[asterisk-users] AGI Script for thinQ CNAM lookup
Hi All, Does anyone have and can share with me an AGI script to dip thinQ for cnam? oR perhaps dialplan curl using curlopts? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDI with minimal features
> I have 2 PBX's, one in each office (say one in New York, one in Boston). I > have mobile users that can show up at either office and connect their soft > phones. > > > > Is there a very simple DUNDI config available which describes how to set > this up? > > Also, can I have the same outbound trunks setup in each office, so that > calls don't have to route across the NY-BOS connection to get out to the > PSTN? > > > > Thanks, Janet Not sure how relevant on newer versions, but yes, pretty easy to setup. http://blog.sinologic.net/wp-content/uploads/2007/08/DUNDi_So_Easy.pdf Good luck!. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicast RTP Paging
>> Thanks Josh. I spun up the latest version 13.6 (current) and added >> chan_rtp.c to channels/ but it will not compile, getting a lot of >> missing declaration parameters, are there any dependencies needed or >> what else should I try. What version of Asterisk did you build >> chan_rtp with Unicast support? Is it in Trunk or SVN testing branch? > > It is currently only available in git master, I haven't tried throwing > it into 13 so I'm uncertain of what all would need to be changed. > > -- > Joshua Colp Works as expected using git master branch, I'm not running git master in production but could really use this functionality. Any ideas on how I could backport/patch UnicastRTP to another branch? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicast RTP Paging
> On 15-10-20 07:18 PM, JR Richardson wrote: >> Hi All, >> >> I playing around with multicast paging, I saw a post from Josh Colp >> about adding unicast support into chan_multicast_rtp but not finding >> details if this is incorporated in dialplan functions or not. >> >> Basically I would like to send a unicast page/rtp stream out to a >> unicast-to-multicast re-director cisco router. >> >> Can anyone point me in the right direction? > > I've got a blog post[1] which explains how I use it for AstriDevCon > streaming, it includes the format. > > Cheers, > > [1] http://www.joshua-colp.com/broadcasting-asterisk-conferences/ > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > Thanks Josh. I spun up the latest version 13.6 (current) and added chan_rtp.c to channels/ but it will not compile, getting a lot of missing declaration parameters, are there any dependencies needed or what else should I try. What version of Asterisk did you build chan_rtp with Unicast support? Is it in Trunk or SVN testing branch? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unicast RTP Paging
Hi All, I playing around with multicast paging, I saw a post from Josh Colp about adding unicast support into chan_multicast_rtp but not finding details if this is incorporated in dialplan functions or not. Basically I would like to send a unicast page/rtp stream out to a unicast-to-multicast re-director cisco router. Can anyone point me in the right direction? Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Tech/Eng Positions Open In Dallas TX
We have a couple of positions open, please contact me off-list if interested. http://www.ntegratedsolutions.com/voice-engineer-dallas/ These are full time positions in Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7940 SIP 8.12 no audio when using Outbound Proxy
Hi All, Simple scenario: 7940 SIPNAT RouterINTERNETAsterisk SIP B2BUA w/Public IP Inbound/outbound calls work fine 2 way audio, features ok, no issues that I can tell so far. 7940 SIP Using Outbound SIP ProxyNAT RouterINTERNETAsterisk SIP w/Public IP Phone registers, call in/out SIP Signaling traversing the proxy ok no audio on phone, SDP messaging is correct. Not using media proxy, media flows between Asterisk and NAT router to phone, no return media from phone to Asterisk. This only occurs on inbound calls to the phone, when the phone makes outbound calls, audio sets up fine 2-way. On inbound call to the phone, I can see media going to the phone but don't hear any on the speaker/handset, no media flowing out of the phone back towards Asterisk so no audio on that end either. I'm testing various phones, the Polycom and Cisco SPA5xx lines work great using outbound proxy. So I'm certain this is a Cisco 7940 problem not accepting RTP due to some internal security check with SIP signaling and media coming from different ip addresses or something like that. So testing a bit more, I put the Cisco 7940 on a Public IP, seems to work fine, audio sets up 2-way inbound and outbound calls. So now I'm thinking it is a NAT issue, but only when using outbound proxy, doesn't make sense, now I'm really confused. Any feedback is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Prepend Message Forwarding Not Working
Hi All, First I've heard of this feature not working from a customer. I did some digging and this is a common bug in several older Asterisk versions, it has more than a few patches in the bug tracker. I've tried a few of them but none will apply to a specific version I'm currently running for a customer, 1.6.0.28. Does anyone have a patch file that will apply to this version or an app_voicemail.c file that is already patched and will compile with this versions to fix this particular bug? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Prepend Message Forwarding Not Working [SOLVED]
Hi All, First I've heard of this feature not working from a customer. I did some digging and this is a common bug in several older Asterisk versions, it has more than a few patches in the bug tracker. I've tried a few of them but none will apply to a specific version I'm currently running for a customer, 1.6.0.28. Does anyone have a patch file that will apply to this version or an app_voicemail.c file that is already patched and will compile with this versions to fix this particular bug? I patched app_voicemail.c manually from the patch file (revision 233691), recompiled and now prepending voicemail works. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Use Allworx Phones With Vanila Asterisk PBX?
Hi All, Got a Customer that bought some Allworx phones but for some reason or another can't use the Allwork PBX. I don't know anything about these phones and can't seem to access the config web interface, http://phone ip/, just gives info, not sure how to configure these phones. Wondering if these will work with vanilla Asterisk system or are they hard wired for Allwork systems only? Any feedback is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Job Posting
Ntegrated Solutions in Dallas, TX is still looking for voice guy. This position is for US hire only, will not sponsor H1B work visa. http://www.ntegrated.net/careers/ Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Tech Job Posting Dallas Texas
Hi All, Ntegrated Solutions is looking for a full-time Asterisk/Telecom Tech and a .net/php developer. http://www.ntegratedsolutions.com/careers/ Forward resume' to j...@ntegrated.com Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk SIP Refer Transfers
Hi All, Using Asterisk 1.6.0.28, having to register some Cisco 7940/60 with SIP firmware 7.4.0. Most functions work from the phone except blind transfer (attended transfer from phone works fine and # PBX transfer works). Blind transfer from the phone uses SIP Refer method. I've seen a bunch of posts about asterisk and SIP Refer, but I can't seem to find the version that this has been fixed, 1.6.2, 1.8. 1.10? I would like to stick with the 1.6.x for config continuity but would upgrade to 1.8 if needed. Or could updating the Cisco 7940/60 SIP firmware to 8.x fix this as well? Any guidance is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto ban IP addresses
I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100sip:100@108.161.145.18;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? You may want to check out this presentation form the last Astricon, it may be relevant: http://www.astricon.net/2012/videos/Automated-Hacker-Mitigation.html Cheers. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.0 disable cdr account logs?
JR Richardson wrote:
My bad. I sent Igor to the boneyard to fetch 1.6.0.28 and it appears to me
that by commenting out lines 309-312 and doing a fresh make you eliminate
the extra files (or make them empty).
Appriciate the suggestion but commenting out 309-312 refused to compile:
cdr_csv.c
/* if (!ast_strlen_zero(cdr-accountcode)) {
if (writefile(buf, cdr-accountcode))
ast_log(LOG_WARNING, Unable to write CSV
record to account file '%s' : %s\n, cdr-a$
*/ }
You need to place the */ after the } or else they are mismatched and
like you have seen, the universe will explode.
Got it, after properly commenting out that section and re-compiling
and reloading cdr_csv.so, I still get the individual account code
CDR's.
Any other suggestions?
Thanks.
JR
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Engineering for the Masses
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Re: [asterisk-users] Asterisk 1.6.0 disable cdr account logs?
Just add noload=cdr_csv.so to modules.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JR Richardson Sent: Friday, October 19, 2012 5:09 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.6.0 disable cdr account logs? Hi All, I would like to disable the cdr account logs but in 1.6.0 but the 'accountlogs=no' switch is not available till 1.8 as far as I can tell. Is the any switch I can turn off int he Mkae file for the cdr_csv.so module to disable accountcode logs? Correct me if I'm wrong, if I noload=cdr_csv.so, won't that disable all csv CDR's. I still want the Master CSV file with account code, what I don't want is a seperate CSV CDR for each accountcode generated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.0 disable cdr account logs?
Hi All, I would like to disable the cdr account logs but in 1.6.0 but the 'accountlogs=no' switch is not available till 1.8 as far as I can tell. Is the any switch I can turn off int he Mkae file for the cdr_csv.so module to disable accountcode logs? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP Realtime Architecture Issue/Bug
I am facing an issue with Peer registration in my asterisk server . I am using asterisk version 1.8.5.0 and using SIP real-time architecture.when i am doing registration it registered fine on asterisk as peer is available in Database. But now i am doing 'sip reload' or 'reload' due to some reason my peer registration is going out and i cannot able to call that peer even though in SIP client it shows me 'registered'. Can any body elaborate on this issue which settings i need to put in sip.conf. I also tried to follow this patch https://issues.asterisk.org/view.php?id=14196 But it allready applied in code base so why it wont work? Here is my sip.conf settings. [general] context=from-internal ; Default context for incoming cal rtcachefriends=no rtupdate=yes rtautoclear=yes rtsavesysname=yes callcounter = yes callevents=yes bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) srvlookup=yes ; Enable DNS SRV lookups on outbound calls pedantic=yes ; Enable slow, pedantic checking for Pingtel tos=184 ; Set IP QoS to either a keyword or numeric val tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos=lowdelay ; lowdelay,throughput,reliability,mincost,none maxexpiry=3600 ; Max length of incoming registration we allow defaultexpiry=120 ; Default length of incoming/outoing registration preferred_codec_only=yes disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw insecure=invite language=en ; Default language setting for all users/peers rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity useragent=dhaval ; Allows you to change the user agent string dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 qualify=yes nat=yes ;canreinvite=yes directmedia=yes directrtpsetup=yes And here is DB fields snapshots. id: 1 name: 201 ipaddr: 172.18.100.243 port: 53624 regseconds: 1328716180 defaultuser: 201 fullcontact: NULL regserver: dhaval useragent: CSipSimple r1133 / b lastms: 554 host: dynamic type: friend context: from-internal permit: NULL deny: NULL secret: 201 md5secret: NULL remotesecret: NULL transport: NULL dtmfmode: NULL directmedia: yes nat: NULL allow: ulaw disallow: g729 insecure: invite callerid: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL Kindly help me to resolve this. Thanks Dhaval The first thing I would try is 'rtcachefriends=yes', that should do it. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Configuration GUI Question
Hi All, There are a lot of existing projects for configuring Asterisk via GUI, so instead of trudging through them all, I'm hopeing to get some guidance. My architecture is ITSP based, we supply hosted PBX's to business customers. A few systems are dedicated PBX's but the majority are virtualized instances. We have been very successful managing the systems for our customers, not a lot of request for user portals or anything like that, so our PBX management consist of command line editing of Asterisk flat files and minimal sql database routines. We have built a few custom user portals for some of our customers using LAMP and have deployed a couple of other user web utilities, CDR search, Operator panel, Queue stats. I would like to implement a more standardized user portal for basic functions like call forward, voicemail password reset, user info change, queue member add/remove, ect I know there are many projects that could do just that, but most of what I'm finding are GUI's that take over the system and have conventions for many more configurable elements than I really need. Most are overkill for what I'm looking for. Because the majority of my PBX's are hosted virtual systems, overhead must be light. I would like to have a centralized management portal that pushes configs out to the PBX's but I'm not apposed to running a GUI on each PBX instance as long as it is light. I would like to be able to customize the interface, brand with my business logos, add or remove configuration elements. I kind of like the Digium Asterisk GUI but I'm just not real familiar with it, just test driving it a bit. What I do like about it is the flat file manipulation, no database needed. Any guidance is much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF issue with 1.8.6.0 and SIP Trunks
Hi All, I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in routing calls to upstream carrier via SIP trunks out. I spent a lot of time in the lab testing 1.8 which included heavily testing DTMF with no issues that came up. It all just seemed to work fine. But then again you can't reproduce every real work scenario in the lab. I'm using rfc2833 inbound and outbound for the new 1.8 call servers. Here is a quick diagram of what is working and what is not: Not working: Customer IP PBXsip trunk rfc2833ast 1.4 rfc2833sip trunkcall server ast 1.8 rfc2833sip trunkupstream carrier Customer PRIcisco PRI gatewaysip trunk rfc2833ast 1.4 rfc2833sip trunk call server ast 1.8 rfc2833sip trunkupstream carrier I can see DTMF RTP events pass through call server to carrier but no response, nothing, nada, zip. Working: Customer SIP Phonesip rfc2833ast 1.4 rfc2833sip trunk call server ast 1.8 rfc2833sip trunkupstream carrier Customer SIP Phonesip rfc2833ast 1.4 rfc2833sip trunk call server ast 1.2 rfc2833sip trunkupstream carrier Customer IP PBXsip trunk rfc2833ast 1.4 rfc2833sip trunk call server ast 1.2 rfc2833sip trunkupstream carrier Customer PRIcisco PRI gatewaysip trunk rfc2833ast 1.4 rfc2833 call server sip trunkast 1.2sip trunkupstream carrier I can see DTMF RTP events pass through to carrier, RTP stream looks the same as the 1.8 server with reliable responses. On both the 1.4 and 1.8 ast servers, these sip.conf parameters are active on peer and global settings: relaxdtmf=yes rfc2833compensate=yes dtmfmode=rfc2833 Now it quickly appears like a problem between the customer PBX and Customer PRI with the SIP trunks to the ast 1.4 servers but it all worked fine before with the 1.2 call servers. After the upgrade of the call servers to 1.8 DTMF is not recognized by the carrier on calls from the customer IP PBX or PRI but is fine with the SIP phones directly registered to the ast 1.4 servers. I found the bug issues with the SRCC change/update issues with DTMF events. It looks like 1.8.6.0 implemented the 'update' and as I read it, should have fixed the issue with the changing SRCC effecting DTMF. But this may not be the case. Specifically, how would I debug RTP/DTMF on the new ast 1.8 server and see if the SRCC is changing between my scenarios described above. Am I on the right track or is there something else I should be looking at? Thanks. JR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can we use MySQL native connector for ARA?
Hello Everyone, The documentation suggests using unixodbc for asterisk realtime. Is there any way we can just use native database clients such as libmysqlclient from MySQL? The native clients tend to be more up-to-date. Thanks in Advance, Nick. I've used the MySQL addon for years with great success, initiated the project to support read/write to separate databases for asterisk clustering. You will get more functionality for complex queries using the odbc connector, but for basic ARA applications, the MySQL addon works fine and I've never had a problem with stability. I also use the cdr_mysql as well. I wrote a couple of papers on asterisk_clustering_with_mysql_replication. They are a bit dated but still relevant. I'll send over if you like. Good luck. JR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 Manager Perl Script Problem
Hi All,
Trying to upgrade some call servers, in the lab making sure all my
applications work, ran into an issue with some manager perl scripts
that pull and reset database info, it seems the command and result
responses have changed but I'm not sure how to resolve. My scripts
are using CPAN Asterisk::Manager; and are working fine on asterisk
1.2.32 but not on Asterisk 1.8.6.0.
Here is the abbreviated script where 1.2.32 is astman1 and 1.8.6.0 is astman2:
#!/usr/bin/perl -w
use strict;
use warnings;
use Getopt::Long;
use Asterisk::Manager;
##setup manager connections##
my $astman1 = new Asterisk::Manager;
$astman1-user('username');
$astman1-secret('password');
$astman1-host('10.10.14.101');
$astman1-connect || die $astman1-error . \n;
my $astman2 = new Asterisk::Manager;
$astman2-user('username');
$astman2-secret('password');
$astman2-host('10.10.14.102');
$astman2-connect || die $astman2-error . \n;
##query databases for cnam count##
$astman1-sendcommand(Action = 'DBGet', Family = 'cnam', Key = 'count');
my @result1 = $astman1-sendcommand(Event = 'DBGetResponse');
my $cnamcount1 = 0$result1[7];
$astman2-sendcommand(Action = 'DBGet', Family = 'cnam', Key = 'count');
my @result2 = $astman2-sendcommand(Event = 'DBGetResponse');
my $cnamcount2 = 0$result2[7];
##total cnam count##
my $cnamtotal = ($cnamcount1+$cnamcount2);
##reset cnam count to 0##
$astman1-sendcommand(Action = 'DBPut', Family = 'cnam', Key =
'count', Val = '0');
my @result101 = $astman1-sendcommand(Event = 'DBGetResponse');
my $cnamreset1 = $result101[1];
$astman2-sendcommand(Action = 'DBPut', Family = 'cnam', Key =
'count', Val = '0');
my @result102 = $astman2-sendcommand(Event = 'DBGetResponse');
my $cnamreset2 = $result102[1];
##disconnect the manager connections##
$astman1-disconnect;
$astman2-disconnect;
print Total CNAM Count for last month is $cnamtotal\n\n;
-end script
The response from the 1.8.6.0 server is Response Success Message
Result will follow but is seems the actual response is not pulled
into $result2. The DBPut command works fine and I get a success
response. I've searched through all the upgrade docs but nothing
mentions command syntax changes.
Any help is appreciated.
Thanks.
JR
--
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Engineering for the Masses
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[asterisk-users] Asterisk SIP Trunk with CUCM Express, Disable Comfort Noise?
Hi All, I have a sip trunk up and running with a CUCM Express, passing calls fine except for a comfort noise error I'm getting on Asterisk: NOTICE[7520]: rtp.c:788 in process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: x.x.x.x I know Asterisk does not support comfort noise. I have no comfort noise on all dial peers in the CUCM express and also have no vad on all the voip peers. That should disable comfort noise but does not. Could this be a bug in IOS or does anyone have any hints on what else to look for? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
From: vip killa Sent: Thu 3/31/2011 8:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and fail2ban Back to the original question, for those of you using Fail2Ban, Does it take an unusually high amount of break-in attempts before attackers are banned? I have it set to 5 attempts in fail2ban but usually, the attacker is able to make over 100 attempts before fail2ban bans them. I've tried this using asterisk's /var/log/asterisk/messages and /var/log/messages with same results. Perhaps someone else is experiencing this or has resolved it, thank you. I have F2B set to ban after 1 attempt. The most I have seen in the logs is 4-5 attemps before ban is applied. I am calling scripts that apply the ban to a cisco access-list, so there is script/telnet/config delay but it is very minimal and works very well. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
I have F2B set to ban after 1 attempt. The most I have seen in the logs is 4-5 attemps before ban is applied. I am calling scripts that apply the ban to a cisco access-list, so there is script/telnet/config delay but it is very minimal and works very well. So I forge one SIP packet and I get you to block the IP address of your SIP trunk (or your IAX trunk)? Cool! -- Tzafrir Cohen Good thing I ignore my own IP blocks JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 80, Issue 73
Back to the original question, for those of you using Fail2Ban, Does it take an unusually high amount of break-in attempts before attackers are banned? I have it set to 5 attempts in fail2ban but usually, the attacker is able to make over 100 attempts before fail2ban bans them. I've tried this using asterisk's /var/log/asterisk/messages and /var/log/messages with same results. Perhaps someone else is experiencing this or has resolved it, thank you. I have F2B set to ban after 1 attempt. The most I have seen in the logs is 4-5 attemps before ban is applied. I am calling scripts that apply the ban to a cisco access-list, so there is script/telnet/config delay but it is very minimal and works very well. JR Speaking blindly as someone who has yet to fool with F2B, I'd rather ban somebody after 5-20 attempts than have the overhead needed to ban them quicker. Guess that's a na?ve view?? Well really I don't see why you would want to ban after 1 attempts. Unless you mis-configure user/pass for the SIP peer, an error message in the log for failure to register or ACL match is a bot or hacker more often than not. It's better to be safe and block first, then pull the ban off the suspect IP once you realize it is legitimate. I'd rather do that than let a hacker try to brute force password hack for 20 or more attempts. Some of these bots are real malicious with attempts coming in the hundreds/sec so my philosophy is to block soon and block often. If you are going to run an automated blocking mechanism, you should get proficient with un-blocking as well for accidental blocking. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 SSH Console Colors Debian Lenny
Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I found this in the init script: #snip-# # Mon Jun 04 2007 Iñaki Baz Castillo i...@in.ilimit.es # - Eliminated SAFE_ASTERISK since it doesn't work as LSB script (it could require a independent safe_asterisk init script). # If you DON'T want Asterisk to start up with terminal colors, comment # this out. COLOR=yes #snop# Commenting out COLOR=yes has no effect. The work around is to use the * 1.4 init script which does call safe_asterisk daemon and things seem to work as expected with the colors. So my question is, will this impact the stability of the system in reference to debian lenny using LSB scripts vs the older init scripts? Or is there another work around to get ssh console colors using the Debian * 1.6.0.28 LSB init script? I also tried 'nocolor = no' in the [options] section of asterisk.conf with no effect. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip attended transfer beep
Hi All, I see some patches about adding atxfer beep sound in the sip channel, but I'm not clear on when this was implemented in what version? I don't see the added function in chan_sip in 1.2.24 or 1.4.21 or 1.6.0.28? Where is this code implemented, what stable release? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician
We have a couple of positions open, please respond to the posting if qualified and interested. http://www.ntegrated.net/resources/job-opportunities/field-service-install-technician http://www.ntegrated.net/resources/job-opportunities/network-engineering-voice These are full time positions in Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP and ANI
Hi All, My research indicates ANI is not really supported with SIP Channels or passed between SIP servers, even with setting function CALLERID(ANI). So the only place this applies is on PRI interfaces, when sending calls out a ZAP PRI you can set the ANI to whatever and CID Number to a different whatever so on the other end of the PRI you will receive the two different values? Is this correct or is there a way to set ANI on an outgoing SIP channel (like to a PRI gateway) and the gateway will see a CID Number and a separate ANI and insert that into the ISDN messaging down the PRI? Thanks for any clarification. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco SIP 8.5 and 9.0 Issues
Hi list, I was wondering if anyone had any solution to either one of two issues I'm having: I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware, it works very well for the most part, but after less then a week of heavy usage, eventually the phone gets into a state where it won't accept or let you place any more calls, the screen flashes no free lines available or something along those lines. (power cycle fixes this). So my preferred solution would be to upgrade to the v9.0(3) firmware, but when that's loaded, the phone won't register with Asterisk anymore, does anyone know if I need to adjust my .cnf.xml file, or is it a bug of some sort? Thanks for any input, -- Gerard Saraber Network Admin. Rarcoa, Inc Use Polycom, but if you really must use cisco phones, downgrade to 7.5. I've got a lot of 79xx phones out there and 7.5 is the last stable release as far as I'm concerned. It just seems to work, no periodic reboots needed, or any other quirkiness like with the newer firmware's. The feature set is not robust, but it is reliable with Asterisk. Keep in mind, Cisco has no incentive to make their SIP firmware work with any other platform other than there servers so I don't really expect it to work properly if at all with Asterisk. 7.5 is the only firmware version that I deploy on a few hundred units and works fine. Good luck. JR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: HUD3 and NON-Trixbox Asterisk?
Hi All, Can anyone clerify that HUD3 is a fonality product, tied into the various trixbox systems? Is there a HUD3 client/server standalone project that can be installed and used with other Asterisk projects? Any comments on using the hudlite client/server package that came out a few years ago? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.38 on a MAX/Lucent/Ascend TNT
Date: Thu, 24 Jun 2010 15:32:39 -0400 From: Ben Winslow winsl...@pa.net Subject: [asterisk-users] T.38 on a MAX/Lucent/Ascend TNT To: asterisk-users@lists.digium.com Message-ID: 4c23b2d7.9090...@pa.net Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello folks, I've been trying to get T.38 over SIP working with calls terminated by a MAX/Lucent/Ascent TNT. As far as I can tell, SIP and T.38 are actually working perfectly; however, I can't get the TNT to properly terminate a FAX call. Does anyone have a working configuration for SIP and T.38 for calls from a TNT or APX? Here's a brief description/diagram of my test setup: Laptop --RS232-- Modem --POTS-- Channel bank (ADIT 600) --CAS T1-- TNT --Ethernet/SIP-- Asterisk --SIP-- t38modem. The TNT is running TAOS 11.0.4 with both the SIP and realtime fax features licensed. I've tried using several modems, and with each I can establish a data call without any problems (although at a maximum of 31200bps/3429 baud using a 33.6 modem, for reasons I haven't dug into yet.) Whenever I try to send a fax from the laptop, however, the call always seems to fail in the first HDLC phase (phase B) with either a timeout or error 23 (COMREC invalid command received.) The modem is connected directly to the channel bank and the channel bank is connected directly to the TNT in an attempt to reduce the number of variables. With my current configuration, the call to Asterisk will come up as a voice call, then be dumped into t38modem when Asterisk receives the 1100Hz CNG tone from the sending modem. At that point, the call is dumped into t38modem which re-invites the TNT with the T.38 options, and the TNT usually sends a T.38 t30-ind of 0x00 (no-signal) or 0x3a (???), although I do occasionally see more promising messages like v29-7200-training or t30-data/hdlc-fcs-ok. Shortly thereafter, the dialing modem will give up and terminate the call. If I try dumping the same call into iaxmodem instead of t38modem, the call actually progresses further -- the real modem receives and decodes the HDLC CSI/DIS from iaxmodem, but the high-speed trainup always fails for calls coming from the TNT. Does anyone have any advice or suggestions? Has anyone actually made T.38 work with one of these devices running ANY TAOS version? Thanks, -- Ben Winslow winsl...@pa.net What version of Asterisk are you running? I've struggled to get this working as well but with TNT TOAS version 14 and Asterisk 6.1.x I managed to get reliable T38 faxes one-way: from fax to PSTN Fax machineSIP T38 ATAAsteriskSIP T38MAX TNTPSTN PRI The other way does not work, the call doesn't switch to T38 and I haven't had time to investigate too deeply. Fax handling in TOAS was greatly improved in 14.0 version, I would suggest you upgrade to that if you can and start there. JR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Being attacked by an Amazon EC2
Perhaps if there was a Asterisk RBL we could all contribute to; for which we could then hook into and drop any connection where a source IP is listed ? -- Thanks, Phil I love the idea of a RBL... count me in for contributing. Especially considering the ridiculous response I received from Amazon. (Basically told me to submit host, destination, port, proto, and log... which of course was already included in the original complaint) I don't think anyone else brought up the Spamhaus DROP project. It's a blacklist of IP addresses and address ranges which are known to ONLY be used for malicious purposes. http://www.spamhaus.org/drop/ We could establish something similar to that for VOIP attacks. It may not be exactly a trivial system to maintain such a list. (removing IP's after X amount of time, disputing false claims etc). Maybe someone could contact spamhaus to create a list for VOIP since they seem to have a nice system in place? Hi All, good discussion, similar to ones we had a year or so ago. The RBL concept is valid, at least to get a repository going that list malicious activity specific to SIP attacks. n I worked with Project Honeypot guys for a while, they are more than willing to assist, as they already have the backend work done for a clearing house identifying hackers. The biggest issue we had a year ago was to create the mechanism in asterisk to push valid log messages out to the database and then determine what to do with that data? I tried to bridge the gap between a few Asterisk developers and the Honeypot developers, ultimately the project stalled and I got busy with other matters. If anyone here would like to pick up the torch and move this along, I can certainly provide info on how far along we got and contact info for the parties involved. Please contact me if you have time to work on this and are interested. I'm sure the Project Honeypot guys will be willing to pick this project back up and work on it. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution
I have a special requirement that insist an Asterisk server, 1.6.1.x,
is used.? I will have 2 SIP trunks coming into the server and I will
have to send calls to these SIP trunks with a round robin distribution
pattern.? I was thinking of using a group count function, if call
count is even send call to SIP Trunk 1, if call count is odd, send
call to SIP Trunk 2.
The decimal portion of ${UNIQUEID} is incremented every time Asterisk
creates a channel. Applying your even/odd logic to this should work
fine.
Thanks Steve, works great:
exten = _X.,1,Set(uniqueidcut=${CUT(CDR(uniqueid),.,2)})
exten = _X.,n,Set(result=${MATH(${uniqueidcut}%2)})
exten = _X.,n,GotoIf($[${result} 0 ]?siptrunk1,1:siptrunk2,1)
I don't have any empirical evidence, but I would suspect a variable
reference (${UNIQUEID}) would be insignificantly faster than invoking a
function that references a variable (CDR(uniqueid)).
Also, for the forseeable future, the Unix epoch will be 10 digits, so I
suspect specifying the character offset in the variable reference (:11*)
will be insignificantly faster than invoking a function (${CUT()}).
And, unless you have a specific need for the decimal portion of the
UNIQUEID, you could roll it all into a single conditional like:
gotoif($[${MATH(${UNIQUEID:11} % 2,int)} 0]?siptrunk1,1:siptrunk2,1)
*) Assuming you're not using the systemname prefix.
I believe you are correct, that would be more efficient.
After mocking this up, the results were not as expected. The even/odd
modulus worked fine using the ${UNIQUEID}, it actually worked too
well. The issue I ran into was each inbound call was consistently
even or odd so all calls went to the same outbound trunk. Each call
would initiate another SIP call out, so the counter would do exactly
what it is supposed to do and increment on each SIP channel. It seems
pretty obvious now that I think about it. So the call distribution to
the outbound trunks will not work based on the incrementing counter of
the ${UNIQUEID}.
After some thought, I decided to send all outbound calls through a
GROUP_COUNT function and distribute calls to the trunks based on
grater-than GottoIf statement like this:
[inbound]
exten = _X.,1,GotoIf($[${GROUP_COUNT(siptrunk1calls)}
${GROUP_COUNT(siptrunk2calls)}
]?siptrunk2,${EXTEN},1:siptrunk1,${EXTEN},1)
[siptrunk1]
exten = _X,1,Set(GROUP()=siptrunk1calls)
exten = _X,n,Dial(SIP/${ext...@siptrunk1,60,)
[siptrunk2]
exten = _X,1,Set(GROUP()=siptrunk2calls)
exten = _X,n,Dial(SIP/${ext...@siptrunk2,60,)
This worked as expected and is evenly distributing inbound calls to
both SIP trunks based on channel usage, with is ultimately desired.
Of course this is not exactly a round robin distribution but works for
what I need.
Thanks.
JR
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Engineering for the Masses
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Re: [asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution
I have a special requirement that insist an Asterisk server, 1.6.1.x, is
used.? I will have 2 SIP trunks coming into the server and I will have
to send calls to these SIP trunks with a round robin distribution
pattern.? I was thinking of using a group count function, if call count
is even send call to SIP Trunk 1, if call count is odd, send call to SIP
Trunk 2.
The decimal portion of ${UNIQUEID} is incremented every time Asterisk
creates a channel. Applying your even/odd logic to this should work fine.
Thanks Steve, works great:
exten = _X.,1,Set(uniqueidcut=${CUT(CDR(uniqueid),.,2)})
exten = _X.,n,Set(result=${MATH(${uniqueidcut}%2)})
exten = _X.,n,GotoIf($[${result} 0 ]?siptrunk1,1:siptrunk2,1)
Thanks.
JR
--
JR Richardson
Engineering for the Masses
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Asterisk send calls to SIP Trunks with Round Robin Call Distribution
Hi All, I know I can do this pretty easily with one of the SIP Proxy/Routers, I already do this using OpenSER as a load balancer. I have a special requirement that insist an Asterisk server, 1.6.1.x, is used. I will have 2 SIP trunks coming into the server and I will have to send calls to these SIP trunks with a round robin distribution pattern. I was thinking of using a group count function, if call count is even send call to SIP Trunk 1, if call count is odd, send call to SIP Trunk 2. Is there another mechanism I could use to accomplish such a need? I appreciate any ideas or guidance. Thanks. JR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dropped Calls
I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference is that one office uses a Digium TDM 8-port card and the other branches use 4-port Rhino cards with only 2 ports in use. What happens is that periodically we will be in a call and the call will just drop. It's usually within the first couple of minutes of the call. The calls can be either incoming or outgoing. The phenomenon affects both the Snoms and the Grandstreams. Along with the dropped call issue, we periodically have a problem where a person we call or a person that calls in cannot hear the person in the our office, but the person in our office can hear the remote person fine. All of the phones are on the same physical network as the asterisk server. There is no NAT, no Firewall, VLAN, etc. between the phones and the server. I have tried running sip debugs on the calls, but on the off chance that my logs catch either a drop or a one-way audio, the sip debug looks like just a normal call. Is there any setting that might cause both one-way audio and dropped calls? Thanks, Brent Davidson Join the club. I've experienced the same with various strains on 1.4.x above 1.4.21.1 (not an issue with this one that I have seen). This issue is truly random and debugging reveals nothing. I run an all SIP environment with same results. My solution was to downgrade to another version or switch to 1.2 or 1.6 depending on what features I need for the system. Sorry I couldn't be of any help, but I feel your frustration. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.1.12 with Grandstream HT502 T38 Fax
Hi All, I'm in the lab with Asterisk 1.6.1.12 and several ATA's testing T38. I hit a snag with the Grandstream HT502. It only seems to nail up a session at 9600bps. The Grandstream GXW4104 nails up consistently at 14400bps. I'm using the same equipment in the same configuration, just switching out the ATA. I have the latest firmware on each unit. Any ideas on what could cause this? The configuration is pretty simple so I don't think I'm missing anything there. I'm guessing there is a built in speed limit on the HT502? Thanks. JR -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license
Zoa wrote: On friday we finally released Attrafax under a GPL2 license. It comes with its own set of modems and built in transparent gatewaying. The solution should be quite stable as long as the line quality is ok. (Some tools for measuring the line quality are included in the release, as well as some fax2mail scripts). There is an example implementation included for Asterisk 1.4, if someone wants to porting it to the new fax backend or more recent asterisk versions and needs some help, let us know. I tested Attrafax this afternoon and was very pleased to see that it worked first time right out of the box. I tested the gateway function with the Asterisk source in the tarbal, Zaptel 1.4.10.1 and a Digium TE405P ver2 4-port T1 card. I really like the console output while processing faxes. Very impressive. Would anyone mind sharing any performance statistics based on real word usage or even high volume lab testing? I'm wondering how many concurrent T38 to PRI faxes could be handled with high end server hardware. Where are the bottlenecks for the software stack, RAM, PCI Bus, Proc Speed, Disc I/O? Would there be a problem running 3 to 4 PRI's full of T38 to SIP Faxes on one server? Could the Attrafax software handle that volume? Thanks in advanced for any feedback. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.x SIP allow incoming calls based on from ip address?
In 1.6, a SIP invite checks for valid [user] then TO: domain=abc.com,context, then if not present [general] context=incoming. This is fine and I think I understand the reasoning behind the new method. But we have lost the ability to route calls based on 'from' ip address. No, 1.6.x should work exactly the same as 1.4. The domains are in 1.4 too, and precedes the [general] section as you say. We still have users and peers. Hmm, well this is not happening like you say. I'm currently using 1.6.0 latest release and the 'host=ip address' is not being looked up. It works in 1.4 but not in this version. I'll do some more debugging and try to figure out what is going on. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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] 1.6.x SIP allow incoming calls based on from ip address?
Hi All, I read some discussions about the new SIP authentication methods for 1.6.X branches and possible addition of new type of user, type=trunk. I'm wondering about the disposition about this. Will it be added? In 1.2 and 1.4 branch, a SIP invite was first checked for a valid [user] then a valid host=ip, then if not present send call to [general] context=incoming. In 1.6, a SIP invite checks for valid [user] then TO: domain=abc.com,context, then if not present [general] context=incoming. This is fine and I think I understand the reasoning behind the new method. But we have lost the ability to route calls based on 'from' ip address. What I'm curious about is if we will be able to configure a sip [user] type=trunk with host=ip address based on FROM: IP Address and direct it to a specific context. sip.conf [provider_1_trunk] type=trunk host=ip address context=provider_1_incoming or something like this: [from ip address] type=trunk context=provider_1_incoming authentication=none Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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] Using SIPPEER status with CUT function?
Hi All,
I'm using Asterisk 1.4 branch and checking the status of some SIP
Peers with the functions ${SIPPEER(101:status)} and the result is OK
(48 ms). Seems to work fine.
Now I would like to use the function CUT to set a variable with the
'OK' portion of the status OK (48 ms) and then do some follow on
stuff if the status is OK.
I'm running into syntax errors in the Set command, I think due to the
spaces in the SIPPEER status.
Any suggestions on how to deal with the 'spaces' in the status?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
--
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Re: [asterisk-users] Using SIPPEER status with CUT function? SOLVED
On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson jmr.richard...@gmail.com wrote:
Hi All,
I'm using Asterisk 1.4 branch and checking the status of some SIP
Peers with the functions ${SIPPEER(101:status)} and the result is OK
(48 ms). Seems to work fine.
Now I would like to use the function CUT to set a variable with the
'OK' portion of the status OK (48 ms) and then do some follow on
stuff if the status is OK.
I'm running into syntax errors in the Set command, I think due to the
spaces in the SIPPEER status.
Any suggestions on how to deal with the 'spaces' in the status?
Disregard, I figured it out:
Set(stat102cut=${CUT(stat102, ,1)})
Just put a space in the deliminator field. I guess I could have tried
that first.
JR
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[asterisk-users] Asterisk 1.4.28 intermittent one way audio?
Hi All, I made the decision recently to update some old call servers from 1.2.x to the latest 1.4.28. I spent 2 weeks in the lab testing every production requirement for these voice servers. Nothing too special, SIP in, database lookup, SIP out, a couple of special applications, write CDR to billing server, nothing intensive at all. I also performed stress testing for call capacity, database performance and media handling. So after all the testing was finished, I was very confident this was the right thing to do. This was a plain jane asterisk install from stable release on debian etch, SIP only, no zaptel or dahdi. I shut down one of my 1.2 voice servers and turned up the new 1.4.28 server in it's place. Within a couple of hours I started getting calls from a couple of customers complaining about intermittent one-way audio issues, they could her the caller, but the caller could not hear them. This was only happening on inbound calls. I collected call examples for the next few days. I was able to trace almost all the calls back to the new 1.4.28 server. No other changes in the infrastructure other than putting in the 1.4.28 server. And the new server is built on the same hardware as the 1.2 servers are on. I could not find any relevant errors on any associated network element or upstream/downstream voice server, I could not find any errors on the 1.4.28 server either. No errors anywhere, nothing, nada, no retransmits, critical packets, rtp warnings, zilch. I pulled the 1.4.28 server out of call rotation, no more customer complaints. The call volume passing through the server is not a lot, 40 to 80 active calls at any time, and maybe 1 to 2 calls per second max, usually a call every 3 or 4 seconds. The one way audio issue occurred at both high and low call volume and was truly random. I captured 14 calls out of several thousand. I'm sure there was a lot more that that, but these were the only ones reported to me for investigation. So my question is, has anyone else experienced intermittent one way audio specific to Asterisk 1.4 that can be identified and resolved. Or maybe suggest another version of 1.4 that does not have an issue like this at these volumes? Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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] AGI perl script set timeout within script?
What about: 1) Fixing the slow responding DNS server? 2) Tweaking /etc/resolv.conf options? 3) Setting up a caching name server on your Asterisk host? 4) Adding the AGI server host name and IP address to /etc/hosts? 5) Using the IP address of the AGI server in your dialplan? Ok, I went with #4 for a bit, then resolved to #5 (pardon the pun), works fine. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- 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] AGI perl script set timeout within script?
Hi All, I'm running an AGI, calling a perl script the does number lookups to a remote server. I would like to put a timeout in the script. The problem I'm running into is if the DNS server is not responding, the script hangs and waits for 30 seconds before returning to the Asterisk dialplan. I would like a timeout of 1 second, then return. Here is my clean script: *** #!/usr/bin/perl $|=1; #Modules to Use ### use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI-ReadParse(); # Set variables according to supplied arguments $number = $ARGV[0]; $AGI-exec(agi,agi://agi.server.com/script.agi?user=usernamenumber=$number); *** Any assistance will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] AGI perl script set timeout within script?
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson
jmr.richard...@gmail.com
wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the Asterisk
dialplan. ?I would like a timeout of 1 second, then return.
On Thursday 07 January 2010 18:59:24 David Backeberg wrote:
* stop using DNS? Problem solved. * put nagios monitoring on your DNS
server? * put in a second DNS server, and tune your DNS timeout to a
very low value in /etc/resolv.conf (read the man page) before jumping
to next server?
Or you could use the Perl language feature, which is called 'alarm'.
Google around for some code samples.
On Thu, 7 Jan 2010, Tilghman Lesher wrote:
Ah, but Perl isn't actually doing the DNS lookup. If you examine his
script, he's merely passing back a name to the Asterisk process, which
is then calling inet_aton(), which is the reason why he cannot control
it from within the script. What he'd actually need to do is to start
using Net::DNS to do the resolution on that name, first, perhaps even
going as far as to connect to the server himself, and relay the channel
between the AGI interface and the remote TCP interface.
Then, he could use alarm() or the Time::Hires module to ensure his own
timeouts override the builtins. But as it stands now, it's all
Asterisk.
If the DNS lookup is being done by Asterisk to resolve the FastAGI server
name. If the DNS lookup is for the (assumed) database server in his script
then the suggestions to use alarm() would do the trick.
I guess we need clarification from the OP.
I tend to agree with Tilghman on this. I tried the perl script eval, alarm,
$SIG{ALRM} functions till I was blue in the face from cussing at the screen.
It does not appear that the perl script is doing the DNS query, otherwise
the eval alarm would timeout and pass control back to asterisk. Another
indication is that '#define MAX_AGI_CONNECT 2000' in res_agi is not being
invoked because the timeout is around 30 seconds. Is that 30 second timeout
built into Asterisk? Can I put an absolute timeout on an agi script from
the dialplan prior to calling the agi application? Maybe I'll fork a macro
with a timeout, yea, that's it, let start forking, something new to cuss at.
Thanks for your input guys.
JR
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Re: [asterisk-users] Realtime mysql extensions mutiple queries for each priority?
On Monday 28 December 2009 23:49:13 Tilghman Lesher wrote:
On Monday 28 December 2009 18:09:15 JR Richardson wrote:
I turned on console debug to see the actual mysql queries and to my
surprise and concern, I see every query for an extension priority
repeated 3 or more times prior to dialplan execution. For instance my
first dialplan activity is all extracted from the database:
context exten pri app appdata
dpdefault14 _991X 1 NoOp INBOUND CALL FROM SIPP
dpdefault14 _991X 2 NoOp TRUNK-${EXTEN:0:2} DID-${EXTEN:2}
dpdefault14 _991X 3 Set CALLERID(number)=600
dpdefault14 _991X 4 Answer
dpdefault14 _991X 5 Goto ${EXTEN:2}|1
Each priority is queried several times before executing. Here is a
sample of the first 2 priorities on a pastebin:
http://pastebin.com/m54c9c41e
I would not think this is normal activity as I can query the database
directly once and get a valid response. I don't have any realtime
mysql connections issues that I can see, no errors in the logs and
console status is:
No, that's normal. The order of queries done is 1) check if the extension
exists, 2) on spawn, retrieve the extension to populate information about
the application into the channel structure, and 3) actually execute the
application. There are 3 queries done for each extension actually executed
in order of priorities and a few more when the extension changes (or
originates). It's not optimal, but it's the way that it works.
At some point, a slight optimization could certainly be done to narrow this
down to a single query from the database, followed by a fairly short
caching period (1 second would be plenty), but that optimization has never
been done.
https://issues.asterisk.org/view.php?id=16521
Needs testing.
--
Tilghman Lesher
Tilghman,
Saying I’m a bit excited right now is an understatement. First of all,
the patch seems to work fine applied to 1.4.28 stable release. The
performance of this patch is extraordinary. Before migrating my static
dialplan to the database I could push 380 calls at 15 to 20 CPS. After
migrating to the database, I could only push a little more than 100
calls and no more than 6 to 9 CPS. With this patch applied, I am
pushing reliably 300 calls at 15 CPS. 7500+ calls without a hiccup.
Not quite as good as a static dialplan, but that is expected. MySQLd
has also decreased utilization, as expected, from 6 to 12, now 1 to 6.
This has got to be an overall performance increase by 50% or more.
I will be patching on my new 1.4 systems going forward. The sooner
this patch gets applied to Asterisk, the better.
Thanks.
JR
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[asterisk-users] Realtime mysql extensions mutiple queries for each priority?
Hi All,
I'm testing some realtime extension apps with Asterisk 1.4.28 and
addons 1.4.10 using res_mysql. Localhost database is 5.0.32 with
Debian Etch. The apps are working fine all syntax is proper, using
Set with (REALTIME) function, Set with (CUT) function, calling a Macro
with s extensions, and using a few pattern matching extensions as
well. I can certainly detail all database rows if needed, but it is
all pretty standard dialplan stuff, nothing too fancy. I built apps
in extensions.conf first to ensure they all work, then migrated them
to the database, they work there as well.
I turned on console debug to see the actual mysql queries and to my
surprise and concern, I see every query for an extension priority
repeated 3 or more times prior to dialplan execution. For instance my
first dialplan activity is all extracted from the database:
context exten pri app appdata
dpdefault14 _991X 1 NoOpINBOUND CALL FROM SIPP
dpdefault14 _991X 2 NoOpTRUNK-${EXTEN:0:2} DID-${EXTEN:2}
dpdefault14 _991X 3 Set CALLERID(number)=600
dpdefault14 _991X 4 Answer
dpdefault14 _991X 5 Goto${EXTEN:2}|1
Each priority is queried several times before executing. Here is a
sample of the first 2 priorities on a pastebin:
http://pastebin.com/m54c9c41e
I would not think this is normal activity as I can query the database
directly once and get a valid response. I don't have any realtime
mysql connections issues that I can see, no errors in the logs and
console status is:
test1-6*CLI realtime mysql status
Connected to astcl...@127.0.0.1, port 3306 with username root for 1
hours, 7 minutes, 28 seconds.
[Dec 28 18:06:50] DEBUG[8664]: res_config_mysql.c:657 mysql_reconnect:
MySQL RealTime: Everything is fine.
test1-6*CLI
Any guidance on trouble shooting this will be appreciated.
Thanks.
JR
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Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9
Cyprus VoIP wrote: This is the reINVITE SDP received from the SIP Proxy: --- Content-Type: application/sdp Content-Length: 353 v=0 o=root 30427 30428 IN IP4 194.98.xxx.xxx s=session c=IN IP4 194.98.xxx.xxx t=0 0 m=image 17548 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:72 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- This is probably originating from a Cisco gateway. Cisco gateways generate T.38 SDPs that do not conform to the T.38 recommendation in one very obvious (and painful) way: they tell us that they can only accept 72 byte packets (T38FaxMaxDatagram), when in fact they can accept packets much larger than that. When you notice that they are also requesting that we use t38UDPRedundancy for error correction, that means that the maximum IFP (single FAX protocol packet) we can include in a UDPTL datagram is around 30 bytes, since we'd need to have room for two of them and a bit of overhead. 30 bytes is a ridiculously small limit for IFPs, and does not allow successful FAXing at any possible bit rate (except for 2400 bits per second using 10 millisecond IFPs, but no FAX stack would do that). I was having similar issues, trying Asterisk 1.6.1.12-rc1 resolved it. http://www.mail-archive.com/asterisk-users@lists.digium.com/msg234015.html Good luck. JR -- JR Richardson Engineering for the Masses ___ -- 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] T38 Passthrough 1.6.1.12-rc1 Good Results
Hi All, I've been knee deep in T38 faxing for a couple of weeks now, trying to find a version of Asterisk that would pass through T38 with an Audiocodes Mediant 1000 and MP203 ATA. I had problems with 1.6.0.x through 1.6.1.10. Tested 6 different versions. Either it just would not work or fail back to G.711, or re-invite with wrong T38FaxMaxDatagram sizes, faxes would work one-way and not the other, and so on, issue after issue. After trading some emails with the dev list, I learned a bit more about what a good T38 negotiation should look like which helped out quite a bit. I started reading the changelog for the various versions of 1.6.0.x and 1.6.1.x, focusing on chan_sip updates and fixes. There have been a lot of updates regarding T38, tweaks, patches, adding functionality, and there was a total re-write of the stack as well. All this within the past few months. Just two days ago 1.6.1.12-rc1 was uploaded, changelog noted a handful of more T38 changes. A particularly interesting one for me was: 2009-11-30 21:55 + [r231694] Kevin P. Fleming kpflem...@digium.com I was getting T38FaxMaxDatagram size miss-matches in the T38 negotiation which was causing failures, IFP byte miss-match and buffer overflow errors. This update has resolved these particular issues and with my specific lab testing, T38 faxing is negotiating faster and completing quicker. So just to check myself, I defaulted my lab setup and rebuilt with just basic configs on the Mediant 1000, the MP203 (behind a NAT) and Asterisk, sent several faxes coming and going with no errors, all T38 negotiated faxes nailed up at 14400. I'll be doing a lot more testing next week, but I'm very happy with the results so far. I wanted to share so you all were aware of the progress that is being made in this particular area and also thank the dev team for responding to the bug tracker, taking suggestions for improvements and doing the coding to make Asterisk the best it can be. I can't wait for T38 gateway. Keep up the good work. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] SIP Change canreinvite=yes/no from dialplan?
Hi All,
Currently I have voice calls from a certain SIP peer coming into an asterisk
server where the specific [SIP] channel is set to 'canreinvite=no'.
I would like to enable reinvites for certain calls, matched on DID. So I'm
wondering if there is a mechanism in the dial plan to turn on/off reinvite
capability or will every call on this channel be forced to use the SIP peer
context for the duration of the call? Is there maybe a new feature in 1.6
that does this?
exten = 5551212,1,Set(canreinvite=yes)
exten = 5551212,2,Dial(SIP/${ext...@othersippeerSIP/$%7bexten...@othersippeer
,,)
Something like that.
Thanks.
JR
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[asterisk-users] Asterisk SIP to Cisco IAD2430 Series?
Hi All, I tried doing SIP from Asterisk to the old Cisco 2400 series IAD's but could not get signalling reliable or DTMF either. I do SIP to Cisco routers with PRI cards in quite a bit, but I guess the old IAD SIP stack is not a robust at the router sip stack so I just could not get it working. I'm wondering if anyone has tried the new Cisco 2430 series IAD's and have been successful and reliable, care to share your experience and sample configs? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] Looking for the asterisk 'off' sound file
Hi All, In the Asterisk extra sound file, there is listed in the text file an 'On' and an 'Off' sound prompt but I only have the 'On' file. I have searched through various versions of 1.6, 1.4 and 1.2, and can not find the file. Does anyone have this prompt and can send it to me, gsm or ulaw? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] strange cisco nat issue
Hi All, I'm running into a reoccurring nat issue, phones (Linksys, Polycom or Cisco) behind a cisco router, 2600 series, ios 12.4. Sometimes 2 or more phones will have the same outside port number and will register to Asterisk with the same port number. As far as I can tell this is only occurring with the SIP port, not RTP ports. To resolve, I have to go into the router and clear the translations for one if the phones, restart the phone and it nails up a new nat session on a different outside port in the router. This seems to happen more often after some power failure on site, when many phones reboot at the same time. This has happened various times at different customers, large and small, different phones, different Asterisk servers. The only thing common in all cases is the cisco routers. We have tried different IOS releases as well. So I'm pretty certain this is an Cisco IOS bug or maybe a router overload? I've searched for a cisco nat bug with no luck. So my question is, has anyone else experienced this type of issue and if so, is there a solution to resolve? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DUNDi + SIP Realtime
Good afternoon gentlemen (and ladies).
A costumer of mine has many servers and each one maps their SIP extensions
to the others via DUNDi. It works like a charm. SIP extensions can only
register at one server, the one they belong to. In case one extension
wants to call other that is registered in another server, DUNDi takes care
of that by calling the other server using IAX2 and G.729 codec. It's
important to clarify that each server today works as a completely
independent PBX, talking to each other using IAX2 that is routed via a
MPLS network. Also, the servers are very distant from each other
physically.
Today the extensions are being mapped in DUNDi like this:
[dundi-internal]
exten = _70XX,1,Noop()
And all extensions configurations are done in sip.conf. No realtime is
being used, yet.
Now the customer wants to take a step further and make it possible that
*every* SIP extension could be able to register in *every* server. That
would make possible for them to use DNS to automatically find the
closest PBX and make the extension register on that one.
So far I considered the following for this project:
- Moving all SIP extensions from individual sip.confs to one MySQL
database, and point all servers to that one
- Configure sip.conf on each machine like this:
regcontext=dundi-internal
rtcachefriends=yes
rtsavesysname=yes
rtupdate=no
rtautoclear=yes
ignoreregexpire=no
That way each time an extension registers, Asterisk would add an extension
to the dundi-internal context, which as you guessed, is the one being
mapped to the other servers. So instead of mapping extensions using
wildcards, the extensions will be mapped individually.
extensions.conf would be something like this:
[internal]
;Tries to make the call using SIP, in the case
;the extension is registered in this server
;If it's not, switches to DUNDi
exten = _,1,Dial(SIP/${EXTEN},60)
exten = _,n,NoOp(DIALSTATUS = ${DIALSTATUS})
exten = _,n,NoOp(FROM_DUNDI = ${FROM_DUNDI})
exten = _,n,GotoIf($[${FROM_DUNDI} = 1]?end:start)
exten = _,n(start),Answer()
exten = _,n,Playback(vm-dialout)
exten = _,n,Goto(dundi-internal-helper,${EXTEN},1)
exten = _,n(end),Noop(Loop detected. Hanging up.)
exten = _,n,Hangup()
[dundi-internal-helper]
switch = DUNDi/dundi_internal
[from-dundi]
exten = _,1,Set(FROM_DUNDI=1)
exten = _,n,Dial(SIP/${EXTEN},60)
So far it's working fine in a test lab with 2 servers running Asterisk
1.6.0.15.
For the gurus out there: is there something that I'm doing terribly wrong,
that would break everything and make the universe collapse into itself
when I apply the same principle on production?
I'll be happy to provide more details in case there are any doubts. I
really appreciate your feedback, no matter what is it. :)
Vin?cius Fontes
www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e
telefonia IP
[JR Richardson]
I used to do this exact thing a few years ago, wrote a couple of papers
about it. Realtime + DINDi works great for this, I would add in MySQL
replication to the mix so each server writes the SIP cache info to a Master
database that is replicated out to all the servers. Each server will have a
copy of the same database and be able to contact the phones if DUNDi queries
become unavailable.
The tricky problem you may run into, if you haven't figured it out yet, is
what to do about voicemail and where the storage will be, distributed
voicemail will be problematic in a dynamic sip ua registration environment
across multiple servers. Centralize voicemail using DUNDi can help this out
as well.
I'll send you some papers off line
Hope this helps.
JR
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[asterisk-users] Using the PBX Directory from a Blackberry
Hi All, A couple of customers called complaining that folks were dialing into their PBX trying to use the Directory to locate users, from a Blackberry, and getting frustrated due to the incompatibility of dialing alpha characters on the the qwerty keyboard and not getting through. The issue of course is the Directory application only recognizes numeric digit tones, not alpha characters (not sure is there is actually tones generated when the alpha characters are pressed, it just doesn't work). Anyhow, on the Blackberry, when you hold down the Alt key and press the alpha character, the device sends out the correct digit tone associated with that character, like on a regular phone keypad. That is how folks can use a Blackberry effectively with the PBX Directory application. Hope this helps. JR -- JR Richardson Engineering for the Masses ___ -- 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] MeetMe and setting conference timeout
I wish to make in some way a timeout mechanism that after X amount of time, it will disconnect the users and kick them out of the conference. How can I do such thing ? This is from my realtime extensions, database formatted. Set the TIMEOUT(absolute)=value to whatever you like in seconds. This will kick this user out after 2 hours. incomingconf136 1 Answer incomingconf136 2 Set TIMEOUT(absolute)=7200 incomingconf136 3 Wait1 incomingconf136 4 MeetMe 136|Mcp| incomingconf136 5 Playbackvm-goodbye incomingconf136 6 Hangup JR -- JR Richardson Engineering for the Masses ___ -- 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 SIP trunk to Cisco IAD2400
Hi All, Does anyone have a config example for setting up SIP trunking to a CIsco IAD2400 and are willing to share? I've done SIP trunking to Cisco 2600's with PRI's but not to the POTS lines on the IAD's, I'm wondering if that is possible and how to specify the DID on the POTS line config for the IAD. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] Strange voicemail problem when call forwarding off local PBX
Hi All, I just experienced a weird issue and though I'd share. I have a pretty standard business PBX setup for a business customer, local extensions, Linksys phones, call comes in and rings local extension exten = 101,1,Dial(SIP/101,20,tr) the physical phone has call forward enabled to the users home, Time Warner residential line service. Intermittently all seems to work except when the home line voicemail picks up. You hear the users home voicemail greeting, but sporadically, instead of hearing a beep after the greeting is finished, the local PBX initiates a transfer. Sometimes you do hear the beep from the residential voicemail and can leave a voicemail. So after a few minutes of pondering the possibilities, I came to the conclusion that the residential voicemail service 'beep' was triggering the local PBX transfer function, but not all the time. I have not come across this before and I have a lot of customers that transfer calls out to cell phones and home lines, other voicemail services. The simple solutions was to disable 't' option in the dial string when calling the local PBX extension 101. The customer uses the phone txfer softkey anyway so no disruption of local function when the user is in the office. Hope this helps. JR -- JR Richardson Engineering for the Masses ___ -- 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] Realtime dialplan application versus REALTIME dialplan function
Hi All,
I'm upgrading some PBX's from 1.2 to 1.4 and having a bit of trouble with
converting the Realtime application to the REALTIME function. I have the
method down and understand simplistically what is going on, at least enough
to get my old 1.2 apps to run in 1.4 functions. I do not understand why
change from the app to the func? What the benefits?
