Summary: | ASTERISK-04330: 2nd call with a Cisco Gateway produces One-Way-Audio | ||
Reporter: | Andreas Anderson (aanderson) | Labels: | |
Date Opened: | 2005-06-03 02:21:59 | Date Closed: | 2011-06-07 14:10:22 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/Interoperability |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) sip-debug.txt | |
Description: | An incoming call (1111111111) from PSTN is forwarded to a SIP-Phone (123) and a cellphone (2222222222) via the same gateway (SIP/123&SIP/2222222222@gateway). If i pick up the call on the SIP-Phone everything is fine, but if i pick it up on the cellphone i have one-way-audio. A sip debug of this case is in Additional Information. The same problem happens, if i'm on a call on the SIP-Phone, and a second caller comes in via PSTN, if i pick up i can hear the PSTN Caller but he can't hear me. | ||
Comments: | By: Olle Johansson (oej) 2005-06-04 07:04:50 All SIP debug output has to be added as a file attachment, not inline in the bug report to make it easier to handle for all of us. Thank yoU! All these devices are inside of a NAT. Is your Asterisk also on the same LAN? By: Andreas Anderson (aanderson) 2005-06-04 08:40:10 debug added as a file, but i cant remove the stuff in "Additional Information". asterisk and the cisco are on the same subnet, nat is disabled in sip.conf. By: Olle Johansson (oej) 2005-06-04 09:14:21 Can you please capture all of the conversation, including the first INVITE. THis trace starts at the "Trying"... By: Andreas Anderson (aanderson) 2005-06-08 08:18:15 It turned out this was a problem with the BRI from the telco, the second B-Channel was "broken", it always produced one-way-audio?? After seven calls to the telco it works again, even if they have "not done anything", some magically elves must have fix it... Sorry for opening a bug on this. I guess thats one with the karma-whip then :-] |