Alcatel Unleashed

hello all

I do have a problem with a Oxe connected to a CCM throught a SIP TG.
I'm having a one way audio problem on calls reaching Oxe extensions through CCD voice guides or automatic attendants:

(1) When a Cisco set calls an Alcatel extension directly, both devices can talk to each other; NO PROBLEM IS FOUND.
(2) When Alcatel calls to Cisco extensión, both devices can talk to each other; NO PROBLEM IS FOUND.

(3) When Cisco extension call a CCD Pilot, a voice guide is played and then the call is transfered to an available agent; Cisco user is able to hear the Alcatel user, but the Alcatel user cannot hear to the Cisco one.

(4) When a Cisco extension calls an Alcatel automatic attendant, and select an Alcatel user through the AA menu; Cisco user is able to hear the Alcatel user, but the Alcatel user cannot hear to the Cisco one.

I have checked that compression resources are available on all GDs; there are no incidents being reported on compression resources.
The problem is happening when calling Alcatel extensions connected to different MGs and using different IP Domains.
IP Domains are configured to use Def. Codec for inter and intradomain calls. The SIP TG is configured for Def. Codec as well.
Default Codec is set to G729.
G729 is being used in the Cisco side.

I have captured traces testing the scenarios above, but I'm not able to see anything incorrect in the signaling, which I guess is good, but something seems to happen to the audio flows.

I'm attaching scenarios (1) and (3) .

can you think what could be the cause of the issue?

thank you very much in advance.

Best regards

Please an output of compvisu lio

here it goes thank you.

dvid9976 wrote:here it goes

compVisuLio.bmp

SARCASM ON:
It is really great to see 2.5 KB text file attached as 810 KB BMP image.
SARCASM OFF

Did you try to check whether you can transfer a succesfull call(both-way audio) to another OXE set (not a CCD agent) in another IP domain?
How is your SIP TG is managed - as ISDN of ABC-F?

dvid9976 wrote:here it goes

compVisuLio.bmp

thank you.

compvisu sys too please and attach as text only

alex wrote:

dvid9976 wrote:here it goes

compVisuLio.bmp

SARCASM ON:
It is really great to see 2.5 KB text file attached as 810 KB BMP image.
SARCASM OFF

Did you try to check whether you can transfer a succesfull call(both-way audio) to another OXE set (not a CCD agent) in another IP domain?
How is your SIP TG is managed - as ISDN of ABC-F?

I thought that the image thing would be a good idea to keep the format.. did not realize about the size, sorry guys x-D

- I stablished a call between Alcatel and Cisco; both sides talked to each other.
- I transfered then the call to another Alcatel under a different IP domain set as default codec for inter/intradomain.
- This second Alcatel phone rings and before the call is answered, the call drops.

TG is configured as ABC-F.

see compvisu sys (in text format this time )

+==============================================================================+
| C O M P V I S U |
+==============================================================================+
| Inter-node protocol H323....... yes |
| RTP Direct..................... yes |
| RTP Direct for H323 terminals.. yes |
| Fast Start..................... yes |
| VAD (Voice Activity Detection): |
| - G723/G729...... no |
| - G711........... yes |
| ECE (Echo Canceller)........... yes |
| - LIO/LIOE........ 16 ms |
| - INTIP/GA/GD..... 128 ms |
| PFE (Post Filter).............. yes |
| Volume ........................ 8 |
| Volume for IP Phone ........... 0dB |
| Volume for other device. ...... 0dB |
| VRE ........................... no |
| Law (Except Media Gateway)..... A law |
| Global compression type ....... G729 |
| Multi-algorithm (for H323) .... yes |
| Compression for INTIP/GD ..... with |
| Compression for IPP ........... with |
| Transit on IP Boards ...........yes |
| ticket Stat IP................. yes |
| IP version..................... IPv4 |
| Transit compatibility.......... no |
| Voip Framing G711 ............. 20 ms |
| Voip Framing G723 ............. 30 ms |
| Voip Framing G729 ............. 20 ms |
| No RBT For Direct RTP H323..... no |
| T38 FAX........................ yes |
+==============================================================================+

to walkaround the problem we have set a qsig between Oxe and CCM until we can find a solution..

thank you very much guys

Best regards

dvid9976 wrote: I thought that the image thing would be a good idea to keep the format..

It's OK. To keep text format use BBCode "Code" tags.

dvid9976 wrote: This second Alcatel phone rings and before the call is answered, the call drops.

TG is configured as ABC-F.

Show the trace for this case.

Also show us an output of "motortrace c" command.

Attached "motortrace c" output.

working to get a maintenance window to test/capture the proposed scenario (the customer is so happy with the Qsig that it is hard to go back to SIP for testing :-S)

thank you

Proxy parameters.
=================
sip stack version 4.0.006.020
initial_timeout 500
timer_t2 4000
recursion 0
min_auth_method 0 NONE=0 DIGEST=2
auth_realm
sipDnsTimerPrimSecond 5000
onlyAuthIncomingCalls 0
quarantine and trusted addresses:
nb_msg_by_period 25
period 3
framework_quarantine_period 1800

Gateway parameters.
===================
url_install 10.1.100.2
url_gw
url_hostname nodo1
num_ss_reseau 10
num_faisc 110
proxy_address not used
DNS_localDomName
DNS_type 0 dnsa=0, dnssrv=1
DNS_primaire Unused
DNS_secondaire Unused
prack_required 0
out_proxy 0 AUCUN=0 INTEGRE=1 EXTERNE=2
proxy_port 5060
proxy_transport 1 TCP=0 UDP=1
sipSubsMinDuration 1800
sipSubsMaxDuration 86400
sipSessionTimer 1800
sipMinSessionTimer 900
SessionTimerMethod 0 re-invite=0, update=1
sipCac 1
SDP_in_180 1
sip_info_enable 0
payload 97
seplos 1

Registrar parameters.
=====================
declared_sets 0
min_expiry_date 1800
max_expiry_date 86400

CPU address.
============
physical local 10.1.100.2
physical twin
role Main 10.1.100.2

Licenses parameter.
=====================
nb max licences 22

hello there

We are back to SIP again but with some differences.
We are using now G711 for Inter/Intra Domain calls and for the SIP TG as well, which seems to be working fine. No longer one way audio is detected and automated attendants / Pilot calls are working fine.

The idea is to test with G729 at any point, to double check and correct the G729 issues.

I will post again as soon as I'm able to test properly and update you with comments, questions or solutions on the findings.

thank you very much again for your interest and your time.