with_sdp():
static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp,
struct ast_rtp *vrtp)
{
struct sip_request req;
if (p-canreinvite == REINVITE_UPDATE)
reqprep(req, p, UPDATE, 0);
else {
// BEGIN POLYCOM CHANGE
p-branch++;
snprintf(p-via, sizeof(p-via), SIP/2.0/UDP
%s:%d;branch=z9hG4bK%08x, inet_ntoa(p-ourip), ourport, p-branch);
// END POLYCOM CHANGE
reqprep(req, p, INVITE, 0);
}
... the rest of the method follows.
Does anyone with any detailed knowledge of other SIP phones know if this will
cause something bad to happen? And, if any Asterisk developers are reading,
could they comment if this will cause problems (memory, etc.)?
Thanks,
Christian
___
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RE: [Asterisk-Users] Errors after re-plugging T1
Hi, not sure if this is your case, but a got rid of my error 500 messages today by changing the machine's motherboard. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Markus Mayer Sent: quarta-feira, 10 de dezembro de 2003 15:18 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Errors after re-plugging T1 Hi, After temporarily pulling the T1 cable out of our Asterisk box, we ended up getting a strange error messages even after the cable was plugged back in. [...] Dec 10 09:01:11 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 Dec 10 09:01:21 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 Dec 10 09:03:42 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 Dec 10 09:03:52 WARNING[1192437440]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 63 failed: Unknown error 500 [...] So I stopped asterisk, unloaded the kernel modules and restarted everything, but still: Dec 10 09:06:16 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: !! No channel map, no channel, and no ds1? What am I supposed to identify? Dec 10 09:06:16 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: !! Unable to add IE 'Channel Identification' Dec 10 09:06:20 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 62 failed: Unknown error 500 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 1 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 2 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 3 Dec 10 09:06:20 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 4 [...] Dec 10 09:18:07 WARNING[1184048960]: File chan_zap.c, Line 5683 (zt_pri_error): PRI: Read on 62 failed: Unknown error 500 Dec 10 09:18:07 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 1 Dec 10 09:18:07 NOTICE[1200825920]: File chan_zap.c, Line 4634 (handle_init_event): Alarm cleared on channel 2 I tried this several times, to no avail. Only rebooting the box helped. The question now is: is there a way to avoid rebooting in a situation like this and still get everything to work again? Rebooting can be a huge pain. Thanks. Regards, Markus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura SPA2000 Asterisk latest firmware (1.0.18)
All, If you currently own a Sipura SPA2000, avoid going to the sipura website and upgrading the firmware. I upgraded my SPA2k a couple of days ago from 1.0.9 (what it came with) to 1.0.18 off the site, and I am having issues with my SPA rebooting itself every 3-10 minutes for no apparent reason. I have been in touch with the *excellent* sipura support folks, and they are working with me to resolve the issue (they can't duplicate it!). They have sent me some debugging firmware and we're in the data gathering phase right now.. Just figured I'd send everyone here a heads up that you might have problems with the latest firmware though. (I shoulda followed the old rule of if it aint broke, don't fix it!). Also, if anyone out there has an SPA2000 and has tried the latest firmware with *success*, I'd be interested in hearing from you, otherwise I'd recommend avoiding it for the time being ;) Thanks! Pat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] External Email Notification -2
I was looking for an assistance to help with the configuration of the email on the unix server not an expert on forum postings. -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 10, 2003 4:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] External Email Notification -2 Kevin wrote: I was hopeful that it was a simple config file entry to accomplish external voice mail notification. Perhaps there is a knowledgeable individual that can help for a small fee... Start with reading the sample voicemail configuration file that is included in your asterisk download. It should be pretty simple to get voicemails forwarded to any e-mail address you have by reading those configuration files. Voicemail by e-mail is a standard feature that only requires basic configuration. And reading the config files is not very complicated and necessary if you want to do anything with Asterisk. When you've read them and found the pages on this subject on the Wiki and in the handbook, please feel free to ask more questions. http://www.voip-info.org http://search.voip-forum.com and the config files in /etc/asterisk /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pridump
two d channels of two separate pris Martin On Wed, 10 Dec 2003, Paulo Mannheimer wrote: Hi All, Can anyone tell me what are the dev1 dev2 parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WAV file volume
We've had exactly the same problem at one of our customers. They have an Adtran TSU600, which is different than the TA750/850's we use everywhere else. I was assuming it was channelbank related, but after finding that the volume on normal calls was fine, I guess it isn't. Does anybody have a solution? -wade -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glenn B. Lawler Sent: Wednesday, December 10, 2003 12:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] WAV file volume We are using Voicemail 2 and use the e-mail with WAV file attachment option. When we record a message from an internal extension, the WAV file sounds fine. However, if a caller from outside leaves a message, the volume on the WAV file is so low it is almost useless. When we access the voicemail through an extension, however, the volume sounds fine. Has anyone had this problem or any idea of a way to fix it? - Glenn Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sip.c update to 1.253
Can someone tell me what this setting is supposed to be? peer-nat = globalnat; It looks like it's inheriting a parameter, but I'm curious, is globalnat an option that we're supposed to set(or let default) in sip.conf? - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura SPA2000 Asterisk latest firmware (1.0.18)
Patrick Cantwell wrote: All, If you currently own a Sipura SPA2000, avoid going to the sipura website and upgrading the firmware. I upgraded my SPA2k a couple of days ago from 1.0.9 (what it came with) to 1.0.18 off the site, and I am having issues with my SPA rebooting itself every 3-10 minutes for no apparent reason. I have been in touch with the *excellent* sipura support folks, and they are working with me to resolve the issue (they can't duplicate it!). They have sent me some debugging firmware and we're in the data gathering phase right now.. Just figured I'd send everyone here a heads up that you might have problems with the latest firmware though. (I shoulda followed the old rule of if it aint broke, don't fix it!). Also, if anyone out there has an SPA2000 and has tried the latest firmware with *success*, I'd be interested in hearing from you, otherwise I'd recommend avoiding it for the time being ;) Thanks! Pat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Hmm... Mine seems to be working just fine. Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA2000 Asterisk latest firmware (1.0.18)
We are in the process of doing testing with the SPA and our SER servers. We have not seen your problem and we are using 1.0.18. We have seen a nasty G.729 codec problem when interoping with the GS Phones. I have consistently reproduced the problem for Sipura and hopefully they will fix it. Next week we will start testing with * and let you know. Are you sending those reboot messages in the SYSLOG server too? Andres. http://www.telesip.net - Original Message - From: Patrick Cantwell [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 10, 2003 5:04 PM Subject: [Asterisk-Users] Sipura SPA2000 Asterisk latest firmware (1.0.18) All, If you currently own a Sipura SPA2000, avoid going to the sipura website and upgrading the firmware. I upgraded my SPA2k a couple of days ago from 1.0.9 (what it came with) to 1.0.18 off the site, and I am having issues with my SPA rebooting itself every 3-10 minutes for no apparent reason. I have been in touch with the *excellent* sipura support folks, and they are working with me to resolve the issue (they can't duplicate it!). They have sent me some debugging firmware and we're in the data gathering phase right now.. Just figured I'd send everyone here a heads up that you might have problems with the latest firmware though. (I shoulda followed the old rule of if it aint broke, don't fix it!). Also, if anyone out there has an SPA2000 and has tried the latest firmware with *success*, I'd be interested in hearing from you, otherwise I'd recommend avoiding it for the time being ;) Thanks! Pat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA2000 Asterisk latest firmware (1.0.18)