To me, the app seemed so elegant with appending a variable name to each
extracted data field within a row. Really convenient and easy, one priority
in the dialplan and all data is extracted and easily used further down the
line.
Initial take on the function is increased priorities in the dialplan to
extract the data then cut it up into specified variable, then cut the
resulting variable into further bits to get the usable data into another
variable so it can be used for the real work.
For example here is the 1.2 dialplan:
exten = 326,1,Realtime(cfwd|exten|${EXTEN}|cf_)
exten = 326,n,GotoIf($[${cf_active} = yes ]?:326|20)
exten = 326,n,Goto(cfaccess,${cf_cfnum},1)
Here is the 1.4 dialplan to accomplish the same thing:
exten = 326,1,Set(row=${REALTIME(cfwd,exten,${EXTEN})})
exten = 326,n,Set(column3=${CUT(row,|,3)})
exten = 326,n,Set(column4=${CUT(row,|,4)})
exten = 326,n,Set(cf_cfnum=${CUT(column3,=,2)})
exten = 326,n,Set(cf_active=${CUT(column4,=,2)})
exten = 326,n,GotoIf($[${cf_active} = no ]?:326|20)
exten = 326,n,Goto(cfaccess,${cf_cfnum},1)
So I have in mind that maybe the function is a bit more versatile than the
old app, but I don't really see it just yet.
Can anyone shed some light on this and expound on the benefits or reasoning
behind the switch in the application usage?
Thanks.
JR
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Re: [asterisk-users] Open Source in an Economic Downturn: Asterisk stories
I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on Sunday in Los Angeles, and the topic of my talk is Open Source in an Economic Downturn. I've got lots of talking points for this talk, but it would be interesting to hear some short anecdotes about how you in the Asterisk community are thriving, or at least surviving, by virtue of the benefits of Open Source. I find that real-world examples are worth more than all of the bullet points in the world, and timely stories from the community would be more interesting than hearing me prattle on. I'm a Texas based service provider, VoIP and Internet for business customers. The last 2 month are actually picking up a bit. We are finding that business folks are really wanting less expensive alternatives for voice and data services, not so much for the new VoIP technology we offer. A year ago, we really had to effectively sell the new technological advantages and promote business enhancing solutions based on voice and data convergence. Now it is all about the bottom line cost. I'm not real concerned for the reason businesses are buying our service, but I'm glad they are. Competitively speaking, we are seeing the LEC (ATT) and other CLEC providers dropping their prices to capture market share. They are getting real aggressive, but it make a good statement about why their prices where high for the past few years. The technology and how it is delivered has not changed much over the past year so the Economic Downturn has affected them enough to reposition their margin strategies. JR -- JR Richardson Engineering for the Masses ___ -- 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 1.4.21.1 intermittent presence working with Polycom
Hi All, I upgraded a PBX from 1.2. to 1.4.21.1 and I'm noticing that the hints for SIP channels are not updating the phones 100% of the time. The hints seem to work for some time, then the notification on the phone will hang in either and on or off state. During this condition, on the PBX, core show hints, indicates the correct presence state for the SIP channel. Also if multiple phones are monitoring the same SIP channel, the presence notification on some phones still work fine, but may hang on one or two phones. We have to reboot the phone for the presence to start working again. We are using the same firmware on the phone that worked fine with the Asterisk 1.2 code, Polycom 650 with 2.1.1. So I'm guessing there is something particular with this version of Asterisk. Any guidance will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] cdr_addon_mysql 'Failed to insert into database' stops * call processing
Hi All, I have some Asterisk 1.2 servers using the cdr_mysql addon (1.2.3) spitting cdr's over to a MySQL database on another server. All is working well except for a strange problem I ran into this morning. During some cdr database maintenance, the cdr table was locked for a few minutes, during this condition all the Asterisk servers stopped processing calls and reported this error: Jan 5 09:57:03 ERROR[938]: cdr_addon_mysql.c:226 mysql_log: mysql_cdr: Failed to insert into database: (1205) Lock wait timeout exceeded; try restarting transaction Jan 5 09:57:54 ERROR[970]: cdr_addon_mysql.c:226 mysql_log: mysql_cdr: Failed to insert into database: (1205) Lock wait timeout exceeded; try restarting transaction Jan 5 09:58:45 ERROR[515]: cdr_addon_mysql.c:226 mysql_log: mysql_cdr: Failed to insert into database: (1205) Lock wait timeout exceeded; try restarting transaction Is there possibly a patch to addons that would relieve this issue? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] Asternic Call Center and Asterisk 1.4 Queues
Hi All, I'm testing the Asterinic Call Center Queue Log Analizer. Working ok except for realtime monitoring. The page updates queue summary and calls waiting, but not Agent status. When an agent is (busy) in [asterisk queue show], the 'state' of the agent in agent status on the web page does not change, always shows 'not in use'. The page does update with 'Last In Call' info after hangup of a call. Any ideas? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] ztdummy: rtc: lost some interrupts at 1024Hz
I'm getting crazy about ztdummy. I have to replicate a PBX where ztdummy is working fine but for some reason I cannot. The two machines have the same kernel, motherboard, the same gcc version and the same zaptel 1.4.8. On the second machine zaptel compiles without errors and ztdummy.ko is generated but when I modprobe it I get the following error in messages: rtc: lost some interrupts at 1024Hz. Since modules sizes are slighly different I copied the working ztdummy on the new machine so the files are the same.nothing changes!! Your mother boards are probably not 100% the same, maybe a chipset is newer and causing an interrupt problem. I've seen this before. Put 'acpi=off' in your kernel boot parameter line in the grub menu.lst like this: title Debian GNU/Linux, kernel 2.6.18-686 root(hd0,0) kernel /boot/vmlinuz-2.6.18--686 root=/dev/sda1 ro acpi=off initrd /boot/initrd.img-2.6.18-686 savedefault Reboot, and that should do it. JR -- JR Richardson Engineering for the Masses ___ -- 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] What syntax to send user:pass in SIP Dial string?
Hi All, I'm trying to get the user:pass embedded in a SIP Dial string instead of calling a SIPuser in sip.conf: Regular way, exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]|30|) Where the 'sipuser' is a context on sip.conf [sipuser] fromuser=sipuser What I would like to do is embed the username:password in the Dial string, something like this: exten = 1234,1,Dial(SIP/[EMAIL PROTECTED]:[EMAIL PROTECTED]|30|) doesn't work though, can't create sip channel. I'm not sure if this can be done? Any guidance will be appreciated. JR -- - JR Richardson Engineering for the Masses ___ -- 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 / t38 gateway
The folks that devloped the fax V.protocols took into acount typical copper problems like noise or echo. But what they never conceived of as even being possible is that a call might shift around in the time domain. Thanks to jitter/latency, the delay time of a call can change in the middle of the call. That isn't possible with copper technologies. This makes faxing over even G.711 a dice roll. IMO, with a sufficiently large buffer and a rock-solid quartz clocking system that goes way beyond what is typically seen, it might be theoretically possible to send a fax over VOIP. Hi All, This is a good discussion. I can support most of the findings here as I have recently spent a lot of time in the lab with T38 equipment from several vendors. Interoperability is a toss up, some ATA's only work with the parent vendor gateways, some gateways are more forgiving and work with Asterisk, some ATA's work straight out of the box, others require a PHD to configure, etc. What I ended up with for a rock solid ON-NET T38 gateway to T38 fax ATA is the Audiocodes Mediant 1000 with Audiocodes ATA's (MPXXX). I emphasis on-net because I control my environment end-to-end from PSTN-Data Center-T1's-Customer-QOS LAN. I was able to reliably push 100's of faxes, multi page, single page, high density, high resolution, natted ATA's, various scenarios successful. I was very excited about T38 and was convinced this was the solution I would build on. So I took one of the ATA's home to test over the Internet (great connection, low hops to my datacenter, low latency, low jitter, lots of bandwidth) and was very disappointed when I could not get more than a 3 page fax through without errors. I started getting protocol and various page errors. I tweaked every T38 parameter that Audiocodes had with zero improvement. So I have to say, my confidence in T38 is very low, at least where open Internet connections are being used. I'm now going to look at some other technology, fax over HTTPS. I will be testing the FaxBack products to see how they stack up. JR - JR Richardson Engineering for the Masses ___ -- 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] How to add contexts in asterisk realtime?
hi for any context ,you must to open /etc/asterisk/extensions.conf and insert this line : exten =Realtime/[EMAIL PROTECTED] and (reload) or (restart now) your asterisk You don't have to restart asterisk, just a 'dialplan reload' will suffice. So really there is no impact to a running system. JR -- - JR Richardson Engineering for the Masses ___ -- 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 T38 and Dialogic DMG 2000
Hi All, I have Asterisk 1.4.21.1 and Dialogic DMG 2000 firmware 6.0.103. Trying to pass t38 fax calls setup like this: fax machineATA (linksys and mediatrix)SIPAsteriskSIPDMG GWPRI From the fax outbound through the PRI works great, seems to be reliable, 40+ faxes with 5 different fax machines tested, plus an efax service. The problem is the other way, PRI inbound to fax, the call doesn't setup between the GW and Asterisk. I have the issued narrowed down to the SIP messaging between the DMG and Asterisk. The DMG invite sends to asterisk: m=audio 49016 RTP/AVP 0 101 [notice the m=audio] a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=image 0 udptl t38 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy And Asterisk responds with the 200 OK: m=image 29475 udptl t38 [notice the m=image] a=T38FaxVersion:0 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPRedundancy I've been corresponding with Dialogic engineering on the messaging and they report that the gateway receiving m=image is not compatible or is telling the gw to immediately setup the call at T38 with is not compatible. The gateway wants to setup the call as audio first, hear the CNG tones and then re-invite to t38. So my question: Is there a way for configuring Asterisk to respond with m=audio instead of m=image? If I disable udptl in Asterisk, call setup fine with audio. Thanks. JR -- - JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need application, CID number match list to call cell phone
Is this a one VIP to one cell number match? Or is it on VIP to multiple cells? On Tue, Aug 26, 2008 at 7:28 PM, JR Richardson [EMAIL PROTECTED] wrote: Hi All, I received a request for a special application and need some guidance. Cust has there own Asterisk PBX with SIP phones, pretty standard setup. They want an after hours application that checks inbound caller ID numbers and matches them to a list, say 5 to 10 numbers of special VIP customers, if there is a match on the list, then forward the call straight to a cell phone, instead of ringing local extension and then to voicemail. The customer also wants to be able to manage this VIP list and the call forward cell phone number themselves, so it needs to be configured, numbers added and deleted, through a web page on the PBX. So I'm thinking I need a dialplan app that has to interface with a MySQL database that holds the list of numbers, so I can build a webpage to add/delete the numbers. [JR Richardson] The info I have is one cell phone, like an on-call cell that gets passed around to on-call individuals. But being able to change this number to a different cell from time to time is required. Thanks. JR ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need application, CID number match list to call cell phone
Hi All, I received a request for a special application and need some guidance. Cust has there own Asterisk PBX with SIP phones, pretty standard setup. They want an after hours application that checks inbound caller ID numbers and matches them to a list, say 5 to 10 numbers of special VIP customers, if there is a match on the list, then forward the call straight to a cell phone, instead of ringing local extension and then to voicemail. The customer also wants to be able to manage this VIP list and the call forward cell phone number themselves, so it needs to be configured, numbers added and deleted, through a web page on the PBX. So I'm thinking I need a dialplan app that has to interface with a MySQL database that holds the list of numbers, so I can build a webpage to add/delete the numbers. Any ideas would be much appreciated. Thanks. JR - JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 T38 UDPTL Pass Through MAX TNT and Linksys 2102
Hi All, I finally got the time to test t38 pass through with a TNT, * 1.4.21.1 and Linksys 2102: PRI TNT SIP Asterisk 2102 SharpFax Faxing either direction, the call sets up with ulaw rtp, when fax tones hit the line, both the TNT and the 2102 switch to t38 and udptl packets fly through Asterisk. All looks good, but, once udptl sets up, every few seconds, I get a warning: 'rtp Read too short' on the Asterisk CLI from the TNT side of the session. Faxes never complete, not even a half page, nothing, transmission just ends. There are only a few parameters on the TNT that effect t38 and I've adjusted them all with no change in the results. Pretty much the same results when testing t38 pass through to a Cisco pri gateway as well. So my question is: Does anyone else have this solution working and wouldn't not mind sharing configs? Thanks. JR -- - JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Intermittent T.38 pass through
Hi All, I've been testing reliability with t.38 faxing pass through with * 1.4.21.1, Linksys ATA's 2102 x 2, Sharp UX-B800SE and Cannon ImageClass D880. cannon 2102 #1 SIP * SIP 2102 #2 sharp Started out with default settings on all devices, configured Asterisk to handle T.38 pass through, the configuration I believe is solid. I get relaiable results faxing from the sharp to the cannon. I get intermittent results, 50% success or failure, faxing from cannon to sharp. The first thing I did was to switch the faxes, so the ATA's and Asterisk remain the same, just switched the faxes to the oposite end of the path. Doing this comfirmed the original results, sharp to cannon is reliable, cannon to sharp is unreliable. What I observe faxing from sharp to cannon: path sets up as ulaw RTP, cannon answers, RTP switched to UDPTL, fax completes Faxing from cannon to sharp: path sets up as ulaw RTP, sharp answers, RTP switches to UDPTL only half the time when UDPTL is active, fax completes, when the path stays with RTP, fax always fails If I understand correctly, Asterisk switches the media stream to UDPTL when it hears valid fax tones on each side of the path, if it only detects fax tones on 1 path leg, then it keeps the media path through RTP. Or is the mechanism switching to UDPTL in the SIP headers? So, I adjusted db levels on the FXS ports, higher and lower, no effect. I increased jitter, reduced jitter, disabled jitter, no effect. Ensured echo can's were off, no effect. Manually set faxes to 14.4bps, ecm off, no effect. Even switched telephone cord, no effect. On these Linksys 2102's, you can predial #99 to force the ATA to enable fax t.38, this works and is reliable, no RTP is setup, just UDPTL. So my question is this: Can I setup Asterisk to only allow t.38 pass through from these ATA's, without the need to use the #99 in every dial string from the fax machine? Thanks. JR -- - JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX t.38 on Asterisk 1.6?
Asterisk 1.6 currently has T.38 origination and termination support. It does not yet have fax gateway support. -- Russell Bryant Russell, Can you please clarify what you mean. I think there is still a bit of confusion as to what termination and gateway and Asterisk 1.6 is all about, capability, functionality, call flow to what application, library requirements, spandsp versioning. And when do you think we can expect to see stable solutions for each. Thanks. JR ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple asterisk approach
I'm not sure if this is the proper way to approach it but i can't figure out
how to setup dundi.
what i did is, i try to determine which server a user is registered, by
calling an agi to query? the realtime db and capture the regserver of? the
user.
e.g.?
exten = _1xx,1,AGI(getserver.php)
exten = _1xx,2,GotoIf($[${REGSERVER} != asterisk-1]?102)
exten = _1xx,3,Dial(SIP/${EXTEN}|30|t)
exten = _1xx,102,Dial(SIP/[EMAIL PROTECTED]|30|t)
exten = _1xx,103,Hangup
then i created peering between the two. so far it is working i can call
extensions that are registered in whatever server. but what i'd like to know
is, would there be a difference on performance on calls when querying a DB to
get the regserver, or is it still adviseable to use dundi for peering.
also i setup DNS SRV for these servers, what if one server fails, should the
user close their phone to re-register to the server that is alive, or will it
automtically register to the other server if the other is unreachable? TIA
Regards
Ron
Use DUNDi, perfect for this. The protocol is very light, no load on
the servers to run it, can handle hundreds of queries a second with no
load. You want to use regcontext and a few other things to make it
all work together. Here are some papers to guide you:
ftp://208.81.55.228/DUNDi_So_Easy.pdf
ftp://208.81.55.228/Using_DUNDi_with_a_Cluster_of_Asterisk_Servers.pdf
Good Luck.
JR
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Engineering for the Masses
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[asterisk-users] Asterisk Openfire Asterisk-IM Plugin Performance Observation
Hi All, I've been playing with Openfire Asterisk-IM plugin installed on the same server with Asterisk 1.4 with MySQL as the Openfire database. Using Spark IM as the client on user machines. It seems to work fairly well, not too bad to install. This first thing I notice is all the packages that must be installed to get Openfire running, java-jre, mysql (needed for asterisk-im to work) and many other dependencies. So now the PBX is over 1.2 Gig for the installation. Typical PBX installs are under 600 Meg. This makes me wonder about server stability, reliability and performance as uptime creeps on and user count increases over 50 to 100+. Can anyone give me feedback on real world experience with this type of setup and any performance issues that my arise? Is it better for production to run Openfire on a separate server than the PBX? My biggest concern is deploying a 100+ user environment with high call volume and high chat volume. Java seems to be a bit resource hungry with the user notifications and call pop ups. I would hate to have the IM server walking over Asterisk and affecting call quality or PBX stability. Thanks. JR - JR Richardson Engineering for the Masses ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911 via MAX TNT
When I send a call out the MAX I get the following -- Got SIP response 484 Address Incomplete back from 172.16.10.230 Any ideas on how to make 911 appear as a ten digit number to the device so that it will pass the number out to the PSTN ? This is not a max tnt problem, the tnt will pass anything you send to it, 911/411/7 digit/10digit/011 international, the question is, does your PSTN provider accept 911 call on the trunk your passing the call to? JR ___ -- 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] asterisk-addons 1.6.0 Command 'realtime mysql status'
Thought I'd post this, see if anyone has experienced this...I am using Asterisk 1.6 from the SVN branch and Asterisk-addons 1.6 from the SVN branch. Mysql is running and I've connected using the information that is in res_mysql.conf but when I try to check the realtime status I get the following: Connected to Asterisk SVN-branch-1.6.0-r117183 currently running on ubuntu (pid = 31371) Verbosity is at least 3 -- Remote UNIX connection ubuntu*CLI module reload res_config_mysql.so -- Reloading module 'res_config_mysql.so' (MySQL RealTime Configuration Driver) == Parsing '/etc/asterisk/res_mysql.conf': == Found == MySQL RealTime reloaded. ubuntu*CLI realtime mysql status Command 'realtime mysql status' failed. Here's a sanitized version of res_mysql.conf [general] dbhost = localhost dbport = 3306 dbname = **db** dbuser = **user** dbpass = **pass** dbsock = /var/lib/mysql/mysql.sock Any thoughts? I'm currently testing this as well. In the res_mysql.conf, use dbhost = 127.0.0.1 instead of 'localhost'. Also try to connect to the mysql database with the user/pass at the command line just to ensure mysql is accepting connections for that user on the local machine. # mysql -u **user** -p In /etc/mysql/my.cnf ensure: bind-address = 0.0.0.0 or bind-address = 127.0.0.1 My test is connecting fine to local and remote databases, I'm use Asterisk 1.6-current and addon-1.6-current from digium ftp, not trunk. Hope this helps. JR -- JR Richardson Engineering for the Masses ___ -- 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] T.38 w/ MAX TNT ASTERISK
We solved our former echo issues... however, as luck would have it.. Faxing is yet a completely different animal. We understand that digium doesn't really support faxing with asterisk... HOWEVER...it seems that ATA manufacturers and the MAX TNT indicate that fax is supported Scenario: TDM/PRIs --MAX TNT -sip- asterisk -sip- ata -fax We should note that the network between the ata and the facility with the max and asterisk is a managed fiber network in a type of campus environment... 1-2ms latency end to end tops... If the ata is reinviting to the MAX TNT shouldn't fax work with T.38... Does anyone have any experience with this configuration ? Thanks, I have been wanting to do this for months, but just can't find the time to work on it. If you do get it going, I would really appriciate knowing how. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] addons-1.6 not seeing installed MySQL packages
Hi All, I'm poking around with 1.6, tried to compile the addon package, but it doesn't see mysql_config installed. I have mysql-client, mysql-common and mysql-server installed. I'm running debian etch. Any suggestions? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] addons-1.6 not seeing installed MySQL packages
I'm poking around with 1.6, tried to compile the addon package, but it doesn't see mysql_config installed. I have mysql-client, mysql-common and mysql-server installed. I'm running debian etch. Any suggestions? I reverted to debian sarge and the addon package did find mysql installed. So I'm guessing debian etch is putting mysql_client in some other place that /usr/sbin/. What I did notice is the addon sample config file for res_mysql.conf doesn't specify how to setup the read/write entries, clarification on that would help also. JR -- JR Richardson Engineering for the Masses ___ -- 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] T38 Passthrough Verification
JR Richardson wrote: I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet ! sip show channels shows the call setup with ulaw. Try setting canreinvite=no for the peer doing T.38. It looks like the code in Asterisk 1.4 will not allow re-invites for an established T.38 passthrough call. I saw a post about the re-invites, so I tried it both ways, canreinvite=yes/no with the same results. Thanks. JR --- JR Richardson Engineering for the Masses ___ -- 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] Lucent Max TNT PRI Agg -- * -- SIP DEV (PHONE or ATA)
Hello... We're attempting to track down an intermittent echo issue. Our setup is phonesipasterisksiptntpri to carriers. We have less than 2 ms latency on the networks (FTTx), totally SIP w/ G711u. The party hearing the echo is the subscriber using sip. The PSTN users does not hear the echo. We should be note that there is zero echo when calling sip to sip with or without reinvites enabled. We have several different phones; linksys, polycom, grandstream (both atas and phones). It's difficult to reproduce the problem regularly so isolation is an issue. I had intermittent echo when I first deployed TNT's as well. It took a while to track down. Adjust the volume on the TNT lower until the echo goes away. Here is what I had to set mine to: In each T1 config: set line-interface voip-gain-control input-pad = 3db-loss set line-interface voip-gain-control output-pad = 3db-loss Hope this helps. JR -- JR Richardson Engineering for the Masses ___ -- 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] T38 Passthrough Verification
Hi All, I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet ! sip show channels shows the call setup with ulaw. Any guidance will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] Anyone have a method of keeping an incremental tally of calls?
