|Summary:||ASTERISK-10556: No ring tone is heard when calling a channel after the calling channel has been answered|
|Reporter:||David Lublink (dlublink)||Labels:|
|Date Opened:||2007-10-17 08:42:54||Date Closed:||2011-06-07 14:03:14|
|Environment:||Attachments:||( 0) sip.debug|
If we have the following dialplan
exten => 101,1,NoOp()
exten => 101,n,Answer()
exten => 101,n,Dial(SIP/david-grandstream|30)
And I call 101 from a SIP device (assuming that the device has access to the context), no ring tone is heard.
Additionally, when calls come in on an IAX channel ( from a third party provider ), the same issue is occuring.
The above example is never used, but is the simplest way to reproduce the problem. The problem is seen in my IVR when someone presses an option, they don't hear anything until the line is answered or sent to voicemail (answered).
It should be noted that this issue also occurs in asterisk 1.2.
****** ADDITIONAL INFORMATION ******
I have downloaded asterisk 1.4.13 and will test to see if the issue occurs in 1.4.13 as well.
|Comments:||By: David Lublink (dlublink) 2007-10-17 08:50:18|
Currently using Debian etch with asterisk compiled from sources.
By: David Lublink (dlublink) 2007-10-17 09:09:18
Here is the sip console with SIP debug activated when I called my number.
By: David Lublink (dlublink) 2007-10-17 09:10:26
It should be noted that I updated my asterisk to 1.4.13 and the sip.debug file is taken from 1.4.13 ( the latest version on the website right now).
By: David Lublink (dlublink) 2007-10-17 09:12:21
For the SIP to SIP no ring problem, I tried the following:
GXP-2000 -> GXP-2000
GXP-2000 -> Linksys PAP2
GXP-2000 -> Sipura 2001
By: Joshua C. Colp (jcolp) 2007-10-17 10:12:42
Please attach a sip debug with console output in it as well. It's hard to piece things together for this scenario without that.
By: Joshua C. Colp (jcolp) 2007-11-28 10:12:58.000-0600
Suspended due to lack of feedback, if this is still an issue and you have the info reopen with it. Peace.