[asterisk-users] Proximity detection versus GSM receiver

Hi, Can anyone tell me the pros and cons of Proximity Detection using bluetooth versus using GSM cell phone with receivers. I like the idea of calls be transferred to my cell phone when I am away from the office and I would like to implement such a system. Thanks Chuck Bunn

[asterisk-users] Proximity Detection: Motorola Q + Bluetooth + Asterisk

eats power. Thanks Chuck Bunn ___ 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

[asterisk-users] Oddball time problem in CID

b browser it is correct. When I hit *98 and then my extension the CID says a time that is some 6 hours off (early)??? I am really confused where could CID be getting this bogus info??? I am using Centos 4.5, Asterisk 1.2.7.1 and Freepbx version 2.3.0.3 Thanks

Re: [asterisk-users] Oddball time problem in CID

Hi Al, That was it, Thank you!!! Al lists wrote: > check tz option in your voicemail.conf > > On 10/5/07, *Chuck Bunn* <[EMAIL PROTECTED] > <mailto:[EMAIL PROTECTED]>> wrote: > > Hi, > > I have a really oddball time problem. When I check the server

[asterisk-users] Problem with Zyxel P-2000W v2 not receiving calls

Hi, I am having an odd problem with one of our Zyxel P-2000W v2 wireless SIP phones. The phone can no longer receive calls. DND is not turned on and the phone has the exact same configuration as the other 2 phones (they each have unique extensions and such but all other settings are the same.

[asterisk-users] One of my incomming lines is busy yet there is no indication in FOP.

Hi, I am having a weird problem with one of my incoming lines. After a reboot everything is fine if I disconnect the line from the wall and reconnect it. After an hour or so the lies goes busy but no indication of this shows up on the Flash Operator panel. I also do not see anything in the t

[asterisk-users] Problem with Zyxel P-2000W v2 not receiving calls

Hi, I am having an odd problem with one of our Zyxel P-2000W v2 wireless SIP phones. The phone can no longer receive calls. DND is not turned on and the phone has the exact same configuration as the other 2 phones (they each have unique extensions and such but all other settings are the same.) Wh

[asterisk-users] One of my incomming lines is busy yet there is no indication in FOP.

Hi, I am having a weird problem with one of my incoming lines. After a reboot everything is fine if I disconnect the line from the wall and reconnect it. After an hour or so the lies goes busy but no indication of this shows up on the Flash Operator panel. I also do not see anything in the the C

Re: [asterisk-users] Problem with Zyxel P-2000W v2 not receiving calls

Hi, Sorry I forgot to mention that the phone is showing registered and 'sip show peers' shows that it is registered. Also the user can make outgoing calls without a problem. thanks Eric "ManxPower" Wieling wrote: Chuck Bunn wrote: Hi, I am having an odd problem wit

Re: [asterisk-users] One of my incomming lines is busy yet there is no indication in FOP.

Hi, I am using 2 TDM400P in a Centos 4.3 box. When we call from a cell phone to the line we get a busy signal... Thanks Lacy Moore - Aspendora wrote: On 1/12/07, *Chuck Bunn* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Hi, I am having a weird problem wi

Re: [asterisk-users] Problem with Zyxel P-2000W v2 not receiving calls

Unmonitored 416(Unspecified)D 0Unmonitored 415/41510.0.0.77D 5060 Unmonitored 414/41410.0.0.69D 5060 Unmonitored 413/41310.0.0.34 D

Re: [asterisk-users] One of my incomming lines is busy yet there is no indication in FOP.

Lacy Moore - Aspendora wrote: On 1/12/07, *Chuck Bunn* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: I am using 2 TDM400P in a Centos 4.3 box. When we call from a cell phone to the line we get a busy signal... There was something similar to this posted a few month

[Asterisk-Users] Syntax error near unexpected token 'fi'

Hi, During boot I am getting an error that says the following: Syntax error near unexpected token 'f'i' /etc/rc3.d/S09zaptel line 92 Any ideas what might be causing this? I am using Fedora 3 with latest Asterisk build Thanks ___ Asterisk-Users mailing l

[Asterisk-Users] Rebooting Asterisk box shows Asterisk failing to shutdown

hi, When I reboot my Fedora 3 box with Asterisk (latest version) I see Asterisk is failing to shutdown properly. All other processes shutdown and show success but Asterisk shows failed. What could be causing this. Thanks ___ Asterisk-Users mailing list

[Asterisk-Users] Low cost box for hosting Asterisk and at least one TDM400p

Hi, Can anyone recommend a very low cost box that could support Asterisk and at least one (preferably two) TDM400p cards and cost less that $150 (preferably under $100). The box should be able to run without a keyboard/mouse or CDROM. It also needs at least one Ethernet port. I know about conve

[Asterisk-Users] TDM400p reliability????

