Re: [asterisk-users] handling jabber status
Hi Philippe,
On Wed, Jun 4, 2008 at 7:36 AM, Philippe Sultan [EMAIL PROTECTED]
wrote:
Hi Matt,
On Wed, Jun 4, 2008 at 1:05 AM, Matthew Gibson [EMAIL PROTECTED]
wrote:
I'd be interested to know more about the status abilities as well, we've
tried to test jabberstatus application, but it doesn't seem to function
as
we expect, it should be returning 0,1,2,3,4,5 based on users current
status,
but switching to away doesn't seem to change it from 0 to 2 .. .
this could be an interesting thread :)
JabberStatus is supposed to retrieve the XMPP status of a buddy, and
store it in a diaplan variable. I just tested it on my Asterisk (1.6)
server.
Here is an example of how to use it :
1234 = {
JabberStatus(asterisk-gmail,[EMAIL PROTECTED],STATUS);
if (${STATUS}=1) {
NoOp(User is online and active, ring his Gtalk client.);
Dial(Gtalk/asterisk-gmail/[EMAIL PROTECTED]);
} else {
NoOp(Prefer the SIP phone);
Dial(SIP/1234);
}
}
Matt, if you're experiencing some problems with this application, for
example on a 1.4 system, do not hesitate to file a bug report.
Thanks for the snippet, I re-wrote it (badly) for regular extensions.conf
usage, and verified it's also working here on 1.6, though I do get a warning
about JabberStatus being depreciated.
--
Thanks,
Matt Gibson
http://www.voipphreak.ca
http://www.mattgibson.ca
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Re: [asterisk-users] Manual Wardialer
On Sat, Apr 26, 2008 at 7:13 PM, Brian J. Murrell [EMAIL PROTECTED] wrote: On Sat, 2008-04-26 at 18:41 -0400, Andreas van dem Helge wrote: Does anyone have a script for manual wardialer for asterisk? not sure if wardialer is the correct term but basically I want to call X number say 555- through 555-0050 and be able to listen to each call and when I hang up or press a key it will dial the next number for me. I guess sort of like scanning an exchange but I want to be on the line and if possible complete / talk on certain calls. Legal issues aside, have you tried this? http://www.softwink.com/iwar/ Thanks, Matt ___ -- 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] Dialplan Visualization, Rating System Unveiled!
Hello Fellow Asterisk Enthusiasts, With all the talk lately of dialplan visualization, we've decided to create a geeky, but fun and useful site. Visit http://www.ratemydialplan.com to share your diagrams of your Asterisk Dialplans and have other users rate them. To get people started with Dialplan Visualization, there are two tools available: Commercial - APSTel Visual Dialplan Pro http://www.apstel.com Opensource - Asterisk Java Project Visualizer http://blogs.reucon.com/asterisk-java/2008/05/10/visualizing_your_dialplan_with_a_graph.html We hope you enjoy using Rate My Dialplan to share and rate Asterisk Visual Dialplans. Visit http://www.ratemydialplan.com to start uploading today, it's completely free! Thanks, RMDP Staff ___ -- 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] Dialplan Visualization, Rating System Unveiled!
On Mon, May 19, 2008 at 3:15 PM, John Signorello [EMAIL PROTECTED] wrote: Matthew Gibson wrote: Hello Fellow Asterisk Enthusiasts, With all the talk lately of dialplan visualization, we've decided to create a geeky, but fun and useful site. Visit http://www.ratemydialplan.com to share your diagrams of your Asterisk Dialplans and have other users rate them. To get people started with Dialplan Visualization, there are two tools available: Commercial - APSTel Visual Dialplan Pro http://www.apstel.com Opensource - Asterisk Java Project Visualizer http://blogs.reucon.com/asterisk-java/2008/05/10/visualizing_your_dialplan_with_a_graph.html We hope you enjoy using Rate My Dialplan to share and rate Asterisk Visual Dialplans. Visit http://www.ratemydialplan.com to start uploading today, it's completely free! Thanks, RMDP Staff -- ___ -- 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 What is the relationship between RMDP and APSTel Visual Dialplan Pro ?? APSTel Visual Dialplan Pro seems to be prominently feature on the to of the page. Hi John, We are working with APSTel to incorporate Dialplan Exporting functions from their software, and the dialplans you see on there now are the sample ones included with their software. If you know of any other software that will help users with dialplan exporting into a visual format, we'll gladly incorporate that into the FAQ on the site like we have with APStel and the Asterisk-Java project. Thanks, Matt -- John Signorello Managing Partner ISPBX LLC Bus: 866 GO ISPBX ext 2000 Dir: 973-841-2061 Cell: 973-534-0888 http://ispbx.com http://cogoblue.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] A couple of newbie questions
Hi Richard, I'm not sure about the sonic wall issues, but for canadian providers try out www.les.net and www.unlimitel.ca We've had good success with both in the past. Thanks, Matt On Thu, May 15, 2008 at 11:38 PM, Richard Spencer [EMAIL PROTECTED] wrote: Hi Everyone, I'm pretty new to asterisk but coming from a call center background; needless to say I am amazed. Here is my current dilemma; but first some info on my setup. I have 3 public IP's from my provider...my LAN sits under one behind a Sonicwall TZ-180, while my trixbox sits on another behind a Linksys home router. The trixbox is running off a Visionman Server and I have port forwarded all ports (-0-6?) to the trixbox. When I am on my LAN, on the same gateway as the trixbox, I can connect using the web interface and ping the box using the public IP; when I am at home on a personal connection, I can do neither. Also, when connected on my LAN, and using a softphone (X-lite), I cannot connect to the trixbox. Is this an issue with the Sonicwall? Also, do anyone have any recommendations for SIP Providers that terminate in Canada? I currently have one DID from voicenetworks.ca and it connects to the trixbox fine and actually bring calls in (I have it set to keep the caller on hold because I can't register the softphone to answer the extension so it busies out.) so I know it has a valid internet connection. Any insight would be great. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Linux distribution to use in Asterisk server
On Fri, May 9, 2008 at 11:19 AM, equis software [EMAIL PROTECTED] wrote: Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? Thanks We use Ubuntu Server on a few of our servers and it's been working fine. We also use Gentoo. Ubuntu is nice and easy for upgrading, but, has some extra fluff that gentoo/slackware doesn't. Thanks, Matt ___ -- 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] Which Cepstral Voice to license
david-8khz and the regular david aren't bad in my experience. On Thu, May 8, 2008 at 4:54 PM, Sanjay Rajdev [EMAIL PROTECTED] wrote: Which Cepstral voice is best for Asterisk? We need to license one. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Cepstral Voice to license
They have demos of all the voices on their site.. On Thu, May 8, 2008 at 6:25 PM, Sanjay Rajdev [EMAIL PROTECTED] wrote: We are looking for a female voice. Regards, Sanjay Rajdev - Original Message - From: Matthew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 9, 2008 2:35:24 AM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] Which Cepstral Voice to license david-8khz and the regular david aren't bad in my experience. On Thu, May 8, 2008 at 4:54 PM, Sanjay Rajdev [EMAIL PROTECTED] wrote: Which Cepstral voice is best for Asterisk? We need to license one. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA FXO / FXS - can forward to sip ?
Yeah, we're looking to get rid of the asterisk box at that location and do it easy if possible. I'm sort of surprised that there are no ATA devices that will forward incoming calls on the FXO to a SIP or IAX destination. Though maybe our usage case is different than most. thanks, matt On Fri, Apr 25, 2008 at 3:22 PM, Arthur [EMAIL PROTECTED] wrote: i can only think of an asterisk box the right dialplan. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATA FXO / FXS - can forward to sip ?
Hi All, Quick question. We have a customer with a T1 located in their data center, and then one TDM card for local calls at their remote offices. We would like to remove the local PBX and TDM card and have them register directly to the main server. For the remote office, that still uses one local telephone number over analogue, we were thinking of getting an ATA device with two FXS and one FXO. The FXO would connect directly to Bell, and the FXS would go to an internal fax machine (outgoing only), and one internal analogue phone. Now, our question is. Since the IVR resides on the server in the datacenter, does anyone know of any ATA devices that will let us forward all calls, over sip (or iax) to the pbx to hit the IVR? We basically only need the local office number for emergencies, and when callers hit it, they should usually get the IVR, unless power is out, in which case the regular analogue phone would work. Anyone have any ideas? Thanks, Matt ___ -- 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] Dialplan Visualization (Extensions.conf or Dialplan Show)
On Sun, Apr 20, 2008 at 11:54 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 21, 2008 at 02:09:26AM +0300, Moshe Brevda wrote: will this do? http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/tb-trixboxgraph-0-0.1.0-2.html btw, it has (almost) nothing to do with trixbox Considering it takes the data from a mysql table called asterisk and hard-wires the default FreePBX (or is it TrixBox CE) password for that table, I'd say it has everything to do with FreePBX. Thanks, but yeah, we had found this already too. It's too tied to FreePBX for our use, we want essentially the same thing but more like asterisk -rx dialplan show | fancygrapherscript.ext :) We're going to look at the graphiz stuff this has incorporated and see if we can do some really crazy regexes to accomplish this.. unless there is a way to make asterisk export the dialplan in xml format or some form of structured format that anyone knows of.. ? thanks, matt ___ -- 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] Dialplan Visualization (Extensions.conf or Dialplan Show)
On Mon, Apr 21, 2008 at 5:11 AM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Mon, Apr 21, 2008 at 04:19:16AM -0400, Matthew Gibson wrote:
On Sun, Apr 20, 2008 at 11:54 PM, Tzafrir Cohen
[EMAIL PROTECTED]
wrote:
On Mon, Apr 21, 2008 at 02:09:26AM +0300, Moshe Brevda wrote:
will this do?
http://yum.trixbox.org/centos/5/RPMS/repodata/repoview/tb-trixboxgraph-0-0.1.0-2.html
btw, it has (almost) nothing to do with trixbox
Considering it takes the data from a mysql table called asterisk and
hard-wires the default FreePBX (or is it TrixBox CE) password for that
table, I'd say it has everything to do with FreePBX.
Thanks, but yeah, we had found this already too. It's too tied to
FreePBX
for our use, we want essentially the same thing but more like
asterisk -rx dialplan show | fancygrapherscript.ext
Here's a quick hack. Just looked at the dot man page. Did't even test
that it is actually valid. But it is probably a good start.
asterisk -rx 'dialplan show' | \
awk -F' '
BEGIN {printf digraph dialplan {\n;};
/^\[ Context/ {context=$2};
/^ Include =/ {printf \t%s - %s\n,context,$2};
END {printf }\n}
'
Only graphs simple inclusions between contexts.
BTW:
asterisk -rx | less
Behaves really strange. no scrolling possible. But I don't see anything
strange in a hexdump.
Awesome. Thanks!
Will post updates as we progress.
Thanks,
Matt
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[asterisk-users] Dialplan Visualization (Extensions.conf or Dialplan Show)
Hello, About 4 years ago there used to be a script floating around to generate dynamic graphs/diagrams of extensions.conf (the asterisk dialplan). It was using GraphViz to perform the graphing. Does anyone have a copy of this script, or a better solution to generate a flowchart of my dialplan? Thanks, Matt ___ -- 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] DUDE!!!!! was RE: Dialplan Visualization(Extensions.conf or Dialplan Show)
On Fri, Apr 18, 2008 at 2:47 PM, BJ Weschke [EMAIL PROTECTED] wrote: Steve Totaro wrote: On Fri, Apr 18, 2008 at 11:09 AM, John Signorello [EMAIL PROTECTED] wrote: excuse me... But did you not just post [asterisk-users] OT Nice IBM 1U Server Gets Along w/Old and New Digium Boards Cheap X305 $199 Did you not provide a link to a COMMERICAL entity? Wasn't your a post a unsolicited post, that is, not in response to a question??? There seems to be two standards here. The fact that you do not work for them is immaterial. If your argument is no commercial reference at all, then how do you explain your post? You may have a point although I was more doing a favor or looking out rather than trying to push my wares on someone where they did not fit. I guess I would say it is different because of intent and presentation. I thought about posting on the biz list but then thought it would better serve many Asterisk users. I certainly was clear in the title that it was a server that cost $199 even if someone did not know that OT was short for Off Topic If anyone has any real objection to my occasional postings of this nature on the users list, then I will certainly keep them to myself. It takes more effort to post something that may be helpful than it does to remain silent. Steve, -1 to not posting stuff like that anymore. I think the guys in the office here have already picked up a couple of these machines here collectively to play. They are a very good deal. Is the post a double standard? Possibly, but it is what it is, and I personally prefer to see it rather than not see it. So, sorry for sparking this big war. Does anyone happen to remember that script? :) I just need it for reverse dialplan graphing, no other utilities required :) Thanks, Matt ___ -- 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] Dialplan Visualization (Extensions.conf or Dialplan Show)
On Fri, Apr 18, 2008 at 9:51 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Apr 18, 2008 at 04:28:29AM -0400, Matthew Gibson wrote: Hello, About 4 years ago there used to be a script floating around to generate dynamic graphs/diagrams of extensions.conf (the asterisk dialplan). It was using GraphViz to perform the graphing. Does anyone have a copy of this script, or a better solution to generate a flowchart of my dialplan? from extensions.conf or from the output of 'dialplan show'? The latter assumes that Asterisk is running, but is also probably simpler to implement. Either or. I would assume from dialplan show might be better because if includes and the like. From extensions.conf might work too if there were no, or limited includes. For the former, I saw a perl module Asterisk::Config on CPAN. There are probably other ways to do that. I'll check it out. Thanks. I know there was a script, I think it was linked from the asterisk cookbook (remember [EMAIL PROTECTED]), but I've since found a few links to the cookbook, and there is no mention in it, I also tried voip-info, and google's cache, but couldn't find it there either. Thanks, Matt ___ -- 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] Cisco 7971
/callerName callerNumberfalse/callerNumber redirectedNumberfalse/redirectedNumber dialedNumbertrue/dialedNumber /forwardCallInfoDisplay /line line button=6 featureID9/featureID featureLabelIntercom/featureLabel proxyYOUR.PBX.IP.HERE/proxy name124/name displayNameIntercom/displayName autoAnswer autoAnswerEnabled3/autoAnswerEnabled /autoAnswer callWaiting3/callWaiting authName124/authName authPassword421/authPassword sharedLinefalse/sharedLine messageWaitingLampPolicy1/messageWaitingLampPolicy messagesNumber*98/messagesNumber ringSettingIdle4/ringSettingIdle ringSettingActive5/ringSettingActive contact124/contact forwardCallInfoDisplay callerNametrue/callerName callerNumberfalse/callerNumber redirectedNumberfalse/redirectedNumber dialedNumbertrue/dialedNumber /forwardCallInfoDisplay /line /sipLines voipControlPort5060/voipControlPort dscpForAudio184/dscpForAudio ringSettingBusyStationPolicy0/ringSettingBusyStationPolicy dialTemplatedialplan.xml/dialTemplate softKeyFilesoftkey.xml/softKeyFile /sipProfile commonProfile phonePassword/phonePassword backgroundImageAccesstrue/backgroundImageAccess callLogBlfEnabled2/callLogBlfEnabled /commonProfile loadInformationSIP70.8-3-3S/loadInformation vendorConfig disableSpeakerfalse/disableSpeaker disableSpeakerAndHeadsetfalse/disableSpeakerAndHeadset pcPort0/pcPort settingsAccess1/settingsAccess garp0/garp voiceVlanAccess0/voiceVlanAccess videoCapability0/videoCapability autoSelectLineEnable0/autoSelectLineEnable webAccess0/webAccess daysDisplayNotActive1,7/daysDisplayNotActive displayOnTime8:00/displayOnTime displayOnDuration10:30/displayOnDuration displayIdleTimeout00:10/displayIdleTimeout spanToPCPort1/spanToPCPort /vendorConfig versionStamp1136931633-57191cee-5ffc-4342-b286-4246b4991890/versionStamp userLocale nameEnglish_United_States/name uid1/uid langCodeen_US/langCode version1.0.0.0-1/version winCharSetiso-8859-1/winCharSet /userLocale networkLocaleUnited_States/networkLocale networkLocaleInfo