similar to: Does Asterisk alter the Headers of INVITE Message

Hi guys, I want to ask if its possible to make calls using one SIP account, The problem is like this : I have an iPhone app and I want all my users to call the same extension which is virtual extension to my call center, so the iPhone app will be using the same SIP account for all users lets say for example: iPhone users uses 6000 at mydomain to call 9000 at my domain(which is the call center)

Hi guys, I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk 1.6 but I cannt make it work, can any body help me plz? Thank you

Hi all, I tried making a video SIP call using Asterisk .... But it didnt work....only voice call works? Regards Faraj Khasib

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

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

Would anybody know how to run a perl script as a daemon that would stay connected to asterisk via AMI? Right now, my AMI script connects to the manager interface, originates a call, disconnects. The script will be run maybe 20+ per minute. It would make more sense to me to have the script run as a daemon and have a persistent connection to asterisk's AMI. Thank you in advance for your input.

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 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

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:

Hi List, I want to secure my server from the hacker's. What is the case by which I can protest it. I have done security of Dialplan, Sip,IAX base security. For linux we are working on Iptables. What else is left so that I will do it too... -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was

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, 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

Hi List, I have stupid question but I want to know it. Why we use the PRI insted of BRI ? Just for the sake of number of lines or any thing else ? And why SIP is used for making calls rather then IAX? Even we know IAX takes 1 channel for making calls? ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Reader -------------- next part -------------- An HTML attachment was