Hi, I have been reading about some of the problems encountered with the TDM400p cards needing a reboot. I am still testing my system so I have not seen this yet. Also my card is showing as a version 2 when I run 'modprobe wtcdm' - is this problem in this card as well. If so can anyone recommend

[Asterisk-Users] Roadmap for Asterisk??

Hi, Is there a road map for Asterisk development past version 1.07. 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.digiu

Re: [Asterisk-Users] Low cost box for hosting Asterisk and at leastone TDM400p

Hi, Yes cheaper than that - do not get me wrong I love Dell hardware but I do not need an installed OS, CDROM, Keyboard/mouse, and floppy. Minus all those I can get in down to $299 at the Dell site (using there 'N' series Optiplex - alternate OS box). I really need it to be a smaller box at a

Re: [Asterisk-Users] Low cost box for hosting Asterisk and atleastone TDM400p - THIN CLIENT MAYBE?

Hi, Actually I guess what I am looking for is semi-sealed box that I can add 1 or 2 PCI cards too. A regular PC work work in most cases since I do not want a keyboard or mouse attached to it. I do not want users screwing with the system. If it is sealed with no monitor/keyboard/mouse then they

[Asterisk-Users] Are there online forums instead of this email forum??

Hi, I am new to Asterisk and the first thing I have noticed about Asterisk and Pingtels open PBX's is that they are using this dinosaur method of running forums. It is a real pain getting every message in the forum and essentially keeping my own database of issues. With that said are there any

[Asterisk-Users] Online forums vs email list...

Hi, I am curious just what the advantages of an email forum over an online one. Thanks for the search tip, but it is still an annoying way to search. Some advantages to an online forum might be the ability to see the number of views on a thread, number of answers on a thread without counting th

[Asterisk-Users] System requirements as users grow...

Hi, I have seen some interesting papers on [EMAIL PROTECTED] and the low end hardware it can support. I there a guide that will tell me hardware requirements verses the number of users, ie amount ram or disk space recomended per user. I also understand there are scalability issues with Asterisk

[Asterisk-Users] Guidelines for sizing hardware...

Hi, I noticed that Asterisk can run on some pretty low end hardware but I have been unable to find any guidelines that will tell me approximately how much memory/disk space is required per user. Has digium or anyone else compiled this. Also I understand there are some scaling issues (everything

[Asterisk-Users] Timecard application

Hi, Does anyone know of a time card application that could be used with Asterisk. I want to be able to dial in to an extension and then type in my employee id. The caller ID would record the number called from and the time called. This would be helpful for time logging of traveling employees such a

[Asterisk-Users] Re: System requirements as users grow...

Sorry I though I turned off the vcard, my mistake... ___ 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] Timecard application

Hi, Does anyone know of a time card application that could be used with Asterisk. I want to be able to dial in to an extension and then type in my employee id. The caller ID would record the number called from and the time called. This would be helpful for time logging of traveling employees su

[Asterisk-Users] Re: Are there online forums instead of this emailforum??

James H. Thompson wrote: The Asterisk-users mailing list is available on Gmane which has a forum like interface as an option: http://dir.gmane.org/gmane.comp.telephony.pbx.asterisk.user Jim James H. Thompson [EMAIL PROTECTED] --

Re: [Asterisk-Users] Timecard application

Hi, Thanks for the help I think it gives me a starting point. Also I do not know to many nurses who can spoof a CID. A client of mine is trying to find an easy way for nurses to record their time and just about anyone can use a telephone. The client is not really interesting in getting employee

Re: [Asterisk-Users] Are there online forums instead of this email forum??

Hi, I really regret bringing the subject up... I guess I hit some nerves so please accept my apology. I have adapted to using the mailing list (Mozilla Thunderbird with filters directing traffic a specific folder, and threading) and it works, not ideally, but it works. The search of goggle work

[Asterisk-Users] Browser based configuration of Asterisk

Hi, I have a Linux Fedora 3 Asterisk only box (2 FXO & 2 FXS ports) with no GUI or WEB server running. I can get to it remotely using Putty but I want to add the capability to at least do Dial Plan configuration via a browser. Do any of the GUI based configurations support such a setup. I reall

[Asterisk-Users] SIP phones to Asterisk using MAC address instead of IP address

Hi, I know this can be done but I guess I am not understanding the few notes I have seen on this - can SIP phones be tied to Asterisk using a PC mac address instead of their IP address (obviously I am using DHCP). If someone could please point to the right Wiki or How to I would greatly appreci

[Asterisk-Users] Distributed services such as voicemail using Asterisk

Hi, Is it possible to distribute services used by Asterisk onto several boxes - similar to Pingtel (Pingtel is not an option for me since I need to tie analog phones into the system). The main service I want to distribute is the voice mail. I know that Mysql (have not tried PostGreSQL yet) can

[Asterisk-Users] Problem registering 'SJPhone'?

Hi, Has anyone had problems registering an SJPone software phone. I get lots of junk mail so I have some filters running in Thunderbird and I have not seen my registration acknowledge come through. Do they (www.sjlabs.com) use some other domain for registration?? Any one else had this problem?

Re: [Asterisk-Users] Problem registering 'SJPhone'?

Hi, I do not know how I missed it (I guess it was a blur among the hundreds of daily spam messages I get) but I misspelled the domain in my address book and thus the white list. It now works... Thanks ___ Asterisk-Users mailing list Asterisk-Users@list

[Asterisk-Users] How do I retrieve voice mail in Asterisk

Hi, I guess I am dense or something but I cannot figure out how to retrieve voicemail using a SIP SJPhone or and Analog phone with Astyerisk. I googled (voicemail +retreive :lists.digium.com) and did not get much. Everything works. I can ring each extension and if it doesn't answer it goes to v

[Asterisk-Users] Syntax checker for Asterisk config files

Hi, Is there some sort of Syntax checker for the Asterisk config files. Also is the /var/log/messages the only file that Asterisk dumps messages too or is there another place. My system works but I want to make sure I am not getting any warnings that might affect stability. Thanks _

[Asterisk-Users] Problem with ARI and seeing voicemail...

in as Admin with ari_password I see all of the call detail but still no voice mail. Any ideas where I might look for my problem. Voicemail is working since I can call the voicemail extension and retrieve messages. I am not using AMP and I have set the standalone flag to true. Thanks Chuck

[Asterisk-Users] ARI - Voicemail not showing - Problem solved!

e and help in solving my problem. If you have not tried the ARI interface you might look at it, my clients love it!! Great job DAN Thanks Chuck Bunn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

[Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...

Hi, I thought I had this problem licked but there still is a rights problem with ARI and Asterisk when using a non-root user (Following the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-root&diff2=25). When I issue the following: chmod --recursive u=rwX,g=rX,o= /var

Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...

more... Thanks Giorgio Incantalupo wrote: Hi Chuck. I had the same problem. I solved it using the externnotify parameter inside voicemail.conf. Just launch a script which changes the /var/spool/asterisk permissions. Giorgio Incantalupo Chuck Bunn wrote: Hi, I thought I had this problem

Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...

Hi, Just so I am clear this patch will work with 1.2.4 and requires manual updating to files and then a recomplie of Asterisk source correct?? Thanks Ben Klang wrote: Hello, I found the same problem very frustrating, mostly because it causes Asterisk to ignore ACLs and umask settings. I

Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...

Hi, Could you post the updated patch for 1.2.4 Thanks Ben Klang wrote: On Thursday 16 February 2006 11:47, you wrote: Just so I am clear this patch will work with 1.2.4 and requires manual updating to files and then a recomplie of Asterisk source correct?? This patch was written agai

[Asterisk-Users] Install instructions for FOP Flash Operator Panel do not make sense...

Hi, Anyone got AFOP working. The install instructions tell you to copy all of the files extracted under the 'html' directory to a subdirectory under your main web directory (in my case this is /var/www/html/panel/) and then the instructions talk about modifying the 'op_server.cfg' file but th

Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...

Incantalupo Chuck Bunn wrote: Hi, I thought I had this problem licked but there still is a rights problem with ARI and Asterisk when using a non-root user (Following the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-root&diff2=25). When I issue the following: c

[Asterisk-Users] Features set in the features.conf stopped working after upgrade.

Hi, I recently moved all of my conf files over to a new Asterisk 1.2.4 server and every works except the features enabled in features.conf. Was there a syntax chnage in 1.2.4? Or is there something else... Here is my features.conf: [general] parkext => 880; W

[Asterisk-Users] Keep getting message in logs that pbx.c cannot find extension context 'default'

Hi, I am getting repeated messages in my logs with the following: * Feb 23 07:56:11 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23 07:56:11 DEBUG[2470] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] Feb 23 07:56:12 NOTICE[24

[Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?

Hi, Is setting the variable _TRANSFER_CONTEXT required in features.conf for Asterisk 1.2.4? It appears from the Wiki that transfers across contexts are not possible when this is set. If it is not set can one trasfer across contexts?? Thanks ___ --B

[Asterisk-Users] Okay can somebody explain this...

Hi, I got rid of the messages I was getting in the CDR (pbx.c: Cannot find extension context 'default') by adding a blank 'default' context at the front of my extensions.conf (I use the context 'extensions-home') this also (well sort of ) fixed my problem with blind transfers. I can blind tra

Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?

transferer TRANSFER_CONTEXT. That context is used to match the extension to dial. So you can set this var to any context you want. Regards On 2/23/06, *Chuck Bunn* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Hi, Is setting the variable _TRANSFER_CONTEXT

Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?

transfer a caller from the spa to the resturant and vise versa. There are seperate phone lines comming in for the spa and resturant as well. Thanks Moises Silva wrote: it seems im not undestanding your question then. Could you provide a practical example? On 2/24/06, *Chuck Bunn* < [EM

[Asterisk-Users] No ring when doing blind transfer.

Hi, I have an odd problem when doing a blind transfer. The transfer is intiated and the transferred caller hears nothing until the timeout. I have tried setting the 'r' and the 'm' variables in the dial command. Nothing happens when I use the 'r' variable when I use the 'm' variable I briefly

[Asterisk-Users] Music on hold volume too high - using built in music on hold.

Hi, I saw this problem mentioned before but the user appeared to be using the MP3 software with asterisk. I am using the native music on hold player in asterisk 1.2 and I too have a volume problem with music on hold. Is this controllable through the 'indications.conf'? I know this file contro

[Asterisk-Users] Zaptel not compiling on lastest Centos 4.2 kernel.

Hi, I made a big mistake on a Centos 4.2 box - I forgot to exclude the kernel from updating. Now zaptel will not do a "make linux26" see below. Is there a way to roll this back or is there a patch to get Zaptel to compile? I have a link to the modules using 'ln -s /lib/modules/uname -r/build

Re: [Asterisk-Users] Failed installing zaptel

Hi, I am having the same exact problem. I am assuming that it was a problem with a kernel update I did. I am in the process of rolling back to an older kernel... I will let you let know if this works. There is also a patch for zaptel but I believe this is for going from 1.3 to 1.4? Thanks H

Re: [Asterisk-Users] Failed installing zaptel

urce from ftp.kernel.org and do make oldconfig with your .config and try again. on Monday 03/13/2006 Chuck Bunn([EMAIL PROTECTED]) wrote > Hi, > > I am having the same exact problem. I am assuming that it was a problem > with a kernel update I did. I am in the process of rolling back to

[Asterisk-Users] All calls in queue go to agent that is down??

Hi, When an agent is logged in and his phone goes off the network (using an SJPhone on a portable PC) and the user had forgotten to log out of the queue all calls go to this agent that is no longer connected to the system. I have tried training and retraining but I need some way to fix this.

Re: [Asterisk-Users] Failed installing zaptel

did a wget for the 2.6.9-22 source from the mirror and it worked like a charm! HTH Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, March 13, 2006 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion S

Re: Spam? Re: [Asterisk-Users] Failed installing zaptel

L PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, March 13, 2006 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Failed installing zaptel Hi, I am having the same exact problem. I am assuming that it was a problem with a kernel update I d

[Asterisk-Users] Can I get to a menu system while in a queue??