Hi All, I thought I read a post a while back of a system call or something in the dialplan whereby a call count can be incremented and spit out to a text file. Not like a group count of active channels. What I would like to accomplish is have an incremental count of a specific dialplan routine that gets called, so after a week or month, I can see how many times a specific dilaplan action has been used. Thanks for any advice. JR -- JR Richardson Engineering for the Masses ___ -- 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] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW
The 601 is powered by PoE with 2 sidecars, so Polycom wants us to put an actual Power Supply on the phone - thinking the voltage is dropping and causing the reboot. I don't buy that, but we are putting one on next Monday. We'll see. That's almost certainly your problem. When you run sidecars with the Polycom 601, you can't rely on PoE - there isn't enough power supplied. Connect your powerpack to the phone and the problem /should/ go away. Semi random reboots are not uncommon on the 601 with sidecars if you're running it on PoE. That makes sense but in my case the 601 w/3 sidecars did not reboot at all and it is run from POE. The 650 just seems to perform much better. JR --- JR Richardson Engineering for the Masses ___ -- 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] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW
JR Richardson Engineering for the Masses -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of asterisk-users- [EMAIL PROTECTED] Sent: Saturday, March 01, 2008 12:00 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 44, Issue 1 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] Oh yes! This has been killing us for about a year. We've had several conference calls with my phone vendor and Polycom and it's still not fixed (or even determined why it is happening). Polycom keeps saying, upgrade to the next version of the firmware. We upgrade, still a problem. (again, for over a year!) In my case, the Polycom 601 actually reboots when we page! When it comes back up, I have a phantom meetme on the Asterisk system and none of the sidecar lights are correct. Sometimes, they simply stop updating completely. Just FYI, go to the CLI and type meetme. You'll get the conference ID and the number of users. Then, type meetme kick confID 01 Using, of course, the conference ID. The 01 is the user that initiated the meetme. So, when you kick 01, the rest go away politely! This keeps us from having to restart Asterisk. We are on Bootrom 3.2.3.0002 and SIP 2.2.0.0047 as of yesterday and we STILL have the problem. Our setup is one Polycom 601 and 25 Polycom 501s that are being paged. The 601 is powered by PoE with 2 sidecars, so Polycom wants us to put an actual Power Supply on the phone - thinking the voltage is dropping and causing the reboot. I don't buy that, but we are putting one on next Monday. We'll see. Our next plan is to get a 650 and see if it can handle the traffic. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, February 29, 2008 9:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the receptionist hangs up the page, the BYE doesn't get back to Asterisk to release all the Page channels so they stay open. I have to restart Asterisk to release all the open SIP Channels. What I think is happening is when all the SIP peers are paged, Asterisk sends 60 hint notifications to the IP 601 and the phone is overloaded and can't respond to SIP POKE or process the BYE message back to Asterisk properly. I'm wondering if I upgrade to a new IP 650 with a faster processor, will this eliminate the issue? Has anyone experienced this or have ideas for resolution or further troubleshooting? Thanks. JR -- JR Richardson Engineering for the Masses We sent a Polycom 650 on sight and replaced the 601. Paging works fine now and that extension has not dropped off the network at all, this customer group pages a lot, probably 20+ times a day. The previous condition with the Polycom 601 is not present with the 650. We made no changes to Asterisk or phone configuration. Both phones were running 2.1.1. Hope this helps. JR --- JR Richardson Engineering for the Masses ___ -- 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] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue
Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the receptionist hangs up the page, the BYE doesn't get back to Asterisk to release all the Page channels so they stay open. I have to restart Asterisk to release all the open SIP Channels. What I think is happening is when all the SIP peers are paged, Asterisk sends 60 hint notifications to the IP 601 and the phone is overloaded and can't respond to SIP POKE or process the BYE message back to Asterisk properly. I'm wondering if I upgrade to a new IP 650 with a faster processor, will this eliminate the issue? Has anyone experienced this or have ideas for resolution or further troubleshooting? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] DUNDi with two servers
JR Richardson Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch = DUNDi/context. dundi lookup number@dundified works great as well as test calls. What is the proper method of handling DUNDi between only two servers? Should I be using a dummy context on one server to handle this? I'm listing the relevant files below for only one server for brevities sake. --- dundi.conf [general] department=Test Lab organization=My Test lab locality=Anywhere stateprov=CA country=US [EMAIL PROTECTED] phone=+55 entityid=00:11:22:33:44:55 cachetime=5 ttl=1 autokill=yes [mappings] dundified = internal,0,SIP,[EMAIL PROTECTED][EMAIL PROTECTED] ,nopartial [55:44:33:22:11:00] model=symmetric host=server2.domain.com inkey=dundikey outkey=dundikey include=dundified permit=dundified qualify=yes order=primary --- extensions.conf [general] static = yes writeprotect = no clearglobalvars = no [globals] [default] include = internal include = parkedcalls [internal] include = external include = parkedcalls switch = DUNDi/dundified exten = 300,1,Dial(SIP/300) exten = 300,n,Hangup() exten = 5551234567,1,Goto(300,1) exten = 301,1,Dial(SIP/301) exten = 301,n,Hangup() exten = 8885551212,1,Goto(301,1) exten = _NXXNXX,1,Dial([EMAIL PROTECTED]) exten = _NXXNXX,n,Hangup() [external] exten = 5551234567,1,Goto(internal,300,1) --- sip.conf [dundified] type=friend dbsecret=dundi/secret context=internal [voipprovider] type=friend host=voipprovider.web dtmfmode=rfc2833 insecure=port,invite disallow=all allow=g729 context=external [300] type=peer callerid=300 username=300 secret=secret host=dynamic context=internal [EMAIL PROTECTED] notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 [301] type=peer callerid=301 username=301 secret=secret host=dynamic context=internal [EMAIL PROTECTED] notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 Thanks in advance! You can't map the [internal] context in dundi.conf because you have the switch = DUNDi/dandified statement in there. That is causing your loop. Map dundi to a dial plan [context] that doesn't have access to the [internal] context. Put your dundi extensions in the new context as a NoOp and things should work fine. JR --- Engineering for the Masses. ___ -- 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] Include in asterisk realtime
I am trying asterisk realtime with mysql database. But i don't know how to put the include entry. Have you some ideas? You have to put the include statements in the static extensions.conf file in the proper [context]. You can't use include=context in the database. JR -- JR Richardson Engineering for the Masses ___ -- 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] Asterisk and MRTG, a little help please...
After reading the sparse info and attempting to get this running, I'm unsuccessful and could use some guidance. I already have a MRTG server up and running serving hundreds of router interface graphs. I would like to add SIP/IAX channel graphs for all our asterisk servers. I'm running asterisk 1.2 and MRTG 2.4.17. I tried the script from http://karlsbakk.net/asterisk/ but get errors that MRTG does not recognize the first few lines, so I think I'm running into MRTG version compatibility issues. Can anyone send me the MRTG scripts that may work with this setup? I have the exact same script from that web page and it runs fine on Asterisk 1.2 and MRTG 2.9.29 (default RH ES3.0 install). I also have other servers on RH9 and MRTG 2.10.13 running without issue as well. What errors are you getting? Unknown option: h Unknown option: 1 Unknown option: 2 ERROR: Line 3 (use strict;) in CFG file (10.10.14.102.cfg) does not make sense I named the script file the IP address of the server.cfg instead of asterisk-mrtg. I call the script from the command line: # env LANG=C /usr/bin/mrtg 10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2 JR -- JR Richardson Engineering for the Masses ___ -- 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] Asterisk and MRTG, a little help please...
6. Re: That script runs fine. You should be able to run it first
manually, if so
please copy and paste the error.
mrtg:/var/www/mrtg# env LANG=C /usr/bin/mrtg
2008-01-28 11:16:01: WARNING: Could not get any data from external
command '/var/mrtg/10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2'
Maybe the external command did not even start. (Illegal seek)
2008-01-28 11:16:01: WARNING: Problem with External get
'/var/mrtg/10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2':
Expected a Number for 'in' but nothing'
2008-01-28 11:16:01: WARNING: Problem with External get
'/var/mrtg/10.10.14.102.cfg -h 10.10.14.102 -1 SIP -2 IAX2':
Expected a Number for 'out' but nothing'
2008-01-28 11:16:01: ERROR: Target[asterisklab2][_IN_] '
$target-[0]{$mode} ' did not eval into defined data
2008-01-28 11:16:01: ERROR: Target[asterisklab2][_OUT_] '
$target-[0]{$mode} ' did not eval into defined data
ntcp-mrtg:/var/www/mrtg#
I can see the script log into the manager interface on the asterisk
server at 10.10.14.102, and there are active SIP channels during
script execution.
Any ideas?
Thanks.
JR
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Engineering for the Masses
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Re: [asterisk-users] Asterisk and MRTG, a little help please...
You need to take a step back and first test the script without using MRTG. Execute it like this: # /opt/bin/asterisk-mrtg -h localhost -u XXX -p -1 SIP -2 Zap 10 10 10 10 You should get 4 lines of numbers. That respresents your SIP and Zap channels. Once you get past this step go back and plug it into your MRTG config. I run the script without using mrtg but I don't get any reply: mrtg:# /var/mrtg/asterisk-mrtg -h [ipaddress] -u [user] -p [pass] -1 SIP -2 IAX2 mrtg:# I can see the script log into the manager on the asterisk server, but doesn't return any figures. i haven't modified anything in the script. I'm wondering if the script is OK and maybe Asterisk manager is not setup correctly or asterisk is not returning proper value. Here is what I have configured in manager.conf: [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [user] secret = pass deny=0.0.0.0/0.0.0.0 permit=[subnet of mrtg server] read = system,call,log,verbose,command,agent,user Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] Asterisk and MRTG, a little help please...WORKING
On 1/28/08, JR Richardson [EMAIL PROTECTED] wrote: You need to take a step back and first test the script without using MRTG. Execute it like this: # /opt/bin/asterisk-mrtg -h localhost -u XXX -p -1 SIP -2 Zap 10 10 10 10 You should get 4 lines of numbers. That respresents your SIP and Zap channels. Once you get past this step go back and plug it into your MRTG config. I run the script without using mrtg but I don't get any reply: mrtg:# /var/mrtg/asterisk-mrtg -h [ipaddress] -u [user] -p [pass] -1 SIP -2 IAX2 mrtg:# I can see the script log into the manager on the asterisk server, but doesn't return any figures. i haven't modified anything in the script. I'm wondering if the script is OK and maybe Asterisk manager is not setup correctly or asterisk is not returning proper value. Here is what I have configured in manager.conf: [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [user] secret = pass deny=0.0.0.0/0.0.0.0 permit=[subnet of mrtg server] read = system,call,log,verbose,command,agent,user I added write = system,call,log,verbose,command,agent,user to manager.conf and things started working. I did not realize write was needed for the script to poll and get a response. no more errors and I get the proper channel count. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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 and MRTG, a little help please...
Hi All, After reading the sparse info and attempting to get this running, I'm unsuccessful and could use some guidance. I already have a MRTG server up and running serving hundreds of router interface graphs. I would like to add SIP/IAX channel graphs for all our asterisk servers. I'm running asterisk 1.2 and MRTG 2.4.17. I tried the script from http://karlsbakk.net/asterisk/ but get errors that MRTG does not recognize the first few lines, so I think I'm running into MRTG version compatibility issues. Can anyone send me the MRTG scripts that may work with this setup? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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 High Availability and Clustering
Hi All, There is a new list available for collaboration in this subject. http://lists.digium.com/mailman/listinfo/asterisk-ha-clustering JR Richardson Engineering for the Masses ___ --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] Realtime: Should I say or should I go (now) ?
And a free hint: if you are going to have to do anything that resembles number porting, swapping extensions, etc.--don't use extensions/phone numbers as SIP usernames. You have to regenerate config files, etc. Make your SIP usernames meaningless and use func_odbc to look up what extension is tied to which device. I second that emotion. I consult with a bunch of people who rolled their own Asterisk systems long ago, and when the try to virtualize their system in various ways they find their hands are tied. It is indescribably confusing once the number in sip.conf gets disengaged from the extensions in the dialplan. I wouldn't say to make the names meaningless, though; there are different ways to use those names so that they have useful meaning. Just don't make them extension numbers; it's like the TCP/IP boundary between layers. See SIP for an example of the problems such a thing can cause :-) B. Within my Realtime Asterisk Cluster, I use Directory Numbers (DN) for all sip/iax devices. These are a 5 or 6 digit number that don't mean a whole lot until I assign an extension to it in the dial plan. So DN 22331 could be exten 101 or exten 1001 and can be updated or changed to a different extension in the dial plan without having to update the device itself, unless the CID needs to be changed. You need very good record keeping to be successful. Also on the phone device, the auth name or account name may be 22331 but the display name will be 1001. To make this change, you need a central provisioning server, update the config file and reboot the phone to update the display name. Hope this helps and doesn't confuse things. JR -- JR Richardson Engineering for the Masses ___ --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] resync linksys SPA9XX config file from Asterisk
Hi All, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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
