Summary: | ASTERISK-17565: Call transfer Avaya - Asterisk ooh323 one way audio | ||
Reporter: | Gianpaolo (red_emotion) | Labels: | |
Date Opened: | 2011-03-16 04:11:17 | Date Closed: | 2011-07-23 01:03:59 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Addons/chan_ooh323 |
Versions: | 1.8.4 | Frequency of Occurrence | |
Related Issues: | |||
Environment: | Attachments: | ||
Description: | I have an Avaya IP Office 403 working in H.323. I managed to connect it to my asterisk server through library ooh323 ... even though I was forced to set as default codec ULAW (which is quite heavy), though the quality is acceptable. My problem born when I do a call transfer to another extension of AVAYA: A = internal of avaya (H.323) n.1 B = internal of avaya (H.323) n.2 C = internal of asterisk (sip) Scenario: A calls B, B runs the call of A to C. B and C (Avaya and Asterisk) are communicating properly, but when they turn the call B of A to C, C hears the voice of A, but A does not hear the voice of C. I read on some forums that might be some problem ooh323, but I do not know where to put hands ... I hope I explained well ... If you have any suggestions let me know Thanks | ||
Comments: | By: Alexander Anikin (may213) 2011-03-16 15:52:21 Hi, can you upgrade your asterisk from 1.4 to latest released 1.8? 1.4 include older version of ooh323 channel driver which have number of bugs. By: Leif Madsen (lmadsen) 2011-04-12 08:50:40 Closed. No response from reporter. Assuming upgrading to 1.8 fixed the issue. |