[asterisk-users] Asterisk DNS SIP issue
The other day my asterisk local SIP clients got hung when my provider had a DNS failure. All registrations went dead (even the ones that were IP addresses) and all sip peers went offline. I know this was know problem at one point is there any update on this when using a FQDN for one of the peer addresses in sip.conf? ___ -- 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] Disappearing B-Channels
Mark, I thought I would also mention that I am still having similar issues even after updating to the latest Asterisk, Zaptel and Libpri. Although I am using a Sangoma, we have similar symptoms with a restart fixing it. I am starting to wonder if I must go back to a Digium card. We originally switched away from the Digium TE110 because of interrupt issues, I think that the interrupt issue has been remedied now. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Greene Sent: Sunday, February 10, 2008 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Disappearing B-Channels I don't think it's my telco, I think it's my TDMoE setup. Does that sound possible? I've never had problems with the circuit until I moved it from a standard digium PRI card to a TDMoE device. Also, if I restart asterisk, all the b-channels come back. Thoughts? On Feb 10, 2008 9:40 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Sunday 10 February 2008 01:44:38 Mark Greene wrote: In my efforts to solve a mystery of asterisk slowly loosing it's ability to take incoming and outgoing calls I set asterisk to restart b-channels every 60 seconds hoping I would find something odd after some time. So now I am looking at the CLI a few hours later and look what happens when asterisk restarts the 23 b-channels I have. pbx1*CLI -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 == Primary D-Channel on span 1 down [Feb 10 01:41:23] WARNING[4102]: chan_zap.c:2401 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! [Feb 10 01:41:24] ERROR[4102]: chan_zap.c:8200 zt_pri_error: !! Got S-frame while link down == Primary D-Channel on span 1 up -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 pbx1*CLI That's the output while I've been writing this email. Those are TWO restarts of the b-channels. Notice I am missing a seizable amount of my 23 b-channels. Where are they going?! How do I find out? I've recompiled my asterisk, zaptel, and libpri to the most recent versions but that's made no difference. You probably have noise on your T1 circuit, which is causing the PRI signalling to become corrupt. If this continues, expect that the T1 circuit will go down from time to time, for a few seconds each time. Your solution is to call your telco and ask for a loopback test. -- Tilghman ___ -- 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.0/1268 - Release Date: 2/9/2008 11:54 AM ___ -- 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 mem leak behavior?
Mark, I thought I would chime in here on your problem. Oddly, I have having the same issue with a PRI with similar symptoms. The odd part is that I have never had an issue like this with a asterisk PRI setup. My setup is a PRI with a Sangoma card with the exact same issue with 1.4.14. After a few days we are unable to communicate with the PRI, The D-channel goes offline as well but the physical circuit stays up with no alarms. It doesn't give one a comfort level with uptime. I had also re-compiled the asterisk 1.4.14 along with zaptel and libpri sources and it still failed. I have since updated to the latest asterisk, zaptel and libpri .17 with the hopes that it will be fixed. I thought perhaps the card may have had an issue but now I am beginning to wonder. Kevin _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Greene Sent: Tuesday, January 29, 2008 9:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk mem leak behavior? I've tried exiting the CLI in hopes that my being in there, though it wouldn't make any sense, was keeping it from restarting. No luck. I've already setup a cron script to restart asterisk at night when there is no traffic going over it. But I hate to just treat the symptoms. I want to solve the problem. It's hard to sleep knowing there is a ghost in one of my machines. It only takes restarting asterisk, nothing else, including zaptel. Once asterisk restarts it's ready to go. I can't make heads or tails of it. There are no PRI errors when all this going on either. Debug shows nothing by usual comm chatter between the system and C/O. - Mark No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.15/1248 - Release Date: 1/28/2008 9:32 PM ___ -- 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 mem leak behavior?
