I'm fairly new to asterisk and I'm trying to play a video file during a
video call without success for a couple of days now.
I've posted a question at stack-overflow describing my problem -
In short, what are the exact specs that a 3gp or mp4 file needs to have
in order to play under asterisk ?*
Any tips, links or suggestions will be very welcome. *Thank you all!
Is there a way to block a specific inbound number? I've found code online for blocking all nocallerid and all 800, etc. but nothing for a specific number. My company is wanting me to block a specific number. Is this possible in Asterisk 1.4 and 1.6 or do I need to go through my Service Provider?
Phone Sys Admin
1-I have a GSM gateway (GOIP) with 8 ports, I used to let every port
register to VoIPSwitch in order to know how many minutes does this GSM
card, ASR ,ACD on each card. It's too simple on VoIPSwitch to add the registrar client to dial plan ,but
in asterisk only I can find trunks How can I do that with asterisk . 2-Do any one know from where I can download a2billing prompts in Arabic for
free. Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext…
Trying to set up a simple sip - sip connection over tcp. Home : 10.0.0 -
Office: 188.8.131.52 Home sip.conf: register =>
[office-going-to-home] ; places calls
type=peer ;; we only call out
remotesecret=password sip show peer office-going-to-home
* Name : office-going-to-home
I'm trying to interface Asterisk with a third party voicemail system. This
voicemail system registers itself as extension 199. This voicemail system
gets the DID number (mailbox) from the SIP To: header field. My problem is creating the SIP INVITE with a To: field that's different
that the Request-Line URI I need to create an INVITE that looks like this in Asterisk: 20:23:12 UDP Packet Received from 127.0.0.1:5060
< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< INVITE sip:firstname.lastname@example.org:5070;target=101%40voicemailserver;cause=486 SIP/2.0 Via: SIP/2.0/UDP 10.11.11.205:5060;rport;branch=z9hG4bK253292 To:
I have a couple of performance/memory related questions:
Is there any downside to using long URIs as far as memory or database (mysql) performance is concerned, e.g. sip:email@example.com? Or is this negligible?
Also is there a performance hit if no pattern matching is used? e.g. exten => _XXX,Noop(... vs exten => 100,Noop(..
exten => 101,Noop(...
exten => 102,Noop(...
exten => 999,Noop(... If a call comes to 999, does Asterisk go through each extension sequentially from 100 to 999 until it finds the matching one? Thanks,
Maybe your logger is not setup properly?! You should get the IP in logs. I
can't think of when you won't get the IP in your logs unless the SIP
packets are manipulated. That IP is from Voxel.net. You don't have a VPS or
service from them do you? 2011/12/29 Michelle Dupuis
> *assume* they simply send an invite, and so they are in the
> external/outside context of my dialplan. So they are trying to reach
In the thread "Interesting attack tonight & fail2ban them" Bruce B
mentioned it would be nice to have input from the Community to come up
with the best set of fail2ban filters. That's a great idea. So let's
start with Bruce's filters (thanks!) and take it from there. Anyone have
any improvements and/or additions? Apologies for the line wrap. No idea
how to prevent that in Thunderbird. The filters are also at
http://pastebin.com/6T9M1W3F Not sure but it may be possible that logging has changed between
Asterisk 1.4, 1.6, 1.8 and…
We are currently using an older version of Eyebeam on our deployment and
keep having an issue with the disappearance of SIP accounts, and after
research found it is a bug on the version we currently have. I am looking for a new softphone solution and I was wondering what
everyone was using out there with your Asterisk deployments. Any
information would be helpful and most appreciated. Our current user base is on Windows XP, but we would like the chosen
solution to be compatible with Windows 7 and MAC as well. I have
1. I checked the log and I don't see any registration attempt, so I *assume* they simply send an invite, and so they are in the external/outside context of my dialplan. So they are trying to reach extensions which don't exist. If they succesfully registered they would be on the internal context, and their calls would have succeeded. (Or am I missing something?). I actually see nothing in the log but the notice (and nothing on the CLI but the notice)...so I assume it is only an invite? 2. I got their IP by turning on SIP DEBUG while they…