Re: [asterisk-users] polarity reverse
I do not have the possibility to check does TDM800 works with asterisk 1.6.2. I checked i have this code in chan_dahdi file. But when I try to call, I get only chan_dahdi.c: Using channel 11 devicestate.c: device 'DAHDI/11-1' state '2' rtp.c: Channel 'DAHDI/11-1' has no RTP, not doing anything channel.c: Not copying variable DIALEDTIME. channel.c: Not copying variable ANSWEREDTIME. channel.c: Not copying variable DIALEDPEERNAME. channel.c: Not copying variable DIALEDPEERNUMBER. DEBUG[9730] channel.c: Not copying variable DIALSTATUS. DEBUG[9730] channel.c: Not copying variable SIPCALLID. DEBUG[9730] channel.c: Not copying variable SIPDOMAIN. DEBUG[9730] channel.c: Not copying variable SIPURI. DEBUG[9548] app_queue.c: Device 'DAHDI/11-1' changed to state '2' (In use) but we don't care because they're not a member of any queue. DEBUG[9730] chan_dahdi.c: Ignore possible polarity reversal on line seizure DEBUG[9730] chan_dahdi.c: Dialing '8685X' DEBUG[9730] chan_dahdi.c: Deferring dialing... VERBOSE[9730] app_dial.c: -- Called 11/8685X DEBUG[9544] devicestate.c: No provider found, checking channel drivers for DAHDI - 11 DEBUG[9544] channel.c: Avoiding initial deadlock for channel '0x1d81910' DEBUG[9730] channel.c: Set channel DAHDI/11-1 to read format slin DEBUG[9544] devicestate.c: Changing state for DAHDI/11 - state 2 (In use) DEBUG[9544] devicestate.c: device 'DAHDI/11' state '2' DEBUG[9730] channel.c: Set channel SIP/XXX-000b to write format slin DEBUG[9730] channel.c: Set channel SIP/XXX-000b to read format slin DEBUG[9730] channel.c: Set channel DAHDI/11-1 to write format slin DEBUG[9548] app_queue.c: Device 'DAHDI/11' changed to state '2' (In use) but we don't care because they're not a member of any queue. DEBUG[9730] chan_dahdi.c: Exception on 15, channel 11 DEBUG[9730] chan_dahdi.c: Got event Hook Transition Complete(12) on channel 11 (index 0) DEBUG[9730] chan_dahdi.c: Sent deferred digit string: T8685Xw DEBUG[9730] chan_dahdi.c: Exception on 15, channel 11 DEBUG[9730] chan_dahdi.c: Got event Dial Complete(9) on channel 11 (index 0) DEBUG[9730] chan_dahdi.c: No echo cancellation requested DEBUG[9730] chan_dahdi.c: Exception on 15, channel 11 DEBUG[9730] chan_dahdi.c: Got event On hook(1) on channel 11 (index 0) DEBUG[9730] channel.c: Hanging up channel 'DAHDI/11-1' DEBUG[9730] chan_dahdi.c: dahdi_hangup(DAHDI/11-1) DEBUG[9730] chan_dahdi.c: Hangup: channel: 11 index = 0, normal = 15, callwait = -1, thirdcall = -1 DEBUG[9730] chan_dahdi.c: Set option TDD MODE, value: OFF(0) on DAHDI/11-1 DEBUG[9730] chan_dahdi.c: Updated conferencing on 11, with 0 conference users VERBOSE[9730] chan_dahdi.c: -- Hungup 'DAHDI/11-1' On Wednesday, April 07, 2010, at 12:17AM, Alec Davis siva...@paradise.net.nz wrote: Does TDM800 with FXO ports work with 1.6.2? You should have also got other 'polarity related messages' during the call setup. One in particluar which prints debug info when a DAHDI_EVENT_POLARITY get fired. Code below. ast_debug(1, Polarity Reversal event occured - DEBUG 2: channel %d, state %d, pol= %d, aonp= %d, honp= %d, pdelay= %d, tv= %d\n, p-channel, ast-_state, p-polarity, p-answeronpolarityswitch, p-hanguponpolarityswitch, p-polarityonanswerdelay, ast_tvdiff_ms(ast_tvnow(), p-polaritydelaytv) ); If it doesn't work with the TDM800, file a bug report. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justas Gulbinskas Sent: Wednesday, 7 April 2010 8:00 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] polarity reverse call not succsessful. I use nokia gsm gw witch have polarity reverse i try on my old asterisk 1.4.17 with digium tdm800 with fxo ports card polarity reverse works fine. But then i connect to asterisk 1.6.2 with sangoma a400 with fxo ports card polarity don't work. polarity reverse is 600 milliseconds set on nokia gsm gw On Apr 6, 2010, at 10:08 PM, Alec Davis wrote: Is the call successfull? The 'Ignore polarity reversal on line seizure' may just be a warning. What equipment, which Telco is the FXO card connected to? Alec Davis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justas Gulbinskas Sent: Wednesday, 7 April 2010 12:03 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] polarity reverse Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no answeronpolarityswitch=yes I use asterisk 1.6.2 and sangoma a400 fxo ports.
Re: [asterisk-users] Asterisk Load Balancing with Redfone/TDMoE driver
hi: it is a good news for customers. we are trying to use that for a big project. hope TDMMoE stable. Best wishes! Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7, elastix, trixbox support. website:www.cnasterisk.com, www.voip88.com From: felipechu...@gmail.com To: asterisk-users@lists.digium.com Date: Thu, 1 Apr 2010 09:28:00 -0300 Subject: [asterisk-users] Asterisk Load Balancing with Redfone/TDMoE driver Redfone uses and improved, in house developed TDMoE driver, officially supported by same Redfone. Redfone´s support site maintains tdmoe driver updated and certified to operate in every zaptel and dahdi versions. Txs Jorge Churio Redfone Communications -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls
hi: how about the codecs? Best wishes! Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7, elastix, trixbox support. website:www.cnasterisk.com, www.voip88.com Date: Wed, 31 Mar 2010 17:20:30 -0500 From: br...@texascountrytitle.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dropped Calls On 3/31/2010 12:16 PM, Philipp von Klitzing wrote: Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? I was suspecting something with either rtptimeout or sip registration timeout, but I'm not sure what. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_ss7 issue
hi: are you sure the CIC is correct? Best wishes! Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7, elastix, trixbox support. website:www.cnasterisk.com, www.voip88.com Date: Sat, 27 Mar 2010 22:53:42 +0530 From: damind...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] chan_ss7 issue Gentler reminderany body to help me pls On Tue, Mar 23, 2010 at 2:20 PM, Kasun Daminda damind...@gmail.com wrote: Dear all, Do you have come acrross with this issue. My ss7 link get fluctuating. It use chan_ss7 version 1.0.95-beta. I have 8 E1s running on a DL380 server. This enable to have calls from sip to ss7 and vice versa. However ss7 links are not stable. linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 2/4, sentseq/lastack: 127/127, total 4034145216, 4031118560 linkset siuc, link l5, schannel 1, sls 1, INSERVICE, rx: 5, tx: 3/3, sentseq/lastack: 95/95, total 4030833616, 4028245568 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 2, tx: 4/4, sentseq/lastack: 127/127, total 4034149872, 4031123216 linkset siuc, link l5, schannel 1, sls 1, INSERVICE, rx: 2, tx: 3/3, sentseq/lastack: 100/100, total 4030838272, 4028250224 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 3/4, sentseq/lastack: 127/127, total 4034154480, 4031127824 linkset siuc, link l5, schannel 1, sls 1, DOWN, rx: 5, tx: 4/4, sentseq/lastack: 100/101, total 4030842880, 4028254832 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 3, tx: 1/4, sentseq/lastack: 127/127, total 4034159456, 4031132800 linkset siuc, link l5, schannel 1, sls 1, DOWN, rx: 0, tx: 0/4, sentseq/lastack: 100/101, total 4030847840, 4028259792 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 4, tx: 4/4, sentseq/lastack: 127/127, total 4034164432, 4031137776 linkset siuc, link l5, schannel 1, sls 1, NOT_ALIGNED, rx: 2, tx: 4/4, sentseq/lastack: 127/127, total 4030852816, 4028264768 ^[[a[r...@localhost ~]# asterisk -rx ss7 link status linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 5, tx: 3/4, sentseq/lastack: 127/127, total 4034169312, 4031142640 linkset siuc, link l5, schannel 1, sls 1, PROVING, rx: 4, tx: 2/4, sentseq/lastack: 127/127, total 4030857696, 4028269632 [r...@localhost ~]# asterisk -rx ss7 link status And I get a log as [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l1'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l1' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l5' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l5'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Excessive poll delay 12718! [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l1' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Full Zaptel input buffer detected, incoming packets may have been lost on link 'l5' (count=64. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l1'. [Mar 23 14:18:40] NOTICE[30389] mtp.c: Empty Zaptel output buffer detected, outgoing packets may have been lost on link 'l5'. [r...@localhost ~]# Can anybody help me on this. It will be great help. Kind Rgds Daminda _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polarity reverse
The feeling I have is the Sangoma A400 driver isn't queueing the event DAHDI_EVENT_POLARITY on each polarity reversal. File a bug report at issues.asterisk.org with all of your details, including sample dialplan and console output. The only option I think you have at the moment is to set answeronpolarityswitch=no and hanguponpolarityswitch=no, this will give you either dead air, or the caller will hear the CTU dialling out (which is comforting). Alec Davis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justas Gulbinskas Sent: Wednesday, 7 April 2010 6:49 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] polarity reverse I do not have the possibility to check does TDM800 works with asterisk 1.6.2. I checked i have this code in chan_dahdi file. But when I try to call, I get only chan_dahdi.c: Using channel 11 devicestate.c: device 'DAHDI/11-1' state '2' rtp.c: Channel 'DAHDI/11-1' has no RTP, not doing anything channel.c: Not copying variable DIALEDTIME. channel.c: Not copying variable ANSWEREDTIME. channel.c: Not copying variable DIALEDPEERNAME. channel.c: Not copying variable DIALEDPEERNUMBER. DEBUG[9730] channel.c: Not copying variable DIALSTATUS. DEBUG[9730] channel.c: Not copying variable SIPCALLID. DEBUG[9730] channel.c: Not copying variable SIPDOMAIN. DEBUG[9730] channel.c: Not copying variable SIPURI. DEBUG[9548] app_queue.c: Device 'DAHDI/11-1' changed to state '2' (In use) but we don't care because they're not a member of any queue. DEBUG[9730] chan_dahdi.c: Ignore possible polarity reversal on line seizure DEBUG[9730] chan_dahdi.c: Dialing '8685X' DEBUG[9730] chan_dahdi.c: Deferring dialing... VERBOSE[9730] app_dial.c: -- Called 11/8685X DEBUG[9544] devicestate.c: No provider found, checking channel drivers for DAHDI - 11 DEBUG[9544] channel.c: Avoiding initial deadlock for channel '0x1d81910' DEBUG[9730] channel.c: Set channel DAHDI/11-1 to read format slin DEBUG[9544] devicestate.c: Changing state for DAHDI/11 - state 2 (In use) DEBUG[9544] devicestate.c: device 'DAHDI/11' state '2' DEBUG[9730] channel.c: Set channel SIP/XXX-000b to write format slin DEBUG[9730] channel.c: Set channel SIP/XXX-000b to read format slin DEBUG[9730] channel.c: Set channel DAHDI/11-1 to write format slin DEBUG[9548] app_queue.c: Device 'DAHDI/11' changed to state '2' (In use) but we don't care because they're not a member of any queue. DEBUG[9730] chan_dahdi.c: Exception on 15, channel 11 DEBUG[9730] chan_dahdi.c: Got event Hook Transition Complete(12) on channel 11 (index 0) DEBUG[9730] chan_dahdi.c: Sent deferred digit string: T8685Xw DEBUG[9730] chan_dahdi.c: Exception on 15, channel 11 DEBUG[9730] chan_dahdi.c: Got event Dial Complete(9) on channel 11 (index 0) DEBUG[9730] chan_dahdi.c: No echo cancellation requested DEBUG[9730] chan_dahdi.c: Exception on 15, channel 11 DEBUG[9730] chan_dahdi.c: Got event On hook(1) on channel 11 (index 0) DEBUG[9730] channel.c: Hanging up channel 'DAHDI/11-1' DEBUG[9730] chan_dahdi.c: dahdi_hangup(DAHDI/11-1) DEBUG[9730] chan_dahdi.c: Hangup: channel: 11 index = 0, normal = 15, callwait = -1, thirdcall = -1 DEBUG[9730] chan_dahdi.c: Set option TDD MODE, value: OFF(0) on DAHDI/11-1 DEBUG[9730] chan_dahdi.c: Updated conferencing on 11, with 0 conference users VERBOSE[9730] chan_dahdi.c: -- Hungup 'DAHDI/11-1' On Wednesday, April 07, 2010, at 12:17AM, Alec Davis siva...@paradise.net.nz wrote: Does TDM800 with FXO ports work with 1.6.2? You should have also got other 'polarity related messages' during the call setup. One in particluar which prints debug info when a DAHDI_EVENT_POLARITY get fired. Code below. ast_debug(1, Polarity Reversal event occured - DEBUG 2: channel %d, state %d, pol= %d, aonp= %d, honp= %d, pdelay= %d, tv= %d\n, p-channel, ast-_state, p-polarity, p-answeronpolarityswitch, p-hanguponpolarityswitch, p-polarityonanswerdelay, ast_tvdiff_ms(ast_tvnow(), p-polaritydelaytv) ); If it doesn't work with the TDM800, file a bug report. Alec -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justas Gulbinskas Sent: Wednesday, 7 April 2010 8:00 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] polarity reverse call not succsessful. I use nokia gsm gw witch have polarity reverse i try on my old asterisk 1.4.17 with digium tdm800 with fxo ports card polarity reverse works fine. But then i connect to asterisk 1.6.2 with sangoma a400 with fxo ports card polarity don't work. polarity reverse is 600 milliseconds set on nokia gsm gw On Apr 6, 2010, at 10:08 PM, Alec Davis wrote: Is the call successfull? The 'Ignore polarity reversal on line seizure' may just be a warning. What equipment, which Telco is the FXO card connected to? Alec Davis
Re: [asterisk-users] FAX 2 mail configuration
How can i use both scripts
On Mon, Apr 5, 2010 at 4:55 PM, Tiago Geada tiago.ge...@gmail.com wrote:
Hi João.
