Hi, there's some news:
First thing, a clarification, in this test I've used sipx v.3.10.2 and
not 3.11.12...
Second, I've updated Cisco vg IOS to version 12.4(15)T9, now the vg
doesn't crash but logging with (debug ccsip error event message )
returns this error:
Sep 17 16:36:48.443:
//366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
SIP: (366) Attribute ptime, level 1 instance 1 not found.
Sep 17 16:36:48.443:
//366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
SIP: (366) Attribute ptime, level 1 instance 1 not found.
Sep 17 16:36:48.447:
//366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
Sep 17 16:36:48.447:
//366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo: Unable
to find the proper instance for FMTP
SIP: (366) fmtp attribute, level 1 instance 0 not found.
Sep 17 16:36:48.447:
//366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo: Unable
to acquire event mask for rfc2833 dtmf relay
Sep 17 16:36:48.451: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued
event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
Sep 17 16:36:48.451: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued
event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Sep 17 16:36:48.463: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Does anyone know what that means?
Nobody uses Cisco gateways? Audiocodes gw are affected from this problem?
The log from vg is in attachment.
Thanks
Gmb ha scritto:
Hi,
i've deployed this scenario:
a sipx v.3.11.12-015003 in HA configuration connected using a SIP
Trunk with a Cisco voice gateway
used for connecting another traditional pbx, this is an hybrid
solution where traditional phones can call
sip phones and viceversa.
Unfortunatly i've noticed that directed call pickup doesn't work when
i try to pick up from a sip phones
a call from traditional phones to sip phones, pick up between sip
phones works properly.
For example I've three phones with extension 100, 200, 201. 100 try to
call 200, 200 is ringing and 201 try to pickup the call:
Traditional phone(100) --> Traditional PBX --> Voicegw --> Sipx -->
Polycom phone(200)
Polycom phone(201) try to pickup callwith *78200
pickup doesn't work and Cisco VG crashes with following error and reboot.
/
System returned to ROM by bus error at PC 0x60FF86F0, address 0x0 at
09:49:55 MET+1 Wed Aug 5 2009
System restarted at 09:51:33 MET+1 Wed Aug 5 2009
System image file is "slot0:c3725-ipvoicek9-mz.124-15.T7.bin"/
I use Polycom soundpoint phones, firmware version 3.1.2 and Voice
Gateway is a Cisco 3700 firmware version 12.4(15)T7
Has anyone else had a similar issue?
Thanks
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route:
<sip:10.102.227.16:5060;lr;sipXecs-rs=%2Afrom%7ENDJFOEU3NUItNUNBQjBCMjI%60%21b08b866ad24dae05908557c321fc97af>
Via: SIP/2.0/TCP
10.102.227.16;branch=z9hG4bK-sipXecs-0296f7b900ac6b1cabf89fd2365a85a7050c
Via: SIP/2.0/UDP
10.102.227.16;branch=z9hG4bK-sipXecs-0293a8552dfca4c6094ef914102e030a6bd6~b55c549992575e7b3cd67523ac2c26c0
Via: SIP/2.0/UDP 10.102.138.226;branch=z9hG4bK27daef301003308F
From: "Riccardo " <sip:[email protected]>;tag=42E8E75B-5CAB0B22
To: <sip:[email protected];user=phone>
Cseq: 1 INVITE
Call-Id: [email protected]
Contact: <sip:[email protected]>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/3.1.2.0392
Accept-Language: en
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 18
Content-Type: application/sdp
Content-Length: 274
Date: Thu, 17 Sep 2009 14:36:46 GMT
Replaces:
[email protected];to-tag=1781CE4-5A8;from-tag=E11B72EC-1F06F5A1
Require: replaces
v=0
o=- 1253198203 1253198203 IN IP4 10.102.138.226
s=Polycom IP Phone
c=IN IP4 10.102.138.226
t=0 0
a=sendrecv
m=audio 2224 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
Sep 17 16:36:48.443:
//366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
SIP: (366) Attribute ptime, level 1 instance 1 not found.
Sep 17 16:36:48.443:
//366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
SIP: (366) Attribute ptime, level 1 instance 1 not found.
Sep 17 16:36:48.447:
//366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
Sep 17 16:36:48.447:
//366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo: Unable to
find the proper instance for FMTP
SIP: (366) fmtp attribute, level 1 instance 0 not found.
Sep 17 16:36:48.447:
//366/5D90487B83AE/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo: Unable to
acquire event mask for rfc2833 dtmf relay
Sep 17 16:36:48.451: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event
from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
Sep 17 16:36:48.451: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event
from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Sep 17 16:36:48.463: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP
10.102.227.16;branch=z9hG4bK-sipXecs-0296f7b900ac6b1cabf89fd2365a85a7050c,SIP/2.0/UDP
10.102.227.16;branch=z9hG4bK-sipXecs-
0293a8552dfca4c6094ef914102e030a6bd6~b55c549992575e7b3cd67523ac2c26c0,SIP/2.0/UDP
10.102.138.226;branch=z9hG4bK27daef301003308F
From: "Riccardo " <sip:[email protected]>;tag=42E8E75B-5CAB0B22
To: <sip:[email protected];user=phone>
Date: Thu, 17 Sep 2009 14:36:48 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Sep 17 16:36:48.463: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP
10.102.227.16;branch=z9hG4bK-sipXecs-0296f7b900ac6b1cabf89fd2365a85a7050c,SIP/2.0/UDP
10.102.227.16;branch=z9hG4bK-sipXecs-
0293a8552dfca4c6094ef914102e030a6bd6~b55c549992575e7b3cd67523ac2c26c0,SIP/2.0/UDP
10.102.138.226;branch=z9hG4bK27daef301003308F
From: "Riccardo " <sip:[email protected]>;tag=42E8E75B-5CAB0B22
To: <sip:[email protected];user=phone>;tag=1784C7C-2699
Date: Thu, 17 Sep 2009 14:36:48 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=3
Content-Length: 0
Sep 17 16:36:48.471: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Contact: <sip:[email protected]>
From: "Riccardo " <sip:[email protected]>;tag=42E8E75B-5CAB0B22
To: <sip:[email protected];user=phone>;tag=1784C7C-2699
Call-Id: [email protected]
Cseq: 1 ACK
Max-Forwards: 20
Via: SIP/2.0/TCP
10.102.227.16;branch=z9hG4bK-sipXecs-0296f7b900ac6b1cabf89fd2365a85a7050c
Content-Length: 0
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