| Summary: | ASTERISK-19301: ooh323 trunk to AVAYA | ||
| Reporter: | M Crespo (mcrespillo) | Labels: | |
| Date Opened: | 2012-02-06 09:11:30.000-0600 | Date Closed: | 2012-02-09 16:22:15.000-0600 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | |
| Versions: | Frequency of Occurrence | Constant | |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Hi team, I'm new here and I don't know if I'm right posting here. I have ASTESRISK 1.6.2.22 and Asterisk addons 1.6.2.3, connected to an AVAYA PBX (I do not manage this PBX). I crated a ooh323 trunk and I can make calls to the AVAYA but I only have one way audio. In the CLI i can see the established channles, but as I said, I have only one way audio. the connection is like this: ASTERISK--------ooh323trunk--------AVAYA audio: ASTERISK--------------------------------->AVAYA ASTERISK<--NO AUDIO IN THIS DIRECTION-----AVAYA The only codec I use is ulaw. in the CLI i can see this: == Using SIP RTP CoS mark 5 -- Executing [4509@internal:1] NoOp("SIP/2999-00000002", "") in new stack -- Executing [4509@internal:2] Dial("SIP/2999-00000002", "OOH323/4509@avaya") in new stack -- Called 4509@avaya -- OOH323/avaya-08b77f40 is ringing -- OOH323/avaya-08b77f40 answered SIP/2999-00000002 [Feb 6 12:08:01] WARNING[1420]: chan_ooh323.c:982 ooh323_indicate: Don't know how to indicate condition 20 on ooh323c_o_3 -- Packet2Packet bridging SIP/2999-00000002 and OOH323/avaya-08b77f40 I read some bugs about this, but nothing like to my problem Any help? Regards. | ||
| Comments: | By: Matt Jordan (mjordan) 2012-02-09 16:22:02.755-0600 Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.4 and 1.6.x branches has ended. For continued maintenance support please move to the 1.8 branch which is a long term support (LTS) branch. For more information about branch support, please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions. After testing with Asterisk 1.8, if you find this problem has not been resolved, please open a new issue against Asterisk 1.8. In addition, chan_ooh323 is an extended support module. As such, it is supported by the Asterisk community, and response time for issues may reflect that. You may want to contact the asterisk-users mailing list for additional help. | ||