SIP THOMSON ST2030

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aflores
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SIP THOMSON ST2030

Post by aflores »

Hello All. We have some Oxe systems, all with SIP phones (Thomson ST2030), all are Rel. 7.1.

Sudenlly, only the SIP phones can not make or receive externals calls.

The solution to that, is rebboting the system, until it crash again.

I have a case open at the Alcatel-Lucent Technical support, but still I'm waiting for a solution, so if some body could giveme some direcction, on what to do, or what can be happens, I will appreciated.

Thanks and best regards to all.
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kike
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Joined: 29 Aug 2007 06:27

Re: SIP THOMSON ST2030

Post by kike »

Check the domains with the command cnx dom.
In some cases, for example when the ST2030 make transfers, there are channels that remain busy after the call end.
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kike
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Re: SIP THOMSON ST2030

Post by kike »

Sorry, I forgot something:
Thomson requires from OXE two free channels for only one call. I don't know if it's still so, but that happened to me in 7.1.
And the solution with the transfers could be to upgrade the software of the Thomson ST2030.
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aflores
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Re: SIP THOMSON ST2030

Post by aflores »

Hi Kike, We have Thomson Software version: 1.54; Boot version: 111; DSP: 101.
It was the version that you upgraded the thomsons?

The command you mentioned, have to ve done during the failure or after the failure?.

I already run the command and get these results:

(32)AMirandin> cnx dom

Fri Oct 5 15:57:24 GMT-4 2007


IP Domain

domain type : IP_R-> IP_REMOTE NIPR-> NO_IP_REMOTE

| number | 0 |
| type | NIPR |
| allowed | ffff |
| used | 0 |
| RIP Intr| G723 |
| RIP Extr| G723 |

| IPP Intr| G723 |
| IPP Extr| G723 |


cnx [ cfg | obj | cr | load | WORD_# ]


Can you please explain a little bit more, what does this comamnd is use for and how to understand the information?.

Thanks you very much for your anwsers.

Regards

AFlores
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kike
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Joined: 29 Aug 2007 06:27

Re: SIP THOMSON ST2030

Post by kike »

Hi.

It is posible to control how many simultaneous calls can be made in each IP domain (domain max voice connec., in IP>IP domain).
The cnx dom command shows you how many free channels do you have in each IP domain for making external calls (external means outside his IP domain). For each domain you will have one column (up is the number of the domain).
Allowed=ffff means that you have no restrictions, and used=n means that there are n external conversations in that moment (intra-domain calls are not restricted).
The rest of the information is clear: the codec that will be used in intra- and extra-domain calls.
In case of no channels free, “used” will be equal to “allowed”.

I see that you have not IP domains declared (except the default value 0, without restrictions), so the problem will be diferent, something with codecs or free compressors, or who knows. When the problem occurs, you can check if you have free compressors with the command “compvisu lio” (look the column of the right), despite of direct RTP they will be used in case of transcoding and for the ringing tone, music on hold, …

The best way to know what happens is to make traces, in order to see the SIP messages between the phone and OXE: ethereal. Or try with the command “traced”.

Regards,
Kike.
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rubrio
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Re: SIP THOMSON ST2030

Post by rubrio »

Hello,
When I was making some tests with those phones I had the same problem, the phones dont work, like if they are unregistered, but in the Oxe still appear registered.

I made it by changing the registering timers to make the oxe send a invite ACK to the phone more frequently, but I dont know exactly what is the correct timer (I put it on 1 second, but it on a real network will generate a lot of traffic).
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dryhouse
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Re: SIP THOMSON ST2030

Post by dryhouse »

Hi Brots,

I have one thomson ST2030 in my OXE system test. The Sip phone make internal and external calls ok, but don´t make calls from ABC destinations. I have trace sip to Thomson ST2030:

INVITE sip:56887@nodomaqueta:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.80.26:5060;branch=z9hG4bK8030259792092547043-5331118
From: "20888801"<sip:20888801@nodomaqueta:5060>;tag=c0a80101-5158ad
To: <sip:56887@nodomaqueta:5060>
Call-ID: 2796fef-c0a80101-0-f@172.26.80.26
CSeq: 1 INVITE
Max-Forwards: 70
Supported: timer, replaces
Session-Expires: 1800
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20888801@172.26.80.26:5060>
User-Agent: THOMSON ST2030 hw3 fw1.66 00-0E-50-4E-84-0E
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 298

v=0
o=20888801 5331118 5331118 IN IP4 172.26.80.26
s=-
c=IN IP4 172.26.80.26
t=0 0
m=audio 16384 RTP/AVP 18 4 8 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


Recv from udp: 172.26.83.10:5060 00:01:24:44:200 (283 bytes)

SIP/2.0 100 Trying
To: <sip:56887@nodomaqueta:5060>
From: "20888801" <sip:20888801@nodomaqueta:5060>;tag=c0a80101-5158ad
Call-ID: 2796fef-c0a80101-0-f@172.26.80.26
CSeq: 1 INVITE
Via: SIP/2.0/UDP 172.26.80.26:5060;branch=z9hG4bK8030259792092547043-5331118
Content-Length: 0