Patrick Cantwell wrote: snip Also, if anyone out there has an SPA2000 and has tried the latest firmware with *success*, I'd be interested in hearing from you, otherwise I'd recommend avoiding it for the time being ;) working fine here as well. was not able to install manually since the sipura site had only a windows .exe last time I checked; however, it got upgraded to 1.0.18 when I signed up for the free month of voicepulse service and they provisioned it through tftp. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
There is a _good_ reason to ask too. I've been experimenting I buy new equipment but I'm still looking to reduce power, heat, noise and space to the bare minimum. No need to buy a CPU that burns 120W of power if you can use a one that uses 45W and lets you get rid of one of the fans. Same with disk drives. More RAM might let you use a low noise/low heat drive rather then that 7200RM noise maker. I'd like to be able to install a notebook sized drive on the * server. No, actually that's a terrible reason to ask. If you are unfamiliar with * you have no business trying to optimize your system. This is typical early optimization that plagues any design or deployment. Just learning * is NOT the place to try and cut corners and save some coin. Get it working, get familiar with it and THEN see where your specific needs are and optimize for them. Early optimization is a problem everywhere, not just in programming. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip.c update to 1.253
most propably the globalnat is nat= defined in the [general] section. Martin On Wed, 10 Dec 2003, Andrew Thompson wrote: Can someone tell me what this setting is supposed to be? peer-nat = globalnat; It looks like it's inheriting a parameter, but I'm curious, is globalnat an option that we're supposed to set(or let default) in sip.conf? - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue bug with call transfer
Kevin Bockman wrote: Yeah I guess you did. I'm not really a programmer so I don't know where it is. All I know is that it's there somewhere and it works. --- Jonathan Tew [EMAIL PROTECTED] wrote: Kevin, I've looked at the source of app_queue.c and can see if the logic for the * key, but nothing related to # in the code. Am I missing something? Thanks, Jonathan Kevin Bockman wrote: --- Jonathan Tew [EMAIL PROTECTED] wrote: We've got the app_queue configured to supposedly allow for a call to be transferred. When the call comes in and an agent answers it (using X-Lite Pro) and then decides to transfer the call (using the SIP phone interface) they get disconnected from their call and after left logged in to the queue system. Obviously we're doing something wrong to transfer the call. We hit * to hangup the call. Is there some other way to transfer the call? I've looked through the source and didn't see it. Thanks, Jonathan ___ Wow, surprised no one answered. Maybe your subject scared everyone off. * is for hangup. # is for transfer. I'm not sure if there is native sip transfer in x-lite or not or if it works or not. Kevin try looking in chan_agent.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
We assume this kind of experimentation is happing in a lab enviroment. Failure there is not a problem it's a data point. In a test system I can take out half the RAM, slow the CPU clock or run the CPU without the cooling fan and just measure what happens. Yes, stupid do do those things in a system people are depending on. --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: There is a _good_ reason to ask too. I've been experimenting I buy new equipment but I'm still looking to reduce power, heat, noise and space to the bare minimum. No need to buy a CPU that burns 120W of power if you can use a one that uses 45W and lets you get rid of one of the fans. Same with disk drives. More RAM might let you use a low noise/low heat drive rather then that 7200RM noise maker. I'd like to be able to install a notebook sized drive on the * server. No, actually that's a terrible reason to ask. If you are unfamiliar with * you have no business trying to optimize your system. This is typical early optimization that plagues any design or deployment. Just learning * is NOT the place to try and cut corners and save some coin. Get it working, get familiar with it and THEN see where your specific needs are and optimize for them. Early optimization is a problem everywhere, not just in programming. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