Hi, Is it possible to get to a menu system while in a call queue. I want users to be able to hit the '*' and be able to goto a menu system from a queue if they so desire. I thought the following would do this but no dice... * extension.conf [general] #include macros.incl #in

[Asterisk-Users] Context confict question??

Hi, If I have an extension in a context and I have another context with the same extension and I include the second context in the first does this cause a conflict or does Asterisk know that there is a 600 extension in each context [big-business] exten => 600,1,Dial(ZAP/1,20) include =>

[Asterisk-Users] What kind of extension numbers can be used in the exit context of a queue?

Hi, In Asterisk 1.2 according to the wiki and I quote: "It is now possible to use multi-digit extensions in the exit context for a queue (although you should not have overlapping extensions, as there is no digit timeout). This means that the EXITWITHKEY event in queue_log can now contain a ke

Re: [Asterisk-Users] Context confict question??

has lower priority. Hope this helps. Andy On 12/2/05, *Chuck Bunn* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Hi, If I have an extension in a context and I have another context with the same extension and I include the second context in the first d

[Asterisk-Users] Can I escape queue with a '*'?

Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listi

Re: [Asterisk-Users] Extension Manual

extension ?? Thanks again in advance - Original Message - From: "Chuck Bunn" <[EMAIL PROTECTED]> To: Sent: Saturday, December 03, 2005 2:06 PM Subject: [Asterisk-Users] Can I escape queue with a '*'? Hi, Can I escape a call queue by pressing a '*'

[Asterisk-Users] Converted mp3 files to raw for musiconhold and still does not work...

Hi, I setup music on hold as directed for Asterisk 1.2 but still no music on hold. Any ideas what I did wrong. I see it start in the CLI but then it immediately stops?? I also see the Hangup occur 20 seconds later as it should according to WitMusicOnHold(20). I used a test setup suggested in

[Asterisk-Users] Why does musiconhold.conf changes require a reboot?

Hi, Why do changes to musiconhold.conf require a reboot. Also if I put mp3's into the /var/lib/asterisk/mohmp3 directory will the be played if I use the -r option? Using Asterisk 1.2 and have run the make config in the /usr/src/asterisk-addons directory. Thanks __

Re: [Asterisk-Users] Re: sound problem in X-Lite phone with asterisk server

Hi, Two things does your codec set in X-lite match what is set in the sip file and have you rebooted since setting up music on hold. I should also ask if ran a make and make install in the asterisk-addons directory, this installs a mp3 player (among other things) in Asterisk 1.2? Vipul Patel

Re: [Asterisk-Users] Transfer/take call to/from other phone

Hi, Push the '#' key followed by the extension for a blind transfer. Thanks Denny Schierz wrote: hi, my asterisk 1.2 works very well with ISDN and SIP, but how can i transfer calls from my phone (for example ISDN, MSN 400) to the another phone (ISDN, MSN 401)? Or when the phone 401 rings, bu

[Asterisk-Users] Restore logging functionality...

Hi, A while back I made the stupid mistake of deleting my log files 'full' and 'messages' for asterisk. I recreated the files by 'touch' filename and I have gone into the Asterisk CLI and tried both 'logger restart' and 'logger rotate' but the logs still show nothing. I run 'logger show chann

Re: [Asterisk-Users] Transfer/take call to/from other phone

Hi, To pick up another persons phone that is ringing dial '*8' followed by their extension. To do an attended transfer dial '*2' followed by the extension... Hope that helps Denny Schierz wrote: hi, Quoting Chuck Bunn <[EMAIL PROTECTED]>: Push the '#

Re: [Asterisk-Users] Restore logging functionality...

ermission to write on the files, delete them , and asterisk will recreate them as user asterisk, or chown them, or change them to 777 best of all, delete them! Marco. Chuck Bunn wrote: Hi, A while back I made the stupid mistake of deleting my log files 'full' and 'messages&#x

[Asterisk-Users] Anyone know anything about the new Linksys One product - does it use Asterisk?

Hi, Does anyone have any details about the Linksys one product that was just announced? Does it use Asterisk? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: htt

[Asterisk-Users] What would prevent logs from being recreated if they are deleted?