Recv from udp: 172.26.83.10:5060 00:01:24:44:210 (515 bytes)

SIP/2.0 480 Temporarily not available
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
User-Agent: OmniPCX Enterprise R9.1 i1.605.31
Reason: SIP;cause=480;text="CONGESTI N "
To: <sip:56887@nodomaqueta:5060>;tag=f163a76bd613e45da4484d214359eaa4
From: "20888801" <sip:20888801@nodomaqueta:5060>;tag=c0a80101-5158ad
Call-ID: 2796fef-c0a80101-0-f@172.26.80.26
CSeq: 1 INVITE
Via: SIP/2.0/UDP 172.26.80.26:5060;branch=z9hG4bK8030259792092547043-5331118
Content-Length: 0



Sent to udp: 172.26.83.10:5060 00:01:24:44:324 (476 bytes)

ACK sip:56887@nodomaqueta:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.80.26:5060;branch=z9hG4bK8030259792092547043-5331118
From: "20888801"<sip:20888801@nodomaqueta:5060>;tag=c0a80101-5158ad
To: <sip:56887@nodomaqueta:5060>;tag=f163a76bd613e45da4484d214359eaa4
Call-ID: 2796fef-c0a80101-0-f@172.26.80.26
CSeq: 1 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
User-Agent: THOMSON ST2030 hw3 fw1.66 00-0E-50-4E-84-0E
Content-Length: 0

The CS rejected the call....any idea??

Thanks and Regards.
may the force be with you....
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dryhouse
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Posts: 165
Joined: 06 Apr 2010 06:11
Location: Madrid,Spain

Re: SIP THOMSON ST2030

Post by dryhouse »

dryhouse wrote:Hi Brots,

I have one thomson ST2030 in my OXE system test. The Sip phone make internal and external calls ok, but don´t make calls from ABC destinations. I have trace sip to Thomson ST2030:

INVITE sip:56887@nodomaqueta:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.80.26:5060;branch=z9hG4bK8030259792092547043-5331118
From: "20888801"<sip:20888801@nodomaqueta:5060>;tag=c0a80101-5158ad
To: <sip:56887@nodomaqueta:5060>
Call-ID: 2796fef-c0a80101-0-f@172.26.80.26
CSeq: 1 INVITE
Max-Forwards: 70
Supported: timer, replaces
Session-Expires: 1800
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:20888801@172.26.80.26:5060>
User-Agent: THOMSON ST2030 hw3 fw1.66 00-0E-50-4E-84-0E
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 298

v=0
o=20888801 5331118 5331118 IN IP4 172.26.80.26
s=-
c=IN IP4 172.26.80.26
t=0 0
m=audio 16384 RTP/AVP 18 4 8 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


Recv from udp: 172.26.83.10:5060 00:01:24:44:200 (283 bytes)

SIP/2.0 100 Trying
To: <sip:56887@nodomaqueta:5060>
From: "20888801" <sip:20888801@nodomaqueta:5060>;tag=c0a80101-5158ad
Call-ID: 2796fef-c0a80101-0-f@172.26.80.26
CSeq: 1 INVITE
Via: SIP/2.0/UDP 172.26.80.26:5060;branch=z9hG4bK8030259792092547043-5331118
Content-Length: 0



Recv from udp: 172.26.83.10:5060 00:01:24:44:210 (515 bytes)

SIP/2.0 480 Temporarily not available
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
User-Agent: OmniPCX Enterprise R9.1 i1.605.31
Reason: SIP;cause=480;text="CONGESTI N "
To: <sip:56887@nodomaqueta:5060>;tag=f163a76bd613e45da4484d214359eaa4
From: "20888801" <sip:20888801@nodomaqueta:5060>;tag=c0a80101-5158ad
Call-ID: 2796fef-c0a80101-0-f@172.26.80.26
CSeq: 1 INVITE
Via: SIP/2.0/UDP 172.26.80.26:5060;branch=z9hG4bK8030259792092547043-5331118
Content-Length: 0



Sent to udp: 172.26.83.10:5060 00:01:24:44:324 (476 bytes)

ACK sip:56887@nodomaqueta:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.80.26:5060;branch=z9hG4bK8030259792092547043-5331118
From: "20888801"<sip:20888801@nodomaqueta:5060>;tag=c0a80101-5158ad
To: <sip:56887@nodomaqueta:5060>;tag=f163a76bd613e45da4484d214359eaa4
Call-ID: 2796fef-c0a80101-0-f@172.26.80.26
CSeq: 1 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
User-Agent: THOMSON ST2030 hw3 fw1.66 00-0E-50-4E-84-0E
Content-Length: 0

The CS rejected the call....any idea??

Thanks and Regards.



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Regards.
may the force be with you....
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