We assume this kind of experimentation is happing in a lab enviroment. Failure there is not a problem it's a data point. Agreed, but then they'd be posting the test result data to the list, not asking what the minimums are. In a test system I can take out half the RAM, slow the CPU clock or run the CPU without the cooling fan and just measure what happens. Yes, stupid do do those things in a system people are depending on. Agreed 100%. If you want to conduct experiments you do so by conducting them, not asking what the minimums are. I dunno, I am kind of in the same boat as Steven on this... if you're gonna experiment then experiment. Don't decide to get into * and the first thing out of your mouth is what's the bare minimum processor+ram you can get to make it work... buy something moderately new (P3, 128M, IDE disk) -- it ain't gonna break the bank, it's gonna be easier to find and likely be far more reliable than that P90 you have in the back room that's been gathering dust for the past 5 years. Sorry I don't mean to be so bitter... I probably shouldn't be posting with this headache. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] was FXO cards
Michael, Digium is working on a 4 port FXO card and hopefully it'll be ready soon. However, if you have to install a system now, your best bet is a multi-port fxo sip gateway; they come in 1,2,4... ports per box. i havn't been following prices lately but you should be able to pick one up for about US$100 / port which works out to be most economical solution for a 6 Co lines situation. cheers, patrick Michael Rowley wrote: Hey guys, appreciate the input. Here are some thoughts. ADSI phones are out of the question. This is a business environment, I can't worry about my employees not knowing how to forward calls, answer calls when away from the multiline phone, and no ADSI phone will handle multiple lines that I have found. I would love to put 6 X100P cards in a case and run asterisk on it. but... I hate the idea of running a channel bank. It just seems unecessary, and more crap to break, set up, and buy. DIalogic 4 and 12 port cards are expensive. The 12 port prohibitively so, so I think I would be better off getting 2 of the 4 port cards. I have found them for 900$ apiece on ebay. That compairs as: Case 1 Channel bank Asterisk box: 500$ (provably less than this, as it is a minimal machine) T100P card: 500$ Channel Bank 600$ (actually, from what I can find, provably more like 1000$ before I am done) Phones: I am going to go with the Mitel Networks 5055, they are about 400$ apiece. Total: 1600$ for the setup (to 2000$) + 2000K for 5 phones. Case 2: FXO cards Asterisk box 500$ Dialogic 4 port card 900$ Phones: same as the above Total 1400$ plus phones, 2K$ for 5 phones. Case 3 proprietary system Mitel 3100 server: 3K Phones: 250$ apiece, 1250$ total. 4250$ total. Case 1 and 2 are ties in my eyes, except the channel bank would provably be cheaper to upgrade to 8 lines. I am just afraid of the channel bank. I just don't know anything about them. If I buy the wrong crap, it gets really expensive fast, plus adds another layer of complexity. Michael Rowley MD FP ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] External Email Notification -2
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Sent: Wednesday, December 10, 2003 5:06 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] External Email Notification -2 I was looking for an assistance to help with the configuration of the email on the unix server not an expert on forum postings. Then you came to the wrong place. If you need help configuring email on a unix server, try the support forum for the mailer installed on your machine. It's not an on-topic discussion here. After you have that working, I'm sure many people on this forum will be more than happy to assist you in getting * to use a properly functioning mailer to send voicemail notification. Here's more unsolicited forum posting advice: the attitude you are taking in the above quoted post does not inspire people to help you in any way. Daryl G. Jurbala BMPC Network Operations Tel: +1 215 825 8401 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXO cards
On Tue, 9 Dec 2003, Barton Hodges wrote: [EMAIL PROTECTED] wrote: On Tue, 2003-12-09 at 15:18, Michael Rowley wrote: Hey guys, has anyone put 6 of the wildcat X100P cards in one machine? I am thinking about putting 6 in one machine, what is everyone elses experience Read the docs. 2 card maximum sane install. Can you point me to the documentation that states this? If I need to connect 3 or 4 pstn lines, are my only choices to add another box and connect them via IAX trunking, or to wait for the 4-port FXO card? Does anyone know when the 4-port card will be released? It is possible but not recommended to put more then 2 x100p's in a box. I have a system with a TDM400 and 4 X100p's. Key is to get a motherbrd that lets you assign IRQ resources since you do not want the above cards to share IRQ's (That said the TDM and an X100 do share an IRQ without a problem but this is a 2.4GHZ machine) Using more then one box is best. As for the FXO modules I have passed out many many times holding my breath! :) John - NetRom Internet Services973-208-1339 voice [EMAIL PROTECTED] 973-208-0942 fax http://www.netrom.com - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
In a test system I can take out half the RAM, slow the CPU clock or run the CPU without the cooling fan and just measure what happens. Yes, stupid do do those things in a system people are depending on. Agreed 100%. If you want to conduct experiments you do so by conducting them, not asking what the minimums are. I dunno, I am kind of in the same boat as Steven on this... if you're gonna experiment then experiment. Don't decide to get into * and the first thing out of your mouth is what's the bare minimum processor+ram you can get to make it work... buy something moderately new (P3, 128M, IDE disk) -- it ain't gonna break the bank, it's gonna be easier to find and likely be far more reliable than that P90 you have in the back room that's been gathering dust for the past 5 years. But... if you place yourself in the position of the newbie, where else could you ask given the documentation that truly doesn't exist (yet). Mark made the comment about a month ago that asterisk is this best kept secret in the world. The flip side of that is jumping on every newbie that comes along and pissing them off enough to leave the list (and the app). For those that have been around here for more then 30 days, you already know that's the nature of this list. If you don't like the questions, delete it and stop cluttering the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
You need to be careful about Intel's cheapo Celeron chips. While they are cheaper than P-4s I just recently saw a well documented comparision that found that the AMD's Duron and Athlon are far more capable for the same or less $$$. This can be important when translating between codecs in *. My new * server is running an Athlon XP 2500+ which only cost me $89 at a local supplier. The whole machine cost only $350. Michael On Wed, 10 Dec 2003 13:58:59 -0600, Rich Adamson wrote: I have been reading asterisks and everything I can get my hands on for the past week. I want to know what class processor is the bare minimum I need for a four port Asterisk installation? As a low-end data point (probably not cool for a reasonable production box), I purchased a eMachine T2240 with a 2.2ghz Celeron, 40 gig drive, 384 meg ram, with integrated 10/100 nic from Circuit City (new, open box, with warranty, $300). Its running asterisk with 2 x100p's, festival, sendmail, apache, mysql, MOH, X11, Gnome, etc, just fine. Asterisk has three iax trunks running, a couple of remote nat'ed 7960's, a few local sip phones, and nothing very fancy for a dial plan. The size of the box (and its architecture) is far more related to voice traffic volumes and uptime then it is anything else. (e.g, if you never place a phone call, you don't need any resources; if you never click the console mouse, gnome is not consuming any cpu resource, etc, etc.) In an idle condition (no calls being processed), top is the heaviest app. Placing a single asterisk demo call from a sip phone (forcing iax2 to Digium) causes asterisk to bump towards the top at about 0.3% cpu utilization with an occasional random peak at 2% cpu. (A single pstn call via x100p to a 7960 averages about 1.0% cpu. Both of these are eyeball inspection of top.) I don't know what you mean by port in your statement, so can't comment on machine size. If you mean four physical pstn lines, I'd have to venture a guess and say the above machine could handle four x100p's. But, as you've already seen from the list, there are probably a dozen different ways to configure * to support any given set of requirements with different cost/benefit trade offs for each. The above just happens to be one of the the cheapest around for new assembled equipment with a so called warranty. Don't think I'd install it at a hospital or police department though. ;) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 People said that there is no economic model for it, but there is: the economic model of flower boxes...I put out flower boxes to raise my self esteem and make my house more attractive. If almost everyone does then the whole town becomes beautiful. The same thing can happen with communications. - Nicolas Negroponte ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
But... if you place yourself in the position of the newbie, where else could you ask given the documentation that truly doesn't exist (yet). Agreed. Perhaps the wiki needs a all you need is a P3-class machine to get started we can get rid of these types of questions. Then I can jump on those who still do. :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Computing horsepower needed
-Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 10, 2003 8:45 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Computing horsepower needed In a test system I can take out half the RAM, slow the CPU clock or run the CPU without the cooling fan and just measure what happens. Yes, stupid do do those things in a system people are depending on. Agreed 100%. If you want to conduct experiments you do so by conducting them, not asking what the minimums are. I dunno, I am kind of in the same boat as Steven on this... if you're gonna experiment then experiment. Don't decide to get into * and the first thing out of your mouth is what's the bare minimum processor+ram you can get to make it work... buy something moderately new (P3, 128M, IDE disk) -- it ain't gonna break the bank, it's gonna be easier to find and likely be far more reliable than that P90 you have in the back room that's been gathering dust for the past 5 years. But... if you place yourself in the position of the newbie, where else could you ask given the documentation that truly doesn't exist (yet). Mark made the comment about a month ago that asterisk is this best kept secret in the world. The flip side of that is jumping on every newbie that comes along and pissing them off enough to leave the list (and the app). For those that have been around here for more then 30 days, you already know that's the nature of this list. If you don't like the questions, delete it and stop cluttering the list. I really have to agree with Rich. I am a newbie. I have been reading the lists and watching the IRC and doing what I can to learn. The general feel that I get is a feeling of intolerance for people trying to learn and understand. Is it intentional? Do the people in the know not want anyone else to be part of a great app? Just wondering. Please try to have some consideration for everyone and follow Rich's suggestion, If you don't like the questions, delete it and stop cluttering the list. I will continue to and use and learn because I think it is a great app. Just my observations. Thanks, Dustin Knuttgen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
I really have to agree with Rich. I am a newbie. I have been reading the lists and watching the IRC and doing what I can to learn. I have no problem with that. The general feel that I get is a feeling of intolerance for people trying to learn and understand. Is it intentional? Do the people in the know not want anyone else to be part of a great app? Just wondering. Please try to have some consideration for everyone and follow Rich's suggestion, If you don't like the questions, delete it and stop cluttering the list. What it seems to be is that the newbies seem to dislike searching the list before asking questions and the repeated questions that really have zero bearing on the actual application. I mean honestly why do people constantly ask what the bare effing minimum system they can get away with is? I am not very experienced with * but that wasn't one of my first questions. I grabbed a system and tried it. I didn't purposely select the 80386DX/33 in my basement. I tried it on my current system and went wow this is pretty damned cool! I then wondered if I could get away with it on a lessor system. I had a P200MMX lying around so I tried it. It seemed to work but ilbc was too CPU hungry. I'm not saying my way is the only way to do things. I am saying that the archives are available and the IRC channel is always around and busy (and anyone who's on there will know I'm pretty helpful there) -- perhaps it's just once in a while the people who are hanging around on the list and IRC get a little fed up with the same nonsensical questions and snap a little. Honestly -- if you're that thin-skinned go buy a PBX or buy commercial support for * and be done with it. The lists and IRC are kind of self-serve. It's the same with every open-source project, IMO. I don't buy it for a second that the people asking what the minimum requirements are were doing it for experimentation or purchasing; they were wondering if their old ancient systems gathering dust would be useful, and then they'd come back in a few days complaining that the systems were hanging, voice was choppy, they couldn't get the cards on separate IRQs and the host of other problems that invariably come with trying to cheap out on a project. And then Asterisk still ends up with a black eye because it sucks. I've been there and seen it enough times to be able to see the pattern emerging. Just the same -- if my posts annoy you, delete them and stop cluttering the list. It works both ways. I will continue to and use and learn because I think it is a great app. Thank you. And honestly, welcome. I will put something about minimum system requirements on the wiki. They'll be a little conservative but at least it has a better chance at bettering the community and reducing these kinds of questions on the list. Can we put this discussion down now, please? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX and PDAs
