We were finally able to connect using the URL option in the final one
hour.
When trying to download the one-x agent softphone, it takes me to the
plds and I believe requires licensing.
I could not find the one-x download on devconnect. Is this available
?
We want to test our SSC application.
Can we use any Sip softphone ? Do we need to configure extensions in
CM ?

thanks

I am trying to connect dev test lab 5 VPN using Open VPN connect and getting connection timeout
attached DevConnect-RemoteLabs.ovpn and screenshot of open vpn error

thanks

I have a general question on the product - IP Office server edition and the webrtc webclient.
Is this supposed to be an on-premise product or can it be used over the internet on the cloud ?

We are trying to use the webclient over the internet but are unable to get media on any call.

FYI, the One-X workplace desktop client works over the internet and we are able to get audio on the call.

The WebRTC WebClient does not.

Is the WebRTC webclient for the IP Office Server Edition inherently not supposed to be used over the internet without VPN ?
Or is it just configuration to enable this ?

Thanks

I have a general question on the product - IP Office server edition and the webrtc webclient.
Is this supposed to be an on-premise product or can it be used over the internet on the cloud ?

We are trying to use the webclient over the internet but are unable to get media on any call.

FYI, the One-X workplace desktop client works over the internet and we are able to get audio on the call.

The WebRTC WebClient does not.

Is the WebRTC webclient for the IP Office Server Edition inherently not supposed to be used over the internet without VPN ?
Or is it just configuration to enable this ?

Thanks

I am trying to use the WebRTC sample app for IP Office.
I have changed the configuration to match my system config.
However it is giving failure to connect.
I have set the GateWay IP and the port to 8443. However when trying to connect it is using 9443.

Any help would be greatly appreciated.

I am trying to log on the IP Office using the sample JS app provided with the SDK download on Chrome.

I have entered the FQDN
Port as 8443
TLS - YEs
Token - No
UserName - extension@domain
PW -

It asks permission to use the mic and speaker which I allow,

MediaEngine: Device unavailable: Mic AvayaClientServices-4.6.0.69.min.js:1 [
MediaEngine: Device unavailable: Speaker
MediaEngine: Device unavailable: Camera
NetworkProvider: CallGatewayProvider,Network error occurred. Status: 400001
CSGPresenceAndCallProvider: CSG presence provider is unavailable
PresenceService: Service is unavailable
CSGPresenceAndCallProvider: Network is unavailable: REST


I have tested with telnet and I can connect to the FQDN on port 8443.

What could be wrong ?

Thanks

Thanks John and Martin.

Can you please confirm if ANI (Caller ID) of an outside caller and the DNIS (The number the caller dialed for an incoming call) would be available on the Sip packets ?

We have setup a test AvayaLab and have been able to make internal calls. We are trying to get external lines connected so that we would be able to get external calls as well. In our internal calls sip captures we do see a CallID and AV-Global-Session-ID on the sip packets. Can these be linked to the call UCIDs ?


P-Asserted-Identity: <sip:9001@avayalab.local>
P-Location: SM;origlocname="Core1";origmedialocname="Core1";orighomelocname="Core1";termlocname="Core1";termmedialocname="Core1";termhomelocname="Core1";smaccounting="true"
Av-Global-Session-ID: 8708b560-8ac7-11e8-828a-005056874e76
Av-Call-Appearance: <sip:9002@avayalab.local>;+avaya-cm-line=1
Endpoint-View: <sip:9001@avayalab.local;gr=ebae5d91dd2490dc2eb2825e9ac128f3>;local-tag=-12965a385b4fa03d60ed88fc_T900110.245.15.8;call-id=8713969c8ac741e897800505687ed2e;remote-tag=8713966a8ac741e8977f0505687ed2e
RSeq: 1
Require: 100rel
Record-Route: <sip:192.168.14.153:5061;transport=tls;lr>
Record-Route: <sip:sm1@192.168.14.152;av-asset-uid=rw-2d44d717;lr;transport=TLS>
Record-Route: <sip:127.0.0.2:15061;transport=tls;lr;ibmsid=local.1531442497112_222061_222477;ibmdrr>
Record-Route: <sip:127.0.0.2:15060;transport=tcp;lr;ibmsid=local.1531442497112_222061_222477;ibmdrr>
Record-Route: <sip:sm1@192.168.14.152;av-asset-uid=rw-2d44d717;lr;transport=TCP>
Accept-Language: en
Contact: <sip:192.168.14.153:5061;transport=tls;gsid=8708b560-8ac7-11e8-828a-005056874e76;epv=%3Csip:9001%40avayalab.local%3Bgr%3Debae5d91dd2490dc2eb2825e9ac128f3%3E>
User-Agent: Avaya one-X Communicator/6.2.12.20 (Engine GA-2.2.0.169; Windows NT 6.2, 64-bit)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY, REFER, INFO, PRACK, PUBLISH, UPDATE
Supported: 100rel, histinfo, join, replaces, sdp-anat, timer
To: <sip:9001@avayalab.local>;tag=87a85668ac741e897720505687ed2e
From: <sip:9002@avayalab.local>;tag=6e39950e5b4fa03b5b605eb8_F900210.245.15.7
Server: Avaya CM/R017x.01.0.532.0 AVAYA-SM-7.1.3.0.713014
Call-ID: 5b_10b9fb-722e373d5b605a37_I@10.245.15.7
CSeq: 92 INVITE
Via: SIP/2.0/TCP 10.245.15.7:49849;branch=z9hG4bK5c_10bc6c31263bbd5b605fb0_I9002
Content-Type: application/sdp
Content-Length: 178


Thanks again.

We want to implement the following method for call recording -

Setup packet mirroring on the agent phones for media RTP
C++ TSAPI application to get call metadata

My questions are as follows -

1. Will we be able to get the RTP information (ports and IPs) on the TSAPI call events ?

2. Do we require any licenses to use TSAPI events ?

3. If TSAPI feed requires additional licences we could decode the Sip packets to get the metadata as well as the RTP information. Will the Sip packet provide all call metadata such as ANI, DNIS, Extension etc. Are there any additional information in the TSAPI call event not available in the Sip packets ?

I thank you in advance for your answer.

Debendra Modi

I may be doing something really stupid but I could not figure out what. I am testing call recording using both MR and SSC on remote labs. To test my application, I dial from one extension to another and record the call. I am unable to send audio on the VPN, and therefore I expect silence audio media to be streamed.

For both MR and SSC, I get the media, however the payloads are vastly different (caputred on wireshark) -

For MR, the payload data is the following -

80007136a73e027c43b47936fe7e7e7ffffefe7efe7fff7e7ffffeff7fff7ffe7efefefe7ffefe7e7e7f7efffeff7e7efefefefe7e7e7efffe7efefefffffeff7eff7ffffffffe7ffefe7eff7f7f7f7efe7efe7efffefeff7ffeff7fff7f7e7ffe7f7f7efefefefffe7eff7e7ffffefe7f7f7efe7ffe7f7f7f7ffffeff7e7efe7efeffff7efe7f7e7eff7efe7efffe7efffffeff7e7f7efffffe7f7ffe7eff7f7efe7ffffefffeff7efeff7f

For SSC, the payload data is -

8000460c0e07801856bd3b6effffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffffff

The size of the payload for both is the same.

For MR, I have 5 extensions configured 40016, 40017..till 40020.

For SSC, I configure 40016, 40017 and 40018 as my calling extensions. I have used 40019 and 40020 as my recording device extensions.

The media parameters are the same for both.

Any hint on how to debug the issue ?

Thanks

Hi John

Thanks for identifying the issue. I am sorry that I did not look hard enough to locate the error.

Thanks again for your time. Everything working OK now after correcting the IP.

Best regards,

Hi John

I had done a capture, but did not get any packets.

I went back to the logs of the 24th and found some additional messages in the log for this failure. Does this message provide any clue to you of what could be wrong -

<132>Oct 24 20:34:31 aeserver DmccMain[20474] -06:00 2017 568 1 com.avaya.aes | :NIO-ChannelServicer Thread: com.avaya.mvcs.h323.MsgStream WARNING - could not decode Q931 message: Failure decoding com.avaya.mvcs.h323.Q931Msg:#012Q.931 Message#012{#012 Call Reference Value: 0#012 Message Type: #012 h323_uu_pdu:#012 {#012 h323_message_body: #012 {#012 nullChoice:#012 }#012 #012 }#012 #012}#012


This message immediately follows my RTP forwarding messages. I have added this log excerpt of the 24th in the attached word file.

Thanks again for your kind help.

Sorry for the delay in my reply as I was waiting to get another remote lab access.

Answers to Martin's questions are -

1. The application actually asked for the recorder to be SSCed into the call
YES

2. The request succeeded
YES

3. The application received a MediaStart event.
YES

Attached is a word document that has the screen shot of the SAT as advised by John and the DMCC trace excerpt for the period. I do not see any error corresponding to the media. The RTP port numbers in the DMCC logs are the same as that I receive on media start.

What could be wrong ?

Thanks

I was on the remote labs earlier today and I was trying to capture media for both MR as well as SSC recording methods. For MR I was using extensions 40016 till 40020 as both monitored devices and recording devices. For SSC I had configured 40016, 40017 and 40018 as MD and 40019 and 40020 as RD's. For the RD I had configured mediainfo as my capturer VPN IP and ports 4502 and 4504 for SSC and for MR 4502, 4504,4506,4508 and 4510. I was using Avaya X Communicator as my agent soft phones to dial calls from one extension to another.

I was also doing a wireshark capture on the VPN adapter. Wireshark screen shots for both SSC and MR is given in the attached document.

Briefly, for MR, I see RTP being sent to port 2788 and UDP packets being sent to my configured ports 4502 and 4506. I am able to capture media.

For SSC I only see RTP on port 2788 and no udp to my configured ports and I am not able to record media.

Is there any reason why RTP is sent on a non-configured port and not getting udp packets for SSC recording method ?

Thanks in advance.

I was searching for my error when I stumbled on this old post.

I was on the remote labs earlier today. In my single step conferencing application, I had configured 3 stations - 40016, 40017 and 40018 as the stations being monitored.

I registered recording devices on extensions 40019 and 40020. I also received an initial hook switch updated event with the hook state as true. (onhook)

I called from 40016 to 40017 and conferenced in 40019. The SingleStep Conference response was -

<?xml version="1.0" encoding="UTF-8"?><CSTAErrorCode xmlns="http://www.ecma-international.org/standards/ecma-323/csta/ed3"><stateIncompatibility>invalidDeviceState</stateIncompatibility></CSTAErrorCode>


What could be wrong ?

Thank you in advance.

We wish to develop a call recording application using the single step conferencing method.

We already have a recorder which can record IP packets. I assume this recorder would act as DMCC softphone devices in the client mode. A recording controller application would receive TSAPI events and conference in the DMCC device when required. We need to develop the recording controller application.

Is my understanding as stated above correct ? Can anyone guide me to sample applications which would be helpful for this development.

Thank you

Dave