We made up a script that sends received faxes trough a smtp server as an
attachment.
the FAX.ael
context FAX
{
s = {
Answer();
Set(TIMEOUT(absolute)=600); // 10 min
Wait(3);
if(${CALLERID(num)}=) { //
Set(Number=withhold); // If
number is private
} //
else {
Set(Number=${CALLERID(num)}); // If
number is NOT private
}
Set(recordFile=${UNIQUEID}_${Number}.tiff);
// Record file to RAM first,
Set(recordPath=/var/log/asterisk/fax/${CALLERID(dnid)}/${STRFTIME(${EPOCH},GMT+0,%F)});
// then run /usr/local/bin/mailfax $1 $2
ReceiveFax(/ramdrive/${recordFile});
Wait(5);
Hangup();
};
h = {
System(/usr/local/bin/faxmail ${recordPath}
${recordFile});
};
}
and the script @ /usr/local/bin/faxmail has got something like:
#!/bin/sh
PATH=/usr/sbin:/sbin:/bin:/usr/bin:/usr/local/bin
if [ -d $1 ]; then
mv /ramdrive/$2 $1;
chmod a+rx $1/$2;
else
mkdir -p $1;
mv /ramdrive/$2 $1;
chmod a+rx $1/$2;
fi
#chmod a+rx /ramdrive/$2;
{
(
sleep 1
echo ehlo tretas.eu
sleep 1
echo AUTH LOGIN
sleep 0
echo -n aster...@tretas.eu|base64
sleep 0
echo -n tretas|base64
echo MAIL FROM: aster...@tretas.eu
sleep 0
echo RCPT TO: tiago.ge...@gmail.com
echo RCPT TO: f...@tretas.eu
sleep 1
echo data
echo Subject: FAX $2
echo FROM: aster...@tretas.eu
echo TO: f...@tretas.eu
sleep 1
echo 'Content-Type: multipart/mixed; boundary=Y3VzY28udHJldGFzLmV1'
echo
echo --Y3VzY28udHJldGFzLmV1
echo 'Content-Type: multipart/alternative;
boundary=Y3VzY28udHJldGFzLmV2'
echo
echo --Y3VzY28udHJldGFzLmV2
echo 'Content-Type: text/plain; charset=ISO-8859-1'
echo
echo Fax em $(date)
echo $1/$2
echo
echo --Y3VzY28udHJldGFzLmV2
echo 'Content-Type: text/html; charset=ISO-8859-1'
echo
echo Fax em $(date)br$1/$2
echo
echo --Y3VzY28udHJldGFzLmV2--
echo --Y3VzY28udHJldGFzLmV1
echo 'Content-Type: image/tiff; name=fax.tiff'
echo 'Content-Disposition: attachment; filename=fax.tiff'
echo Content-Transfer-Encoding: base64
echo X-Attachment-Id: 0.1
echo
sleep 1;
cat $1/$2|base64
sleep 1;
echo --Y3VzY28udHJldGFzLmV1--
echo .
#echo quit
) | telnet smtp.tretas.eu 25
}
Boa sorte!
On 30 March 2010 16:29, Joao Gomes Pereira gomespere...@startel.ptwrote:
Hello
Im trying to configure Fax2Mail in my Asterisk 1.4.23.1 server, wich
receievs the Faxes through a SIP trunk.
I found a lot of solutions in voip-info.org
So, I would like to know what's the best free Fax2Mail solution and if I
really need to install Dahdi or Zaptel.
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
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[asterisk-users] Dynamic agent showing as Invalid
Hi, I have written some very simple dialplan logic for our call centre agent system so that when we log an agent into the queue they login as something like: Local/4...@roamingagent/n We have the occasional problem whereby Asterisk sees an agent as Invalid: Local/4...@roamingagent/n with penalty 10 (dynamic) (Invalid) has taken no calls yet Can anyone shed any light on what asterisk would consider Invalid? If I restart asterisk the problem goes away for some time and when it reoccurs it isn't always the same agent. All other agents using the methods to login/out are working fine. Just logging this agent out and back in again doesn't correct the problem. The RoamingAgent code just looks up the SIP extension for any given agent from the asterisk database and sends the call there. We're using Asterisk 1.6.2.0. Thanks Jordan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RES: Cache sound files for faster processing
I haven't ever try any ram disk before. --- On Tue, 4/6/10, Flavio E. Goncalves fla...@voffice.com.br wrote: From: Flavio E. Goncalves fla...@voffice.com.br Subject: [asterisk-users] RES: Cache sound files for faster processing To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Tuesday, April 6, 2010, 9:56 AM Did you tried the good old ram disk? Flavio E. Goncalves www.asteriskguide.com -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de David Backeberg Enviada em: Tuesday, April 06, 2010 12:50 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Cache sound files for faster processing On Tue, Apr 6, 2010 at 12:36 AM, huu giang huugiang...@yahoo.com wrote: Dear List, Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk, it will load from the memory and so Asterisk can process much more call at a time than with faster speed it is not caching. Thanks, Aside from the suggestions, you could try out an SSD drive, which is both expensive compared to a traditional hard drive and very fast. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cache sound files for faster processing
Thanks Steve for your information. As you said, I don't need care for caching sound files ?, Linux is responsible for the job ?, So at the first time, Asterisk will load sound files from hard disk, and after that, it will load from RAM. Thanks. --- On Tue, 4/6/10, Steve Edwards asterisk@sedwards.com wrote: From: Steve Edwards asterisk@sedwards.com Subject: Re: [asterisk-users] Cache sound files for faster processing To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, April 6, 2010, 7:15 AM Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk On Mon, 5 Apr 2010, Luki wrote: Not directly, but it's not really needed. A long as the machine has enough RAM, the files will be served from RAM by the operating system. Sure there is the overhead of opening/closing files and reading them, but on modern OS this overhead is negligible if the files are cached (asterisk may even use mmap, but I'm not sure). You can also make a ram disk (say via tmpfs), copy the sounds there and symlink the sound directory to that location. However, I don't think you will gain much. A bit off topic, but recently I was trying to improve the performance of a MythTV frontend (a Linux home theater application). I tried tmpfs and /dev/ramx and neither yielded noticeable improvement. My informal conclusion is that Linux does a good enough job at managing memory that tweaking is probably not worth it. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI + Dial + stream file ?
Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI + Dial + stream file ?
Use L() option in Dial application while originating the call. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial On Wed, Apr 7, 2010 at 7:00 PM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Regards, Godson Gera http://godson.in -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI + Dial + stream file ?
There is a parameter L which you can use in the dial command. More about it you can see on voip-info.org, but it'll be something like this: Dial(SIP/223,60,L(11000:1)) The first 11000 means 11 minutes allowed duration of the call and after 10 minutes it'll play message You have one minute. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-07 9:39 AM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI + Dial + stream file ?
Thank you Godson Zeeshan ! :-) Mickael. Zeeshan Zakaria a écrit : There is a parameter L which you can use in the dial command. More about it you can see on voip-info.org http://voip-info.org, but it'll be something like this: Dial(SIP/223,60,L(11000:1)) The first 11000 means 11 minutes allowed duration of the call and after 10 minutes it'll play message You have one minute. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-07 9:39 AM, Mickael MONSIEUR mickael.monsi...@gmail.com mailto:mickael.monsi...@gmail.com wrote: Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls
On 3/31/2010 12:16 PM, Philipp von Klitzing wrote: Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? I was suspecting something with either rtptimeout or sip registration timeout, but I'm not sure what. Hi, I have had similar issue. I have downgraded from 1.6 to 1.4 and issue got solved. Never managed to find what is going on. It was happening only if all were true: - linksys phone or pap - asterisk 1.6 - use certain VOIP provider. Solution: moved to 1.4 I hope thsi helps. Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PSTN issues
Hope some can help me. I have a PSTN coming into TDM400 into Asterisk. We also have direct telephones connected to the PSTN bypassing the Asterisk. When a call comes in on the PSTN the direct connected phones ring first and if no one picks up , Asterisk picks and get routed to internal sip phones. I am not able to find what I should tune to make the calls always go through asterisk without the direct telephones ringing. Things used to work right, suddenly, I have this problem after a recent storm. Thanks, -braman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN issues
Hi Try usecallerid=no usually the asterisk server will wait two rings before answering when usecallerid is set to yes. Flavio E. Goncalves www.asteriskguide.com 2010/4/7 Balu Raman brama...@gmail.com Hope some can help me. I have a PSTN coming into TDM400 into Asterisk. We also have direct telephones connected to the PSTN bypassing the Asterisk. When a call comes in on the PSTN the direct connected phones ring first and if no one picks up , Asterisk picks and get routed to internal sip phones. I am not able to find what I should tune to make the calls always go through asterisk without the direct telephones ringing. Things used to work right, suddenly, I have this problem after a recent storm. Thanks, -braman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] D-Channel Span Up without Down
I am getting a bunch of Primary D-Channel on span 1 up but there was not a down message before that. Is this normal? Confidentiality Statement Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or entity to whom it is addressed. Any review, retransmission, dissemination to unauthorized persons or other use of the original message and any attachments is strictly prohibited. If you received this electronic transmission in error, please reply to the above-referenced sender about the error and permanently delete this message. Thank you for your cooperation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN issues
On Wed, 7 Apr 2010, Balu Raman wrote: I have a PSTN coming into TDM400 into Asterisk. We also have direct telephones connected to the PSTN bypassing the Asterisk. When a call comes in on the PSTN the direct connected phones ring first and if no one picks up , Asterisk picks and get routed to internal sip phones. I am not able to find what I should tune to make the calls always go through asterisk without the direct telephones ringing. Things used to work right, suddenly, I have this problem after a recent storm. 0) A more specific subject will get more specific answers. Your description sounds like: You have an incoming PSTN connected to a junction strip. You have telephones and your Asterisk server connected to the junction strip. Ringing will be present at the telephones and the Asterisk server at the same time because they are connected together at the same point. There is no way for Asterisk to ring before the telephones unless your telephones don't ring immediately. If you want Asterisk to handle all incoming calls, the telephones will need to be connected to the TDM400 as well as the incoming PSTN. What changes were introduced because of the storm? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] D-Channel Span Up without Down
On Wed, Apr 7, 2010 at 11:26 AM, Jason Walker jason.wal...@amgsrv.comwrote: I am getting a bunch of Primary D-Channel on span 1 up but there was not a down message before that. Is this normal? Confidentiality Statement Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or entity to whom it is addressed. Any review, retransmission, dissemination to unauthorized persons or other use of the original message and any attachments is strictly prohibited. If you received this electronic transmission in error, please reply to the above-referenced sender about the error and permanently delete this message. Thank you for your cooperation. No, the font size is not normal. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN issues
On Wed, Apr 7, 2010 at 11:07 AM, Balu Raman brama...@gmail.com wrote: Hope some can help me. I have a PSTN coming into TDM400 into Asterisk. We also have direct telephones connected to the PSTN bypassing the Asterisk. When a call comes in on the PSTN the direct connected phones ring first and if no one picks up , Asterisk picks and get routed to internal sip phones. I am not able to find what I should tune to make the calls always go through asterisk without the direct telephones ringing. Things used to work right, suddenly, I have this problem after a recent storm. Thanks, -braman Can you turn the ring volume all the way down on the POTS phones? That is the only way I can think of that would prevent them from Ringing Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone coming to Paris next week for AstriEurope?
Randy R wrote: Several regulars from the VUC will be there, some of us are arriving Tuesday night. Anyone else considering the trip? Post here or contact me off list so we can meet. /r I'll be there... For those that don't know me, I work a lot on chan_dahdi/libss7/libpri/DAHDI. I'm not sure what my schedule is going to be like there, but I'd love to hear about any meetups that may happen if I can fit it in. Matthew Fredrickson Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone coming to Paris next week for AstriEurope?
On Wed, Apr 7, 2010 at 6:40 PM, Matthew Fredrickson cres...@digium.com wrote: I'll be there... For those that don't know me, I work a lot on chan_dahdi/libss7/libpri/DAHDI. I'm not sure what my schedule is going to be like there, but I'd love to hear about any meetups that may happen if I can fit it in. Should be fun, Matt, glad to hear you'll be there! r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk hangup all outging calls after 32 seconds
Hi again... No ideas? El mar, 30-03-2010 a las 20:05 -0500, Ing CIP. Alejandro Celi Mariátegui escribió: (Sorry, but my english is not good) Hi, I have a problem with my new asterisk instalation. I search in google but I couldn't find nothing. Here's the thing. Before, we have 2 asterisk servers, each one with a E1 card. one with a Digium TE105 and the another with a A104 and we have a very simple setup. A Linux IBM X4300 Server is running CentOS 5.4 + Asterisk 1.6.2 (one month ago I download the latest ones). No analog cards. I can make calls between my extension perfectly; When I make outgoing calls using the PRI (any E1 PRI) in some phones the call hangup without any reaseon exactly after 32 seconds Where can be the problem or the error? Regards, -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe -- Este mensaje ha sido analizado por MailScanner en busca de virus y otros contenidos peligrosos, y se considera que está limpio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe -- Este mensaje ha sido analizado por MailScanner en busca de virus y otros contenidos peligrosos, y se considera que est� limpio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk hangup all outging calls after32 seconds
Post your CLI output from one of these calls. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ing CIP. Alejandro Celi Mariátegui Sent: Wednesday, April 07, 2010 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk hangup all outging calls after32 seconds Hi again... No ideas? El mar, 30-03-2010 a las 20:05 -0500, Ing CIP. Alejandro Celi Mariátegui escribió: (Sorry, but my english is not good) Hi, I have a problem with my new asterisk instalation. I search in google but I couldn't find nothing. Here's the thing. Before, we have 2 asterisk servers, each one with a E1 card. one with a Digium TE105 and the another with a A104 and we have a very simple setup. A Linux IBM X4300 Server is running CentOS 5.4 + Asterisk 1.6.2 (one month ago I download the latest ones). No analog cards. I can make calls between my extension perfectly; When I make outgoing calls using the PRI (any E1 PRI) in some phones the call hangup without any reaseon exactly after 32 seconds Where can be the problem or the error? Regards, -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe -- Este mensaje ha sido analizado por MailScanner http://www.mailscanner.info/ en busca de virus y otros contenidos peligrosos, y se considera que está limpio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing CIP. Alejandro Celi Mariátegui a...@linux.org.pe -- Este mensaje ha sido analizado por http://www.mailscanner.info/ MailScanner en busca de virus y otros contenidos peligrosos, y se considera que está limpio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Agent Callback methods?
