NO VOICE IN OXO - OXO BREAK IN THROUGH SIP LINK

[Kanon79]

Hi evereybody.

I got 2 oxos with upgrade from R7 to R10. These oxos had a H.323 link. When the oxo´s had in release 7 All external incoming calls arrives to the AA of Site A and via ARS table and collective speed dialing external users can achieve extension from site B without problems. Now when I made the upgrade to R10 i changed from H.323 to SIP by customer desicion. After the change all calls between Users of Site A and Site B are OK but when an external user call to AA from site A and ask for an extension of site B the call is procesed and ring in the suscriber set but when answer the call there's no voice. After many tests i found that if the user of site B answer the call, detect no voice and then put the call in hold and take it back again .... the voice start to working.
Also if i set a dinamic routing from site A to site B and an external user ask for the extensión forwarded the call go to site B without problems.

Also I tried to use H323 in R10 but when i change de protocol the IP trunk crash and all comms on site A. and site B. fail

Any advice, what can it happens?

The Oxo of Site A has 2 Pri board and 30 SIP Channels
The Oxo of site B has 1 Pri board for outgoing calls and 30 SIP channels

Numbering from Site A: 1XXX
Numbering from site B: 2XXX


I attach a trace of one call from an external number to ext. 2552 in site B. via AA from site A. (Is the same call, but samples was taken in each oxo)
I found this error in trace from site B: voip/USR_VOIP/code/gw_siptrunk/lib/NET_API/SIP_Call.cppSIP - SendSuccess processing error( unable to find transaction ReInvite in Invite ) any ideas?

trace_call_siteB.txt

trace_call_siteA.txt

Thanks a lot.

Excuse me for my bad English.


Regards

[cavagnaro]

As maaaaaaany times before, network issue. Check the forum for any of those posts where this is discussed.

[fe80]

Hi,
I can't recognize a network issue in the traces provided. We can also assume the network is okay because H232 has worked well. It would be nice to have instead of txt-Traces of the SIP dialog binary pcap-files including the audio streams for analysis.

[cavagnaro]

H323 uses a total different network and port behavior than SIP, comparing them is useless. Yes we discussed this previously on the forum.
You have a symptom. No sound, means? No RTP traffic. Means? No route. Means? Network issues.


Enviado de meu C6603 usando Tapatalk

[Kanon79]

Hi Cav, fe;

Thanks for the answer; I'm agree that could be a network issue but the customer isn´t because he argument that if the internal calls between both sites are OK, theres no reason for the external calls transfered by AA dont go ok.... I already talk to ISP and firewall admin and they only said .... "all is ok here .... nothing is block or being dropped so is not our problem" ... and the customer belive them. the only thing that I note was in a trace route from OXO A to OXO B and viceversa the steps are not the same. Maybe this is the problem .... I'll open a E-sr and let you now the answer.

Regards.

[Kanon79]

tracert.pdf

[vad]

About RTP and codec - you have incoming call to OXO A via SIP and transfer call to OXO B via SIP?
In this case allowed or not RTP traffic OXO B-provider? OXO A - provider no problem, OXO A - OXO B no problem, but what about OXO B - provider?

[Kanon79]

Hi Vad;
The Incoming calls to Oxo A use ISDN Pri Access and the transfer between Oxo A and Oxo B use SIP. Codecs in both sites are by default and Direct RTP is disable in both sites. oxo in site B has a Pri only to outgoing calls. Yesterday I create a virtual extension in oxo of site A and make a forward to another ext. in site B via ARS. I call the Customer PBX, then the AA answer the call so I ask for the virtual ext. number, the call was diverted to the ext. in oxo B and the comm is ok. So this is confusing me. So by now i created lots of virtual ext. in OXO A while i solve this issue. In some cases this could be a solution but the customer has over 120 ext. in site B and 180 ext. in site B, so I can't create more than 250 users in OMC.

Regards.

[fe80]

This is an other hint showing the network is okay. Except virtual extensions are able to solve network issues. To find out what is going on you have to do TCP Dump Traces on both sites (with and without your virtaul ext.-workaround) and analyze the pcap files with Wireshark.

[Kanon79]

HI Fe80; I post 2 traces .cap files from each OXO at same time, I see a RTP port change whitout reason from site B to A. any ideas. (flow call to ext. 2552). Im still waiting for ESR answer .... by now they suggest me to upgrade the CPUs to the last versión in BPWS .... we will see!!

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