Kevin, After upgrading to the latest build of everything have you seen the problem anymore? Don't know yet, waiting for it to break ( not a good feeling as you know) What's your hardware and software configs? Maybe we can find a similarity in our systems. It's a dell poweredge with a basic config, 23b + 1 d, Sangoma a101, hw-d-channel. What HW are you using and are you using any SPANDSP? - Mark ___ -- 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] Polycom Remotely Cancel Call Forward
I have a remote user on a Polycom IP Phone who has set call forwarding by accident and is away from the phone. Does anyone know of a way to remotely un-forward the phone? I tried to reboot the phone but that didn't work and removing the mac-phone.cfg caused problems ___ -- 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] Polycom Remotely Cancel Call Forward
Great suggestion, thanks. The boot failed with the mac-phone.cfg removed. I re-touched the file and followed your suggestion. Any way of removing the call forwarding feature via the xml configs? Kevin Kiely wrote: I have a remote user on a Polycom IP Phone who has set call forwarding by accident and is away from the phone. Does anyone know of a way to remotely un-forward the phone? I tried to reboot the phone but that didn't work and removing the mac-phone.cfg caused problems Remove the XML element tag from within mac-phone.cfg that it updated with the forwarding information and then reboot it again. -- Bird's The Word Technologies, Inc. http://www.btwtech.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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.5/1228 - Release Date: 1/16/2008 9:01 AM ___ -- 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] Polycom Remotely Cancel Call Forward
I guess I was interested in Disabling the forwarding feature completely via the config. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Thursday, January 17, 2008 7:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom Remotely Cancel Call Forward When setting a forward on the phone, the phone will upload to your ftp server a modified macaddr-phone.cfg XML file that (amongst other locally made changes) contains an OVERRIDE statement similar to this: OVERRIDE reg.1.fwdContact= reg.1.fwdStatus=1 ... / Change the .fwdStatus attribute to 0, then reboot the phone (sip notify polycom-check-cfg peername). That will removed the forward just fine, at least in my setup here. Works the other way as well: modify the XML file to list a valid .fwdContact and set .fwdStatus to 1, then reboot the phone. That phone won't ring again until the forward is disabled :) ___ -- 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.19.5/1228 - Release Date: 1/16/2008 9:01 AM ___ -- 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] Polycom 330 beep on new VM
The only problem with this workaround is that on the Polycom 550 (backlit display) the backlit goes bright every 30 seconds then back to dim. Any work around for that? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Johnson Sent: Friday, December 21, 2007 2:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [asterisk-users] Polycom 330 beep on new VM This is pretty easy to suppress using the configuration files. Check: http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio On Dec 21, 2007 11:55 AM, Ugo Bellavance [EMAIL PROTECTED] wrote: Hi, I have a Polycom 330 that emits a beep every 30s or so when there is a message waiting. Is there a way to disable that? It is pretty annoying. Regards, Ugo ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.5/1190 - Release Date: 12/19/2007 7:37 PM ___ -- 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] Ultrastmonkey? Ultramonkeyast? Astrimonkey? High Availability and Asterisk
I'll second the interest in this topic... a walkthrough on this topic especially with OpenSER would be great.. Kevin Kiely -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Monday, October 08, 2007 12:25 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ultrastmonkey? Ultramonkeyast? Astrimonkey? High Availability and Asterisk On 10:52, Mon 08 Oct 07, JR Richardson wrote: Hi All, snip / I'm happy to facilitate and document these solutions and share my successes and failures. It would be great to see them documents :) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.14.4/1056 - Release Date: 10/7/2007 6:12 PM ___ --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] ATS X10001P
Per the earlier recommendation, I picked up one of the ATS X10001P to evaluate. I was able to configure the LAN for access, however, I don't see where to enter the sip credentials. I have accessed the web interface with root/test and don't see any sip configuration information. I also accessed via Telnet and see more info but no place for the realm or Sip credentials. Am I missing something? Thanks in advance.. Kevin Kiely -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Friday, September 14, 2007 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DECT SIP phones On Thursday 13 September 2007 19:05:51 Stephen Bosch wrote: I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some annoying issues anyway. Can anyone suggest a solid alternative DECT SIP phone that is available in North America? I don't know how solid you would consider them, but I have repurposed the ATS X10001P phones that are sold for use with Lingo into phones that can be used with Asterisk. At $70US, I suspect they are the least expensive SIP DECT phones available. http://asterisk.drunkcoder.com/hacks/ats-config/ -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.485 / Virus Database: 269.13.16/1004 - Release Date: 9/12/2007 5:22 PM ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Followme app_followme
When using app_followme, I am receiving the following warnings on the console. We are calling the followme app with no options for additional voice announcements. Is anyone else experiencing this issue with 1.4.11? -- Executing [EMAIL PROTECTED]:1] FollowMe(SIP/101206006-b72223d8, 101206002) in new stack [Sep 13 12:10:37] WARNING[1447]: file.c:563 ast_openstream_full: File /var/spool/asterisk/followme.1189699837.464 does not exist in any format [Sep 13 12:10:37] WARNING[1447]: file.c:813 ast_streamfile: Unable to open /var/spool/asterisk/followme.1189699837.464 (format 0x4 (ulaw)): No such file or directory -- SIP/101206006-b72223d8 Playing 'followme/pls-hold-while-try' (language 'en') ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Followme app_followme
Easy now... I didn't see the closed posts after searching that's why I looked elsewhere. When I searched for the posts on the bug list they weren't there and was able to see the resolution. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Parker Sent: Thursday, September 13, 2007 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Followme app_followme Kevin Kiely wrote: When using app_followme, I am receiving the following warnings on the console. We are calling the followme app with no options for additional voice announcements. Is anyone else experiencing this issue with 1.4.11? -- Executing [EMAIL PROTECTED]:1] FollowMe(SIP/101206006-b72223d8, 101206002) in new stack [Sep 13 12:10:37] WARNING[1447]: file.c:563 ast_openstream_full: File /var/spool/asterisk/followme.1189699837.464 does not exist in any format [Sep 13 12:10:37] WARNING[1447]: file.c:813 ast_streamfile: Unable to open /var/spool/asterisk/followme.1189699837.464 (format 0x4 (ulaw)): No such file or directory -- SIP/101206006-b72223d8 Playing 'followme/pls-hold-while-try' (language 'en') Kevin, You have already posted 2 duplicate bugs reports which were closed and a very clear answer was given as to why. I honestly do not know how much more clear I can make this. Yes, it was a problem in 1.4.11. However, this has ALREADY been fixed in svn. It will be in the next release. If you would like to have this fix, you can run the latest version of svn branch 1.4. -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.485 / Virus Database: 269.13.16/1004 - Release Date: 9/12/2007 5:22 PM ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MAKE Menuselect
Does anyone know a way in Asterisk 1.4 to select the options from the menuselect menu from the command line? ___ --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] MAKE Menuselect
There have been a lot of updates to the asterisk source recently. I thought the only way to additional options from the menuselect was to run the make menuselect and select the 'optional' install items. Is there an easier way to upgrade asterisk without recompiling the new tarball and re-selecting the additional options. -Original Message- From: dave cantera [mailto:[EMAIL PROTECTED] Sent: Sunday, July 22, 2007 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MAKE Menuselect kevin, make menuselect - creates an xml file... let me look to see where it is [EMAIL PROTECTED] asterisk-1.4.5]# ls -l menu* Current Directory is /usr/local/src/asterisk-1.4.5 -rw-r--r-- 1 root 2065 Jun 25 18:36 menuselect.makedeps -rw-r--r-- 1 root 1654 Jun 25 18:36 menuselect.makeopts -rw-r--r-- 1 root 37350 Jun 25 18:34 *menuselect-tree* look in menuselect-tree, and... hmm... this looks promising for trying to figure it out... Current Directory is /usr/local/src/asterisk-1.4.5/menuselect -rw-r--r-- 1 root 31131 Aug 19 2006 example_menuselect-tree daveC Kevin Kiely wrote: Does anyone know a way in Asterisk 1.4 to select the options from the menuselect menu from the command line? ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.12/910 - Release Date: 7/21/2007 3:52 PM ___ --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] MultiParking
Does anyone have the multiparking feature enabled in asterisk 1.4? or suggest multiple parking lots? ___ --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] Multiple Parking Lots
Anyone using any variation of Multiparking, Parking Valet or servicing Call Parking with Multiple Tennants? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Parking Lots
I should have included using a multi parking feature with asterisk 1.4? -Original Message- From: Kevin Kiely [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 17, 2007 9:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Multiple Parking Lots Anyone using any variation of Multiparking, Parking Valet or servicing Call Parking with Multiple Tennants? ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.8/904 - Release Date: 7/16/2007 5:42 PM ___ --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] MultiParking
app_valetparking listed here http://www.freeswitch.org/asterisk_stuff/ Indicates support for Asterisk 1.4. The documentation listed suggests an install like so: cd /usr/src/asterisk cp contrib/scripts/astxs /usr/bin/ cd apps wget http://www.bkw.org/app_valetparking.c cd .. astxs -install apps/app_valetparking.c However astxs doesn't seem to be present in asterisk 1.4 Does anyone have this working with 1.4? and any suggestions on how to install? _ From: Darryl Dunkin [mailto:[EMAIL PROTECTED] Sent: Monday, July 16, 2007 10:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MultiParking Look at app_valetparking here: http://www.voip-info.org/wiki/index.php?page=Asterisk+addons _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Kiely Sent: Monday, July 16, 2007 16:47 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] MultiParking Does anyone have the multiparking feature enabled in asterisk 1.4? or suggest multiple parking lots? No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.6/902 - Release Date: 7/15/2007 2:21 PM ___ --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] Parking Valet
app_valetparking listed here http://www.freeswitch.org/asterisk_stuff/ Indicates support for Asterisk 1.4. The documentation listed suggests an install like so: cd /usr/src/asterisk cp contrib/scripts/astxs /usr/bin/ cd apps wget http://www.bkw.org/app_valetparking.c cd .. astxs -install apps/app_valetparking.c However astxs doesn't seem to be present in asterisk 1.4 Does anyone have this working with 1.4? and any suggestions on how to install? Thanks Kevin ___ --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] Group Function
I cant figure this out. I have seen this same example many places but the
group never gets incremented. Am I missing something?
exten = 99,1,Set(GROUP(99) = G99)
exten = 99,2,GotoIf($[${GROUP_COUNT(99)}0]?103)
exten = 99,3,dial(SIP/qoqieoeiwq)
exten = 99,103,Hangup
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RE: [asterisk-users] Follow me on multiple numbers..
I tried to look at the code in Trixbox but when the option 'confirm' is selected in the follow me properties screen, no code is generated and the call goes dead. Is there a trick to get the code generated? _ From: Philippe Lindheimer [mailto:[EMAIL PROTECTED] Sent: Saturday, March 17, 2007 12:20 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Follow me on multiple numbers.. On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote: Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell phone answering machine picks up the phone right away so my home phone never rings. I also want the caller to be able to leave a voicemail and the cell phone answering machine messes it all up. I have call screening setup so the call gets answered by the cell phone answering machine and it never accepts the call. I would appreciate if someone can help me with the setup. You can create a follow-me with 1.2 that requires you to confirm the call before answering the channel. If you need an example, go have a look at the code I generate in the dialplan in freepbx to do that exact thing when you choose call confirm. No need to go to 1.4 just for that. philippel _ Never http://us.rd.yahoo.com/evt=49938/*http:/tools.search.yahoo.com/toolbar/feat ures/mail/ miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. Check it out. -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.11/723 - Release Date: 3/15/2007 11:27 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4 Follow-Me Application
Ok, bug report submitted. 0009307 -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Friday, March 16, 2007 6:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.4 Follow-Me Application On 3/16/07, Kevin Kiely [EMAIL PROTECTED] wrote: I am having an issue with the follow me application in 1.4 The application description (below) indicates that if the specified followmeid profile doesn't exist in followme.conf, execution will be returned to the dialplan and call execution will continue at the next priority. That's not happening for me and the execution terminates not continuing to the next priority in the dialplan. Can anyone confirm this? Thanks, Kevin exten = 502,1,Followme(cell|s) exten = 502,2,Playback(goodbye) exten = 502,3,Hangup -- Executing [EMAIL PROTECTED]:1] FollowMe(SIP/2101-b6e17f60, cell|s) in new stack [Mar 16 23:29:34] WARNING[10814]: app_followme.c:954 app_exec: Profile requested, cell, not found in the configuration. == Spawn extension (from-sip, 502, 1) exited non-zero on 'SIP/2101-b6e17f60' [Description] FollowMe(followmeid|options): This application performs Find-Me/Follow-Me functionality for the caller as defined in the profile matching the followmeid parameter in followme.conf. If the specified followmeid profile doesn't exist in followme.conf, execution will be returned to the dialplan and call execution will continue at the next priority. Options: s- Playback the incoming status message prior to starting the follow-me step(s) a- Record the caller's name so it can be announced to the callee on each step n- Playback the unreachable status message if we've run out of steps to reach the or the callee has elected not to be reachable. Returns -1 on hangup I can't confirm it just now but I can certainly fix it if you post a bug on bugs.digium.com about it. :) Thanks! -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.11/723 - Release Date: 3/15/2007 11:27 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Follow-Me Application
I am having an issue with the follow me application in 1.4 The application description (below) indicates that if the specified followmeid profile doesn't exist in followme.conf, execution will be returned to the dialplan and call execution will continue at the next priority. That's not happening for me and the execution terminates not continuing to the next priority in the dialplan. Can anyone confirm this? Thanks, Kevin exten = 502,1,Followme(cell|s) exten = 502,2,Playback(goodbye) exten = 502,3,Hangup -- Executing [EMAIL PROTECTED]:1] FollowMe(SIP/2101-b6e17f60, cell|s) in new stack [Mar 16 23:29:34] WARNING[10814]: app_followme.c:954 app_exec: Profile requested, cell, not found in the configuration. == Spawn extension (from-sip, 502, 1) exited non-zero on 'SIP/2101-b6e17f60' [Description] FollowMe(followmeid|options): This application performs Find-Me/Follow-Me functionality for the caller as defined in the profile matching the followmeid parameter in followme.conf. If the specified followmeid profile doesn't exist in followme.conf, execution will be returned to the dialplan and call execution will continue at the next priority. Options: s- Playback the incoming status message prior to starting the follow-me step(s) a- Record the caller's name so it can be announced to the callee on each step n- Playback the unreachable status message if we've run out of steps to reach the or the callee has elected not to be reachable. Returns -1 on hangup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Follow me on multiple numbers..
Before you take the jump, take a look at my post earlier regarding 1.4 and follow me _ From: Ritesh Agrawal [mailto:[EMAIL PROTECTED] Sent: Friday, March 16, 2007 9:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Follow me on multiple numbers.. Thanks a lot. I am still on 1.2 and a bit worried about the unknowns of 1.4 on a production system. What do you guys think? Is it time to make a jump??? Any pointers for easy migration. I have a bunch of addons already on 1.2 and hence the concern. R On 3/16/07, Bruce Reeves [EMAIL PROTECTED] wrote: Have you looked at the Follwo-me feature in 1.4? It can require the answering channel to accept the call. You might take a look. On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote: Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell phone answering machine picks up the phone right away so my home phone never rings. I also want the caller to be able to leave a voicemail and the cell phone answering machine messes it all up. I have call screening setup so the call gets answered by the cell phone answering machine and it never accepts the call. I would appreciate if someone can help me with the setup. Thanks. R PS: I am always open to making a donation to a charity of your choice. Please help me and help the charity :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.11/723 - Release Date: 3/15/2007 11:27 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Broadvoice registration problems
Any ideas? Did anyone experience something like that? Thx Yes, unfortunately, all the time. There answer is if it works with a sip softphone client than it's not their problem. It does work with the softphone client. -Original Message- From: Bartosz Wegrzyn - maillists [mailto:[EMAIL PROTECTED] Sent: Thursday, December 14, 2006 3:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Broadvoice registration problems Hello, I have two broadvoice accounts. Lately, very often my broadvoice accounts are in unregistered state. When I log into asterisk I see: voip*CLI sip show registry Host Username Refresh State sip.broadvoice.com:5060 [EMAIL PROTECTED] 120 Request Sent sip.broadvoice.com:5060 [EMAIL PROTECTED] 120 Request Sent xxx and yyy are my ohone numbers, when I do tcpdump on my router, I see only packets comming out my router to broadvoice but nothing comming back on port 5060 23:06:41.398952 IP adsl-myip.dsl.chcgil.ameritech.net.5060 147.135.12.128.5060: UDP, length 411 23:06:42.306975 IP adsl-myip.dsl.chcgil.ameritech.net.5060 147.135.12.128.5060: UDP, length 411 23:06:42.398951 IP adsl-myip.dsl.chcgil.ameritech.net.5060 147.135.12.128.5060: UDP, length 411 23:06:44.306932 IP adsl-myip.dsl.chcgil.ameritech.net.5060 147.135.12.128.5060: UDP, length 411 23:06:44.398921 IP adsl-myip.dsl.chcgil.ameritech.net.5060 147.135.12.128.5060: UDP, length 411 the only way to get registered is to change the proxy in the host file, after some time same story happens, I am in register sent state and nothing is coming back from the broadvoice, changing proxy in the host file again solves problem, For me it sounds like broadvoice servers does not want talk with my servers for some unknown reason, I called them but their support is unqualified, and I could not get any answer, Any ideas? Did anyone experience something like that? thx Bartosz Wegrzyn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.15.18/586 - Release Date: 12/13/2006 6:13 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Broadvoice incoming DTMF problems
Is anyone having problems and Broadvoice with incoming DTMF not being recognized from a caller originating on the PSTN connection to Broadvoice? Broadvoice tech support confirmed this issue as a result of their carrier connections and suggested a work around in the dial plan(SIPDtmf). This does work but breaks DTMF for BroadVoice callers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom Call Parking
This parking patch looks like a good idea. I applied the patch but it
doesn't seem to work. The patch install was successful and I modified my
features.conf like the features.conf.sample suggested. I don't see any
mention of the k or K in the 'show application dial'. Any ideas? Did I miss
a step here?
Kevin
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 9:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Call Parking
Hi Paul -
It'd be great if I didn't have to enter the
digits and press the Park button again.
If you're interested in easier parking you might want to check out the patch
at:
http://bugs.digium.com/bug_view_page.php?bug_id=7090
You can do one-touch parking with it.
When my users had to manually enter the park extension, not one of
them used the parking feature. Now that they can just press one
button to park, it is a very often used feature. The new Polycom sip
firmware (2.01) also allows remapping a speed dial to another key, so
they can do one-button park pickup from one of the unused keys like
the Services key.
- Noah
On 10/3/06, Paul Dugas [EMAIL PROTECTED] wrote:
On Tue, 2006-10-03 at 16:22 -0400, Paul Dugas wrote:
Does anyone have any info on using the call-park feature on Polycom
phones? All I can find is that it must be supported by the SIP
server. It doesn't appear to have any related configuration settings
or other such clues as to how to use it.
Did some sniffing and found the Polycom trying to transfer the call to a
callpark extension. Found some old postings on this list that
discussed it and found this little gem:
exten = callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|
SIP/${DIALEDPEERNUMBER}|incoming,s,7)
With the call-park feature enabled (search for it in the Polycom
sip.cfg) I get a Park soft-key when I'm on a call. Press it, enter
any number, and press it again. I get the call back announcing the
parking space. Works good. It'd be great if I didn't have to enter the
digits and press the Park button again.
Paul
--
Paul Dugas, Computer EngineerDugas Enterprises, LLC
[EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park
http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA
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RE: [asterisk-users] Polycom Call Parking
I tried unsuccessfully to get this to work. I am using AAH 2.7 which has
asterisk 1.2.5.
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Call Parking
Hi Kevin -
This parking patch looks like a good idea. I applied the patch but it
doesn't seem to work. The patch install was successful and I modified my
features.conf like the features.conf.sample suggested. I don't see any
mention of the k or K in the 'show application dial'. Any ideas? Did I
miss
a step here?
I don't think the 'k' options will show in the Dial() application, but
you should be able to use it anyway.
It's possible, though, that the patch won't work with the version of
asterisk that you're using. It was designed for /trunk, and not for
the tarball 1.2.x releases. I actually don't use that patch. I wrote
a different one that's designed for the 1.2.x tarball releases.
- Noah
-Original Message-
From: Noah Miller [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 9:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom Call Parking
Hi Paul -
It'd be great if I didn't have to enter the
digits and press the Park button again.
If you're interested in easier parking you might want to check out the
patch
at:
http://bugs.digium.com/bug_view_page.php?bug_id=7090
You can do one-touch parking with it.
When my users had to manually enter the park extension, not one of
them used the parking feature. Now that they can just press one
button to park, it is a very often used feature. The new Polycom sip
firmware (2.01) also allows remapping a speed dial to another key, so
they can do one-button park pickup from one of the unused keys like
the Services key.
- Noah
On 10/3/06, Paul Dugas [EMAIL PROTECTED] wrote:
On Tue, 2006-10-03 at 16:22 -0400, Paul Dugas wrote:
Does anyone have any info on using the call-park feature on Polycom
phones? All I can find is that it must be supported by the SIP
server. It doesn't appear to have any related configuration settings
or other such clues as to how to use it.