Since AgentCallbackLogin() was apparently removed from 1.6, does anyone have anything to replace that functionality? Thanks- Joe -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] D-Channel Span Up without Down
HahahahaI definitly agree with Steve. On Wed, Apr 7, 2010 at 11:44 AM, Steve Totaro stot...@first-notification.com wrote: On Wed, Apr 7, 2010 at 11:26 AM, Jason Walker jason.wal...@amgsrv.comwrote: I am getting a bunch of Primary D-Channel on span 1 up but there was not a down message before that. Is this normal? Confidentiality Statement Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or entity to whom it is addressed. Any review, retransmission, dissemination to unauthorized persons or other use of the original message and any attachments is strictly prohibited. If you received this electronic transmission in error, please reply to the above-referenced sender about the error and permanently delete this message. Thank you for your cooperation. No, the font size is not normal. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk hangup all outging calls after 32 seconds
Have you modified extensions_custom.conf? From the bash command line try: grep TIMEOUT\(absolute\) /etc/asterisk/extensions_*.conf and post the results. Greg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
Hi Guys, Currently, I have a Sangoma A400 installed with 20 ZAP PSTN analogue lines. The first line is giving me problems due to rain (probably coroded line). My server using FreePBX dials out with g0 (group 0 which includes all 20 lines) and it happens that the bad line is the very first line. Can I simply put ; in zapata.conf like this to seclude the first zap line from getting calls in or out? ;context=from-zaptel ;group=0 ;signalling = fxs_ks ;channel = 1 context=from-zaptel group=0 signalling = fxs_ks channel = 2 Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
bruce bruce wrote: Can I simply put ; in zapata.conf like this to seclude the first zap line from getting calls in or out? I'm not familiar with FreePBX, but I'd say that's logical. Make the change and then from a console type zap show channels, only 2 though 20 should be showing. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
Yes you can. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-07 2:42 PM, Doug Lytle supp...@drdos.info wrote: bruce bruce wrote: Can I simply put ; in zapata.conf like this to seclude the first zap line ... I'm not familiar with FreePBX, but I'd say that's logical. Make the change and then from a console type zap show channels, only 2 though 20 should be showing. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
Since this is hopefully a temporary problem, it would be simpler to simply do zap destroy channel 1 from a CLI prompt. But yes, the ; in the conf will comment these lines until you undo it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Wednesday, April 07, 2010 1:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls bruce bruce wrote: Can I simply put ; in zapata.conf like this to seclude the first zap line from getting calls in or out? I'm not familiar with FreePBX, but I'd say that's logical. Make the change and then from a console type zap show channels, only 2 though 20 should be showing. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic agent showing as Invalid
Jordan Kirby escribió: Hi, I have written some very simple dialplan logic for our call centre agent system so that when we log an agent into the queue they login as something like: Local/4...@roamingagent/n We have the occasional problem whereby Asterisk sees an agent as Invalid: Local/4...@roamingagent/n with penalty 10 (dynamic) (Invalid) has taken no calls yet Can anyone shed any light on what asterisk would consider Invalid? If I restart asterisk the problem goes away for some time and when it reoccurs it isn't always the same agent. All other agents using the methods to login/out are working fine. Just logging this agent out and back in again doesn't correct the problem. The RoamingAgent code just looks up the SIP extension for any given agent from the asterisk database and sends the call there. We're using Asterisk 1.6.2.0. Thanks Jordan Normally asterisk shows local channels as Invalid when there is not a valid extension and context where it points to. In your case, there may not be a 4223 extension on the RoamingAgent context. Try fixing and reloading your dialplan, and reloading the app_queue.so module for it to recheck the local channel member definitions. Hope it helps. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls
On 4/7/2010 2:45 AM, asterisk card support wrote: hi: how about the codecs? Best wishes! Asterisk Support group(sangoma, digium...), providing asterisk conf, pri, ss7, elastix, trixbox support. website:www.cnasterisk.com, www.voip88.com I have the phones and asterisk limited to ulaw only. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rebooting Polycom's - Could not create address for 'XXXX'
We are running a PBX box for our small company's phone system. There are two phones on our network that are not working though. When I reboot a phone that works correctly, like this (note 4907 is the phones extension): asterisk -rx sip notify polycom-check-cfg 4907 It returns this: Sending NOTIFY of type 'polycom-check-cfg' to '4907' and the phone proceeds to reboot normally. However if I try to do this with a phone that is not connecting, like extension 4912, I get this: Could not create address for '4912' And /var/log/asterisk/full just spits out: [Apr 7 15:20:41] VERBOSE[2899] logger.c: -- Remote UNIX connection [Apr 7 15:20:41] VERBOSE[12115] logger.c: -- Remote UNIX connection disconnected Not much help. I can't seem to see any other Logs that are logging anything for this phone. Does anyone have any clue where I can start to determine the problem? Where should I be looking? On this PBX box, all the Polycom provisioning configs and settings are in /home/PlcmSpIp/ I'm new to the PBX/Asterisk thing cause I'm taking over for the previous IT guy and trying to figure out the mess he made with this. Any help would be appreciated. Jake -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] testexpr2
On Tuesday 06 April 2010 15:31:20 Richard Kenner wrote: Why aren't you using check_expr in the utils directory? Aren't they two different things? I thought check_expr looks at a whole file for syntax errors while testexpr2 just parses one expression and returns its value. But if testexpr2 doesn't exist anymore, shouldn't the documentation be updated? check_expr2 in utils/ has long since replaced testexpr2. As far as the documentation, it could probably stand to be updated. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to log into separate file
Hi all, I want to have a separate file to log what i need for my dialplan without all output from Asterisk. By this way, i can easily to trace problems caused by my dialplan. How can i do that? Thanks in advance. Quyps -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rebooting Polycom's - Could not create address for 'XXXX'
As I fairly know, you cannot send a SIP message if the telephone is not registered to you (In case host=dynamic in configuration), since you don't know which is the telephone address or if it's behind NAT. If not registered you should try with an outside script like this: http://www.voip-info.org/wiki/view/Polycom+reboot+hardphone+script Regards Jose Flores Galicia floj...@gmail.com BriefCode Code Based Training 2010/4/7 Jake Wilson jake.wil...@answeron.com We are running a PBX box for our small company's phone system. There are two phones on our network that are not working though. When I reboot a phone that works correctly, like this (note 4907 is the phones extension): asterisk -rx sip notify polycom-check-cfg 4907 It returns this: Sending NOTIFY of type 'polycom-check-cfg' to '4907' and the phone proceeds to reboot normally. However if I try to do this with a phone that is not connecting, like extension 4912, I get this: Could not create address for '4912' And /var/log/asterisk/full just spits out: [Apr 7 15:20:41] VERBOSE[2899] logger.c: -- Remote UNIX connection [Apr 7 15:20:41] VERBOSE[12115] logger.c: -- Remote UNIX connection disconnected Not much help. I can't seem to see any other Logs that are logging anything for this phone. Does anyone have any clue where I can start to determine the problem? Where should I be looking? On this PBX box, all the Polycom provisioning configs and settings are in /home/PlcmSpIp/ I'm new to the PBX/Asterisk thing cause I'm taking over for the previous IT guy and trying to figure out the mess he made with this. Any help would be appreciated. Jake -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help with a pika warp asterisk appliance problem.
I've tried the forums at pikawarp.org and it seems no one is there anymore. Is there someone here that can help. last week I decided it was time to upgrade the rom on the machine to try to get the newest freepbx. so I followed the instructions to upgrade. I resetup the extensions and all the other setting like it was before. I did this over the network so I go onsite to double check that it is working correctly. It isn't all the fxs lines that go into the fxs card have bad static - some so bad you can't even tell it is a dial tone. the fxo card seems ok and the fxs line that is on the main board is ok. So I figured the rom didn't falsh correctly so I downloaded it again. This time to my local machine put it the files on the sd card - deleted the file that keeps it from loading over and over and let it flash again. -- with the same results. Something it not setting up properly for the fxs card in the system - or has damaged it. fxs/1 is on the main board - it works dial tone clear and sound during a call clear fxs/2 can't tell these is dial tone fxs/3 can hear dial tone thru bad static fxs/4 can hear dial tone thru bad static - not quite as bad as 3 fxs/5 dial tone sounds ok - but if you call you start hearing clicks and pop and bursts of static. Does anyone have any ideas? This upgrade says it has a special procedure and changes the layout of the files it uses - so I am not sure I can downgrade again. I've asked on the Pikawarp.org forum but so far no answer. if it goes a few more days I will have to try something - the people in that building have been using the 1 phone and thier cell phones for a few days now - but I really need to find a fix for this. Thanks for listening and if you have any ideas I would sure like to hear them. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to log into separate file
On Thu, 8 Apr 2010, Pham Quy wrote: I want to have a separate file to log what i need for my dialplan without all output from Asterisk. By this way, i can easily to trace problems caused by my dialplan. You can control how much and where Asterisk logs by fiddling with logger.conf. Personally, I like logging through syslog so I can configure all of my Asterisk hosts to send all of their logging to a central log host. I also use syslog() in my AGIs so everything is in the same place. For production, I use: syslog.local0 = error When I'm debugging something, I'll crank up the volume like: syslog.local0 = debug,dtmf,error,event,info,\ notice,verbose,warning If this doesn't give you the level of logging you need, you could always call an AGI or do something really ugly like: exten = *,n,system(logger -t foo mumble) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
