similar to: SIP / RTCP statistics logging

G'day. Has anyone here used the Dualphone SIP products (their LAN range) together with Asterisk sufficiently to comment on them? http://dualphone.net/ https://www.dualphone.com.au/product_info.php/products_id/51 I am interested, specifically, in these questions: * are they generally reliable and good quality, especially audio quality? * do they work well with Asterisk for

G''day. I am wondering what the current best practice for building a single file out of distributed fragments is with puppet. Specifically, my problem: 1. Install munin-node on arbitrary machines. 2. Install ''munin.conf'' as a single file on one machine, containing a configuration stanza for every machine that munin-node is installed on. The current best practice

Anyone had any experience getting puppet to update multiple OS''s on a single server? For example, for a set of blades Network booting from a primary server, the OS for each blade would be stored on the primary server. For example: /pxe/host1/<normal OS directory structure> /pxe/host2/<normal OS directory structure> ..... /pxe/hostn/<normal OS directory structure> Can

Hello All, I''m new to puppet, and I''d like to know: Is there a formal best practices guide for syncing { /etc/passwd, shadow, group, hosts} across clients from the master? For instance; is it a better practice to make a hard link to these files and share the link, as opposed to just sharing the files directly via a target in fileserver.conf? Inquiring minds want to know...

I have this in nodes.pp $puppetmaster_fqdn = generate("/usr/bin/facter","fqdn") and this in a template http://<%= puppetmaster_fqdn %>:8080 When puppet runs, this is the result: http://puppet.home :8080 Anybody any clue to whats causing this? I''ve tried -%> -- You received this message because you are subscribed to the Google Groups "Puppet

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this

Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error" I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be

Hi, according to discussion on asterisk IRC, where people said, that macros will be depracated, I tried to migrate from macros to contexts and Gosub but if I try to use gosub in extensions.ael, ael compiler complains, that I shouln't use Gosub app, but I can't find ael keyword, that will be Gosub equivalent, or can I ignore this ael warnings? thanks PJ LOG: lev:3 file:pval.c

Hi, Does anybody have some ideas - how to play a sound file on channel, after that bridged channel got hanged up? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. atis at iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

Hi, Does anybody know if there is a way for a call goes back to transferer if unanswered ? Thanks Luis A P Barbosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070912/1e356013/attachment.htm

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from

I want to have a PC-based real-time VoIP bulk call generator (including both SIP signaling and RTP generation) for stress testing and precise analysis of the VoIP network equipment. Do any one knows a free program can do that . Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with

Hi i have problem with AddQueueMember logic. I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) and now i want to call to this Agent: exten => _1XX,1,Dial(Agent/${EXTEN:1}) call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. This doesn't work, How can i do this on Asterisk 1.4(not

Hi, Do you have any example showing how to use SendFAX ? I can see several examples of ReceiveFAX but not a single one showing SendFAX. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081127/b41ca08b/attachment.htm

G''day. We recently found some issues with the `links => follow` setting in recursive file copying; the designed behaviour is that it should allow you to determine if the master serves a symlink in a module as a symlink, or as the content of the file that the symlink points to. The full details are here: https://projects.puppetlabs.com/issues/12418 The short version is that toggling

Hi, First of all, this is my very first message to this list, so please bear with me while I''m getting used to it. Sorry for any non-standard way to ask a question here which I can be using without prior knowledgment. I would like to be able to set up some config parameters on a given node which is going to be config-managed by Puppet and, for various reasons, I would like to be able

Hi guys I've just read this about the upcoming release of * 1.6: ?Better reporting through a new call event logging capability in Asterisk 1.6 will allow complete tracking of events that take place during a call. The goal, according to Fleming, is to provide more detail than traditional CDR (Call Detail Recording) features offer and to allow for more granular tracking and auditing.? That

Hi, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long

Hi, I have an simple queue and agents defines with memeber => SIP/123. If for example Agent "SIP/123" has an call, the queue didnt care and tries to send additional calls to this agents. So Iam loosing time. SIP/123 (In use) has taken no calls yet How to stop this, especially when the device is not able to send an BUSY back. Use LOCAL channels and parse 'show queues' or