Re: [asterisk-users] app-conference
On 8/28/07, fateme fatah [EMAIL PROTECTED] wrote: Hi: I think app-conference is used where there isn't zaptel hardware,in the other word when we use zaptel hardware we shouldn't use app-conference for conference call sevice and we should use meetme application and load ztdummy.Is it true? Best regards. Hi yes app_conference need some timer source app_meetme can use ztummy but on highload expect to use hardware source ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip provider settings problem, please help
- Original Message - From: Jody Gugelhupf [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, August 27, 2007 3:55 PM Subject: [asterisk-users] voip provider settings problem, please help hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when picked up, but after a while it stops working, incoming calls for this provider are not shown anymore in the CLI, but from other providers it always works, but the phone is ringingn nevertheless when calling my skypho account...when i then turn off the pbx and restart after sumthing like 2 hours my skypho account is working fine again, the incmiong calls are shown in the asterisk CLI, but after, i don't know let's say an hour or so it again stops working, incoming calls for my skypho account can not be seen in the asterisk CLI, then if i turn off the pbx for an hour or so it works again, so i thought it must be a setting issue, maybe something with the register? althought it always shows it registered when i use 'sip show registry' someone has an idea what i have to set or do to have it working permanently? what could be the problem here? i got no clue whatsoever and i have been using asterisk only since half a year, please help me, i'm totaly desperate, thx in advance jody :) Smell's like a NAT issue. Are you behind NAT ? As some one else mentioned try to set qualify=yes as well as register often. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple servers using realtime
We have a similar set up. I would recommend also using SER and load balancing so you can load balance your calls out between your asterisk box's. - Original Message - From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, August 22, 2007 8:12 PM Subject: Re: [asterisk-users] Multiple servers using realtime Seems to me that as long as all the contacts / reachability info / URIs are distinct for each user, there is not a problem with using one big database, and that it certainly presents less of a maintenance headache. It also provides easier migration path to future options you may want to explore that *do* take advantage of its shared aspect. On Wed, 22 Aug 2007, Peder @ NetworkOblivion wrote: I am in the process of setting up several * servers using realtime and connecting to mysql. I am trying to figure out if I should just use one database and one set of tables for all of the user data. Or if I should have separate databases for each * box. The boxes are independent of each other in that customerA only connects to box A. They will never fail over to box B or anything like that. I want to use realtime for queues,voicemail, sippeers and extensions. The only issue that I have come up with so far is that a common voicemail table would cause each box to try and send out mwi indicators since it appears each * box pulls all of the voicemail boxes from the DB every 10 seconds. Any thoughts on whether I should go with one DB, or separate per box DB's? There is one mysql box, I am not referring to mysql on each box, I am referring to whether I should use separate DB's within the one mysql box for each * box. Thanks. Peder ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prepaid Billing: A2Billing, AstBill, ASTCC
Dear Tzafrir; Sorry I did not understand what do you mean by: Does it work with '-T' and 'use strict'? Do u mean the ASTPP or the prepaid billing? Where I have to run '-T' and 'use strict'? You do not think that I need to do download the prepaid billing software or it come with Asterisk? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460 Have you looked at ASTPP? Have not looked in a while but Darren had plans to integrate it into OSCommerce and some other neat features. I think he based it on the original ASTCC but has made some major improvements. Does it work with '-T' and 'use strict'? Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: DELL Platforms
snip I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM and a couple 250gig SATA drives. Totally VoIP so I cannot comment on cards or interrupts, but so far it has been flawless. I would like to see how many G729/ULAW conversions it could handle. How would I go about benchmarking that? Thanks, Steve /snip I would advise against the SC Series because there is no RAC card option. You never know that you needed one till you need to format a box from your hotel room ;) I have used some Poweredge 1850's with Asterisk (VOIP Only) and I have been real happy. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] deadagi and billsec or answeredtime
Hello,
I want to create php rate script and I'm using Deadagi. But I allways get
billsec 0 , or nothing. Can you help me to solve this problem...
My extension.conf:
exten = _123,1,DeadAgi(rate.php)
exten = _123,2,hangup
And my simple test php script rate.php
#!/usr/local/bin/php -q
?php
include_once (dirname(__FILE__)./phpagi.php);
$AGI = new AGI();
$AGI-answer();
$AGI-stream_file('demo-thanks');
$AGI-stream_file('vm-goodbye');
$AGI-hangup();
$billsec = get_var($AGI,CDR(billsec));
debug(Billsec: $billsec, 1);
function debug($string, $level=3) {
global $AGI;
$AGI-verbose($string, $level);
}
function get_var( $agi, $value) {
$r = $agi-get_variable( $value );
if ($r['result'] == 1) {
$result = $r['data'];
return $result;
}
else
return '';
}
?
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Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Marc Patino Gómez wrote: Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc Try a Sangoma card? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: DELL Platforms
Dovid B wrote: snip I am running an SC1435 with two dual core Opteron 2212, four gigs of RAM and a couple 250gig SATA drives. Totally VoIP so I cannot comment on cards or interrupts, but so far it has been flawless. I would like to see how many G729/ULAW conversions it could handle. How would I go about benchmarking that? Thanks, Steve /snip I would advise against the SC Series because there is no RAC card option. You never know that you needed one till you need to format a box from your hotel room ;) I have used some Poweredge 1850's with Asterisk (VOIP Only) and I have been real happy. Not so much a problem since the CoLo is staffed 24/7 and glad to help. Soon to add KVMoIP. There is a rudementary (have not played with it yet) /The latest industry-standard Intelligent Platform Management Interface (IPMI) 2.0 Baseboard Management Controller (BMC) allows remote, out-of-band management over a network or serial connection with any industry-standard IPMI management program. /I will play with this feature a bit before investing in KVMoIP. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
That is why I suggested Sangoma. Ask them if you can return it if it does not fix your problem. It is alot easier than disabling things in BIOS and hunting for the elusive noises. Digium would have you believe that the problem is the Dell box but if a Sangoma card works perfectly in the same box, then where is the actual problem? Anyways, if you are set on Digium, call their support and give them SSH. They may be your best bet in fixing the issue. Thanks, Steve Marc Patino Gómez wrote: Hi Steve, All my cards are Digium, I tried diferent Digium cards and I had the same problem. Regards, Marc Steve Totaro wrote: Marc Patino Gómez wrote: Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc Try a Sangoma card? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Authenticating SIP user in LDAP database instead of SIP.conf file
On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Gavin, Sorry for having miss pelt your name twice... Thank you once again for your prompt reply. Is this the correct version of the driver for Asterisk 1.2.x : res_config_ldap-v0.7.tar.gz from the link http://bugs.digium.com/view.php?id=5768 If you use an old version of res_config_ldap with Asterisk version 1.2, I'm pretty sure you'll be asked to upgrade to Asterisk 1.4 if you seek any help via the lists or bug tracker. If you can use the latest release of Asterisk, you should. Thank you for your time and patience, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: On 27/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin! ;-) As of today I am using the res_config_ldap of Astirectory in my test Asterisk system to connect to a test LDAP database of my University. Things seem to be working fine so far. Now I'm faced with the task of installing this in the productive system. Before doing so, I'd sure like to consider trying the RealTime database driver that you people have developed. Why so? because I trust your judgment. Thanks, but you should still test it yourself. I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. This would mean removing Astirectory module, installing the new driver and loading the new schema into LDAP. In my view, the latter part shouldn't be a concern because the old attributes and object classes (Astirectory) should in no way interfere with the new ones. Besides the old object classes could be deleted from LDAP. Also the former part shouldn't be of much concern either. Nope, you are correct. My only concern as of now is in the installation of the RealTime database driver because the 'readme' file does not say anything about the installation. It only says about the configuration after installation. From the link: http://svn.digium.com/svn/asterisk/team/group/res_config_ldap/ Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. This Digium version is for 1.4.x, not 1.2 Thanks in advance, Abhishek On 8/27/07, Gavin Henry [EMAIL PROTECTED] wrote: I see it is res_config_ldap. You'd be much better using the latest version in the bug tracker. On 27/08/07, Gavin Henry [EMAIL PROTECTED] wrote: On 26/08/07, Abhishek M S [EMAIL PROTECTED] wrote: Dear Mr Galvin, Gavin ;-) Thank you for the links. Had gone through the bug tracker before though. I was specifically referring to the schema for the driver 'Astirectory' and not the one related to the real time LDAP driver for Open LDAP. It's for any LDAP Compliant Directory Server. In the 'Astirectory' documentation there's a file defining the schema for LDAP which is incomplete. By incomplete I mean the Syntax and few other fields are not defined let alone the schema being a static file. I do understand that for Open LDAP a static file schema should be defined. Not really. in the RealTime driver you can specify which LDAP attributes map to which Asterisk Config settings. The only reason why I preferred Astirectory over the LDAP real time driver was the fact that there is no mapping required for SIP users and peers. OK, maybe I need to go and read more about Astirectory. Regards Abhishek On 8/24/07, Gavin Henry [EMAIL PROTECTED] wrote: Please see the official tracker in the Digium buglist: http://bugs.digium.com/view.php?id=5768 Here are the schemas we did for OpenLDAP: http://bugs.digium.com/file_download.php?file_id=14842type=bug http://bugs.digium.com/file_download.php?file_id=14841type=bug Also, for Novell eDirectory, see: http://forge.voicerd.org/frs/?group_id=7release_id=17 Gavin. -- http://www.suretecsystems.com/services/openldap/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Hi Steve, All my cards are Digium, I tried diferent Digium cards and I had the same problem. Regards, Marc Steve Totaro wrote: Marc Patino Gómez wrote: Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc Try a Sangoma card? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Hi Steve, Thanks for your advice, I will order a Sangoma card and test the box. A part from this, you know any other point to recomend Sangoma cards versus Digium cards? Many thanks, Marc Steve Totaro wrote: That is why I suggested Sangoma. Ask them if you can return it if it does not fix your problem. It is alot easier than disabling things in BIOS and hunting for the elusive noises. Digium would have you believe that the problem is the Dell box but if a Sangoma card works perfectly in the same box, then where is the actual problem? Anyways, if you are set on Digium, call their support and give them SSH. They may be your best bet in fixing the issue. Thanks, Steve Marc Patino Gómez wrote: Hi Steve, All my cards are Digium, I tried diferent Digium cards and I had the same problem. Regards, Marc Steve Totaro wrote: Marc Patino Gómez wrote: Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc Try a Sangoma card? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager Interface, response types
Hi everyone, I am writing a project that uses the Asterisk Manager Interface to monitor events. I just wanted to confirm if the types of messages sent back by the AMI are - Event - Response - Status If there are any other can anyone please point them to me or point me to some documentation where I could read about this Thanks. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk call waiting with SIP
Hi any body have idea about asterisk call waiting with SIP if we use asterisk1.4.5 hard phone with extensions. i need dial plan logic for this which capable to activate deactivate such feature. with ATA IP phone it is possible but with normal hardphone SIP in asterisk is it possible to indicate call waiting? plz help me for this Amit ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
I will let you judge that for yourself. I suggest that you email someone at Sangoma sales *directly* (not a reseller) to ask about buying the card and if it can be returned if it does not work properly. Explain your issue with the Digium card briefly. See how it turns out for you. Thanks, Steve Totaro Marc Patino Gómez wrote: Hi Steve, Thanks for your advice, I will order a Sangoma card and test the box. A part from this, you know any other point to recomend Sangoma cards versus Digium cards? Many thanks, Marc Steve Totaro wrote: That is why I suggested Sangoma. Ask them if you can return it if it does not fix your problem. It is alot easier than disabling things in BIOS and hunting for the elusive noises. Digium would have you believe that the problem is the Dell box but if a Sangoma card works perfectly in the same box, then where is the actual problem? Anyways, if you are set on Digium, call their support and give them SSH. They may be your best bet in fixing the issue. Thanks, Steve Marc Patino Gómez wrote: Hi Steve, All my cards are Digium, I tried diferent Digium cards and I had the same problem. Regards, Marc Steve Totaro wrote: Marc Patino Gómez wrote: Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc Try a Sangoma card? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] deadagi and billsec or answeredtime
On 8/28/07, Giedrius Augys [EMAIL PROTECTED] wrote:
Hello,
I want to create php rate script and I'm using Deadagi. But I allways get
billsec 0 , or nothing. Can you help me to solve this problem...
It seems that there is problem with Answer(). Does it get executed
from AGI? Do you hear voice prompts?
You can try adding Answer() before DeadAgi().
Regards,
Atis
My extension.conf:
exten = _123,1,DeadAgi(rate.php)
exten = _123,2,hangup
And my simple test php script rate.php
#!/usr/local/bin/php -q
?php
include_once (dirname(__FILE__)./phpagi.php);
$AGI = new AGI();
$AGI-answer();
$AGI-stream_file('demo-thanks');
$AGI-stream_file('vm-goodbye');
$AGI-hangup();
$billsec = get_var($AGI,CDR(billsec));
debug(Billsec: $billsec, 1);
function debug($string, $level=3) {
global $AGI;
$AGI-verbose($string, $level);
}
function get_var( $agi, $value) {
$r = $agi-get_variable( $value );
if ($r['result'] == 1) {
$result = $r['data'];
return $result;
}
else
return '';
}
?
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Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org
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Re: [asterisk-users] IAX2 trunking scalability
Hi, I thought I'd give a follow up to this discussion for the archives... Currently I'm trunking 30 channels of g.729 traffic (no transcoding going on, the call comes in and goes out as g.729) and it takes about 350 kbps bandwith bidirectional. So on average each call takes 11.5 - 12 kbps of bandwith. The solution seems stable and the QoS is identical... so for the price (2 commodity PCs...), IAX2 trunking is well worth the effort since it reduces bandwith usage by a factor of 2. Cheers, Jean-Michel. -- Jean-Michel Hiver - YKOZ +262 (0)692 828 070 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] deadagi and billsec or answeredtime
I solve this problem. I'm not sure, but you get billsec when you use Dial
application. Using dial app , I get billsec.
2007/8/28, Atis [EMAIL PROTECTED]:
On 8/28/07, Giedrius Augys [EMAIL PROTECTED] wrote:
Hello,
I want to create php rate script and I'm using Deadagi. But I allways
get
billsec 0 , or nothing. Can you help me to solve this problem...
It seems that there is problem with Answer(). Does it get executed
from AGI? Do you hear voice prompts?
You can try adding Answer() before DeadAgi().
Regards,
Atis
My extension.conf:
exten = _123,1,DeadAgi(rate.php)
exten = _123,2,hangup
And my simple test php script rate.php
#!/usr/local/bin/php -q
?php
include_once (dirname(__FILE__)./phpagi.php);
$AGI = new AGI();
$AGI-answer();
$AGI-stream_file('demo-thanks');
$AGI-stream_file('vm-goodbye');
$AGI-hangup();
$billsec = get_var($AGI,CDR(billsec));
debug(Billsec: $billsec, 1);
function debug($string, $level=3) {
global $AGI;
$AGI-verbose($string, $level);
}
function get_var( $agi, $value) {
$r = $agi-get_variable( $value );
if ($r['result'] == 1) {
$result = $r['data'];
return $result;
}
else
return '';
}
?
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--
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org
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Pagarbiai / Best Regards,
Giedrius Augys
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Re: [asterisk-users] deadagi and billsec or answeredtime
On 8/28/07, Giedrius Augys [EMAIL PROTECTED] wrote:
I solve this problem. I'm not sure, but you get billsec when you use Dial
application. Using dial app , I get billsec.
Well, Dial usually changes ANSWER status, because you usually answer
phone that is ringing. If you won't answer, you won't get billsec.
Regards,
Atis
2007/8/28, Atis [EMAIL PROTECTED] :
On 8/28/07, Giedrius Augys [EMAIL PROTECTED] wrote:
Hello,
I want to create php rate script and I'm using Deadagi. But I allways
get
billsec 0 , or nothing. Can you help me to solve this problem...
It seems that there is problem with Answer(). Does it get executed
from AGI? Do you hear voice prompts?
You can try adding Answer() before DeadAgi().
Regards,
Atis
My extension.conf:
exten = _123,1,DeadAgi(rate.php)
exten = _123,2,hangup
And my simple test php script rate.php
#!/usr/local/bin/php -q
?php
include_once (dirname(__FILE__)./phpagi.php);
$AGI = new AGI();
$AGI-answer();
$AGI-stream_file('demo-thanks');
$AGI-stream_file('vm-goodbye');
$AGI-hangup();
$billsec = get_var($AGI,CDR(billsec));
debug(Billsec: $billsec, 1);
function debug($string, $level=3) {
global $AGI;
$AGI-verbose($string, $level);
}
function get_var( $agi, $value) {
$r = $agi-get_variable( $value );
if ($r['result'] == 1) {
$result = $r['data'];
return $result;
}
else
return '';
}
?
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Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org
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Pagarbiai / Best Regards,
Giedrius Augys
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Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? - www.BEST.eu.org
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[asterisk-users] Linksys (PAP2) delay time between hung up and line release
Hi, I have a PAP2 with 2 phone ports. When I call them everything works fine until I hung up the call. There is about 30-40 seconds until I can call to that extension again. Before that it gives me busy messages. Extension config: exten = 199,1,Dial(SIP/199,30) exten = 199,102,Hangup Any suggestions? Thanks -- Mejor, Mozilla Firefox http://www.mozilla-europe.org/es/products/firefox/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Motherboard with SATA RAID1 support That's a mulit-port SATA controller with RAID in the driver (software). 256 MB RAM Use a little more RAM. digium PRI/E1 card Is there any reason you aren't using Sangoma cards? 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) None. 2. If I use Hardware based RAID 1, what would be the impact to the system? A PCI slot. 3. According to your practical experiance what is the ideal solution among both options? Software RAID works fine. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Saftware RAID1 or Hardware RAID1 with Asterisk (Andrew Joakimsen)
Dear Andrew, Thanks for your kind responce. Regards, vidura. = Motherboard with SATA RAID1 support That's a mulit-port SATA controller with RAID in the driver (software). 256 MB RAM Use a little more RAM. digium PRI/E1 card Is there any reason you aren't using Sangoma cards? 1. If I use Software RAID, what would be the impact to my deployment? ( problems that I have to face with regard to the call flow ) None. 2. If I use Hardware based RAID 1, what would be the impact to the system? A PCI slot. 3. According to your practical experiance what is the ideal solution among both options? Software RAID works fine. -- Thanks Regards, Vidura Senadeera, Network Engineer, Debug Solutions Sri Lanka. Tel - +94114520036 Mobile - +9466596 Web - www.debug.lk ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip provider settings problem, please help
hi Anselm :)
thx for your tip, though i have qualified turned on, anyhow here are my
complete sip.conf and
extensions.conf, thx for any help :)
sip.conf
[general]
allowoverlap = yes
realm = mydomain.tld
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
tos = lowdelay
disallow = all
allow = alaw,ulaw,gsm,ilbc,g729
trustrpid = no
dtmfmode = auto
externip = XXX.XXX.XXX.XXX
localnet = 10.0.0.0/255.255.0.0
nat = yes
canreinvite = yes
rtcachefriends = yes
fromdomain = sshn.net
qualify = yes
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
register = user:[EMAIL PROTECTED]
extension.conf:
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no
[globals]
TRUNKOPTIONS =
EMERGENCY = 0
EMERGENCY_TRUNK =
TRANSFERS_CTX = DefaultOutgoingRule
CALLBACK_CTX = DefaultOutgoingRule
DISA_CTX = DefaultOutgoingRule
DISA_PASSWD =
DYNAMIC_FEATURES = automon
TRUNK =
SIP/3124XSIP/9083XXXSIP/069929SIP/webcalldirectDESIP/webcalldirectNLSIP/iXcallSIP/messagenet
OUTGOING_PREFIX =
[_all_]
include = _all-extensions_
include = _all-resources_
include = _all-applications_
include = _catch-all_
[_catch-all_]
exten = _X.,1,AGI(dial.php|entity=group=5extension=${EXTEN})
exten = _X,1,AGI(dial.php|entity=group=5extension=${EXTEN})
[app-AgentCallbackLogin_92]
exten = *100,1,AGI(dial.php|entity=1group=6extension=*100)
[_all-applications_]
include = app-AgentCallbackLogin_92
include = app-AgentCallbackLogout_94
include = app-audiorecorder
include = app-callermailbox
include = app-cancel-CFB-calling-extension
include = app-cancel-CFNR-calling-extension
include = app-cancel-CFU-calling-extension
include = app-CFB-calling-extension
include = app-CFNR-calling-extension
include = app-CFU-calling-extension
include = app-dnd-off
include = app-dnd-on
include = app-mailbox
[app-AgentCallbackLogout_94]
exten = *101,1,AGI(dial.php|entity=2group=6extension=*101)
[app-audiorecorder]
exten = *99,1,AGI(dial.php|entity=3group=6extension=*99)
[app-callermailbox]
exten = *98,1,AGI(dial.php|entity=4group=6extension=*98)
[app-cancel-CFB-calling-extension]
exten = *91,1,AGI(dial.php|entity=5group=6extension=*91)
[app-cancel-CFNR-calling-extension]
exten = *93,1,AGI(dial.php|entity=6group=6extension=*93)
[app-cancel-CFU-calling-extension]
exten = *73,1,AGI(dial.php|entity=7group=6extension=*73)
[app-CFB-calling-extension]
exten = *90,1,AGI(dial.php|entity=8group=6extension=*90)
[app-CFNR-calling-extension]
exten = *92,1,AGI(dial.php|entity=9group=6extension=*92)
[app-CFU-calling-extension]
exten = *72,1,AGI(dial.php|entity=10group=6extension=*72)
[app-dnd-off]
exten = *79,1,AGI(dial.php|entity=11group=6extension=*79)
[app-dnd-on]
exten = *78,1,AGI(dial.php|entity=12group=6extension=*78)
[app-mailbox]
exten = _*98.,1,AGI(dial.php|entity=13group=6extension=_*98.)
[macro-agentcallbacklogin]
exten = s,1,AgentCallbackLogin(${CALLERID(num)}||${CALLERID(num)[EMAIL
PROTECTED])
[macro-agentcallbacklogout]
exten = s,1,Answer
exten = s,n,System(asterisk -rx agent logoff Agent/${CALLERID(num)})
exten = s,n,Playback(agent-loggedoff)
exten = s,n,Playback(vm-goodbye)
[macro-audiorecorder]
exten = s,1,AGI(record.php)
[macro-reroute]
exten = s,1,Goto(${ARG2},${ARG1},1)
[macro-callback]
exten = s,1,Wait(2)
exten = s,n,AGI(agi-callback.agi,${CALLERID(num)},${ARG1},${ARG2},${ARG3})
[macro-cancel-CFB-calling-extension]
exten = s,1,DBdel(CFB/${CALLERID(num)})
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,Playback(call-fwd-on-busy)
exten = s,n,Playback(de-activated)
[macro-cancel-CFNR-calling-extension]
exten = s,1,DBdel(CFNR/${CALLERID(num)})
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,Playback(call-fwd-no-ans)
exten = s,n,Playback(de-activated)
[macro-cancel-CFU-calling-extension]
exten = s,1,DBdel(CFU/${CALLERID(num)})
exten = s,n,Answer
exten = s,n,Wait(1)
exten = s,n,Playback(call-fwd-cancelled)
[macro-CFB-calling-extension]
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,BackGround(ent-target-attendant)
exten = s,n,Read(toext,then-press-pound)
exten = s,n,Wait(1)
exten = s,n,Set(DB(CFB/${CALLERID(num)})=${toext})
exten = s,n,Playback(call-fwd-on-busy)
exten = s,n,Playback(for)
exten = s,n,Playback(extension)
exten = s,n,SayDigits(${CALLERID(num)})
exten = s,n,Playback(is-set-to)
exten = s,n,SayDigits(${toext})
[macro-CFNR-calling-extension]
exten = s,1,Answer
exten = s,n,Wait(1)
exten = s,n,BackGround(ent-target-attendant)
exten = s,n,Read(toext,then-press-pound)
exten = s,n,Wait(1)
exten = s,n,Set(DB(CFNR/${CALLERID(num)})=${toext})
exten = s,n,Playback(call-fwd-no-ans)
exten = s,n,Playback(for)
exten = s,n,Playback(extension)
exten = s,n,SayDigits(${CALLERID(num)})
exten = s,n,Playback(is-set-to)
exten = s,n,SayDigits(${toext})
[macro-CFU-calling-extension]
exten = s,1,Answer
exten = s,n,Wait(1)
[asterisk-users] calls being forwarded to neighbor?? please help, thx :)
hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box has ip 10.0.0.13, asterisk machine (machine 2) has ip 10.0.0.20 and machine one is on 10.0.0.1, which is the connected to internet, anyhow everything is more or less working fine, though sometimes i see strange things in the asterisk CLI, e.g. in the part below there is the line: -- Now forwarding SIP/9083XXX-0816b208 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/486-081d4738) i don't get the 'Local/247110358' part, why the hell is that number there? that is the phone number of my neighbor (though without international code), why is it showing there and trying to forwrd it there? below is the whole CLI output i had when i saw this and furhter down the buttom are my sip.conf and extension.conf, i was also trying to set up my voicemail, but somehow that also doesn't work... anyhow nay help is appreciated, thx a bunch katie-jody :D -- Executing NoOp(SIP/9083XXX-0816b208, Incoming-s. CallerID:0031648978254 0031648978254 - Calling:s) in new stack -- Executing AGI(SIP/9083XXX-0816b208, incoming.php|answered=schannel=48rule=4uniqueid=1188247273.6) in new stack -- Launched AGI Script /usr/share/asterisk/agi-bin/incoming.php -- AGI Script Executing Application: (macro) Options: (incoming-call-to-extension|SIP/486|30||voiceone/custom/hello) -- Executing NoOp(SIP/9083XXX-0816b208, 0031648978254 0031648978254) in new stack -- Executing Ringing(SIP/9083XXX-0816b208, ) in new stack -- Executing Dial(SIP/9083XXX-0816b208, SIP/486|30|tw) in new stack -- SIP Seeding peer from astdb: '486' at [EMAIL PROTECTED]:5060 for 3600 -- Called 486 -- SIP/486-081d4738 is ringing == Spawn extension (macro-incoming-call-to-extension, s, 3) exited non-zero on 'SIP/9083XXX-0816b208' in macro 'incoming-call-to-extension' == Spawn extension (macro-incoming-call-to-extension, s, 3) exited non-zero on 'SIP/9083XXX-0816b208' -- Executing NoOp(SIP/31247110460-081e3ed8, Incoming-s. CallerID:Judit 31247110570 - Calling:s) in new stack -- Executing AGI(SIP/31247110460-081e3ed8, incoming.php|answered=schannel=46rule=2uniqueid=1188247365.8) in new stack -- Launched AGI Script /usr/share/asterisk/agi-bin/incoming.php -- AGI Script Executing Application: (macro) Options: (incoming-call-to-extension|SIP/486|||) -- Executing NoOp(SIP/31247110460-081e3ed8, Judit 31247110570) in new stack -- Executing Ringing(SIP/31247110460-081e3ed8, ) in new stack -- Executing Dial(SIP/31247110460-081e3ed8, SIP/486||tw) in new stack -- Called 486 -- SIP/486-081e9418 is ringing == Spawn extension (macro-incoming-call-to-extension, s, 3) exited non-zero on 'SIP/31247110460-081e3ed8' in macro 'incoming-call-to-extension' == Spawn extension (macro-incoming-call-to-extension, s, 3) exited non-zero on 'SIP/31247110460-081e3ed8' -- Executing NoOp(SIP/9083XXX-0816b208, Incoming-s. CallerID:0031247110570 0031247110570 - Calling:s) in new stack -- Executing AGI(SIP/9083XXX-0816b208, incoming.php|answered=schannel=48rule=4uniqueid=1188247411.10) in new stack -- Launched AGI Script /usr/share/asterisk/agi-bin/incoming.php -- AGI Script Executing Application: (macro) Options: (incoming-call-to-extension|SIP/486|30||voiceone/custom/hello) -- Executing NoOp(SIP/9083XXX-0816b208, 0031247110570 0031247110570) in new stack -- Executing Ringing(SIP/9083XXX-0816b208, ) in new stack -- Executing Dial(SIP/9083XXX-0816b208, SIP/486|30|tw) in new stack -- Called 486 -- SIP/486-081d4738 is ringing -- Got SIP response 302 Moved Temporarily back from 10.0.0.13 -- Now forwarding SIP/9083XXX-0816b208 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/486-081d4738) -- Executing AGI(Local/[EMAIL PROTECTED],2, dial.php|entity=group=5extension=247110358) in new stack -- Launched AGI Script /usr/share/asterisk/agi-bin/dial.php -- AGI Script Executing Application: (ChanIsAvail) Options: (SIP/31247110460SIP/9083XXXSIP/0699291034SIP/webcalldirectDESIP/webcalldirectNLSIP/iXcallSIP/messagenet) -- AGI Script Executing Application: (macro) Options: (dialout|247110358) -- Executing Set(Local/[EMAIL PROTECTED],2, TOUCH_MONITOR=20070827-224353_0031247110570-247110358) in new stack -- Executing NoOp(Local/[EMAIL PROTECTED],2, CID_NAME : Katie) in new stack -- Executing NoOp(Local/[EMAIL PROTECTED],2, CID_NUMBER: 0031247110460) in new stack -- Executing NoOp(Local/[EMAIL PROTECTED],2, CID_CLIR : 0) in new stack -- Executing NoOp(Local/[EMAIL PROTECTED],2, TRUNK : SIP/31247110460) in new stack -- Executing Set(Local/[EMAIL PROTECTED],2,
Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Marc Patino Gómez wrote: Hi Steve, Thanks for your advice, I will order a Sangoma card and test the box. A part from this, you know any other point to recomend Sangoma cards versus Digium cards? Many thanks, Marc 5 year warranty, to name one. Sangoma says their cards will work in ALL modern machines. If they can't make it work ( never seen that ) they will refund. If you have problems, and you give them SSH, they will fix it. John Novack Steve Totaro wrote: That is why I suggested Sangoma. Ask them if you can return it if it does not fix your problem. It is alot easier than disabling things in BIOS and hunting for the elusive noises. Digium would have you believe that the problem is the Dell box but if a Sangoma card works perfectly in the same box, then where is the actual problem? Anyways, if you are set on Digium, call their support and give them SSH. They may be your best bet in fixing the issue. Thanks, Steve Marc Patino Gómez wrote: Hi Steve, All my cards are Digium, I tried diferent Digium cards and I had the same problem. Regards, Marc Steve Totaro wrote: Marc Patino Gómez wrote: Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc Try a Sangoma card? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my co-pilot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Hi John, thanks for this usefull info Marc John Novack wrote: Marc Patino Gómez wrote: Hi Steve, Thanks for your advice, I will order a Sangoma card and test the box. A part from this, you know any other point to recomend Sangoma cards versus Digium cards? Many thanks, Marc 5 year warranty, to name one. Sangoma says their cards will work in ALL modern machines. If they can't make it work ( never seen that ) they will refund. If you have problems, and you give them SSH, they will fix it. John Novack Steve Totaro wrote: That is why I suggested Sangoma. Ask them if you can return it if it does not fix your problem. It is alot easier than disabling things in BIOS and hunting for the elusive noises. Digium would have you believe that the problem is the Dell box but if a Sangoma card works perfectly in the same box, then where is the actual problem? Anyways, if you are set on Digium, call their support and give them SSH. They may be your best bet in fixing the issue. Thanks, Steve Marc Patino Gómez wrote: Hi Steve, All my cards are Digium, I tried diferent Digium cards and I had the same problem. Regards, Marc Steve Totaro wrote: Marc Patino Gómez wrote: Hi list, I have a terrible noise issue with Dell SC1430 + Digium TE110P. The digium card is not sharing interrupts with any other device, as I saw in Dell's BIOS and also with lspci -vb command. After changing coax wire, UTP, balum, digium card ... I have found that the problem is in Dell box, so now I'm running the same Asterisk config in other server with the same Digium card and there is no noise in PRI. Any advice to solve the problem with Dell box? Regards, Marc Try a Sangoma card? Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] calls being forwarded to neighbor?? please help, thx :)
On Aug 28, 2007, at 8:24 AM, Jody Gugelhupf wrote: -- Now forwarding SIP/9083XXX-0816b208 to 'Local/ [EMAIL PROTECTED]' (thanks to SIP/486-081d4738) Because SIP/486 issued a 302 redirect to 247110358. Check the phone for the forwarding setting. /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
On 8/28/2007 at 9:30 AM, John Novack [EMAIL PROTECTED] wrote: Marc Patino Gómez wrote: Hi Steve, Thanks for your advice, I will order a Sangoma card and test the box. A part from this, you know any other point to recomend Sangoma cards versus Digium cards? Many thanks, Marc 5 year warranty, to name one. Sangoma says their cards will work in ALL modern machines. If they can't make it work ( never seen that ) they will refund. If you have problems, and you give them SSH, they will fix it. John Novack Digium has done this, for me, as well. However, in either case, I have reservations about letting others wack away at my machines, especially if one cannot see what they are doing. No so much not trusting them, but not learning a thing along the way. When I voiced that concern to the Digium techs, they set up a thing called screen (I think it was) to allow me to see and or interact with their session. joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astricon Meetup
Everyone, I will be attending Astricon in Phoenix and would like to have a little get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Exchange ideas and go over some use cases and see how we can all work together to improve our understanding of the dynamics of how everything works together. * Scaleability * Reusability of code * Standards (VoiceXML, MRCP and more) If anyone is interested please email me off list and we'll plan on having a meeting of minds. Thanks, Brian West FreeSWITCH.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't create audio conversationbetweensoftphonesthrough Asterisk
Hello, I do not think that the presence bit will be crucial to our application. Thanks for your help. I will keep you posted if I get any progress. Thanks, Denis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gerald A Sent: Monday, August 27, 2007 5:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't create audio conversationbetweensoftphonesthrough Asterisk Hi, On 8/27/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Thanks very much for the help, I appreciate it. Recently, one of my co-workers and I have altered the code to just register with the Asterisk server and place an audio call. This gets rid of the subscription part of the application, so I do not get the 489 Bad Event error anymore. I believe the 488 Not Acceptable Here error occurs when the invite is being sent. After the sdp body and header information are created, they are sent as an invite for the audio call. The problem seems to be some part of the invite that we are sending. I have a hunch that it may have to do with the codecs that the Jain-phone chooses. I will continue looking into this. Glad to hear you were able to get some traction with the voice calling. Is the presence bit something that is critical to your custom app? I'm going to be fiddling with some soft phone stuff soon, so I am still planning on taking a peek at Jain just for the heck of it. Keep me updated on your progress, and if you need any assistance, give me a shout. Thanks, Gerald. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 Confusion
I've purchsed 5 g729 licenses from digium, but am a little confused about why things are acting the way they are. When I do show g729 I see: 0/0 encoders/decoders of 5 licensed channels are currently in use My sip.conf starts out: [general] disallow=all allow=ulaw allow=g726 then my hosts look like this: [6016716] username=6016716 accountcode=75415 type=friend secret=obsurified qualify=yes port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Test VoIP Accounts 6016716 All works fine, however if I make a change so it reads: [6016716] disallow=all allow=g729 username=6016716 accountcode=75415 type=friend secret=obsurified qualify=yes port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Test VoIP Accounts 6016716 and reload, strange things begin to happen. A show g729 shows this: 5/0 encoders/decoders of 5 licensed channels are currently in use and suddenly I can not hear anything if I try to make a call.From observation, it almost seems like other units on the network are using the g729 codecs, but doesn't my sip.conf prohibit g729 unless expressly allowed?! Why would allowing g729 under one extension allow everyone else to suddenly start using g729? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HDL F10 brazilian doorbell device + TDM2400
Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not living there so I do not know if its a matter of signalling type or what. Is there anbody who tried this stuff or similar? Thanks Giorgio ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
Joe Acquisto wrote: On 8/28/2007 at 9:30 AM, John Novack [EMAIL PROTECTED] wrote: Marc Patino Gómez wrote: Hi Steve, Thanks for your advice, I will order a Sangoma card and test the box. A part from this, you know any other point to recomend Sangoma cards versus Digium cards? Many thanks, Marc 5 year warranty, to name one. Sangoma says their cards will work in ALL modern machines. If they can't make it work ( never seen that ) they will refund. If you have problems, and you give them SSH, they will fix it. John Novack Digium has done this, for me, as well. However, in either case, I have reservations about letting others wack away at my machines, especially if one cannot see what they are doing. No so much not trusting them, but not learning a thing along the way. When I voiced that concern to the Digium techs, they set up a thing called screen (I think it was) to allow me to see and or interact with their session. joe a. All politics aside, Your best bet if getting the audio issues fixed is your top priority is to have two plans. One should always have at least an A and a B plan in anything. A. Call Sangoma and order a card B. Call Digium tech support If Digium gets it working acceptably then cancel with Sangoma. If they do not, then you already have a fix on the way. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Confusion
and reload, strange things begin to happen. A show g729 shows this: 5/0 encoders/decoders of 5 licensed channels are currently in use I think you have a loop of some kind. As you can see none of those call are actually established since no decoders are in use. Try to debug and see why those 5 calls are acually not connected in the first place. and suddenly I can not hear anything if I try to make a call.From observation, it almost seems like other units on the network are using the g729 codecs, but doesn't my sip.conf prohibit g729 unless expressly allowed?! Why would allowing g729 under one extension allow everyone else to suddenly start using g729? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Andres http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Confusion
Matt wrote: I've purchsed 5 g729 licenses from digium, but am a little confused about why things are acting the way they are. When I do show g729 I see: 0/0 encoders/decoders of 5 licensed channels are currently in use My sip.conf starts out: [general] disallow=all allow=ulaw allow=g726 then my hosts look like this: [6016716] username=6016716 accountcode=75415 type=friend secret=obsurified qualify=yes port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Test VoIP Accounts 6016716 All works fine, however if I make a change so it reads: [6016716] disallow=all allow=g729 username=6016716 accountcode=75415 type=friend secret=obsurified qualify=yes port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Test VoIP Accounts 6016716 and reload, strange things begin to happen. A show g729 shows this: 5/0 encoders/decoders of 5 licensed channels are currently in use and suddenly I can not hear anything if I try to make a call.From observation, it almost seems like other units on the network are using the g729 codecs, but doesn't my sip.conf prohibit g729 unless expressly allowed?! Why would allowing g729 under one extension allow everyone else to suddenly start using g729? What version of Asterisk is this? Show channels and sip debug might help you track down the offenders (you might want to log to a text file because the SIP stuff flies by too fast. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon Meetup
oohs no! Whats up, haven't heard much out of you lately. Chris Brian West wrote: Everyone, I will be attending Astricon in Phoenix and would like to have a little get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Exchange ideas and go over some use cases and see how we can all work together to improve our understanding of the dynamics of how everything works together. * Scaleability * Reusability of code * Standards (VoiceXML, MRCP and more) If anyone is interested please email me off list and we'll plan on having a meeting of minds. Thanks, Brian West FreeSWITCH.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] server recommentation (unique requirements)
Howdy. I've been having trouble finding a fairly modern server that meets the following requirements: - Molex power connectors (don't want to use the Digium FXS power supply) - 4 PCI slots, one full-length (TDM2400P, 2xT100P, additional NIC) - dual power supplies - preferably dual CPUs = 1GHz - preferably rack-mountable (3-4RU) - CentOS-friendly We'd also like to stay away from older HP servers. Any recommendations would be greatly appreciated. Thanks. Stephen Kratzer Network Engineer CTI Networks, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distributed System
Hi all, I'm kind a New to Asterisk.But I'm a Network Administrator with 5 years of experiance. I want to know for an installation with 90 clients, If I don't want to have just 1 server for it, then how is it possible to distribute it among about 3 servers. Should I do it in a cluster (kernel level) or something with SER? Best Regards, Seysan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 and TDM800 fxo stop answering
Stefano Arata wrote: Hi, I have two asterisk with the Digium TDM400 installed on the first and the TDM800 installed on the second. Both systems are linux Debian 4.0 whith kernel 2.6.18 and asterisk 1.2.24. Often the cards stop answering calls, and I can't make or receive calls; I need to reboot the system or manually reload the zaptel modules to restore it. I've tried zaptel versions 1.2.18, 1.2.19 and 1.2.20 too but the problem remains. I can't find any error in the asterisk log files nor in the syslog but I've found this suggestion http://www.voip-info.org/wiki/view/Asterisk+automatic+daily+restart on wiki, that suggests to set up a cron job to restart the driver daily, but this doesn't work for me. Are there other solutions to this problem? First off, could you try zaptel-1.2.20.1? I made a fix that could possibly be related to this and it was either in 1.2.20 or 1.2.20.1. If that doesn't fix, could you please contact Digium tech support so we can make sure your problem is fixed. Thanks :-) -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Confusion
Well does g729 have to run on both legs of a call? For instance, when I have 5/0 and I make a call from a SIP device... I get 5/1.. I can't hear any audio on my SIP phone, however if I call someone they can hear me. On 8/28/07, Andres [EMAIL PROTECTED] wrote: and reload, strange things begin to happen. A show g729 shows this: 5/0 encoders/decoders of 5 licensed channels are currently in use I think you have a loop of some kind. As you can see none of those call are actually established since no decoders are in use. Try to debug and see why those 5 calls are acually not connected in the first place. and suddenly I can not hear anything if I try to make a call.From observation, it almost seems like other units on the network are using the g729 codecs, but doesn't my sip.conf prohibit g729 unless expressly allowed?! Why would allowing g729 under one extension allow everyone else to suddenly start using g729? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Andres http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.4.4 compiling problems
Hi! I have this error compiling Zaptel 1.4.4 make: *** No rule to make target `xpp/xpp_usb.ko', needed by `install-modules'. Stop. The Zaptel 1.2.5 compile ok. Any ideas?? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed System
On Aug 28, 2007, at 10:14 AM, Seysan wrote: Hi all, I'm kind a New to Asterisk.But I'm a Network Administrator with 5 years of experiance. I want to know for an installation with 90 clients, If I don't want to have just 1 server for it, then how is it possible to distribute it among about 3 servers. Should I do it in a cluster (kernel level) or something with SER? I would recommend SER plus Asterisk. I have had great success with using Asterisk with OpenSER. Best Regards, Seysan /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Confusion
Show channels doesn't show me the codec a call is using, unless I'm missing it somewhere... is there someway I can find out which channels are using what codec? On 8/28/07, Steve Totaro [EMAIL PROTECTED] wrote: Matt wrote: I've purchsed 5 g729 licenses from digium, but am a little confused about why things are acting the way they are. When I do show g729 I see: 0/0 encoders/decoders of 5 licensed channels are currently in use My sip.conf starts out: [general] disallow=all allow=ulaw allow=g726 then my hosts look like this: [6016716] username=6016716 accountcode=75415 type=friend secret=obsurified qualify=yes port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Test VoIP Accounts 6016716 All works fine, however if I make a change so it reads: [6016716] disallow=all allow=g729 username=6016716 accountcode=75415 type=friend secret=obsurified qualify=yes port=5060 nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=Test VoIP Accounts 6016716 and reload, strange things begin to happen. A show g729 shows this: 5/0 encoders/decoders of 5 licensed channels are currently in use and suddenly I can not hear anything if I try to make a call.From observation, it almost seems like other units on the network are using the g729 codecs, but doesn't my sip.conf prohibit g729 unless expressly allowed?! Why would allowing g729 under one extension allow everyone else to suddenly start using g729? What version of Asterisk is this? Show channels and sip debug might help you track down the offenders (you might want to log to a text file because the SIP stuff flies by too fast. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDL F10 brazilian doorbell device + TDM2400
gincantalupo wrote: Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not living there so I do not know if its a matter of signalling type or what. Is there anbody who tried this stuff or similar? It sounds like there might be an issue here related to not having disconnect supervision enabled. Can this device provide come sort of disconnect supervision? -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] server recommentation (unique requirements)
Stephen Kratzer wrote: Howdy. I've been having trouble finding a fairly modern server that meets the following requirements: - Molex power connectors (don't want to use the Digium FXS power supply) - 4 PCI slots, one full-length (TDM2400P, 2xT100P, additional NIC) - dual power supplies - preferably dual CPUs = 1GHz - preferably rack-mountable (3-4RU) - CentOS-friendly We'd also like to stay away from older HP servers. Any recommendations would be greatly appreciated. Thanks. Stephen Kratzer Network Engineer CTI Networks, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dell PowerEdge 4400 should fit what you are asking for. The 2950 might also. Jonn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon Meetup
haha you going to be there? /b On Aug 28, 2007, at 9:30 AM, Chris Childress wrote: oohs no! Whats up, haven't heard much out of you lately. Chris Brian West wrote: Everyone, I will be attending Astricon in Phoenix and would like to have a little get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Exchange ideas and go over some use cases and see how we can all work together to improve our understanding of the dynamics of how everything works together. * Scaleability * Reusability of code * Standards (VoiceXML, MRCP and more) If anyone is interested please email me off list and we'll plan on having a meeting of minds. Thanks, Brian West FreeSWITCH.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E911 mf camma Trunks
I just set up a t1 with 2 camma mf 911 trunks on it, and I am having a issue
with it. We can call 911 which is routed out these new trunks, and it goes
to the 911 center, but they are not getting the ANI and hence no record
found. Our LEC is Embarq, and they say they can see the call come in and
send:
KP-911-ST and then KP-0-911-ST rather then KP-0-ANI-ST
I turned on all the debug and logging that I could and still can't tell
exactly what it is sending to them.
My question is has any one had this issue, is anyone else even using the
e911, and if so how do you have it set up? I'll post my relevant configs
below:
ZAPTEL.conf
===
#Sangoma A102 port 2 [slot:11 bus:1 span: 2]
span=2,0,0,esf,b8zs
em=25-48
===
ZAPATA.conf
===
;911 group
group = 2
restrictcid=yes
signalling = e911
channel = 25-26
===
EXTENSIONS.conf
===
[globals]
DIGG1=Zap/g1
DIGG2=Zap/g2
; 911
exten = 911,1,NoOp(911 ANI CALLER ID INFO IS: ${CALLERID(ani)} REGULAR
CALLER ID: ${CALLERID(all)})
;exten = 911,2,Dial(${DIGG2}/${EXTEN})
exten = 911,2,Dial(${DIGG2}/${CALLERID(ani)})
===
I've tried it with either one of those ${EXTEN} which just does 911, and the
${CALLERID(ani)} both have the same result no number transmitted over the
911 trunks, keep in mind we still get to the 911 center with no problem.
Here is the DEBUG from the last call I did:
DEBUG
[Aug 28 15:05:25] DEBUG[3207] pbx.c: Function result is '3086327836'
[Aug 28 15:05:25] DEBUG[3207] pbx.c: Function result is ' 3086327836'
[Aug 28 15:05:25] DEBUG[3207] pbx.c: Launching 'NoOp'
[Aug 28 15:05:25] DEBUG[3207] pbx.c: Function result is '3086327836'
[Aug 28 15:05:25] DEBUG[3207] pbx.c: Launching 'Dial'
[Aug 28 15:05:25] DEBUG[3207] chan_zap.c: Using channel 25
[Aug 28 15:05:25] DEBUG[3207] rtp.c: Channel 'Zap/25-1' has no RTP, not
doing anything
[Aug 28 15:05:25] DEBUG[3207] channel.c: Not copying variable
STACK-from-pbx-911-2.
[Aug 28 15:05:25] DEBUG[3207] channel.c: Not copying variable
STACK-from-pbx-911-1.
[Aug 28 15:05:25] DEBUG[3207] chan_zap.c: Dialing '3086327836'
[Aug 28 15:05:25] DEBUG[3207] chan_zap.c: Deferring dialing...
[Aug 28 15:05:25] DEBUG[2347] channel.c: Avoiding initial deadlock for
channel '0x81dda38'
[Aug 28 15:05:26] DEBUG[3207] chan_zap.c: Exception on 41, channel 25
[Aug 28 15:05:26] DEBUG[3207] chan_zap.c: Got event Wink/Flash(3) on channel
25 (index 0)
[Aug 28 15:05:26] DEBUG[3207] chan_zap.c: Exception on 41, channel 25
[Aug 28 15:05:26] DEBUG[3207] chan_zap.c: Got event Hook Transition
Complete(12) on channel 25 (index 0)
[Aug 28 15:05:26] DEBUG[3207] chan_zap.c: Got hook complete in MF FGD,
waiting for wink now on channel 25
[Aug 28 15:05:27] DEBUG[3207] chan_zap.c: Exception on 41, channel 25
[Aug 28 15:05:27] DEBUG[3207] chan_zap.c: Got event Dial Complete(9) on
channel 25 (index 0)
[Aug 28 15:05:27] DEBUG[3207] chan_zap.c: No echo cancellation requested
[Aug 28 15:05:27] DEBUG[3207] chan_zap.c: Exception on 41, channel 25
[Aug 28 15:05:27] DEBUG[3207] chan_zap.c: Got event Ring/Answered(2) on
channel 25 (index 0)
[Aug 28 15:05:29] DEBUG[3207] chan_zap.c: Exception on 41, channel 25
[Aug 28 15:05:29] DEBUG[3207] chan_zap.c: Got event Dial Complete(9) on
channel 25 (index 0)
[Aug 28 15:05:29] DEBUG[3207] chan_zap.c: No echo cancellation requested
[Aug 28 15:05:29] DEBUG[3207] chan_zap.c: No echo training requested
[Aug 28 15:05:30] DEBUG[3207] chan_zap.c: Exception on 41, channel 25
[Aug 28 15:05:30] DEBUG[3207] chan_zap.c: Got event Dial Complete(9) on
channel 25 (index 0)
[Aug 28 15:05:30] DEBUG[3207] chan_zap.c: No echo cancellation requested
=
VEBOSE
=
[Aug 28 15:05:25] -- Accepting AUTHENTICATED call from 64.187.80.26:
requested format = ulaw,
requested prefs = (ulaw),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
[Aug 28 15:05:25] -- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/pbx1-15,
911 ANI CALLER ID INFO IS: 3086327836 REGULAR CALLER ID: 3086327836)
in new stack
[Aug 28 15:05:25] -- Executing [EMAIL PROTECTED]:2] Dial(IAX2/pbx1-15,
Zap/g2/3086327836) in new stack
[Aug 28 15:05:25] -- Called g2/3086327836
[Aug 28 15:05:30] -- Zap/25-1 answered IAX2/pbx1-15
[Aug 28 15:05:56] -- Hungup 'Zap/25-1'
==
Is there some way to see in logging if we are transmitting, the ANI via mf
properly ( as in any mf tone logging ), I can't seem to make it log the KP
lines it sends at all. Can someone point this out in the code, as I can't
seem to find where in the code it builds these strings.