On Dec 10, 2003, at 1:52 PM, Leif Madsen wrote: On Wed, 2003-12-10 at 10:47, Steve Underwood wrote: Hi all, Does anyone know of any work in progress on IAX based telephony for PDAs? Putting iaxcomm on a Zaurus or iPAQ, for example. If someone where doing this, I would be the most ecstatic, happy person in the entire whole big wide world. Seriously... someone PLEASE make an IAX client for my Zaurus :) (I would if I had programming knowledge... doh!) Try compiling testcall (included in libiaxclient) for your zaurus. It might work. On the other hand, there's probably some floating point in there that may make it unacceptably slow -- you might need to disable some of the DSP features for it to perform. The zaurus supports the OSS sound device API, right? Then, you may soon be able to compile iaxcomm for your zaurus, depending on the stability of the embedded wx ports. Or, someone can make a native qt interface or whatever the zaurus uses.. -SteveK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Computing horsepower needed
On Wed, 2003-12-10 at 21:29, Andrew Kohlsmith wrote: But... if you place yourself in the position of the newbie, where else could you ask given the documentation that truly doesn't exist (yet). Agreed. Perhaps the wiki needs a all you need is a P3-class machine to get started we can get rid of these types of questions. Then I can jump on those who still do. :-) Regards, Andrew The address of the Wiki/FAQ should probably be added to the footer of the Mailing list. It helps with other lists - though newbies (like me) still tend to ignore it... http://www.voip-info.org/wiki-Asterisk Jon Carnes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729
Hi guys, Just installed G.729 (from digium) codec and after starting asterisk getting the following warning: [codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator) WARNING[1082809536]: File asterisk.c, Line 234 (listener): Select retured error: Interrupted system call WARNING[1082809536]: File asterisk.c, Line 234 (listener): Select retured error: Interrupted system call == Detected 1 licensed G.729 transcoders WARNING[1074420608]: File translate.c, Line 219 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator 'g729tolinb' from format G729A to SLINR, cost 9 == Registered translator 'lintog729b' from format SLINR to G729A, cost 140 Can someone shed some light on this please. Alex. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP Codec Error(s) - Is there really a solution for this or these?
Hi All, I have add the below error ever since installing and running with * for the past 6 months. It only occurs on calls from * to a H.323 gateway. I am using chan_h323. I have searched HI and LOW for a solution within the archives and elsewhere. When the error presents on the console - there is a millisecond pause in the active voice call . . . . NOTICE[1265529664]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec 19 received NOTICE[1265529664]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec 19 received Thanks in desparation for any ideas Regards, Steven Thomas Technical Project Manager Network Connectivity Services, IBM Australia Ph: 0404 099 262 NH011, IBM Centre, St Leonards, 2065 Internet: [EMAIL PROTECTED] Visit us at http://www.ibm.com/services/au/its
[Asterisk-Users] A solution to free line notification
Barton Hodges wrote:
I've been messing around with a free line notification
where an extension is dialed for a second when a line becomes
available. I can't seem to get the h extension to continue
when the local party hangs up. I've seen references to other
people having the same problem in the list archives, and the
solution presented was to use AGI.
I finally figured out how to get this to work. Thanks to one of Steven
Critchfield's emails today, I found out about sample.call and
/var/spool/asterisk/outgoing which is what I needed to control the
dialing. It seems that if you call a macro from within an h
extension, only one, or a few select lines get called before the macro
returns. I messed around with different alternatives until I found one
that worked. I would give anything for control structures and
user-defined functions within the dialplan. A nice little for() loop
would tidy things up nicely. Is AGI what I need to be using? I wasn't
sure how to do things such as Dbget(), except through the Exec() call.
