similar to: view the current calls and their codec

Hello list- I'm having an extremely odd issue with an installation of mine. The system is running * 1.2.12.1 and currently handles around 100 handsets. With the exception of a few Grandstream DTA's, all devices are Polycom 320, 430, or 601's. After a recent power outage, I'm having an extremely odd issue with one of the handsets. One of the Polycom 601 units simply reboots every

Greetings list- I'd like the ability to hangup all calls for a particular extension from the system CLI. I understand this can probably be scripted using the AMI but I'm not familiar on how to do it. Help! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105

I have an Infortrend RAID box I'd like to see as one big 6TB partition, but I only can get 2.2TB partitions to work. I was trying to do this with an Adaptec controller but apparently they are only (any of them) 48 bits wide. Does anybody have a working system for SCSI/Centos over 2.2TB? Milt Mallory Topix.com 650-461-8316 Always consider the issues of progressive enhancement and

Can anyone recommend one to run under CentOS? Dnk Sent from my iPhone

Hi., do you think that sbr policy in queue strategy will be useful? Bye ---------- Forwarded message ---------- From: nik600 <nik600 at gmail.com> Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com> Hi to all isn't there any plan to add the Skills Based Routing strategy in

Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937]

This may be of interest -- as a tool we can use to test our systems and as a weapon that may be used against us :) http://warvox.org/ A brief read-over looks like it uses iaxclient and ruby to war dial a range of numbers and record audio samples to be analyzed to identify if the call was answered by a modem, fax machine, human, etc. The calls are placed through a PSTN termination

Greetings All- I'm about to embark on some remote management testing and need a way to login to a remote system running CentOS 4.x/5.x via SSH, su to root (using a password), then execute a command. I currently login to the boxes using key based SSH like this: ssh -i ~/remote_key admin@$REMOTEIP Then, I SU to root. However, if I try to do this automatically like this: ssh -i ~/remote_key

Hi to all the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? This is a piece of my sip.conf: [202] type=friend secret=202 host=dynamic ; This device registers with us username=202 ; Username to use when calling this device before registration limitonpeers = yes call-limit = 2 busy-level = 1 The directive busy-level is ignored.... I've also tried

Hello folks, I have a 20 channel fractional PRI and I would like to dedicate one of the lines for a Fax service (in and outbound). Is this possible with Asterisk and what conf would I need for that? Thanks, -JE

I have an interesting observation which I thought I'd pass along to save other people from spending time trying to 'fix' it. One of my clients uses Charter's so called "business phone service". They provide 'analog' phone lines over IP. In general, they've worked OK. End users were saying that the phone are "cutting out" at times. What

thanks John yeah, your approach is much siple, i've tried it but i'm not able do detect DTMF tones. it seems that on calls that i receive DTMF tones are handled correctly, but on calls generated from Asterisk to the world when the called side sends some DTMF digits they are not detected: -- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new

Hi to all i'm using PlayDTMF with AJAM, after the authentication, i make a request like this: host:8088/asterisk/mxml?action=PlayDTMF&Channel=SIP/200-sdadsadkioah&Digit=1 the result is: <ajax-response> <response type='object' id='unknown'><generic response='Success' message='DTMF successfully queued' /></response>

hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The

Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

hello , all i want free g729 codec for asterisk i tried so many moduless on asterisk.hosting.lv but cant find any related codec to my machine i cant understand where to start my asterisk version is 1.6.0.5 and following are output of cpu cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 6 model : 23 model name : Intel(R) Core(TM)2 Duo CPU

Hi, I have one in stock - got it from a client who wanted to get rid of all his old IT equipment. Looks strange, did not have enough time to play with it ....Tried it once, looked hard to configure. It stays unused in the storage room. Peter On 29.4.2010 10:20, Tim Nelson wrote: > Greetings all- > > I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks

Greetings all- I've stumbled upon a TDMoE gateway for FXO/FXS called the SpiderMux. It looks rather interesting. Has anyone used one? Where did you purchase it? Pricing? Operational issues? http://spidermux.com/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105

Dear all i'm planning a transparent migration from a courier server that provides both IMAP and POP3 access to users to a remote dovecot server with both IMAP and POP3 access. I have to migrate about 2500 users for 250 GB of space. I'm using dovecot 2.2.5.4 on debian6 squeeze. To make a transparent migration i have to maintain old IMAP UIDs and POP3 UIDs, so i've read