Hi Kaleem, Below are my response
is there any reason you are going for SRTP? Customer work in Financial sector. Hence its pre requisite.
I always advise customers to first setup normal link and fully test then move to SRTP. You are right. We don’t see any such issue with RTP media. Things get complicated with Encryption/secure media but that’s mandatory requirement.
when you are saying freeswitch not responding with key, are you seeing the message in FS Debug being generated? Correct. I see new key in debug logs.
then next step would be to see this in traceSM logs and see if SM100 is getting these and how is SM responding to these and if sending to CM or not. I was able to identify this issue using TraceSM. As there is no defined response for ACK, FS not sending anything back to SM. Hence I wanted to know if I can trigger UPDATE or 200 OK with my initial email to the group.
Prashanth From: kaleem rehman [mailto:k4kaleem@gmail.com] Sent: 08 September 2015 12:52 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold Hi Prashanth, is there any reason you are going for SRTP? I always advise customers to first setup normal link and fully test then move to SRTP. when you are saying freeswitch not responding with key, are you seeing the message in FS Debug being generated? if not then you will need to raise a JIRA. if you can, then next step would be to see this in traceSM logs and see if SM100 is getting these and how is SM responding to these and if sending to CM or not. thanks, kaleem On Tue, Sep 8, 2015 at 11:08 AM, <freeswitch-users-request@lists.freeswitch.org<mailto:freeswitch-users-request@lists.freeswitch.org>> wrote: Send FreeSWITCH-users mailing list submissions to freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request@lists.freeswitch.org<mailto:freeswitch-users-request@lists.freeswitch.org> You can reach the person managing the list at freeswitch-users-owner@lists.freeswitch.org<mailto:freeswitch-users-owner@lists.freeswitch.org> When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: OneWay audio after hold->unhold (Prashanth Devarajappa) ---------- Forwarded message ---------- From: Prashanth Devarajappa <Prashanth.Devarajappa@enghouse.com<mailto:Prashanth.Devarajappa@enghouse.com>> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Cc: Date: Tue, 8 Sep 2015 10:07:27 +0000 Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold Thanks for your message.
Best Route to do this is via session Manager as this is the supported method. First create a trunk link between Session manager and Freeswitch
This is how we have setup our environment and got this far. Prashanth From: kaleem rehman [mailto:k4kaleem@gmail.com<mailto:k4kaleem@gmail.com>] Sent: 08 September 2015 10:46 To: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 111, Issue 82 Hi Prashanth / Bob, Avaya CM supports sip but this is mainly used for Aura VM, Experience portal and SM and more of Avaya only devices, direct conenctions to Asterisk , FS & 3CX work but I always experienced issues due to Avaya SIP Limitations and also officially this is not supported Best Route to do this is via session Manager as this is the supported method. First create a trunk link between Session manager and Freeswitch, then register extension directly to SM and make calls between from this extension and Freeswitch, once everything is working then create a trunk between CM & SM, or use existing and setup digit routing on SM and CM. if you have any issues setting up SM and its entities then let me know. this way will be best and far most easy one. thanks, kaleem On Tue, Sep 8, 2015 at 9:54 AM, <freeswitch-users-request@lists.freeswitch.org<mailto:freeswitch-users-request@lists.freeswitch.org>> wrote: Send FreeSWITCH-users mailing list submissions to freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request@lists.freeswitch.org<mailto:freeswitch-users-request@lists.freeswitch.org> You can reach the person managing the list at freeswitch-users-owner@lists.freeswitch.org<mailto:freeswitch-users-owner@lists.freeswitch.org> When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: OneWay audio after hold->unhold (Bote Man) 2. Re: Compiling under SmartOS (Stanislav Sinyagin) 3. Re: OneWay audio after hold->unhold (Prashanth Devarajappa) ---------- Forwarded message ---------- From: Bote Man <bote_radio@botecomm.com<mailto:bote_radio@botecomm.com>> To: "'FreeSWITCH Users Help'" <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Cc: Date: Mon, 7 Sep 2015 18:03:17 -0400 Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold 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. Thanks! Bote 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 ☺ Prashanth From: Bote Man [mailto:bote_radio@botecomm.com] Sent: 05 September 2015 04:50 To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@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! Bote On Fri, Sep 4, 2015 at 12:09 PM, Prashanth Devarajappa <Prashanth.Devarajappa@enghouse.com<mailto:Prashanth.Devarajappa@enghouse.com>> wrote: Hello, 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 ? Regards Prashanth _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ---------- Forwarded message ---------- From: Stanislav Sinyagin <ssinyagin@gmail.com<mailto:ssinyagin@gmail.com>> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Cc: Date: Tue, 8 Sep 2015 08:55:37 +0200 Subject: Re: [Freeswitch-users] Compiling under SmartOS See the update at https://freeswitch.org/jira/browse/FS-7967 I fixed the compilation problems, and now there's a runtime issue. On Mon, Aug 17, 2015 at 10:40 AM, Stanislav Sinyagin <ssinyagin@gmail.com<mailto:ssinyagin@gmail.com>> wrote:
I see there are some people on the list, working with SmartOS.
The current master fails to compile: https://freeswitch.org/jira/browse/FS-7967
Your input will be appreciated.
I just started looking around and getting the feeling what SmartOS is. I worked with Solaris quite a lot, but that was almost 10 years ago.
---------- Forwarded message ---------- From: Prashanth Devarajappa <Prashanth.Devarajappa@enghouse.com<mailto:Prashanth.Devarajappa@enghouse.com>> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Cc: Date: Tue, 8 Sep 2015 08:53:42 +0000 Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold Hi Bob, I am told that our Avaya is “avaya communication manager is the 8300” So far with our testing, we have not seen any issues related to transfer. (May be we have not done extensive transfer tests) Prashanth From: Bob Hartwig [mailto:bobjectsfreeswitch@gmail.com<mailto:bobjectsfreeswitch@gmail.com>] Sent: 07 September 2015 17:42 To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Subject: Re: [Freeswitch-users] OneWay audio after hold->unhold Is this an Avaya AS5300 by any chance? Those have some serious interop issues when it comes to transfers. I had to struggle mightily to get a client's product (not Freeswitch) to interoperate with it. I'm really interested to hear if this gets resolved for Freeswitch. On Fri, Sep 4, 2015 at 11:09 AM, Prashanth Devarajappa <Prashanth.Devarajappa@enghouse.com<mailto:Prashanth.Devarajappa@enghouse.com>> wrote: Hello, 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 ? Regards Prashanth _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org