Did some sniffing and found the Polycom trying to transfer the call to a
callpark extension. Found some old postings on this list that
discussed it and found this little gem:
exten = callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|
SIP/${DIALEDPEERNUMBER}|incoming,s,7)
With the call-park feature enabled (search for it in the Polycom
sip.cfg) I get a Park soft-key when I'm on a call. Press it, enter
any number, and press it again. I get the call back announcing the
parking space. Works good. It'd be great if I didn't have to enter the
digits and press the Park button again.
Paul
--
Paul Dugas, Computer EngineerDugas Enterprises, LLC
[EMAIL PROTECTED]phone: 404-932-1355 522 Black Canyon Park
http://dugas.ccfax: 866-751-6494 Canton, GA 30114 USA
--
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you should destroy the e-mail and any attachments or copies.
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RE: [asterisk-users] Dell hardware ...
This information seems to indicate there is a problem with the 1850 and the onboard nic. http://connection-telecom.com/support.html -Original Message- From: Ed Greenberg [mailto:[EMAIL PROTECTED] Sent: Saturday, September 16, 2006 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Dell hardware ... Precisely our configuration. Dell 1850 with 4 port PRI digium cards. No issues on my last two consulting jobs. /edg --On Tuesday, September 12, 2006 4:14 PM +0200 Arjan Kroon [EMAIL PROTECTED] wrote: Hi, Alan, We use Dell 1850 (about 20 server) and we have 4 ports PRI Digium cards in it and it works perfect. It is almost PlugPlay. greetings ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.12.4/449 - Release Date: 9/15/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Expansion Module
Has anyone used the Polycom expansion module with multiple lines? My application is for 20 lines and read there was a limit of 7 at one point. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] polycom expansion module
I am considering a Polycom expansion module for the IP601 for a DSS/BLF application. I had read that there was a limitation as to the number of lines that could be monitored with the hint command. Can anyone tell me if they are using this with multiple lines, I need to monitor 20 extensions? Thanks Kevin -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.12.3/446 - Release Date: 9/12/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 911 Testing
Be careful here... Our local PSAP is handled by the fire department. I had one of our guy's make a test call and we were told that this test must be coordinated and scheduled in advance with the chief. They want no test calls. It would probable be safest to check before making the call as they could consider it an abuse of the emergency system. It seems like a catch 22. -Original Message- From: Shane Young [mailto:[EMAIL PROTECTED] Sent: Sunday, August 13, 2006 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Leif Neland Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 911 Testing Quoting Leif Neland [EMAIL PROTECTED]: According to what I've read somewhere, at least our 911 (112) has an answering machine, saying Alarm central, one moment and a few seconds delay, before the call actually is signaled to the dispatcher, to filter out misdials and crank calls. So if you hang up quickly, they'll never know or be bothered. In Minnesota (probably most places in the US) Once you have dialed 911, even if it was in error, you should stay on the line until a dispatcher answers. If you don't they'll consider it a 911 hangup and attempt to call you back. If they can not reach you, they will dispatch a law enforcement officer (and in some areas, other emergency services). The usual call flow I've experianced is this: I Dial 911 They answer Minneapolis 911 I say This is Shane from company x making a 911 test call. They will either say ok or Please Hold if they have other calls waiting. Once they have said ok, I'll say I want to confirm you see my number as xxx-xxx- and my address is y They will almost always say Yes, that's what we have I'll say Thank you They will say Good Bye and hang up. I'll hang up. --Shane This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.10.9/417 - Release Date: 8/11/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Follow Me
Is there a way to patch an existing Asterisk 1.2.5 version with the follow me application? -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Saturday, February 25, 2006 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Follow Me On 2/25/06, Nilesh Londhe [EMAIL PROTECTED] wrote: I could not find followme app listed when I tried show applications on the CLI. Is this app patch incorporated into asterisk 1.24 release tree? If not, what are the plans for the future? On 2/24/06, Dinesh Nair [EMAIL PROTECTED] wrote: On 02/23/06 23:08 Darrick Hartman said the following: True, but why not accept the app? It sure makes the dial plan alot nothing wrong with that, i wasnt suggesting rejecting the application or anything. just pointing out that scripting it within the dialplan makes it more flexible for more people, especially those who cant code in C to change how it behaves. It's not part of the main tree yet. I don't really know whether or not it will make it into v1.4. I hope so, but it's not up to me. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Permit/Deny function question
I am trying to limit IAX connectivity to a server with the permit/deny combination. In this example to allow ip 123.123.123.123 but it's not working. If I remove the mask on the deny parameter it allows all hosts. With the deny statement like below it blocks all connections even using a mask or no mask with the permit IP. What am I missing? allow=all context=from-external secret=sample type=user deny=0.0.0.0/0.0.0.0 permit=123.123.123.123 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Enter CDR Account code during call
Does anyone have any suggestions as how to enter a CDR Account code during a call? I know it can be done in the extension logic before the answering the call, but I wanted to optionally enter an account code on certain calls without prompting on every call before or after the call? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk server to provide virtuals IPBX
App_valetparking is a great (and necessary) addition to asterisk. Does app_valetparking.c work with the current release of asterisk? I tried to install it on Asterisk 1.0.9 and I get errors following the instruction in the wiki? app_valetparking.c:678: dereferencing pointer to incomplete type -Original Message- From: Olle E Johansson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 2:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX Christopher L. Wade wrote: On Wed, Dec 21, 2005 at 10:45:29AM -0500, C F wrote: The workaround for the parking limitation is app_valetparking.so from http://www.pbxfreeware.org/app_valetparking.c instructions on how to install is on the wiki. On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Hello, Is Asterisk able to provide virtuals IPBX ? I mean one hardware server which handle one IPBX per enterprise . A lot of service providers do that. One caveat is the parking function, that only supports one parking lot for all virtual PBXs. /O There is also a work in progress in svn to add context support to the builtin asterisk parking. I forget which developer is working on it but it should be hard to find if you check the asterisk-commits archive on lists.digium.com. That would be me :-) It is in the multiparking branch. /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App_valetparking
App_valetparking is a great (and necessary) addition to asterisk. Does app_valetparking.c work with the current release of asterisk? I tried to install it on Asterisk 1.0.9 and I get errors following the instruction in the wiki? app_valetparking.c:678: dereferencing pointer to incomplete type -Original Message- From: Olle E Johansson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 2:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX Christopher L. Wade wrote: On Wed, Dec 21, 2005 at 10:45:29AM -0500, C F wrote: The workaround for the parking limitation is app_valetparking.so from http://www.pbxfreeware.org/app_valetparking.c instructions on how to install is on the wiki. On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Hello, Is Asterisk able to provide virtuals IPBX ? I mean one hardware server which handle one IPBX per enterprise . A lot of service providers do that. One caveat is the parking function, that only supports one parking lot for all virtual PBXs. /O There is also a work in progress in svn to add context support to the builtin asterisk parking. I forget which developer is working on it but it should be hard to find if you check the asterisk-commits archive on lists.digium.com. That would be me :-) It is in the multiparking branch. /O ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Join wav Files in Linux
Does anyone know how to join two .wav audio files via the command line in Linux for playback with Asterisk? ___ 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] Dell PowerEdge SC420 interrupt issue
We are getting HDLC errors on a PRI with a Dell PowerEdge SC420. I suspect it may be an interrupt issue. Can anyone recommend a low cost name brand server that will not share the interrupts or have the issues that the Dell PowerEdge SC420. Thanks ___ 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] Comedian Mail Voicemail Goto Function
A while back I used a Meridian Mail system that had a goto function to go to a specific message in your inbox/folder. I found this feature useful as I tend to keep a fair amount of messages in my boxes and it's helpful to advance to a more recent message like ( for example in the advanced menu 3-4-5 goto 99#) will bring you to the last message or #99. I was curious if anyone else had an interest in such a feature? ___ 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] DHCP vendor strings
I have an interest in using vendor strings in my DHCP scope to assign different IP's for my Polycom and Cisco phones. Has anyone used this approach and may have some examples of the dhcp.conf with the strings? Thanks ___ 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 behind Cisco Router
I am having a problem communicating with my asterisk box behind a Cisco router. I am running NAT on the inside and wanted to port forward to the asterisk IP but it is not working. I must be missing something.. This is the NAT statement I am using: ip nat inside source static udp 10.2.1.50 4569 interface Serial0 4569 ip nat inside source static udp 10.2.1.50 22 interface Serial0 7022 I realize that this may not be a totally asterisk question, so if anyone can offer a suggestion, a reply off-list is appreciated. Kevin ___ 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] Empty voicemail attachments?
I had the same problem on the HEAD version and went to STABLE to resolve it. -Original Message- From: Andrew C. Brown [mailto:[EMAIL PROTECTED] Sent: Friday, April 15, 2005 6:14 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Empty voicemail attachments? I have Asterisk newly setup. When a voicemail is left, Asterisk emails out the voicemail as a wav file attachment (in voicemail.conf, append=yes) but the attachment always ends up having a size of 0 bytes and no content to play. Diagnosis steps taken so far: I used Kmail to manually send myself a local email with an attachment and that came through whole. I looked at the wav files stored in the spool directories and they are complete and playable as expected But the content of the file doesn't make it through email. Any ideas anyone can offer as a possible cause of this problem? Thanks a mil voicemail.conf: format=WAV|wav|gsm append=yes [EMAIL PROTECTED] [default] 222 = 1234,Joe,[EMAIL PROTECTED] ___ 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 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] broadvoice
Exactly the point. Not sure if there is a limitation but it is generally provisioned the same as with most of there other offering which support call waiting. I don't think they want to advertise that multiple channels can be used, I think they monitor the monthly minutes and make an assessment from there. -Original Message- From: Kerry Garrison [mailto:[EMAIL PROTECTED] Sent: Monday, April 04, 2005 2:48 PM To: 'Matt'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] broadvoice I called them about this and the vauge answer I got was that you get 2 connections per account in order to allow the equivilant of a line with call waiting. While there is no hard-wired limitation that I know of, it is best not to abuse it so as to prevent them from enforcing one. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Monday, April 04, 2005 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] broadvoice Hi, I'm currently routing my asterisk server out over broadvoice.. it seems I can do multiple outgoing and incoming calls does anyone know if broadvoice actually allows this or not? ___ 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 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 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] Concurrent calls: best provider?
