similar to: Call audio leaking between calls

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

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

Has anyone seen the following in 1.4 (1.4.17) We have istances when the number of sip channels in use multiples up (eg: we have 40 channels in use, and then it will jump to 80, then 100+ and it will keep going upwards) and in doing this, all the channels which are in use at that time are simply cut off or frozen. The only way for us to get everything back to normal is via a hard restart of

I've just installed 1.8.3-rc3 on a test server as we really needed that deadlock involving REFER fix on our server but now I'm having an odd issue with one way audio with a specific type of call. If I do extension to extension calls there is full 2 way audio. If I route in an incoming call through numbers provided by our SIP provider there is no inbound audio (mobile to * SIP extension)

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

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

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

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

Hi We've been running asterisk 1.4.17 (deb package) in a production environment for some while now and are finally taken the plunge to update to 1.4.38 (Ubuntu servers). All of this is using the RealTime Architecture I have upgraded the asterisk version in one of our test environments and blind transferring seems to have suddenly stopped working. It was working fine under 1.4.17 So, call

We have a client that has sporadic problems with the *8 pickup facility. The server they are using is 1.8.5 and they are using Snom phones. Every now and then when they try to do a pickup from another phone they get a forbidden message on the phone and I can see the following in the logs. [Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL [Aug 8 11:51:53] ERROR[19314] astobj2.c:

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

Hi Does anyone have any rough idea how far away 1.8.3 is? We can't deploy 1.8 yet because of this issue https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a

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

Hi Is there an option on MeetMe that means the conference room is only available if an admin user is logged in? I've had a look the the application from the asterisk cli but I can't really see what I'm after. Currently using 1.4.17 (deb package) Soon moving up to 1.8.2 (rpm package) Thanks in advance -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

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

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

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

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

Hi I have some question about mysql realtime. 1) Anyone know exactly if there is a specific order to declare sip table column for realtime ? In which file can I find that order ? 2) In my extconfig.conf, [settings] are : sipusers => mysql,general,siptable sippeers => mysql,general,siptable so means that I use realtime dynamic exactly ? Is it normal if some parameters from sip.conf still