[Freeswitch-users] OneWay audio after hold->unhold

Bote Man bote_radio at botecomm.com
Tue Sep 8 02:03:17 MSD 2015

Actually, yes I need all of the settings on the Avaya side. 


I am not using encryption and I never could get the Avaya CS1000M to talk to the FreeSWITCH installation. I think the closest I could get was the signaling would allow the call to be answered, but I got no audio to pass either direction. 


I think the Avaya puts information in unexpected elements of the SIP message and I had no more time to troubleshoot it because it was not my primary task.


There must be some small setting that I am missing and I would love to find the missing piece of the puzzle.







From: Prashanth Devarajappa
Sent: Monday, 07 September, 2015 09:47
Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold


Avaya PBX Version Control

§  Media encryption over IP:Y

§  Session Manager Element Manager 6.3

§  System Manager Release 6.3.5

§  Communication System Management 6.3.10


In IP-Network-region :

§  Codec set matches group for encryption

§  Intra-region IP-IP Direct Audio: yes

§  Inter-region IP-IP Direct Audio: yes


In CM, signaling-group of type: SIP

§  Transport Method: TLS


Let me know if you need further Avaya config details.


FreeSwitch :

Its standard FS with custom (code/config) changes to enable 


·         Support for RFC 5939 : Capability Negotiation

·         Support for UNENCRYPTED_SRTCP

·         Support IP Shuffling with in ACK…… to be done J







From: Bote Man [mailto:bote_radio at botecomm.com] 
Sent: 05 September 2015 04:50
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold


We would be VERY appreciative if you could provide the configurations on the Avaya and FreeSWITCH that allow them to talk to each other. In my experience Avaya does not play well with others, and I have tried and given up.

Thank you in advance!



On Fri, Sep 4, 2015 at 12:09 PM, Prashanth Devarajappa <Prashanth.Devarajappa at enghouse.com> wrote:



I am connecting to Avaya switch using SIP trunk interface from FS with TLS & SRTP connection. This works well for inbound and outbound calls. I have an issue with media, when I try hold -> unhold operation from FS for an outbound call.


When FS tries to unhold the call by sending Re-INVITE, avaya responds with 200 OK. Then sends INVITE w/o SDP to shuffle the media(IP Shuffling, to make media flow directly b/w endpoints). FS responds with 200 OK + SDP as expected. Now avaya sends ACK + SDP where SDP has the updated connection info(IP and port of endpoint) where media should be sent. Along with this it also changes the crypto master key as per RFC due to change in connection info(c=).


On seeing this ACK, FS update the connection info(which is good) and change its own crypto master key in response. however this new master key is not sent to far end, meaning FS can decode the media it receives but other end can't decode the media FS is sending to it resulting in one way audio.


Has anyone come across this issue ?  


Is there any way to trigger Re-INVITE or UPDATE from FS so other end gets new master key ?






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