T1 PRI -Original Message- From: Scott Wolfe [mailto:[EMAIL PROTECTED] Sent: Monday, April 04, 2005 3:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Concurrent calls: best provider? This brings up the question. What is the best service for concurrent calls? In the case where I have a small business I might have 10-15 people needing to call out and they could all be on at the same time. -Scott - Original Message - From: Kerry Garrison [EMAIL PROTECTED] To: 'Matt' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, April 04, 2005 11:48 AM Subject: RE: [Asterisk-Users] broadvoice I called them about this and the vauge answer I got was that you get 2 connections per account in order to allow the equivilant of a line with call waiting. While there is no hard-wired limitation that I know of, it is best not to abuse it so as to prevent them from enforcing one. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Monday, April 04, 2005 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] broadvoice Hi, I'm currently routing my asterisk server out over broadvoice.. it seems I can do multiple outgoing and incoming calls does anyone know if broadvoice actually allows this or not? ___ 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 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 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 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] BroadVoice configuration changes for Outbound
Not that I need to stick up for Broadvoice and yes, they are not very good at returning emails, but, for me, I have used Broadvoice on several asterisk systems at different locations and haven't had any problems. It works great and is very flexible. -Original Message- From: Daryll Strauss [mailto:[EMAIL PROTECTED] Sent: Saturday, March 05, 2005 7:00 PM To: Gabriel Gunderson; Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] BroadVoice configuration changes for Outbound On Sat, 5 Mar 2005 15:02:47 -0700, Gabriel Gunderson [EMAIL PROTECTED] wrote: May I suggest: 1) Updating your website that tells how to configure Asterisk for Broadvoice. 2) Answering emails to [EMAIL PROTECTED] 3) Emailing your users that signed up as BYOB when you think a change might break stuff. Gabriel you're being WAY too nice. BroadVoice, if you're going to offer a BYOD plan, you can't go making changes that break the customers setup without notifying them first and giving them sufficient time to update their systems first! If you do mistakenly make a change that breaks your customers you reverse the change and then go back to step 1. Finally, posting on the Asterisk list to tell people you broke your system isn't sufficient. What if the user isn't reading asterisk-users. Heaven knows it's tough to keep up with this list. You really need to reverse this change, notify all your customers directly, and then put it in to place a week later. This is really amateur behavior. Telephone is critical infrastructure. You need to provide a stable base and you need to start communicating with your customers. You've got to tell us when you're making changes and you've got to start answering support email. Put a trouble ticket in place. Heck, you can buy one off the shelf. I really like your service. It works well for me, but this sort of behavior is infuriating. I want to see you survive and be successful. This sort of behavior isn't going to do that. - |Daryll CC: support and David Epstein ___ 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 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] BroadVoice Help
Try this: dtmfmode=inband register = [number]:[EMAIL PROTECTED] [broadvoice] type=peer fromuser=[number] host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice reinvite=no canreinvite=no pedantic=yes qualify=yes disallow=all allow=alaw -Original Message- From: Manjit Riat [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 25, 2005 7:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] BroadVoice Help Is the Broadvoice service up? I just signed up with them and started receiving calls in no time but could not make calls. And after a few minutes I cannot even place calls. register = [number]:[EMAIL PROTECTED] [broadvoice] type=peer fromuser=[number] host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband any help would be appreciated.. whenever I place a call from broadvoice I get Executing Dial(SIP/cisco-acfa, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 404 Not Found back from 147.135.8.128 -- SIP/broadvoice-7099 is circuit-busy Where XXX is the number I am trying to place. - Whenever I try placing a call to broadvoice it sends me straight to voice .. (The party you are trying to reach is busy..) Thanx. -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.3 - Release Date: 1/24/2005 ___ 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] BroadVoice Help
For me, yes, a lot better in many ways. -Original Message- From: Chris Ford [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 25, 2005 8:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BroadVoice Help Does broadvoice offer better service than voice pulse? Chris Ford CMF International Technologies LLC. [EMAIL PROTECTED] - Original Message - From: Kevin Kiely To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, January 25, 2005 7:51 PM Subject: RE: [Asterisk-Users] BroadVoice Help Try this: dtmfmode=inband register = [number]:[EMAIL PROTECTED] [broadvoice] type=peer fromuser=[number] host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice reinvite=no canreinvite=no pedantic=yes qualify=yes disallow=all allow=alaw -Original Message- From: Manjit Riat [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 25, 2005 7:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] BroadVoice Help Is the Broadvoice service up? I just signed up with them and started receiving calls in no time but could not make calls. And after a few minutes I cannot even place calls. register = [number]:[EMAIL PROTECTED] [broadvoice] type=peer fromuser=[number] host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband any help would be appreciated.. whenever I place a call from broadvoice I get Executing Dial(SIP/cisco-acfa, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 404 Not Found back from 147.135.8.128 -- SIP/broadvoice-7099 is circuit-busy Where XXX is the number I am trying to place. - Whenever I try placing a call to broadvoice it sends me straight to voice .. (The party you are trying to reach is busy..) Thanx. -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.3 - Release Date: 1/24/2005 ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.3 - Release Date: 1/24/2005 ___ 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] Voicepulse Caller ID Name
Does anyone use VoicePulse Inbound service and receive Caller ID Name? I receive caller ID number but no name. Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