nameUnited_States/name uid64/uid version1.0.0.0-1/version /networkLocaleInfo deviceSecurityMode1/deviceSecurityMode idleTimeout0/idleTimeout idleURL/idleURL authenticationURLhttp://YOUR.PBX.IP.HERE/cisco/authenticate.php/authenticationURL directoryURLhttp://YOUR.PBX.IP.HERE/cisco/directory.php/directoryURL informationURLhttp://YOUR.PBX.IP.HERE/cisco/help.php/informationURL messagesURL/messagesURL proxyServerURL/proxyServerURL servicesURLhttp://YOUR.PBX.IP.HERE/cisco/services.php/servicesURL dscpForSCCPPhoneConfig96/dscpForSCCPPhoneConfig dscpForSCCPPhoneServices0/dscpForSCCPPhoneServices dscpForCm2Dvce96/dscpForCm2Dvce transportLayerProtocol4/transportLayerProtocol capfAuthMode0/capfAuthMode capfList capf phonePort3804/phonePort processNodeNameccm-beta-5-1/processNodeName /capf /capfList certHash/certHash encrConfigfalse/encrConfig natReceivedProcessingfalse/natReceivedProcessing natEnabledfalse/natEnabled natAddress/natAddress /device Thanks, Matt On Fri, Mar 28, 2008 at 2:58 PM, J. Oquendo [EMAIL PROTECTED] wrote: Matthew Gibson wrote: What are you trying to do? I run a 7970 here with SIP. Get it to work ;) I can get the phone to register but something via way of NAT (I'm not using it) is getting in the way. I was hoping to find an example SEPxxx.xml file from someone using the 7971. Firmware is 8.3.3 -- J. Oquendo SGFA #579 (FW+VPN v4.1) SGFE #574 (FW+VPN v4.1) wget -qO - www.infiltrated.net/sig|perlhttp://www.infiltrated.net/sig%7Cperl http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x3AC173DB ___ -- 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
Re: [asterisk-users] Cisco 7971
http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+md5secret then in your sip.conf [ext] ... ;secret=123 md5secret=MD5SECRET Thanks, Matt On Sat, Mar 29, 2008 at 1:13 PM, Patrick [EMAIL PROTECTED] wrote: On Sat, 2008-03-29 at 05:25 -0400, Matthew Gibson wrote: Make sure you are using md5secret for your password, and turn off the regular secret. Here's my file working on a 7970 with SIP 8.3.3 [snip big cisco config file] Maybe it has a different name but I don't see any option containing md5 in the config you pasted. What is the md5 option called? I would like to setup md5 authentication between my 7961 on SIP 8.3.3 with Asterisk 1.4.18. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7971
What are you trying to do? I run a 7970 here with SIP. Thanks, Matt On Thu, Mar 27, 2008 at 7:02 AM, J. Oquendo [EMAIL PROTECTED] wrote: Anyone have some up-to-date (within the past 3 months) on Asterisk and the 7971. Searched voip-info, Google, etc., etc., to no avail. Documentation I found was scattered, vague. Thanks in advance. -- J. Oquendo SGFA #579 (FW+VPN v4.1) SGFE #574 (FW+VPN v4.1) wget -qO - www.infiltrated.net/sig|perlhttp://www.infiltrated.net/sig%7Cperl http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x3AC173DB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage - HP Question
I've had good luck with these guys: http://rackmountsetc.com/ supermicro have never failed me yet. On Tue, Mar 25, 2008 at 6:03 PM, Jesse Molina [EMAIL PROTECTED] wrote: If you want barebones where you add your own processor, RAM, hard drives, and options, try SuperMicro brand servers. They are thousands of dollars less than the big (fat) names like IBM and HP/Compaq, but very good quality. I've built several clusters of computers with SuperMicro systems. They are great if you want to do barebones, clusters, or other special projects. It just takes a little more time to do the assembly work. Try newegg.com for some sample pricing. On Tue, Mar 25, 2008 at 04:38:43AM -0400, Al Baker wrote: Steve - Where are you buying your HPDL380 's ? I may need quite a # of these for a new project. What factors were the Most Important to you in selecting this product ? What, if anything, is there any you do NOT like about these boxes ? How many have you deployed ? What is the largest box you have deployed ? Who does your hardware maintenance ? HP ?? Thx for sharing !!! Steve Totaro wrote: On Tue, Feb 26, 2008 at 3:10 PM, Matt [EMAIL PROTECTED] wrote: I've had it with Dell server garbage.They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver.They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I change underwear. [end rant]. Can anyone recommend an IBM or Gateway server that you have used with Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has room for one or two PCI-express interface cards? HP DL380 is my baby. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BLF on Cisco 7970
Hello All, I've been trying to get BLF working with Asterisk 1.6-LatestBeta, and My Cisco 7970 (Latest SIP Firmware). Has anyone successfully completed this? I got the patch to merge in from http://www.voip-info.org/wiki/view/Asterisk+Presence+for+Cisco+79x1+Phones With a bit of hackery to that code, it successfully compiled chan_sip for me. So, With 1.6 we have TCP Enabled on Asterisk, Phone functions and registers fine. I think all that is left is the SEPMAC.xml configuration required for BLF on the softkeys. Can anyone else share some insights to this with me? Also, Along the same lines, does anyone have documentation on what the possible FeatureID's are for the line/line configuration in the SEPMAC.xml ? So Far I know 9 shows a little phone, and 22 shows the Speed Dial keypad. I don't want to try every possible number here just to find out though :) Thanks, Matt ___ -- 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] Adding Cepstral to Asterisk
Wojciech Tryc wrote: I am not following... Why would you need to integrate Cepstral directly into Asterisk? Just to be able to call it as Asterisk app from your dialplan? I am running Cepstral and calling it through the System call. You could try the howto located here: http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt for cepstral integration into asterisk. It makes it app_cepstral, instead of using system calls. Mat ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Weird Problem - SIP/POLYCOM/DTMF
Hi All, I've been having a weird issue with one of my servers and it's Asterisk installation. The server is running Slackware, and Kernel 2.6.12. I'm running the latest CVS-HEAD edition of *. I also have 4 other asterisk servers with the same software configuration, but different hardware and they have no issues like this. I have one block of users behind a PF firewall at the office. This office asterisk server is connected to the Master server which has the T1 at one of our other locations. All of the phones (Polycom IP500) are using G729 default and allow Ulaw/Alaw to connect to the Master Server, and the remote users are also registerd to the office server using Ulaw/Alaw. Dtmfmode has been verified that it's rfc2833 for g729 and inband for the ulaw registrations. The users at work do not seem to be affected by this problem, only users on home networks not sure if that plays a part or not. It was verified that the channels were indeed all using G729 and set to rfc2833 while making the calls to the Master Server. The issue is that once in a while the phones will stop being able check voicemail because the server receives doubled, tripled, quadruled or more digits from the user checking the voicemail box. Voicemail still works on same phone not using G729 registered to office server. I verified that setting relaxdtmf in sip.conf to yes or no and it does not seem to make a difference, nor does stopping/starting asterisk or unloading zapata drivers and reinserting then starting asterisk again. I also tinkered with rxgain and txgain to no avail. The only way I have been able to fix the issue is to reboot the server. I also noticed that if I reboot the server, and check voicemail it works fine, but if I stop asterisk and then start it again I once again have the same issue and it seems to only be fixed if I reboot again. Anyone have any ideas? Matt Gibson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SoundPoint IP Attendant Console
Chris Coulthurst wrote: So the IP 601 is the 600 with a few extras? Looks like Polycom dropped the ball again -- yet another pretty phone with NO BACK LIGHT. Does the design team at Polycom have their brains unscrewed? When I was at Spring VON 04, I was talking with the guy from Polycom and asked about a backlight on their phones. His answer was somewhere along the lines of not in the foreseeable future .. mind you plans may have changed by now. Matt G ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sending fax
Sorry to interrupt :) But I believe what you guys are searching for lays here: http://www.inter7.com/?page=astfax Thanks, Matt Wiley Siler wrote: Google can translate if that helps... w ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk in India?
Hi, Is anyone successfully using Asterisk in India hooked up to the PSTN? I have tried defaultzone=us and no tones would work at all when calling the IVR, but if i set defaultzone=uk most but not all of the buttons work. Does anyone have any tips or tricks for getting TDM / PSTN connectivity from asterisk in India? Tia, Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