Here are snippits to show how it was done:
/var/lib/asterisk/agi-bin/fln.agi:
#!/bin/sh
[ $# -gt 0 ] || exit 0;
echo -e Channel: ${1}
WaitTime: 1
Callerid: Free Line Notification (000) 000-
Context: default
Extension: s
Priority: 1 /var/spool/asterisk/outgoing/fln.$$
/etc/asterisk/extensions.conf:
[from-inside]
include = to-internal
include = app-freeline
exten = h,1,Macro(hangup)
[check-fln]
exten = s,1,DBget(TECH=FLN/${EXT})
exten = s,2,ChanIsAvail(Zap/1Zap/2Zap/3Zap/4)
exten = s,3,DBdel(FLN/${EXT})
exten = s,4,AGI(fln.agi,${TECH}/${EXT})
exten = s,5,Goto(macro-hangup,s,${PRI})
exten = s,102,Goto(macro-hangup,s,${PRI})
exten = s,103,Goto(macro-hangup,s,${PRI})
exten = s,104,Goto(macro-hangup,s,${PRI})
[macro-hangup]
exten = s,1,SetVar(PRI=4)
exten = s,2,SetVar(EXT=111)
exten = s,3,Goto(check-fln,s,1)
exten = s,4,SetVar(PRI=7)
exten = s,5,SetVar(EXT=112)
exten = s,6,Goto(check-fln,s,1)
exten = s,7,SetVar(PRI=10)
exten = s,8,SetVar(EXT=113)
exten = s,9,Goto(check-fln,s,1)
exten = s,10,Wait(1)
exten = s,11,Hangup
[macro-goodbye-hangup]
exten = s,1,Playback(vm-goodbye)
exten = s,2,Macro(hangup)
[app-freeline]
exten = _*98,1,Cut(CHAN=CHANNEL,-,1)
exten = _*98,2,Cut(TECH=CHAN,/,1)
exten = _*98,3,Cut(EXT=CHAN,/,2)
exten = _*98,4,DBput(FLN/${EXT}=${TECH})
exten = _*98,5,Answer
exten = _*98,6,Playback(contrib/activated)
exten = _*98,7,Playback(vm-for)
exten = _*98,8,Playback(vm-extension)
exten = _*98,9,SayDigits,${CALLERIDNUM}
exten = _*98,10,Macro(goodbye-hangup)
exten = _*99,1,Cut(CHAN=CHANNEL,-,1)
exten = _*99,2,Cut(EXT=CHAN,/,2)
exten = _*99,3,DBdel(FLN/${EXT})
exten = _*99,4,Answer
exten = _*99,5,Playback(contrib/de-activated)
exten = _*99,6,Playback(vm-for)
exten = _*99,7,Playback(vm-extension)
exten = _*99,8,SayDigits,${CALLERIDNUM}
exten = _*99,9,Macro(goodbye-hangup)
___
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Re: [Asterisk-Users] RTP Codec Error(s) - Is there really a solution for this or these?
On Wednesday 10 December 2003 23:15, Steven Thomas wrote: Hi All, I have add the below error ever since installing and running with * for the past 6 months. It only occurs on calls from * to a H.323 gateway. I am using chan_h323. I have searched HI and LOW for a solution within the archives and elsewhere. When the error presents on the console - there is a millisecond pause in the active voice call . . . . NOTICE[1265529664]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec 19 received NOTICE[1265529664]: File rtp.c, Line 418 (ast_rtp_read): Unknown RTP codec 19 received First, the solution: the provider needs to turn off Comfort Noise. Second, the vendor of the product needs to update their equipment to use the RIGHT codec number for Comfort Noise, which is 13. RTP 19 has been reserved (i.e. deprecated) for several years and SHOULD NOT BE USED. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unknown RTP codec 19
On Wednesday 10 December 2003 11:08, SW wrote: Hi Folks, I had success with making SIP to H323 g/w calls using chan_h323 and G.729. Calls go through well though it takes incredibly long time to get connected which I thing due to fast start being disabled at the remote end, will ask remote side to fix it. There is a Notice message which appears on my * console which bothers me. While in a conversation, I get the following; File rtp.c line 418 (ast_rtp_read): Unknown rtp codec 19 received. I read in the mail archive that rtp codec 19 is comfort noise. Should I ignore this notice ? Or is there a better way to tell * asterisk to handle comfort Noise ? Asterisk does not handle comfort noise. Turn it off on the other side of the connection, if possible. Also, bug your vendor to update their software, as RTP 19 has been reserved for several years. RTP 13 is the correct codec number for comfort noise. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