Hi, Please excuse the cross post but these seems to be one of those issues that may be answered by a developer or someone with direct administrative knowledge of the deep workings of Asterisk. I have deleted my log files expecting them to be recreated by Asterisk 1.2 but nothing happens after

[Asterisk-Users] '#' (fast foward) and '*' (Rewind) not working in VoicemailMain

Hi, '#' (fast forward) and '*' (Rewind) not working in VoicemailMain with Asterisk 1.2.1 Do I have to do something in the dialplan to make this work? I have '##' set as a blind transfer and '*2' set as a attended transfer in features.conf. Per the Wiki Voicemailmain has the following settings

[Asterisk-Users] Blind transferred user does not hear phone ring while waiting for phone to be picked up.

Hi, Please excuse the double post but I am about to report this as a bug and I want to verify that others are having the same problem. Also I have seen numerous bugs reported that are not bugs but just misconfiguration, etc. and I do not want to burden the developers with a frivolus bug repor

[Asterisk-Users] Best way to automatically stop and start Asterisk nightly

Hi, I am planning on restarting asterisk nightly as I seem to be experiencing some sort of memory leak (Asterisk slows down over time). I have reviewed the Asterisk suggestions for management and one item is the routine rebooting of Asterisk. Since I have Asterisk 1.2.1 what is the recommend

Re: [Asterisk-Users] Blind transferred user does not hear phone ring while waiting for phone to be picked up.

OOPs I forgot to mention I am using Asterisk 1.2.1 and I had the same problem with 1.0.9 and 1.2.0 Chuck Bunn wrote: Hi, Please excuse the double post but I am about to report this as a bug and I want to verify that others are having the same problem. Also I have seen numerous bugs

Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

Hi, What are you codec and dmtfmode settings in sip.conf and in the sip phone settings. If you dmtfmode is set to 'inband' and you are using anything other than ulaw or alaw codec it wont work. Also since your hear the phone sometimes you may be experiencing QOS issues on your network. Doe yo

Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

, Chuck Bunn <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>: What are you codec and dmtfmode settings in sip.conf and in the sip phone settings. I use gsm. If you dmtfmode is set to 'inband' and you are using anything other than ulaw or alaw codec it wont

Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

is used that packets could time out and you would never know it since UDP is dumb and has no packet loss recovery mechanism. What is the topology of your network. Is the Asterisk box and the client on the same backbone and switch? Thanks Evil Skymarshal wrote: Hi Chuck, 2005/12/17, Chuck

Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1

recovery mechanism. What is the topology of your network. Is the Asterisk box and the client on the same backbone and switch? Thanks Evil Skymarshal wrote: Hi Chuck, 2005/12/17, Chuck Bunn <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>: What are you codec and dmtfmode setti

Re: [Asterisk-Users] Best way to automatically stop and start Asterisk nightly

the problem. I will use your command to see if 'asterisk.pid' inflates over time... Thanks Again Tzafrir Cohen wrote: On Wed, Dec 14, 2005 at 08:49:06AM -0700, Chuck Bunn wrote: Hi, I am planning on restarting asterisk nightly as I seem to be experiencing some sort of memory leak

[Asterisk-Users] Automatic logoff of all agents at set time

Hi, Is there a way to force the logoff of all agents at a set time say 8:00PM or do I have to do an agentcallbacklogin with each agents credentials? I am using Asterisk 1.2 The wiki shows an extension that the agent calls to preform the logoff - I need something that is completely automated a

Re: [Asterisk-Users] Automatic logoff of all agents at set time

Hi, I understand how GototIfTime works but that still leaves agents logged in and if an agent is absent the next day calls will go to an agent that is not there. Thanks Michiel van Baak wrote: On 16:22, Tue 27 Dec 05, Chuck Bunn wrote: Hi, Is there a way to force the logoff of all

Re: [Asterisk-Users] Automatic logoff of all agents at set time

Hi, I have not been able to find anything about persistent agents in any wiki? Where does this command go and what is its syntax? Thanks Michiel van Baak wrote: On 18:06, Tue 27 Dec 05, Bud Bach wrote: But, if the agents don't log out for some reason, they will still be logged in the ne

Re: [Asterisk-Users] Automatic logoff of all agents at set time

Lopez wrote: It is set in the queues.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Wednesday, December 28, 2005 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Automatic

Re: [Asterisk-Users] Automatic logoff of all agents at set time

Hi, Oh sorry I am using asterisk 1.2.1 Thanks Kevin P. Fleming wrote: Chuck Bunn wrote: When I add 'persistentmembers=no' in queues.conf and reload I get a message in the message log file saying unknown keyword 'persistentmembers'. I got the syntax from http://www.voi

[Asterisk-Users] SJPhone with external ringer

Hi, Does anyone know if it is possible to setup an SJPhone with an external ringer of some sort. One of the operators may not always be at her desk and when she is not wearing a headset she cannot hear the phone ring - is there some way to fix this? Thanks ___

[Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express

Hi, Does anyone know if the Digium Wildcard will work on a PCI Express or PCI-X motherboard. Specifically I am looking at the Dell 850 1U rack server for use with Asterisk. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk

Re: [Asterisk-Users] Will Digium Wildard work with PCI-X or PCI Express

er and Systems Developer Kevin Bockman wrote: Chuck Bunn wrote: Does anyone know if the Digium Wildcard will work on a PCI Express or PCI-X motherboard. Specifically I am looking at the Dell 850 1U rack server for use with Asterisk. They will work in PCI-X of course but not PCI Express.

[asterisk-users] Examples of handeling input from phones with PHP

Hi, Can anyone direct me to where I might find examples of handling interactive input from a phone using PHP and AGI. I want to have someone dial an extension and then have the system request input from the user, take that input and put it into a database. Thanks ___

[asterisk-users] Can anyone recommend a large button sip phone for the elderley.

Hi, Can anyone recommend a large button/type sip phone (VOIP) that an older person could use. I have a client that needs to have large button phones for elderly residents in her facility. Thanks ___ --Bandwidth and Colocation provided by Easynews.co

Re: [asterisk-users] Can anyone recommend a large button sip phone for the elderly.

Hi Cory, Do you know if any of the ATA to SIP converters that can handle visual indicators (flashing light in addition to ring sound) and for that matter can Asterisk handle visual indicators for ringing? Thanks ___ --Bandwidth and Colocation provi

[asterisk-users] Is there a Blue tooth wireless headset that will work with asterisk?

Hi, Does anyone know if there is a blue-tooth wireless headset that works with asterisk and/or a SIP software phone on the PC? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update opt

Re: [asterisk-users] Can anyone recommend a large button sip phone for the elderly.

Hi, So am I to understand that the visual indicator responds the same way a ring would and thus if Asterisk tells a phone to ring the visual indicator uses that signal and does not require a separate signal? I guess I am use to seeing visual indicators in hotels that blink when there is a mes

Re: [Asterisk-Users] Ouch ... error while writing audio data: :Brokenpipe

Hi, What was the editing of the zapata.conf file that fixed your problem??? Thanks Bob Hrbek wrote: Ok...a little editing of the zapata conf files and ...you da man. That is starting it. Thanks, Bob - Original Message - From: "Steve Totaro" <[EMAIL PROTECTED]> To: "Asterisk U

[Asterisk-Users] Cannot telnet to port 5038 on asterisk

Hi, I cannot do the following: telnet 127.0.0.1 5038 I get connection refused and this is preventing AMP from installing. I had this working when I was using FC3 but I had to upgrade to FC4 for another application. So I am running PHP5, MYSQL 4 with FC4 and asterisk is running (I had this pr

Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk

5038 is the correct port and it is enabled. Thanks [EMAIL PROTECTED] wrote: On 10/17/2005, "Chuck Bunn" <[EMAIL PROTECTED]> wrote: Hi, I cannot do the following: telnet 127.0.0.1 5038 Is telnet enabled? Brett __

Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk

ing to the telnet port) -- Mike -Original Message- [EMAIL PROTECTED] Sent: Monday, October 17, 2005 12:28 AM Subject: Re: [Asterisk-Users] Cannot telnet to port 5038 on asterisk On 10/17/2005, "Chuck Bunn" <[EMAIL PROTECTED]> wrote: Hi, I cannot do the fol

[Asterisk-Users] Credit card machines, Asterisk and Digium any issues?

Hi, We want to be able to use any of 4 outgoing lines (POTS connected to FXO cards on a Digium TDM400P card) for a credit card machine (you know those little machines that have a phone line attached and are in many small establishments) We have 4 FXS for analog phones and 2 credit card machin

  1   2   >