SX10 SIP configuration
Hello
I just got a cisco telepresence SX10 device uses the SIP protocol only and I have no experience with this at all. Older devices that I worked with used H323 Protocol which is very easy to configure. Looking for help on how to configure the SIP. Your help will be greatly appreciated
Thank you
The Radvision MCU was implemented to support SIP?
Tags: Cisco Support
Similar Questions
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MCU 8510 - TMS - SX10 (SIP only)
Hi all
I have a question where we have a new SX10 which I added to the MSDS and can make calls to a fine. However when I add it to a conference using the MCU it tells me that the connections settings will change, then deletes all the connections in total of parameters. Addition of 10 other endpoints at the same conference without the SX10 and it works fine (although it is trying to connect on H323 which is fine).
Why when I add the SX10 at the same conference must kill the connection settings for all units and there is no way to change?
Why can't I just MCU dial the SX10 via SIP only and the rest stay that H323?
This is what it looks like before you add the SX10 unit
It's what happens after you have added the SX10 and select OK to popup its will change the parameters of connection
Anyone has any idea why?
I just did some tests, make sure that the endpoint you are trying to add to the Conference has the same allow incoming and outgoing, planning of the options checked in TMS as your MCU.
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Failure of HTTP post when connecting via SIP
Hi all
I'm doing HTTP post, but I get corrupted in the response data.
This problem only occurs on the device (4.5) and only if you use the SIP configuration
(the problem does not when you use a device via Wifi 4.6)
Here is my code:
HttpConnection connection = (HttpConnection)Connector.open(url + ";deviceside=false");connection.setRequestProperty("x-rim-transcode-content", "*/*");connection.setRequestMethod(HttpConnection.POST);connection.setRequestProperty("Content-Type","multipart/form-data; boundary=@---------------------------123");connection.setRequestProperty(HttpProtocolConstants.HEADER_CONTENT_LENGTH, String.valueOf(data.length)); OutputStream os = connection.openOutputStream();os.write(data); if (connection.getResponseCode()==HttpConnection.HTTP_OK){ StringBuffer result = new StringBuffer(); byte[] buffer = new byte[2000]; int i = 0; DataInputStream in = connection.openDataInputStream(); while ((i = in.read(buffer)) != -1){ result.append(new String(buffer, 0, i,"ISO-8859-1")); }}Does someone know what I'm doing wrong here?
(BTW, the system cut the suffix HTTP "ConnectionType =" which should be equal to "mds" + "»' + 'public')
I guess that your assumption that the "transcode-including" in the request header method caused the problem was correct.
I changed the method request header, it shows the HttpConnection API:
... // Set the request method and headers c.setRequestMethod(HttpConnection.POST); c.setRequestProperty("If-Modified-Since","29 Oct 1999 19:43:31 GMT"); c.setRequestProperty("User-Agent","Profile/MIDP-2.0 Configuration/CLDC-1.0"); c.setRequestProperty("Content-Language", "en-US"); ...and the problem has been corrected
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SIP: Failure cannot connect to...
Hello
I just got a Quickset SX20 video system for my business. Therefore, I created a SIP account with access to the server of proxy to 'getonsip '.
I went into my settings on the SIP configuration, but I got this error: "Failed: unable to connect to 69.57.179.234:5060" as you can see in this screenshot:
Here is my system SIP configuration:
I opened port 5060 for TCP/UDP protocol on my firewall. IP address of the proxy is getonsip.com, they gave me a free Sip account. I tried other free providers, but it never worked once! (I have always sort of various errors, such as DNS or something...)
Thank you very much.
Why? Firstly your end point says that it cannot connect, check the error you posted yourself.
So, if you try to connect to the ip address you have posted at least from here only get a connection refused
$ nmap -sT -p 5060,5061 69.57.179.234
Starting Nmap 5.00 ( http://nmap.org ) at 2013-07-05 15:58 CEST
Interesting ports on nile2.junctionnetworks.com (69.57.179.234):
PORT STATE SERVICE
5060/tcp closed sip
5061/tcp closed sip-tls
Nmap done: 1 IP address (1 host up) scanned in 0.80 seconds
Some thoughts of othe:
* even if it is a "sip" supplier could not provide all the necessary capabilities for video
* If you need NAT traversal problems could place
Difference can be the feature, service, features, functionality, stability, location...
Like today, you have a problem with the supplier then ask them to fix it so that everything works for you ;-)
Cordially capabilities, a lot of video-conference calls are always placed on h323, real
video providers will probably offer you transparent connections supporting h323 and sip
and interoperability between the two.
Location/features: in some scenarios (as your endpoint is behind a nat) media may need to be
supported by the provider. This means that they must support video media that is quite
intense bandwidth, so don't not even support that (apart from technical limitations such as the sharing of content/bfcp).
Also depending on where you are and the provider, you can add delay substential.
Like nothings free, ask yourself what is the reason behind this provider to offer the service
I'm not saying that you will not find any provider of good and free and functional, but at least I'm not aware of anything.
Please note the answers using the stars below and set it to the response if it is.
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E20 hang with Sip Config - TE4.1.1 - Strange
We see that some of our E20 - most of them on the internet is suspended as soon as endpoint is started.
If the end point is without a network cable, then the E20 is stable.
If the SIP configuration is deleted, the E20 is stable.
As soon as we put in the SIP config and records the E20 on SIP, the unit freezes.
We see root that enforcement Tandberg failure - root cli tsh says 'unable to connect request.
Is this a known issue with TE4.1.1.
If we disable SIP and only use H323 - device is stable.
VCS - X7.2.1
Hello
the log files you sent from the E20 show a problem with the Marvell chip that is inside the E20 ethernet controller.
Feb 26 07:59:02 (none) principal: FPGA programmed OK
Feb 26 07:59:02 (none) main: marvell.c: ioctl (9,-2146669310,...) ==-1, errno == 14, wrong address
Feb 26 07:59:02 (none) main: platform/marvell/marvell.c:132: marvell_ioctl: Assertion ' 0 & 'Marvell ioctl failed' ' failed.
Feb 26 07:59:02 (none) main: signal received SIGABRT (6) wire 0x4082a4c0, 1704 TID
Feb 26 07:59:02 (none) principal: records:
Feb 26 07:59:02 (none) principal: R0: 00000000 R1: 000006a 8 R2: R3 00000006: 000006a 8
Feb 26 07:59:02 (none) principal: R4: R5 00000006: 40826bdc R6: 40826000 R7: 0000010 c
Feb 26 07:59:02 (none) principal: R8: 00000bdc R9: 00000000 R10: FP 4082a4c0: be976974
Feb 26 07:59:02 (none) principal: PC: 407206f8 IP: be976978 SP: be97695c LR: 407206 c 4
Feb 26 07:59:02 (none) principal: ERR: 00000000 CPSR: 20000010 FAULT: TRAP 00000000: 00000000
Feb 26 07:59:02 (none) main: OLDMSK: 00000000
Failure to our tracking system is down, so couldn't find a DDT corresponding to this, but to me, it sounds like the hardware.
Will check with engineering.
The recent change you speak may cause some E20s crashing. There is an open on this default:
CSCue59199"target ="_blank"> CSCue59199 - Boot: error SIPAUTH: cannot delete the signature (gState = 1)"
When we are challenged on a presence SIP subscribe message, the E20 crashes. When this happens, you will see a message like "SUBSCRIBE SUBSCRIBE got proxy-challenged in 407 authorization."
Now the files of historical newspapers that sent you to the specific E20 does not match this DDT. You have other E20s crashing when you activate the SIP? If so, can you send files of historical log of such a device, which now no longer crashes with disabled SIP?
Here's the sequence before the crash.
Could you gather some debug SIP and try to activate the SIP on 1 E20 please?
The tsh c:
The ctx sippacket debug log 9
When the aircraft crashes, disable SIP again and transfer of log files to check.
3 Mar 11:43:43 (no) principal: admin user (1001) executed successfully the command ' / presence / to subscribe/Start URI: 91000001' sweet-infusion - 7.cisco.com.
"3 Mar 11:43:43 (no) principal: 8241.76 SipStack I: SipEv: Active subscribe to"
' type 'presence', unsolicited = 0
3 Mar 11:43:43 (no) principal: 8241.76 I: SipUa added GRUU OK SipStack
3 Mar 11:43:43 (no) principal: 8241.77 SipPacket PacketDump: Proto: SIP, name: SUBSCRIBE
SIP/2.0, Direction: Outgoing or remoteAddress is set: 10.106.93.69:5061, GroupEntity:
Time: 8241765 content:!
3 Mar 11:43:43 (no) principal: 8241.77 SipPacket SUBSCRIBE
SIP/2.0
3 Mar 11:43:43 (no) principal: 8241.77 SipPacket Via: SIP/2.0/TLS 171.69.87.88:5061; branch = z9hG4bKb13dc154a422586a9cb016c9d7ac0a3f.1; rport
3 Mar 11:43:43 (no) principal: 8241.77 SipPacket Call-ID:
3 Mar 11:43:43 (no) principal: 8241.77 SipPacket CSeq: SUBSCRIBE 101
3 Mar 11:43:43 (no) principal: 8241.78 SipPacket Contact:
3 Mar 11:43:43 (no) principal: 8241.78 SipPacket to:
; tag = 92c18ad11538b1bd
3 Mar 11:43:43 (no) principal: 8241.78 SipPacket to:
3 Mar 11:43:43 (no) principal: 8241,78 SipPacket Max-Forwards: 70
3 Mar 11:43:43 (no) principal: road of SipPacket of 8241.78:
3 Mar 11:43:43 (no) principal: 8241.78 SipPacket User-Agent: TANDBERG/257 (TE4.1.1.271887Beta1)
3 Mar 11:43:43 (no) principal: 8241.78 SipPacket Expires: 3600
3 Mar 11:43:43 (no) principal: 8241.78 SipPacket event: presence
3 Mar 11:43:43 (no) principal: 8241.78 SipPacket Accept: application/pidf + xml
3 Mar 11:43:43 (no) principal: 8241.78 SipPacket Content-Length: 0
3 Mar 11:43:43 (no) principal: SipPacket 8241.78
3 Mar 11:43:43 (no) principal: SipPacket 8241.78 >!
3 Mar 11:43:43 (no) principal: 8242.02 SipPacket PacketDump: Proto: SIP, name: SIP/2.0 407 Proxy Authentication Required, Direction: inbound, RemoteAddress: 10.106.93.69:5061, GroupEntity:
Time: 8242018 content:!
3 Mar 11:43:43 (no) principal: 8242.02 SipPacket SIP/2.0 407 Proxy Authentication Required
3 Mar 11:43:43 (no) principal: 8242.02 SipPacket Via: SIP/2.0/TLS 171.69.87.88:5061; branch = z9hG4bKb13dc154a422586a9cb016c9d7ac0a3f.1; received = 171.69.87.88; rport = 36924
3 Mar 11:43:43 (no) principal: 8242.02 SipPacket Call-ID:
3 Mar 11:43:43 (no) principal: 8242.03 SipPacket CSeq: SUBSCRIBE 101
3 Mar 11:43:43 (no) principal: 8242.03 SipPacket to:
; tag = 92c18ad11538b1bd
3 Mar 11:43:43 (no) principal: 8242.03 SipPacket to:
; tag = aaa367e278a7ece0
3 Mar 11:43:43 (no) principal: 8242.03 SipPacket server: TANDBERG/4120 (X7.2.1)
3 Mar 11:43:43 (no) principal: 8242,03 SipPacket Proxy-Authenticate: Digest realm = "akgvcsc1.ciscolab.com", nonce = "bb98866b8387ba58270573abceefb75e19dcde22e51b036493413ef0b5cd", opaque = "AQAAAK5Mba8WRqO56xvFJpIWzjCx51zZ", stale = FALSE, algorithm = MD5, qop = "auth"