I'm using
Asterisk 1.4.11
Zaptel 1.4.3
libpri 1.4.1
WANPIPE 3.1.1
Thank you for any help you can give.
==
Andrew Ott Email: [EMAIL PROTECTED] or [EMAIL PROTECTED]
Network Admin/Webmaster Web:
Re: [asterisk-users] G729 Confusion
Matt wrote: Well does g729 have to run on both legs of a call? If the call is established and there is audio both ways then yes. If the call has not been answered yet then you will only see 1 encoder used. In your case somebody is using up 5 encoders and it is probably from calls coming into the box from the PSTN side since Asterisk is having to use 5 encoders to 'encode G729 from ulaw or another codec'. If this is a test platform then you might have a loop. If this is a production system with a lot of users then try to track down the offenders with a 'sip show channels' and look at the 'Form' column to see who is using G729. For instance, when I have 5/0 and I make a call from a SIP device... I get 5/1.. I can't hear any audio on my SIP phone, however if I call someone they can hear me. That is expected since when you tried to make the call there were no encoders left. Your phone encodes the call and Asterisk is able to decode it and deliver the audio to the other party but not the other way around. On 8/28/07, Andres [EMAIL PROTECTED] wrote: and reload, strange things begin to happen. A show g729 shows this: 5/0 encoders/decoders of 5 licensed channels are currently in use I think you have a loop of some kind. As you can see none of those call are actually established since no decoders are in use. Try to debug and see why those 5 calls are acually not connected in the first place. and suddenly I can not hear anything if I try to make a call.From observation, it almost seems like other units on the network are using the g729 codecs, but doesn't my sip.conf prohibit g729 unless expressly allowed?! Why would allowing g729 under one extension allow everyone else to suddenly start using g729? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Andres http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Confusion
On Tue, Aug 28, 2007 at 10:39:09AM -0500, Andres wrote: For instance, when I have 5/0 and I make a call from a SIP device... I get 5/1.. I can't hear any audio on my SIP phone, however if I call someone they can hear me. That is expected since when you tried to make the call there were no encoders left. Your phone encodes the call and Asterisk is able to decode it and deliver the audio to the other party but not the other way around. Does that mean 1) Don't buy 729 licenses in odd numbers ? or 2) Asterisk will attempt to complete a call (rather than correctly returning reorder) when it can't allocate a codec for both directions of the call. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HDL F10 brazilian doorbell device + TDM2400
Hi Matthew, I asked HDL some docs about their product but nothing is explained about signalling. I've tried reversepolarity and busydetect without success. I know 4 things: a) it works with analogic PBX b) both F10 and called party phone are connected to the PBX so no telco line is involved c) the F10 user hears a strange long noise instead of a hangup noise (even if Asterisk gives an Hangup message) d) I'm using the immediate=yes mode (it is not a normal analogic phone) and this is mandatory to make F10 work correctly I'm sorry I do not have more infos. Giorgio Matthew Fredrickson wrote: gincantalupo wrote: Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not living there so I do not know if its a matter of signalling type or what. Is there anbody who tried this stuff or similar? It sounds like there might be an issue here related to not having disconnect supervision enabled. Can this device provide come sort of disconnect supervision? -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.4 compiling problems
On Tue, Aug 28, 2007 at 12:00:47PM -0300, equis software wrote: Hi! I have this error compiling Zaptel 1.4.4 Any reason you don't use 1.4.5.1 ? make: *** No rule to make target `xpp/xpp_usb.ko', needed by `install-modules'. Stop. The Zaptel 1.2.5 compile ok. Any ideas?? What kernel? What distribution? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting tones
(With regard to your final question) As far as I can tell, EAGI is AGI with the extra file descriptor devoted to the linear pcm audio stream. As such, I would assume, but have never tested, that in order to send DTMF OUT from your AGI app, you would need to use the AGI functions, i.e. EXEC SENDDTMF options should work. EXEC PLAYBACK and STREAM FILE might work for you too if you've got dtmf tones stored in files. Moj Robert Prince wrote: Hello folks, I'm interested in detecting tones on specific frequencies with specific timing; for example, I'd like Asterisk to dial out and when the channel starts/call connects, listen for a 1200Hz tone that plays for 100ms. Is this doable with Asterisk using something already extant? After looking through documentation, mailing lists, and some of the source I had the idea that I might be better off using EAGI for this, and coding the actual listener in C. If EAGI were the right way to go, would I be able to respond/send tones back (e.g., DTMF tones) on the audio stream? Or would it go to STDOUT from the EAGI app's perspective? Thanks and cheers, Robert Prince ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell SC1430 + Digium TE110P = Digital Noise in PRI
On Tue, Aug 28, 2007 at 09:51:41AM -0400, Joe Acquisto wrote: When I voiced that concern to the Digium techs, they set up a thing called screen (I think it was) to allow me to see and or interact with their session. gnu screen is a standard prorgram available in most distributions. I usually try to install it the first thing I get to a system before wacking anything. And then ask the other party to run: screen -x You also learn then that '#' begins a comment on bash, and that 'yes' is a dangerous thing to say in an interactive session. This can help to see what what the other guy is doing in case you generally trust it to have good intentions and not plant a backdoor on your system or something... (That is not to say that I would allow strangers get root access to my laptop :-) ) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Confusion
Ok, So does someone want to explain this: (About 5 hours after I enabled G729... reloaded... then had the problem... then disabled G729.. and reloaded I still have): 1*CLI show g729 5/0 encoders/decoders of 5 licensed channels are currently in use Yet, no one is using these: iax2 show channels shows everyone is on ulaw. sip show channels shows everyone is on ulaw. On 8/28/07, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Tue, Aug 28, 2007 at 10:39:09AM -0500, Andres wrote: For instance, when I have 5/0 and I make a call from a SIP device... I get 5/1.. I can't hear any audio on my SIP phone, however if I call someone they can hear me. That is expected since when you tried to make the call there were no encoders left. Your phone encodes the call and Asterisk is able to decode it and deliver the audio to the other party but not the other way around. Does that mean 1) Don't buy 729 licenses in odd numbers ? or 2) Asterisk will attempt to complete a call (rather than correctly returning reorder) when it can't allocate a codec for both directions of the call. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.4 compiling problems
My linux kernel version is v2.6.15 (Gentoo) I think my kernel need some usb modules. At night I try to compile anwe kernel with usb options. I'll try to use Zaptel 1.4.5.1 On 8/28/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Aug 28, 2007 at 12:00:47PM -0300, equis software wrote: Hi! I have this error compiling Zaptel 1.4.4 Any reason you don't use 1.4.5.1 ? make: *** No rule to make target `xpp/xpp_usb.ko', needed by `install-modules'. Stop. The Zaptel 1.2.5 compile ok. Any ideas?? What kernel? What distribution? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel hardware for timing (was: Re: app-conference)
ram wrote: app_meetme can use ztummy but on highload expect to use hardware source A thing that was on my mind for quite some time now: Would it be beneficial to have a Zaptel compatible card in a system just as a timing source, even if it's not connected to a PRI? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.4 compiling problems
On Tue, Aug 28, 2007 at 02:07:58PM -0300, equis software wrote: My linux kernel version is v2.6.15 (Gentoo) I think my kernel need some usb modules. At night I try to compile anwe kernel with usb options. I'll try to use Zaptel 1.4.5.1 So you probably don't have USB support in the kernel. The zaptel build system still does not know how to detect that from your kernel config. So Just disable the module xpp. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNOW Web GUI
As Tzafrir stated, it will NOT work with 1.2.x. Where is this html.conf, which README? I will update it. I will write a brief page on setting up the *GUI for all who want to know.. There are SOME GUI's that work with 1.2, however, I almost guarantee none of them are client side, such as this one. -bk Steve Totaro wrote: Will this work on 1.2.x? I just installed it and did make samples. The README references a file called html.conf which does not exist and also abruptly ends with the word to on a blank line. Besides that, what would the URL be for AsteriskNow? Is that customizable in the elusive html.conf file? Any GUIs that are easily installed on existing systems and work with 1.2.x? Thanks, Steve bkruse wrote: svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow thegui; cd thegui; sh configure; make sudo make install ; clear ; echo 'completed' -bk Yann JOUANIN wrote: You can do it from svn server , I think there is a page in the wiki Best, yann *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Jeremy Mann *Envoyé :* vendredi 24 août 2007 17:30 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* [asterisk-users] AsteriskNOW Web GUI Is the web GUI for AsteriskNOW able to be loaded on an existing server(that was installed from ubuntu-server and asterisk loaded from source)? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Confusion
Jay R. Ashworth wrote: On Tue, Aug 28, 2007 at 10:39:09AM -0500, Andres wrote: For instance, when I have 5/0 and I make a call from a SIP device... I get 5/1.. I can't hear any audio on my SIP phone, however if I call someone they can hear me. That is expected since when you tried to make the call there were no encoders left. Your phone encodes the call and Asterisk is able to decode it and deliver the audio to the other party but not the other way around. Does that mean 1) Don't buy 729 licenses in odd numbers ? When you buy 1 license, you get 1 encoder and 1 decoder. So odd or even numbers are fine. or 2) Asterisk will attempt to complete a call (rather than correctly returning reorder) when it can't allocate a codec for both directions of the call. Yes, Asterisk will complete the call and you will have no audio if you have no free licenses. Cheers, -- jra ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Confusion
Matt wrote: Ok, So does someone want to explain this: (About 5 hours after I enabled G729... reloaded... then had the problem... then disabled G729.. and reloaded I still have): I can't explain that. I have only seen stuck encoders/decoders on very old versions of Asterisk. I remember on one version, if the call ended in voicemail, your encoder would be stuck and would never be freed. You would have to restart Asterisk to free the licenses. But that was corrected a long time ago. What version are you running? 1*CLI show g729 5/0 encoders/decoders of 5 licensed channels are currently in use Yet, no one is using these: iax2 show channels shows everyone is on ulaw. sip show channels shows everyone is on ulaw. Andres http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.4 compiling problems
Ok, but how can I do that?? Sorry I'm new in Linux/Asterisk world! Thanks Tzafrir On 8/28/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Aug 28, 2007 at 02:07:58PM -0300, equis software wrote: My linux kernel version is v2.6.15 (Gentoo) I think my kernel need some usb modules. At night I try to compile anwe kernel with usb options. I'll try to use Zaptel 1.4.5.1 So you probably don't have USB support in the kernel. The zaptel build system still does not know how to detect that from your kernel config. So Just disable the module xpp. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT
On 7/9/07, Noah Miller [EMAIL PROTECTED] wrote: Hi Stefan - What I want to accomplish: - calls within the LAN are re-invited (RTP goes from endpoint to endpoint) - asterisk detects when a call is going beyond the local LAN (over the NAT), and then stays in the middle. I'm wondering if this is hard to do and how I'm supposed to configure this. I don't really know how hard it would be to do what you describe, but if you're interested in getting the results you want with a minimum of effort, just keep asterisk in the media path all the time. Set canreinvite=no, and your calls should work consistently whether they stay inside the NAT or go outside. This is what I ended up doing. Until I ran into issues again with outgoing calls. Current setup = asterisk 1.4.11, installed on a host connected to the internet (internet route able IP-address) and my internal network ( 192.168.254.254). SIP phones are on the internal network, STUN and such hasn't been configured. SIP.conf: externhost = external hostname -- ddns.org canreinvite = no localnet = 192.168.254.0/24 ; nat = option is not set Outgoing call to our sip provider ends up being setup like this: outbound RTP stream: SIP phone (192.168.254.104) -- asterisk internal (192.168.254.254) asterisk external (internet IP) -- asterisk external (internet IP) (!!!) inbound RTP stream: SIP provider (internet IP) -- asterisk external (internet IP) asterisk internal (192.168.254.254) -- SIP phone (192.168.254.104) I have no idea why asterisk is trying to send the outbound RTP stream to itself. Removing the externhost and localnet settings doesn't help either. Neither does setting nat = yes, even in the example below. SIP.conf: externhost = external hostname -- ddns.org canreinvite = nonat localnet = 192.168.254.0/24 ; nat = option is not set. Outgoing call to our sip provider ends up being setup like this: outbound RTP stream: SIP phone (192.168.254.104) -- asterisk internal (192.168.254.254) asterisk external (internet IP) -- SIP provider (internet IP) inbound RTP stream: SIP provider (internet IP) -- asterisk external (internet IP) asterisk internal (192.168.254.254) -- SIP phone (192.168.254.104) The inbound RTP stream goes well for +/- 1 second, then the SIP provider responds to a re-invite sent by my asterisk box to send the trafic to 192.168.254.104 (the SIP phone on my internal network). outbound RTP stream: SIP phone (192.168.254.104) -- asterisk internal (192.168.254.254) asterisk external (internet IP) -- SIP provider (internet IP) inbound RTP stream: SIP provider (internet IP) -- SIP phone (192.168.254.104) I don't understand the logic of Asterisk sending the re-invite for inbound RTP stream. I would be more logical if Asterisk would send an invite for the outbound RTP stream: outbound RTP stream: SIP phone (192.168.254.104) -- SIP provider (internet IP) inbound RTP stream: SIP provider (internet IP) -- asterisk external (internet IP) asterisk internal IP (192.168.254.254) -- SIP phone (192.168.254.104) Does the logic have anything to do with in which order the interfaces are defined on the box? In my case, ETH0 = 192.168.254.254, ETH1 = internet IP. I can't find any configuration examples of my kind of setup, where a dual-homed host running asterisk has one NIC on the Internet and one on the internal (RFC1918 space) network. All examples I've bumped into have either the asterisk box behind a NAT router (i.e. it only has a RFC1918 IP-address) or the asterisk box is on a real IP. with kind regards, Stefan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed System
The question I always have when someone mentions distributing the load across multiple machines is how do you handle contexts for phones on different machines? I want all of my phones to dial into [companyA-phones]. I have to define it in two different places (or more depending on the number of boxes). Also, say I have a single company and I want a single auto attendant with dial by name? If users go to two different boxes, then voicemail dial by name will break because voicemail won't check both boxes for the name. Also, what about dialing a peer. Say all of my phones are 2xx. If I am 201 and I dial 202, how is my dialplan setup so that it knows that 202 is on box 2, versus box 1 where I am registered? I think having several boxes works fine if you are doing home user type stuff where you don't have lots of users within one context, but if you have offices with several people, I just see lots of potential issues. I could be wrong, but I've never been able to figure out a way around it. Brian West wrote: On Aug 28, 2007, at 10:14 AM, Seysan wrote: Hi all, I'm kind a New to Asterisk.But I'm a Network Administrator with 5 years of experiance. I want to know for an installation with 90 clients, If I don't want to have just 1 server for it, then how is it possible to distribute it among about 3 servers. Should I do it in a cluster (kernel level) or something with SER? I would recommend SER plus Asterisk. I have had great success with using Asterisk with OpenSER. Best Regards, Seysan /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATrpms/Fritz FCPCI CAPI/Fedora 7
HI all, Has anyone succesfully installed an AVM Fritz! card on Fedora 7 using the drivers at ATrpms recently? http://atrpms.net/dist/f7/fcpci/ I tried with a clean F7 build on my EPIA 5000 yesterday, after modifying /etc/capi.conf (removing the coment # in front of fcpci line) I received the following error when executing 'capiinit' - FATAL: Error inserting fcpci (/lib/modules/2.6.22.4-65.fc7/updates/drivers/isdn/fritz/fcpci.ko): Unknown symbol in module, or unknown parameter (see dmesg) ERROR: failed to load driver fcpci After some searcing I found this article - https://bugs.launchpad.net/ubuntu/+source/linux-restricted-modules-2.6.22/+bug/121978 I am a little stumped however what to do next and indeed if this is the cause of the problem, can anyone offer some guidance ? Thanks in advance. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E911 mf camma Trunks
Andrew Ott wrote:
ZAPATA.conf
===
;911 group
group = 2
restrictcid=yes
signalling = e911
channel = 25-26
===
...
