similar to: Full transfer details on inbound calls

Displaying 20 results from an estimated 7000 matches similar to: "Full transfer details on inbound calls"

2009 Aug 20
8
mysql sip realtime
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
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 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
2011 Feb 28
2
Asterisk 1.8.3-rc3 and one way audio
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)
2011 Jan 05
1
Blind Transfer not working - 1.4.38
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
2010 Dec 17
2
Asterisk Freeze In 1.4 realtime
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
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
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 Aug 11
5
Trouble with *8 Pickup
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:
2010 Jul 28
2
Answered call not bridged
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
2010 Mar 29
5
Continue a dialplan when the client hang up the call
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
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
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
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
2011 Feb 11
3
Asterisk 1.8.3
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
2011 Feb 03
1
MeetMe and admin users
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
2011 Feb 22
3
Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
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
2011 Feb 22
3
Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 Now Available
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
2009 Dec 11
3
Calls Dropping
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:
2011 Apr 13
11
Realtime SIP & peer status
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: