similar to: AMI perl daemon

I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -------------- next part

Hi all, I am trying to send an extra header in SIP INVITE Message , i.e (email="me at me.com") but when I check the Message at the target that header is not there So I is Askterisk altering the Message and Is there away to include extra headers for SIP INVITE Message? Thank u

Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110408/8908db5f/attachment.htm>

I have "externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl" so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the

I'm wondering if there is a simply way to perform a voicemail callback feature using AGI. For instance, a caller leaves a voicemail, the voicemail will then call the owner of the voicemailbox determined by a database look up. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus

Hi All; It look like there are some free (open source) tools that are used for Asterisk reporting special for call center (to see number of agents logged in, number of calls now, .. etc), and to be used as dashboard. Can someone direct me for something really is suitable and stable? Regards Bilal

I know over time SIP OPTIONS message handling has changed and I've seen some write ups that seem to indicate that an s extension in the default context is needed now to get them to work. It's probably my error in any case. So, what am I doing wrong or what do I need to do to get the sip ping to work? Bruce Ferrell Just for fun, I created a sip peer called ping at a fixed address

Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple

I just completed building a feature rich asterisk voicemail system using perl, php, and mysql. My only concern is that the system i built will not be able to handle the call volume needed. Let me start by explaining my setup. Incoming call -> route.agi (perl -> mysql lookup) -> AGI -> voicemailbox (using mysql odbc) or terminate with wrong number message if a message is left in a

Hello! I want to thank everyone who helped me out with tips for load balancing asterisk machines in a cluster. I have encountered a new problem that is related to SIP diversion headers in the INVITE. I make calls through the manager interface and now want to add a SIP-Diversion header that changes the CallerID of a number that is not available on the trunk, the CallerID to be visible externally

Hello, I'm using SIP realtime with MySQL DB. Is it possible to get the status of the SIP peer (free / calling) from this realtime DB ? If not, is there another way to obtain the call state of a SIP peer ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Does anyone know if Asterisk is a good tool to be used for a large quantity of g711 and g729 transcoding? What is the best alternative for that? -- Woody Dickson woodydickson at gmail.com <woody.dickson at gmail.com> US and Worldwide Termination ============ Contact me for the following offering ============ USA Onnet - 0.0049/min USA Offnet - 0.011/min USA Mobile starting

Hi all, Sorry if this is a little off topic, but I just want to know a thing here. What system is used for sending out the caller's number in the US? Here in Sweden we use DTMF to send the number out. I just need to know what is used in the US since I don't think I will be able to use an American caller ID device over here. Many thanks for any info, Christian

Hello, I'm interested in knowing if anyone out there has successfully connected Asterisk to a Cisco UC 560 via SIP trunking? We have a client of ours that we put in an Asterisk install, one of their sister companies who we don't control is putting in a Cisco UC 560. From my looking I think it can be done, but the vendor is telling them it can't. Thought I'd ask around here and see

Hi Guys! We were using queuemetrics since long time with asterisk 1.2 but recently we have install 1.8 asterisk and but there is a big different in queue_log its saying SIP/XXXX instead of Agent/XXXX that is obvious behaviors. so do i need to change Agent/XXXX to SIP/XXXX in queuemetrics ? or is there any workaround to keep business running same like it was before. -S --------------

I have a couple of questions about asterisk 1.6: Can you change codecs mid-call upon re-invite? Can you handle the SDP offer-answer in the 200-ACK instead of the usual INVITE-200? Thanks in advance for any insight. Gary -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Apologies if this is a duplicate - been having mail server issues and I don't think I managed to send it when I tried this morning. It seems there is no .conf syntax highlighting script available for gedit. I'm thinking of putting one together myself, but don't want to reinvent the wheel. So I'm just enquiring if anyone knows of one that already exists that i've missed.

Hi, long time ago, I came up with an optimal configuration set for my environment - good detection and little false positives. Unfortunately some people are always being detected as Answering Machines. I'm not up to re-adjust my precious balance of initial_silence/max_words/... , so I'm thinking about to check if the pickup time is equal to the pickup time when the same phone number was

Hello, There are some parameters in the ISUP data (coming into the network via SIP-T packets) that need to be translated into SIP headers to be used by asterisk for proper call routing. What gateways are available to accomplish this? Thanks, Elliot -------------- next part -------------- An HTML attachment was scrubbed... URL: