similar to: Speaking of Polycom phones...updated ROM: ouch!

I've got a problem w/ zap calls being dropped right after they are answered. I have a log file: http://pastebin.com/368526 Everything looks OK except for the DEBUG[25563] chan_zap.c: Exception on 9, channel 1 that seems to come up quite often. As soon as the other end of the Zap answers (my cell phone), and I can even hear a half second of noise, the line goes dead and gets hungup. In

I've got an issue w/ 1.2.0beta1, where if I call VoiceMailMain from a sip phone, and then either put in incorrect passwords or just hang up, I never get a Spawn Extension that hangs up the call, and my sip phone is not capable of making any more calls until I restart the daemon. Can anybody help me fix this? -- Jesse Keating GameHouse -- Systems Engineer

When I hang up in voicemail, Asterisk seems to stop responding. (hangup vs pressing # to disconnect). After that, no calls can be made until I restart Asterisk. In IRC, a developer seemed to think it had to do with me using switch => in my dial plan. Basically I never see the calling extension get the -1 signal. Can somebody help me figure out why this is happening and how I can fix it

I tried following the Wiki page regarding the Polycom 501 and having the same extension appear on all 3 line buttons (just like my cisco) but I'm having no luck. Has anyone else had success in doing this? Perhaps someone who has been successful can update the wiki? Thanks, Matthew http://www.voip-info.org/tiki-index.php?page=Polycom+Soundpoint+IP+501

Hello, Sorry, I know I read this somewhere but now I can't find it when I need it. I'd like to force a call to go out one line if we dial '9' first and then the number. Same for '8' only I will force it out a different line. There is a parameter or a method to allow the dialtone to come back after pressing the first 9... but I can't remember how to do it. Anyone know?

Hello, Can anyone point me in the direction of software to monitor channel usage on voice T1s? Using a TE410. The wiki documentation seems geared to SIP channel usage.... Thanks Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051004/8450d0c2/attachment.htm

I have installed Asterisk and i have configured with two SJPhones; i am able to make calls between these two phones.I am planning to develop a application basically web based application from which the administrator able to trace the call logs or call summary, i mean from which user agent to user agent call is going , and what is the staus, if second user tranfers the call to the third

Is there a way I can have "voice mail check" calls coming from my internal users automatically get to the right extension, without having the user enter their extension? I'm thinking that I could have the local SPA boxes translate, or have each user live in a context where the extension in question exists uniquely per user, but both of these seem kludgey. Thanks in advance for

I am using version 1.4.1.0040 and 2.6.1.0003. I have read about newer versions out there, how can I get them? I'm having an intermittent issue regarding DHCP with one of my phones, and I recall when I loaded 1.4.1 on this one phone, it failed once, then it succeeded the second time. I am wondering if a firmware update will help. Or is there any known issues with Fedora Core 2 dhcpd and IP

I'm testing out some IP501 phones and I ran into an issue. WHen I try to add a new contact into the directory, I am not able to. A window blinks really fast but the entry isn't saved. When you exit the Contact Directory system you get a 'Busy! Please try again' window. What the heck could be going on? -- Jesse Keating GameHouse -- Systems Engineer

I saw several threads requesting an Asterisk forum to complement the email list. i.e. http://lists.digium.com/pipermail/asterisk-dev/2004-February/003103.html I recently created an Asterisk forum within TMC's popular VoIP forums for everyone to use. http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15

Grasping at straws here...is anyone using a Dell PowerConnect 2224 24-port unmanaged 10/100 switch in a deployment? I have two separate asterisk installations with bad one-way audio where the only common elements left are the Dell switches and Polycom IP-500 phones. Two different ITSPs - one location uses IAX2 to the outside world, the other SIP. Two different bandwidth providers, one SDSL 1.5

My Polycom IP301 hangs on "Processing Cfg..." Here is the boot log: 0930155446|so |4|00|---------- Initial log entry ---------- 0930155446|so |4|00|+++ Note that bootrom log times are in GMT +++ 0930155446|wdog |4|00|Initial log entry 0930155446|cfg |4|00|Initial log entry 0930155446|copy |4|00|Initial log entry 0930155446|cdp |4|00|Initial log entry 0930155446|cdp

Is it possible to convert a VMWare image to KVM? As I have been building a few test machines on Vmware Fusion and would like to migrate some to a KVM server. Thanks in advance. Matt Keating Linux System Admin ? Dennis Interactive 30 Cleveland St, London, W1T 4JD Tel: 020 7907 6823 (direct line)? Fax: 020 7907 6600 (fax)? P Please consider the environment before printing this e-mail NOTE: The

What is the best way to benchmark disk IO? I'm looking to move one of my servers, which is rather IO intense. But not without first benchmarking the current and new disk array, To make sure this isn't a full waste of time. thanks

Hello, Does anybody have the latest Boot ROMs for the IP500 and IP 600 Polycom phones. I have one of each and can't find the Boot ROM v 3 anywhere to download. I would also love a good sample phone.cfg and sip.cfg files from an Aussie asterisk user to look at. Also the ip500 is having problems trying to load the bootrom 2.6.2 ? Any ideas? Kind regards Michael Felder IT Medic

Hey all, I'll give my reseller a call for support in the morning, but I usually have better/faster luck on the list. I've got a SoundPoint IP500 that I upgraded to BootROM 2.6.2 and SIP image 1.5.2 on someone elses advice, I forgot to change out the old config for the new when I loaded the image up (I guess the config changed a bunch between 1.5.2 and 1.3.1) I was prompted with an error

Hi, all. Every now and then, some of my users get "Error" on their phones. A reboot fixes it, but it's quite annoying/inconvenient. I'm running Asterisk 1.2.4, and have the following firmware, etc.: Bootrom: 2.6.2.0032 BootBlock: 2.5.0(11500_030) SIP application: 1.6.2.0041 Any ideas as to why this might be happening? Thanks! -Ken

Im setting up a very basic rails app and have a question about the needed migrations & associations. Basically, my app is an occasion reminder service that emails users when occasions that they select or input are coming up. Occasions will be selected from a checkbox type list or alternatively manually input by the user (name and date of the occasion). Users can have_many occassions.

Could somebody explain the below patch which is added in openssh6.8.P1?. During the netconf test at the end customer site it causes performance degradation. Would this cause any other issues if we revert this patch in openssh 6.8.P1? commit 57d10cbe861a235dd269c74fb2fe248469ecee9d Author: markus at openbsd.org <markus at openbsd.org> Date: Mon Jan 19 20:16:15 2015 +0000 upstream