similar to: Muted calls occasionally dropping after 30 seconds

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 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 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

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

Hello, We have a problem that calls seem to be dropping for no reason. Is there any way to write a debug log to disk so that I can check it as soon as a call is lost? It happens randomly once or twice a day to different callers. Many thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get the 'no matching peer' error when we get a dictionary SIP attack. Now the logs always show a 'wrong password' when there actually isn't a matching peer. We even have alwaysauthreject = yes in our sip.conf. Has anyone else noticed this phenomenon? Thanks in Advance Ish -- Ishfaq Malik

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 all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately. At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -------------- next part -------------- An HTML

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'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

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 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 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

The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and

The Asterisk Development Team has announced security releases for Asterisk branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and

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

Hello, I'm using SIP realtime with MySQL DB. Is it possible to get the status of the SIP peer (free / calling) from this realtime DB ? If not, is there another way to obtain the call state of a SIP peer ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: