similar to: How does format_mp3 work?

Hi Is there any way to change the preferred audio playback format in asterisk (I'm using 1.8.25.0) i.e. first check for gsm, if doesn't exits then check for slin? Regards Ish -- Ishfaq Malik 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: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex

Hi I'm using asterisk 1.8.7.0 and adding a fail over trunk in case my primary goes down. I'm wondering what the best method of checking if the primary being up is. Is DIALSTATUS suitable for this or is there any good SIP headers to look at after the Dial step? Thanks in Advance Ish -- Ishfaq Malik <ish at pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44

Hi Does anyone know what the different return codes from AGI script execution mean? I'm getting a lot of AGI Script <script-name> completed, returning 4 I'm using asterisk 1.8.7.0 Thanks in advance -- 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:

Hi In asterisk 1.8.7.0, an inbound call that was transferred to another peer would have 2 cdr entries. In asterisk 1.8.18.0 this same activity has a single cdr entry. This is a rather large and fundamental change to be enacting halfway through a single family branch, was there any reason why this happened? It means we can't upgrade without doing significant extra development and testing.

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

Hello asterisk-users, We've been having intermittent issues with chan_sip - it stops responding to cli requests, trying to reload chan_sip from cli doesn't seem to have any effect, initiated calls carry on for a short period, but no new SIP requests are processed ('sip show channels' hangs forever, server stops responding to SIP OPTIONS, or any other SIP messages). We have updated

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 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

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

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 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

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

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

Hi How much impact on performance do DONT_OPTIMISE and BETTER_BACKTRACES have on a busy (13000+ entries in cdr for yesterday) server? I'm trying to decide whether to have them on in case of crashes or not. -- 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 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 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 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