I've tried it with either one of those ${EXTEN} which just does 911, and the
${CALLERID(ani)} both have the same result no number transmitted over the
911 trunks, keep in mind we still get to the 911 center with no problem.
Have you tried restrictcid=no ? I believe it should still send the ANI
(and only block the Caller ID information), but I'd suggest you try
turning it off, especially since I doubt the call takers will complain
if they get more information than they need, rather than no information
at all...
Trevor
--
Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please
visit http://www.digitalcon.ca/voip9-1-1/ to find out more!
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[asterisk-users] Fax Problems with SpanDSP
Hi list, I'm running current SpanDSP http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz with Asterisk 1.2.22 somewhat successfully. Most Fax machines do work but I have problems with people having Tobit FaxWare and Shamrock CapiFax. http://www.tobit.com/login/mrd.asp?CategoryID=120 http://www.shamrock.de/ I've got black bars over the pages. In Tobit some content is Ok, other is covered by the black bars. Anyone else has simliar problems? I talked to Tobit and they said there should be an option somewhere in SpanDSP to disable Fax header crossbars. But I found none. Can anybody help me with this issue. Please no switch to Hylafax mails, because I'm very happy with SpanDSP, it integrates nicely and works most time. Thank you, Regards Christian Peter ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problems with SpanDSP
Christian Peter wrote: Can anybody help me with this issue. Please no switch to Hylafax mails, because I'm very happy with SpanDSP, it integrates nicely and works most time. *chuckle* Lee. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Confusion
We are running 1.2.6 Asterisk version. On 8/28/07, Andres [EMAIL PROTECTED] wrote: Matt wrote: Ok, So does someone want to explain this: (About 5 hours after I enabled G729... reloaded... then had the problem... then disabled G729.. and reloaded I still have): I can't explain that. I have only seen stuck encoders/decoders on very old versions of Asterisk. I remember on one version, if the call ended in voicemail, your encoder would be stuck and would never be freed. You would have to restart Asterisk to free the licenses. But that was corrected a long time ago. What version are you running? 1*CLI show g729 5/0 encoders/decoders of 5 licensed channels are currently in use Yet, no one is using these: iax2 show channels shows everyone is on ulaw. sip show channels shows everyone is on ulaw. Andres http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed System
Peder @ NetworkOblivion wrote: The question I always have when someone mentions distributing the load across multiple machines is how do you handle contexts for phones on different machines? I want all of my phones to dial into [companyA-phones]. I have to define it in two different places (or more depending on the number of boxes). Also, say I have a single company and I want a single auto attendant with dial by name? If users go to two different boxes, then voicemail dial by name will break because voicemail won't check both boxes for the name. Also, what about dialing a peer. Say all of my phones are 2xx. If I am 201 and I dial 202, how is my dialplan setup so that it knows that 202 is on box 2, versus box 1 where I am registered? I think having several boxes works fine if you are doing home user type stuff where you don't have lots of users within one context, but if you have offices with several people, I just see lots of potential issues. I could be wrong, but I've never been able to figure out a way around it. Realtime + MySQL does it. That needs some extra work but it's possible. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problems with SpanDSP
Christian Peter wrote: Can anybody help me with this issue. Please no switch to Hylafax mails, because I'm very happy with SpanDSP, it integrates nicely and It just show you how many people on this list are pleased with HylaFAX+ 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Confusion
Matt wrote: We are running 1.2.6 Asterisk version. That is clearly a bug then. You should open a bug report. There are other similar strange things like the one in: http://bugs.digium.com/view.php?id=9526 On 8/28/07, Andres [EMAIL PROTECTED] wrote: Matt wrote: Ok, So does someone want to explain this: (About 5 hours after I enabled G729... reloaded... then had the problem... then disabled G729.. and reloaded I still have): I can't explain that. I have only seen stuck encoders/decoders on very old versions of Asterisk. I remember on one version, if the call ended in voicemail, your encoder would be stuck and would never be freed. You would have to restart Asterisk to free the licenses. But that was corrected a long time ago. What version are you running? 1*CLI show g729 5/0 encoders/decoders of 5 licensed channels are currently in use Yet, no one is using these: iax2 show channels shows everyone is on ulaw. sip show channels shows everyone is on ulaw. Andres http://www.telesip.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problems with SpanDSP
On Aug 28, 2007, at 3:49 PM, Doug Lytle wrote: Christian Peter wrote: Can anybody help me with this issue. Please no switch to Hylafax mails, because I'm very happy with SpanDSP, it integrates nicely and It just show you how many people on this list are pleased with HylaFAX+ Doug -- I'm rather pleased with it. /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problems with SpanDSP
On Tue, Aug 28, 2007 at 10:11:03PM +0200, Christian Peter wrote: Hi list, I'm running current SpanDSP http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz with Asterisk 1.2.22 somewhat successfully. Shouldn't you have used spandsp 0.0.3 with asterisk 1.2 ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load testing/burn-in for Sangoma quad PRI card
Hello all - I'm about to deploy an asterisk server here at work. Before deploying, I'd like to do an extended load test on the system. I currently have T1 crossover cables connecting ports 1-2 and 3-4. Would there be an easy way to script generating a bunch of calls across these spans? I envision generating 23 calls over the 1-2 span and 23 over the 3-4 span. I'd like to start the calls and then let them stay connected for several days to make sure things are in order. This number of calls would be a *lot* higher load than this system would ever see, but I just want to be safe. Is there currently any script out there that would facilitate this sort of testing? Here's my current config: linux-2.6.21 asterisk-1.4.10 zaptel-1.4.4 wanpipe-3.1.3 libpri-1.4.1 Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Password Issue
Asterisk Users, I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System 2.9.18-4-amd64. A TDM03B is installed on the Debian System. Every time, I try to change my voicemail pin via the Sip phone, the voicemail.conf does not get modify and I see this warning message on the Asterisk command line: [Aug 29 00:12:23] WARNING[19142]: app_voicemail.c:799 vm_change_password: variable has bad format. == Saving '/etc/asterisk/voicemail.conf': Saved == Parsing '/etc/asterisk/users.conf': Found == Saving '/etc/asterisk/users.conf': Saved -- SIP/225-00719470 Playing 'vm-passchanged' (language 'en') -- SIP/225-00719470 Playing 'vm-options' (language 'en') Now, when I manually change the pin in the voicemail.conf file there is no problem. I tried looking on internet for any information, but I found nothing useful. Does anybody have any insight on why I can't change my voicemail pin via the Sip phone? Thanks in advance. Here is my voicemail.conf file: maxsilence = 10 silencethreshold = 128 maxlogins = 3 emaildateformat = %A, %B %d, %Y at %r tz = central dialout = outbound sendvoicemail = yes callback = outbound review = yes nextaftercmd = yes [zonemessages] eastern = America/New_York|'vm-received' Q 'digits/at' IMp central = America/Chicago|'vm-received' Q 'digits/at' IMp central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours' military = Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' european = Europe/Copenhagen|'vm-received' a d b 'digits/at' HM [default] 200 = 201 = 1234 225 = 1234 _ Find a local pizza place, movie theater, and more .then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theatercp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=950607encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed System
Is there anywhere that we can look into for Realtime + MySQL that you mentioned? or about SER? Thanks On 8/28/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Peder @ NetworkOblivion wrote: The question I always have when someone mentions distributing the load across multiple machines is how do you handle contexts for phones on different machines? I want all of my phones to dial into [companyA-phones]. I have to define it in two different places (or more depending on the number of boxes). Also, say I have a single company and I want a single auto attendant with dial by name? If users go to two different boxes, then voicemail dial by name will break because voicemail won't check both boxes for the name. Also, what about dialing a peer. Say all of my phones are 2xx. If I am 201 and I dial 202, how is my dialplan setup so that it knows that 202 is on box 2, versus box 1 where I am registered? I think having several boxes works fine if you are doing home user type stuff where you don't have lots of users within one context, but if you have offices with several people, I just see lots of potential issues. I could be wrong, but I've never been able to figure out a way around it. Realtime + MySQL does it. That needs some extra work but it's possible. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NAT
On 8/28/07, Stefan van der Eijk [EMAIL PROTECTED] wrote: On 7/9/07, Noah Miller [EMAIL PROTECTED] wrote: Hi Stefan - What I want to accomplish: - calls within the LAN are re-invited (RTP goes from endpoint to endpoint) - asterisk detects when a call is going beyond the local LAN (over the NAT), and then stays in the middle. I'm wondering if this is hard to do and how I'm supposed to configure this. I don't really know how hard it would be to do what you describe, but if you're interested in getting the results you want with a minimum of effort, just keep asterisk in the media path all the time. Set canreinvite=no, and your calls should work consistently whether they stay inside the NAT or go outside. This is what I ended up doing. Until I ran into issues again with outgoing calls. Current setup = asterisk 1.4.11, installed on a host connected to the internet (internet route able IP-address) and my internal network (192.168.254.254). SIP phones are on the internal network, STUN and such hasn't been configured. SIP.conf: externhost = external hostname -- ddns.org canreinvite = no localnet = 192.168.254.0/24 ; nat = option is not set Outgoing call to our sip provider ends up being setup like this: outbound RTP stream: SIP phone (192.168.254.104) -- asterisk internal (192.168.254.254) asterisk external (internet IP) -- asterisk external (internet IP) (!!!) inbound RTP stream: SIP provider (internet IP) -- asterisk external (internet IP) asterisk internal (192.168.254.254) -- SIP phone ( 192.168.254.104) I have no idea why asterisk is trying to send the outbound RTP stream to itself. Removing the externhost and localnet settings doesn't help either. Neither does setting nat = yes, even in the example below. nat = yes solved it in the example above. SIP.conf: externhost = external hostname -- ddns.org canreinvite = nonat localnet = 192.168.254.0/24 ; nat = option is not set. Outgoing call to our sip provider ends up being setup like this: outbound RTP stream: SIP phone (192.168.254.104) -- asterisk internal (192.168.254.254) asterisk external (internet IP) -- SIP provider (internet IP) inbound RTP stream: SIP provider (internet IP) -- asterisk external (internet IP) asterisk internal (192.168.254.254) -- SIP phone (192.168.254.104) The inbound RTP stream goes well for +/- 1 second, then the SIP provider responds to a re-invite sent by my asterisk box to send the trafic to 192.168.254.104 (the SIP phone on my internal network). outbound RTP stream: SIP phone (192.168.254.104) -- asterisk internal (192.168.254.254) asterisk external (internet IP) -- SIP provider (internet IP) inbound RTP stream: SIP provider (internet IP) -- SIP phone (192.168.254.104) I don't understand the logic of Asterisk sending the re-invite for inbound RTP stream. I would be more logical if Asterisk would send an invite for the outbound RTP stream: outbound RTP stream: SIP phone (192.168.254.104) -- SIP provider (internet IP) inbound RTP stream: SIP provider (internet IP) -- asterisk external (internet IP) asterisk internal IP (192.168.254.254) -- SIP phone (192.168.254.104) Does the logic have anything to do with in which order the interfaces are defined on the box? In my case, ETH0 = 192.168.254.254, ETH1 = internet IP. I can't find any configuration examples of my kind of setup, where a dual-homed host running asterisk has one NIC on the Internet and one on the internal (RFC1918 space) network. All examples I've bumped into have either the asterisk box behind a NAT router ( i.e. it only has a RFC1918 IP-address) or the asterisk box is on a real IP. with kind regards, Stefan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load testing/burn-in for Sangoma quad PRI card
Having calls connected for that duration is worthless testing... What you need to do is call setup and tear down many times per second... I recommend trying to accomplish 20-30cps at 1ms to 10ms variable durations. That will expose any bugs quickly. And that my friend is how you expose any bugs... leaving calls up for days is easy... its the setup and tear down that you'll have bugs in. /b On Aug 28, 2007, at 4:11 PM, Erik Anderson wrote: Hello all - I'm about to deploy an asterisk server here at work. Before deploying, I'd like to do an extended load test on the system. I currently have T1 crossover cables connecting ports 1-2 and 3-4. Would there be an easy way to script generating a bunch of calls across these spans? I envision generating 23 calls over the 1-2 span and 23 over the 3-4 span. I'd like to start the calls and then let them stay connected for several days to make sure things are in order. This number of calls would be a *lot* higher load than this system would ever see, but I just want to be safe. Is there currently any script out there that would facilitate this sort of testing? Here's my current config: linux-2.6.21 asterisk-1.4.10 zaptel-1.4.4 wanpipe-3.1.3 libpri-1.4.1 Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed System
On 8/28/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Realtime + MySQL does it. That needs some extra work but it's possible. Or DUNDi. JR just posted a quick tutorial on getting that up and running: ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problems with SpanDSP
On Wed, 2007-08-29 at 00:03 +0300, Tzafrir Cohen wrote: On Tue, Aug 28, 2007 at 10:11:03PM +0200, Christian Peter wrote: Hi list, I'm running current SpanDSP http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz with Asterisk 1.2.22 somewhat successfully. Shouldn't you have used spandsp 0.0.3 with asterisk 1.2 ? Actually, as Steve Underwood has gently reminded the list several times, he recommends SpanDsp 0.0.2 for Asterisk 1.2 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Password Issue
Hello John, I think it is not the problem with your Asterisk, it is with your Phone (IP Phone or Softphone) Check the dtmf format on that. I think it is set to inbound, then change it to rfcxx. Then it should work fine. Regards, AFShin On 8/28/07, John Meksavan [EMAIL PROTECTED] wrote: Asterisk Users, I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System 2.9.18-4-amd64. A TDM03B is installed on the Debian System. Every time, I try to change my voicemail pin via the Sip phone, the voicemail.conf does not get modify and I see this warning message on the Asterisk command line: [Aug 29 00:12:23] WARNING[19142]: app_voicemail.c:799 vm_change_password: variable has bad format. == Saving '/etc/asterisk/voicemail.conf': Saved == Parsing '/etc/asterisk/users.conf': Found == Saving '/etc/asterisk/users.conf': Saved -- SIP/225-00719470 Playing 'vm-passchanged' (language 'en') -- SIP/225-00719470 Playing 'vm-options' (language 'en') Now, when I manually change the pin in the voicemail.conf file there is no problem. I tried looking on internet for any information, but I found nothing useful. Does anybody have any insight on why I can't change my voicemail pin via the Sip phone? Thanks in advance. Here is my voicemail.conf file: maxsilence = 10 silencethreshold = 128 maxlogins = 3 emaildateformat = %A, %B %d, %Y at %r tz = central dialout = outbound sendvoicemail = yes callback = outbound review = yes nextaftercmd = yes [zonemessages] eastern = America/New_York|'vm-received' Q 'digits/at' IMp central = America/Chicago|'vm-received' Q 'digits/at' IMp central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours' military = Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' european = Europe/Copenhagen|'vm-received' a d b 'digits/at' HM [default] 200 = 201 = 1234 225 = 1234 _ Find a local pizza place, movie theater, and more….then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theatercp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=950607encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load testing/burn-in for Sangoma quad PRI card (off list)
On 8/28/07, Brian West [EMAIL PROTECTED] wrote: What exactly are your needs? I can provide you some sipp scripts that might help you. Brian - thanks for the reply. If you read my email, I believe I make it fairly clear what my needs are. I have a 4-port Sangoma PRI card installed. Crossover cables are connected between ports 1-2 and ports 3-4. I'd like to generate a bunch of calls over those spans. -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed System
Realtime and DUNDi covers all the bases. On 8/28/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: The question I always have when someone mentions distributing the load across multiple machines is how do you handle contexts for phones on different machines? I want all of my phones to dial into [companyA-phones]. I have to define it in two different places (or more depending on the number of boxes). Also, say I have a single company and I want a single auto attendant with dial by name? If users go to two different boxes, then voicemail dial by name will break because voicemail won't check both boxes for the name. Also, what about dialing a peer. Say all of my phones are 2xx. If I am 201 and I dial 202, how is my dialplan setup so that it knows that 202 is on box 2, versus box 1 where I am registered? I think having several boxes works fine if you are doing home user type stuff where you don't have lots of users within one context, but if you have offices with several people, I just see lots of potential issues. I could be wrong, but I've never been able to figure out a way around it. Brian West wrote: On Aug 28, 2007, at 10:14 AM, Seysan wrote: Hi all, I'm kind a New to Asterisk.But I'm a Network Administrator with 5 years of experiance. I want to know for an installation with 90 clients, If I don't want to have just 1 server for it, then how is it possible to distribute it among about 3 servers. Should I do it in a cluster (kernel level) or something with SER? I would recommend SER plus Asterisk. I have had great success with using Asterisk with OpenSER. Best Regards, Seysan /b ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed System
This fails to take into account total failure of a machine. NAT mappings and various other variables that are not covered by Dundi or realtime... Best thing is to use OpenSER in the front then failure isn't a huge issue. /b On Aug 28, 2007, at 4:40 PM, Bruce Reeves wrote: Realtime and DUNDi covers all the bases. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] server recommentation (unique requirements)
We are at about 250 days of 24/7 uptime now. It would be more but we had a long power outage and the UPS's ran out. We are using Sangoma cards though. You can easily substitute the 2U for a 3U but I don't think you need it. Qty 1 Supermicro SC823T-R500LP, 2U, redundant 500W ps w/ PFC, 6x1 SATA hot swap bays, DVD-RW/Floppy http://www.supermicro.com/products/chassis/2U/823/SC823T-R500LP.cfm Qty 1 Supermicro PDSME+, Intel 3100 Mukilteo-2p chipset, Dual LAN, 2x 64-bit 133MHz PCI-X, 2x 64-bit 100MHz PCI-X, optional KVMoIP card http://www.supermicro.com/products/motherboard/Xeon3000/3010/PDSME+.cfm -Original Message- From: Stephen Kratzer [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 28, 2007 7:54 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] server recommentation (unique requirements) Howdy. I've been having trouble finding a fairly modern server that meets the following requirements: - Molex power connectors (don't want to use the Digium FXS power supply) - 4 PCI slots, one full-length (TDM2400P, 2xT100P, additional NIC) - dual power supplies - preferably dual CPUs = 1GHz - preferably rack-mountable (3-4RU) - CentOS-friendly We'd also like to stay away from older HP servers. Any recommendations would be greatly appreciated. Thanks. Stephen Kratzer Network Engineer CTI Networks, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Password Issue
Seysan, I tried changing the DTMF format to RFC2833, but it did not help. Any other suggests? From: Seysan [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail Password Issue Date: Tue, 28 Aug 2007 15:26:47 -0600 Hello John, I think it is not the problem with your Asterisk, it is with your Phone (IP Phone or Softphone) Check the dtmf format on that. I think it is set to inbound, then change it to rfcxx. Then it should work fine. Regards, AFShin On 8/28/07, John Meksavan [EMAIL PROTECTED] wrote: Asterisk Users, I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System 2.9.18-4-amd64. A TDM03B is installed on the Debian System. Every time, I try to change my voicemail pin via the Sip phone, the voicemail.conf does not get modify and I see this warning message on the Asterisk command line: [Aug 29 00:12:23] WARNING[19142]: app_voicemail.c:799 vm_change_password: variable has bad format. == Saving '/etc/asterisk/voicemail.conf': Saved == Parsing '/etc/asterisk/users.conf': Found == Saving '/etc/asterisk/users.conf': Saved -- SIP/225-00719470 Playing 'vm-passchanged' (language 'en') -- SIP/225-00719470 Playing 'vm-options' (language 'en') Now, when I manually change the pin in the voicemail.conf file there is no problem. I tried looking on internet for any information, but I found nothing useful. Does anybody have any insight on why I can't change my voicemail pin via the Sip phone? Thanks in advance. Here is my voicemail.conf file: maxsilence = 10 silencethreshold = 128 maxlogins = 3 emaildateformat = %A, %B %d, %Y at %r tz = central dialout = outbound sendvoicemail = yes callback = outbound review = yes nextaftercmd = yes [zonemessages] eastern = America/New_York|'vm-received' Q 'digits/at' IMp central = America/Chicago|'vm-received' Q 'digits/at' IMp central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours' military = Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' european = Europe/Copenhagen|'vm-received' a d b 'digits/at' HM [default] 200 = 201 = 1234 225 = 1234 _ Find a local pizza place, movie theater, and more .then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theatercp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=950607encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ More photos, more messages, more storageget 2GB with Windows Live Hotmail. http://imagine-windowslive.com/hotmail/?locale=en-usocid=TXT_TAGHM_migration_HM_mini_2G_0507 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed System
Seysan wrote: Is there anywhere that we can look into for Realtime + MySQL that you mentioned? Maybe http://www.voip-info.org/wiki/view/Asterisk+RealTime http://www.asteriskguru.com/tutorials/realtime_pgsql.html Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI cards, Digium vs. Sangoma
Your inabiity to configure a card doesn't make the competitors card better just because you said so. On 8/27/07, shadowym [EMAIL PROTECTED] wrote: They know what they are doing and do a lot of it. I don't have to give an Who is 'they'? opinion myself. There is enough evidence all over for people to draw the proper conclusions for themselves. Enough? really? where? On the interweb? Why does evidence count more than experience. If you don't have any experience state so. I drew my conclusions from my experience, since it seems that you don't have any you had to draw you conclusions based on evidence that for some reason you cant disclose. Is the evidence top secret? -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Sunday, August 26, 2007 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma On 8/26/07, shadowym [EMAIL PROTECTED] wrote: Well there are a couple fine examples of FUD if I do say so myself. Just do a search and see what cards the 'serious' companies out there are using. Nuff said. Can you define 'serious'? -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Sunday, August 26, 2007 8:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma Eric ManxPower Wieling wrote: Sangoma cards are complicated to set up, have a history of kernel (and zaptel) VERSION issues. i.e. It seems like the zaptel or kernel version I'm running on a machine is always something newer than is supported by the Sangoma drivers. Never had any issues once I got it compiled. You're forgetting one. I'm terrified of upgrading zaptel or the kernel from remote with the systems I have Sangoma cards on. I have, on many occasions, had kernel panics when trying to shut down wanrouter. I don't have this 'fear' with Digium cards. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNOW Web GUI
The README is here: svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow /Configuration = You may install sample configuration files by doing make samples. Also you will need to edit your Asterisk configuration files to enable the GUI properly, specifically: 1) In http.conf: [general] enabled = yes enablestatic = yes/ I am looking at Thirdlane's solution now. Very impressive and modest cost. Thanks, Steve bkruse wrote: As Tzafrir stated, it will NOT work with 1.2.x. Where is this html.conf, which README? I will update it. I will write a brief page on setting up the *GUI for all who want to know.. There are SOME GUI's that work with 1.2, however, I almost guarantee none of them are client side, such as this one. -bk Steve Totaro wrote: Will this work on 1.2.x? I just installed it and did make samples. The README references a file called html.conf which does not exist and also abruptly ends with the word to on a blank line. Besides that, what would the URL be for AsteriskNow? Is that customizable in the elusive html.conf file? Any GUIs that are easily installed on existing systems and work with 1.2.x? Thanks, Steve bkruse wrote: svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow thegui; cd thegui; sh configure; make sudo make install ; clear ; echo 'completed' -bk Yann JOUANIN wrote: You can do it from svn server , I think there is a page in the wiki Best, yann *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Jeremy Mann *Envoyé :* vendredi 24 août 2007 17:30 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* [asterisk-users] AsteriskNOW Web GUI Is the web GUI for AsteriskNOW able to be loaded on an existing server(that was installed from ubuntu-server and asterisk loaded from source)? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Password Issue
While I can't say this won't work the way you have it, I CAN say it's not the way mine is set up and it's not a way I've SEEN it ever set up. Could it just be complaining that you've got nothing on the right side of the = for mailbox 200? Or could it be complaining that you don't have anything past the pin number on the other lines? Try: 201 = 1234,Name or 201 = 1234,Name,email I'm thinking it's my first suggestion, though. To test that, try adding another without a pin number: 199 = and see if you then get two of the variable has bad format error messages Moj John Meksavan wrote: Here is my voicemail.conf file: [default] 200 = 201 = 1234 225 = 1234 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it possible to register without sending the password
Dear Philipp; How can I add the verbose and debug to the consol entry in the logger.conf to be able to take logging about the attempt of registeration for the sip end point? Regards Bilal If secret enabled, then some endpoints can not register (maybe due to compatibility in reading the negotiation packets), so what is the solution? I'm sure they can. Maybe you could tell the list which endpoints don't work? Also in SIP registration: why I do not see the log for registration packets periodically while I can see this in IAX2? Is it related to my v tracing level? Probably. How about you try with more vvv? If you *really* need to see what's going on you might add verbose and debug to the console= entry in logger.conf. But that's probably not what you want. Last point: I noticed that some endpoints that are not able to register (when secret is required), then I was not able to see any log at the asterisk side while SIP client still not registered. At least, it should display the fail for registeration, why does not display it? Is it related to my v tracing level? Where in the same tracing level, I am able to see the registeration fail if the endpoint sent an wrong username. For example if the context was [bilal_sip] and the endpoint username was bilal_1000 then I see a the message (log) that declare that registeration from bilal_1000 failed (ofcourse because bilal_1000 is not configured while bilal_sip is configured in the sip.conf). Could you send the part of your sip.conf? Sounds like a configuration issue. Regards, Philipp Kempgen Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545433 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load testing/burn-in for Sangoma quad PRI card (off list)
Erik Anderson wrote: On 8/28/07, Brian West [EMAIL PROTECTED] wrote: What exactly are your needs? I can provide you some sipp scripts that might help you. Brian - thanks for the reply. If you read my email, I believe I make it fairly clear what my needs are. I have a 4-port Sangoma PRI card installed. Crossover cables are connected between ports 1-2 and ports 3-4. I'd like to generate a bunch of calls over those spans. -erik Google SIPP and add some dialplan magic. Another more creative tool would be to place an ad in the Penny Saver or whatever your local equivalent is for a free 42 inch LCD TV, you haul and list your number. I bet that would generate alot of calls. You could put them through and IVR, then a queue, and finally a meetme room. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it possible to register without sending the password
bilal ghayyad wrote: How can I add the verbose and debug to the consol entry in the logger.conf to be able to take logging about the attempt of registeration for the sip end point? console = notice,warning,error,debug,verbose as explained in /etc/asterisk/logger.conf Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Load testing/burn-in for Sangoma quad PRI card (off list)
Erik Anderson wrote: On 8/28/07, Brian West [EMAIL PROTECTED] wrote: What exactly are your needs? I can provide you some sipp scripts that might help you. Brian - thanks for the reply. If you read my email, I believe I make it fairly clear what my needs are. I have a 4-port Sangoma PRI card installed. Crossover cables are connected between ports 1-2 and ports 3-4. I'd like to generate a bunch of calls over those spans. -erik Also, you seemed to miss Brian's main point, keeping calls up is not going to tax your box or prove anything really, you want to create as many short calls as possible. Run BOINC in the background for a CPU burn-in test. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI cards, Digium vs. Sangoma
Please don't feed the trolls. ;-) C F wrote: Your inabiity to configure a card doesn't make the competitors card better just because you said so. On 8/27/07, shadowym [EMAIL PROTECTED] wrote: They know what they are doing and do a lot of it. I don't have to give an Who is 'they'? opinion myself. There is enough evidence all over for people to draw the proper conclusions for themselves. Enough? really? where? On the interweb? Why does evidence count more than experience. If you don't have any experience state so. I drew my conclusions from my experience, since it seems that you don't have any you had to draw you conclusions based on evidence that for some reason you cant disclose. Is the evidence top secret? -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Sunday, August 26, 2007 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma On 8/26/07, shadowym [EMAIL PROTECTED] wrote: Well there are a couple fine examples of FUD if I do say so myself. Just do a search and see what cards the 'serious' companies out there are using. Nuff said. Can you define 'serious'? -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Sunday, August 26, 2007 8:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI cards, Digium vs. Sangoma Eric ManxPower Wieling wrote: Sangoma cards are complicated to set up, have a history of kernel (and zaptel) VERSION issues. i.e. It seems like the zaptel or kernel version I'm running on a machine is always something newer than is supported by the Sangoma drivers. Never had any issues once I got it compiled. You're forgetting one. I'm terrified of upgrading zaptel or the kernel from remote with the systems I have Sangoma cards on. I have, on many occasions, had kernel panics when trying to shut down wanrouter. I don't have this 'fear' with Digium cards. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Password Issue
Mojo, Thanks for helping me with this issue. You must have a NAME and EMAIL address after putting in the voicemail pin. I just migrate to Asterisk 1.4.x from 1.2.13, so I am still trying to get use to all the new stuff in the newer version. In Asterisk 1.2.13, it is not necessary to have a name and email address. Thanks again for your help in resolving this issue. Best Regards, John From: Mojo with Horan Company, LLC [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail Password Issue Date: Tue, 28 Aug 2007 14:10:19 -0800 While I can't say this won't work the way you have it, I CAN say it's not the way mine is set up and it's not a way I've SEEN it ever set up. Could it just be complaining that you've got nothing on the right side of the = for mailbox 200? Or could it be complaining that you don't have anything past the pin number on the other lines? Try: 201 = 1234,Name or 201 = 1234,Name,email I'm thinking it's my first suggestion, though. To test that, try adding another without a pin number: 199 = and see if you then get two of the variable has bad format error messages Moj John Meksavan wrote: Here is my voicemail.conf file: [default] 200 = 201 = 1234 225 = 1234 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ More photos, more messages, more storageget 2GB with Windows Live Hotmail. http://imagine-windowslive.com/hotmail/?locale=en-usocid=TXT_TAGHM_migration_HM_mini_2G_0507 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel causes kernel crash - zt_init_tone_state
Hi, I've been avoiding investigating this issue for a while; I used to revert to a previously compiled version of zaptel a previous kernel (as at some point I think I stopped being able to compile the older zaptel against the newer kernels) and all was well. However I've now upgraded kernels again and it seems silly to hide from the problem - here goes, let's try and fix it! At some point (and I'm afraid the exact point is lost in the mists of time) I upgraded zaptel as per normal, but that time the upgrade wasn't entirely successful and the new zaptel installation caused frequent crashes of my server. I think this was around the time of Zaptel 1.4.1 / 1.4.2, but I can't be entirely sure. A kernel call trace of when my system crashes, along with copies of various config files, is here: http://pastebin.ca/674091 but I have included highlights below. I'm a little bit stuck really, as I don't know where to start with debugging this. I'd really appreciate any clues or suggestions - I've configured my kernel for serial port console so that I can capture the call traces, and the problem is reproducible by simply picking up a Zap channel handset. Sometimes it takes longer to crash, sometimes it's quicker - and I'm getting some really strange dialtones varying from complete silence, through loud single tones, right up to weird multiple tones similar to a fax or modem. I'm running a stock FC6 system, and I don't believe it's a hardware issue as it has always worked fine under previous versions of zaptel. My hardware is a TDM400 with an (unused) X100P as well - see below for output when my machine boots up. Any suggestions gratefully received - where do I start?! Thanks, Jonathan -- If we knew what it was we were doing, it would not be called research, would it? - Albert Einstein Zaptel Echo Canceller: MG2 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXS/DPO Module 3: Installed -- AUTO FXS/DPO Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) wcfxo: DAA mode is 'FCC' Found a Wildcard FXO: Wildcard X101P Zaptel Transcoder support loaded Call Trace: [f89ae977] __zt_transmit_chunk+0x38d/0x10ff [zaptel] [f8822173] uhci_scan_schedule+0x6c7/0x75f [uhci_hcd] [f89b503f] zt_transmit+0x3a1/0x48e [zaptel] [f89b50b4] zt_transmit+0x416/0x48e [zaptel] [f89897d2] wctdm_interrupt+0x91f/0x9b5 [wctdm] [f895eb18] wcfxo_interrupt+0x2ee/0x5e3 [wcfxo] [c04541e6] handle_IRQ_event+0x1a/0x3f [c04553f5] handle_fasteoi_irq+0x64/0x98 [c0455391] handle_fasteoi_irq+0x0/0x98 [c04071f7] do_IRQ+0xac/0xd1 [c041ad1f] smp_apic_timer_interrupt+0x74/0x80 [c040592b] common_interrupt+0x23/0x28 [c0403281] mwait_idle_with_hints+0x3b/0x3f [c0403285] mwait_idle+0x0/0xa [c04033c9] cpu_idle+0x96/0xb7 [c0764a8e] start_kernel+0x316/0x31e [c0764227] unknown_bootoption+0x0/0x202 === Code: 40 54 00 00 00 00 c7 40 58 00 00 00 00 eb 14 89 50 44 c7 40 50 00 00 00 00 89 48 54 8b 44 24 08 89 43 58 5b c3 c7 00 00 00 00 00 8b 4a 04 89 48 04 8b 4a 08 c7 40 0c 00 00 00 00 89 48 08 8b 4a EIP: [f89ab285] zt_init_tone_state+0x6/0x2c [zaptel] SS:ESP 0068:c07cbf10 Kernel panic - not syncing: Fatal exception in interrupt ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNOW Web GUI
I am looking at Thirdlane's solution now. Very impressive and modest cost. The asterisk GUI is free :] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AsteriskNOW Web GUI
Steve, That is http.conf, not html.conf. You can type make checkconfig to check your asterisk configuration now. -bk Steve Totaro wrote: The README is here: svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow /Configuration = You may install sample configuration files by doing make samples. Also you will need to edit your Asterisk configuration files to enable the GUI properly, specifically: 1) In http.conf: [general] enabled = yes enablestatic = yes/ I am looking at Thirdlane's solution now. Very impressive and modest cost. Thanks, Steve bkruse wrote: As Tzafrir stated, it will NOT work with 1.2.x. Where is this html.conf, which README? I will update it. I will write a brief page on setting up the *GUI for all who want to know.. There are SOME GUI's that work with 1.2, however, I almost guarantee none of them are client side, such as this one. -bk Steve Totaro wrote: Will this work on 1.2.x? I just installed it and did make samples. The README references a file called html.conf which does not exist and also abruptly ends with the word to on a blank line. Besides that, what would the URL be for AsteriskNow? Is that customizable in the elusive html.conf file? Any GUIs that are easily installed on existing systems and work with 1.2.x? Thanks, Steve bkruse wrote: svn co http://svn.digium.com/svn/asterisk-gui/branches/asterisknow thegui; cd thegui; sh configure; make sudo make install ; clear ; echo 'completed' -bk Yann JOUANIN wrote: You can do it from svn server , I think there is a page in the wiki Best, yann *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Jeremy Mann *Envoyé :* vendredi 24 août 2007 17:30 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* [asterisk-users] AsteriskNOW Web GUI Is the web GUI for AsteriskNOW able to be loaded on an existing server(that was installed from ubuntu-server and asterisk loaded from source)? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
