similar to: Finding out asterisk settings from console

Displaying 20 results from an estimated 20000 matches similar to: "Finding out asterisk settings from console"

2011 May 19
3
Manager logged on/off messages
Hi Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062
2009 Nov 10
1
Call audio leaking between calls
Hi Has anyone ever had experience of phones on the same office network being able to hear other concurrent call's audio whilst on calls of their own? We're getting this for the first time and I'm at a bit of a loss as to where to start to look. We're using 1.4.17 Any pointers would be much appreciated! Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660
2009 Aug 25
2
Authenticating SIP peer on IP address only
Hi I know this is far from best practice but is it possible to authenticate a sip peer on the IP address it's coming from so that it doesn't need to use a UN and Pass? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062
2009 Nov 16
2
Odd Local Channel and 0 billsec issue
Hi I've been noticing an odd issue with our servers (1.4.17) where a large number of one particular customer's (we operate a hosted VoIP platform) calls go through a Local channel rather than the SIP channel and whenever this happens our asterisk CDR is recording a billsec value of 0. Our outgoing calls to POTS are sent through a separate carrier and we get a daily CDR off them in
2009 Oct 16
1
Check if a variable is set
Hi Is there any way to check if a variable is set in asterisk? I've had a look around and can't find a purpose built function for it. I'm going to be using it to see if an argument has been passed with a macro or not (e.g. see if ${ARG3} is set or not) Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062
2011 May 20
1
*8 pickup and CLI presentation
Hi When we use the *8 feature to pick up a call on another extension, the phone will only display *8 and *8 is what is stored in the phones memory. Is there anything we can do so that when we use *8 the incoming caller's CLI will be presented on the screen of the phone and in the phones memory? We are using Snom phones but I'm sure this is an asterisk rather than phone issue... Thanks
2011 Dec 23
1
GotoIfTime days query
Hi I'm using 1.8. Is there a way you can specify staggered days in a single GotoIfTime command e.g. mon|wed|fri? Thanks in Advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062
2010 Sep 29
2
Alert-Info advice
Hi guys I'm using asterisk 1.4 and going on to Snom phones. I'm trying to add a sip header to make the Snom phone use a different ring tone on one particular incoming number. I have added the following to the dial plan of the incoming context +------+------------------+-------+----------+--------------+-------------------------+ | id | context | exten | priority | app
2011 May 23
1
AJAM XML output not valid xml
Hi I'm using asterisk 1.8.3.2 and have been implementing AJAM. I've noticed the final '>' is missing from every response I've had so far. Here is an example <ajax-response> <response type='object' id='unknown'><generic response='Success' message='Authentication accepted' /></response> </ajax-response Has anyone
2009 Jul 10
1
Lagged Extension
Hi There I have an extension which is in a different country and is constantly lagged (about 800ms). When anyone tries to call this extension we get a No route to destination message. Now I would have thought that the server should be able to find a route to the destination seeing as the peer poke finds it's way there. Or is that lag too much to create a SIP channel? Thanks in advance
2010 Dec 15
1
Transferring problem within Queues
Hi We are using asterisk 1.4.17 for the apt repository on an Ubuntu server and we're getting an odd problem with one customer using a Queue The queue is called in the dialplan with the options Tn The queue only has one member. Occasionally and starting to get more frequently the caller ends up being initially answered by the wrong extension (i.e. one that is not a member of the queue) Has
2012 Jul 24
2
Finding the position of a character in a string
It there a native asterisk dialplan function which will tell me the position of a specific character in a given string? eg if I wanted to find what position the '@' was at in ${SIPURI} Thanks in advance Ish -- Ishfaq Malik <ish at pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w:
2010 Jul 01
2
Brute force attacks
Hi We've just noticed attempts (close to 200000 attempts, sequential peer numbers) at guessing peers on 2 of out servers and thought I'd share the originating IPs with the list in case anyone wants to firewall them as we have done 109.170.106.59 112.142.55.18 124.157.161.67 Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -------------- next part
2010 Feb 10
1
Muted calls occasionally dropping after 30 seconds
Hi I'm having a very odd phenomenon happening on our production server (1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds after the SIP phone hits the mute button but it doesn't happen all the time. I've done a sip debug while watching this happen and that doesn't show anything other than a BYE message being sent out of the blue. The rtptimeout and
2011 May 26
3
UK English sounds packs
Hi Does anyone know if there are any free UK accented English sounds packs? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062
2011 Aug 02
3
MixMonitor and attended transfers
Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording
2011 Mar 16
2
chan_sip.c:3115 __sip_xmit of 0x108d33c0 (len 523) to xxx.xxx.xxx.xxx:0 returned -1: Invalid argument
Hi Does anyone know what this error is about? I've had 0 success in trying to find any reference to it on the internet Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062
2010 Apr 13
2
Full transfer details on inbound calls
Hi We're using asterisk 1.4.17 using RealTime and my boss has decided that we should keep a track of the full history of incoming calls i.e. who and when they were transferred to. The asterisk CDR only holds the initial answering channel for any call and not any further transfers that may have happened. The idea we are toying with is getting the time and the originating channel from the
2010 Mar 12
2
ExtenSpy Problem
Hi I'm trying to get ExtenSpy to work but it wont, I'm dialling a number from my mobile which comes into our server and answering the number on a particular SIP extension which all works fine. I'm then dialling an exten from my own SIP extension which executes the ExtenSpy for the correct extension but I hear nothing. Here is the output in the CLI -- Executing
2011 Feb 08
2
Call Recording audio file quality query
Hi We're getting requests coming in for higher quality audio in our call recordings. We currently use MixMonitor and everything is being saved in it's native 8000Hz, 16 bit wav format. I have seen information on using Monitor and specifying a conversion to mp3 when the call ends and the 2 channels get mixed but surely the 2 channels are already saved as 16bit 8000Hz wav files so the