3 Mar 11:43:43 (no) principal: 8242.04 SipPacket Content-Length: 0
3 Mar 11:43:43 (no) principal: SipPacket 8242.04
3 Mar 11:43:43 (no) principal: SipPacket 8242.04 >!
3 Mar 11:43:43 (no) principal: permission to SUBSCRIBE got proxy-challenged in 407 SUBSCRIBE
3 Mar 11:43:43 (no) main: signal received SIGSEGV (11) wire 0x4084d450, TID 1809
3 Mar 11:43:43 (no) principal: illegal memory access to: 0x8
3 Mar 11:43:43 (no) principal: records:
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SPA9000/SPA941 Conference issue
HI - we have a SPA9000 and in the course of running, but we are having problems with the Conference feature.
In three Conference, there is always a phone that can not hear another thing everybody hears that conveys this phone.
Internally using SPA941 Conference functionality works very well but is not with two external callers.
We have a phone Polycon a FXS power and to make the phone Conference functionality works very well also.No idea why this is happening?
Thank you!
Hello! Thanks for your reply.
After making a few changes on the SPA941, I got this job.
That's essentially what I changed to make it work. Some of these setting changes have been suggested by our provider callcentric.com
Within your SIP configuration page:
RTP packet size: 0.020
Name of the G729a Codec: G729
Name of the G729ab Codec: G729
Manage VIA received: Yes
Manage VIA rport: Yes
Insert VIA received: Yes
Insert VIA rport: YesWithin your Ext configuration page:
NAT Mapping Enable: Yes
NAT Keep Alive Enable: Yes
Favorite Codec: G.729a
Use Pref Codec only: No. -
Hello
Can someone help me with this:
I have UC520 and lots of SPA 942 phones. How to grow? And is it possible at all?
-
Best regards
Damir
Hi Damir,
The SPA900 phones are only compatible with SIP, so you should configure the SIP on the CPU in order to record the SPA942 phones.
Please refer to this post for some sample SIP configurations: https://supportforums.cisco.com/message/3330441
Thank you
-john
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CUCME no calls incoming, outgoing calls okay
Hello everyone,
I'm setting up a CUCME with SIP trunk, I can make calls outside, but I can´t receive everything from the outside, it's my second time as a SIP configuration
I ve use debug command voice dialpeer all to check was happening, but I can´t find the problem.
This is my config:
IP server host sip - A.B.C.D
!
voip phone service
list of approved IP addresses
IPv4 A.B.C.D 255.255.255.252
!
translation of the voice-rule 1
rule 1 / 325277\ (\) / / 1\1 /.
!
voice translation-profile IN
translate 1 called
!
Dial-peer voice 1 voip
Description * incoming SIP trunk call *.
entrants IN translation-profile
session protocol sipv2
session target sip-Server
incoming called-number.
codec voice-class 1
voice-class sip dtmf-relay rtp - nte force
DTMF-relay rtp - nte
No vad
!
ePhone-dn 1
number 100
Description of RECEPTION
!
ePhone 2
address Mac YYYY. BENAMER. CCBC
ePhone-model 1
type 7942
Keep-Conference
button 1:1
NOTE: The IP address are hidden, just for safety
Here is the output from my debug/tests:
voice translation rule 1 32527700 #test
Matched with rule 1
Original number: 32527700 translated number: 100
Number of origin type: no number translation type: no
Original number plan: no number plan translated: no
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Number = 32527700, called number = 32527700, Peer Info Type = DIALPEER_INFO_SPEECH
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Associate the rule of = DP_MATCH_DEST; Called number = 32527700
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String = 32527700, expanded String = 32527700 number = 32527700T
Timeout = TRUE, incoming = FALSE, Peer Info Type = DIALPEER_INFO_SPEECH
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result =-1
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No outbound dial-peer does not; Result = NO_MATCH(-1)
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring = 32527700, saf_enabled = 1 saf_dndb_lookup = 1, dp_result =-1
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result = NO_MATCH(-1)
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Number = 59513212, called number =, Voice-Interface = 0 x 0.
Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,
Peer Type Info = DIALPEER_INFO_SPEECH
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Associate the rule of = DP_MATCH_ANSWER; Number = 59513212
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Incoming = TRUE, expand = FALSE
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String =, expanded string =, number of calls = 59513212T
Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result =-1
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:[email protected]/ * /.
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Associate the rule of = DP_MATCH_ORIGINATE; Number = 59513212
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Incoming = TRUE, expand = FALSE
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String =, expanded string =, number of calls = 59513212T
Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Result =-1
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:[email protected]/ * /.
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result = NO_MATCH(-1) after all rules attempt Match
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result =-1
* Jan 29 16:17:14: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeer:[email protected]/ * /.
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Number = 59513212, called number = 32527700, Voice-Interface = 0 x 0.
Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,
Peer Type Info = DIALPEER_INFO_SPEECH
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Associate the rule of = DP_MATCH_VIA_URI; URI = SIP:A.B.C.D:5060
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Incoming = TRUE, expand = FALSE
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String =, expanded string =, number of calls =
Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result =-1
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Associate the rule of = DP_MATCH_REQUEST_URI; URI = sip:[email protected]/ * /: 5060; user = phone
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Incoming = TRUE, expand = FALSE
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String =, expanded string =, number of calls =
Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result =-1
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Associate the rule of = DP_MATCH_TO_URI; URI = sip:[email protected]/ * /; user = phone
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Incoming = TRUE, expand = FALSE
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String =, expanded string =, number of calls =
Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result =-1
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Associate the rule of = DP_MATCH_FROM_URI; URI = sip:[email protected]/ * /; user = phone
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Incoming = TRUE, expand = FALSE
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String =, expanded string =, number of calls =
Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Result =-1
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Associate the rule of = DP_MATCH_INCOMING_DNIS; Called number = 32527700
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:
Incoming = TRUE, expand = FALSE
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchCore:
Dial String = 32527700, expanded String = 32527700 number =
Timeout = TRUE, incoming = TRUE, Peer Info Type = DIALPEER_INFO_SPEECH
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/MatchNextPeer:
Result = Success (0); Incoming dial-peer = 1 is set in correspondence
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchPeertype:[email protected]/ * /.
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerCore:
Result = Success (0) after DP_MATCH_INCOMING_DNIS; Incoming dial-peer = 1
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpMatchSafModulePlugin:
dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
* Jan 29 16:17:14: //-1/F0EA1F0180EB/DPM/dpAssociateIncomingPeerSPI:[email protected]/ * /.
Can someone help me?
Thanks in advance!
I looked on the other leg of the SIP messages appeal, here's the fault for the where incoming call is being failed because the session timer is too small, has received from the SBC (provider)
Call ID:
Call ID: [email protected]/ * /.
INVITE RECEIEVED SBC - SIP
* Jan 29 16:53:19: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
GUEST sip: 32527700 @(WAN): 5060; user = phone SIP/2.0
Via: SIP/2.0/UDP (SIP_SERVER): 5060; Branch = z9hG4bK776928550196f0d843ca0b092
Call ID: [email protected]/ * /.
From:; tag = 6e8b9968-CC-25
TO:
CSeq: 1 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, see
Max-Forwards: 70
Supported: 100rel, timer
User-Agent: Huawei SoftX3000 V300R601
Session time-out: 300
Min - SE: 90
Contact:
Content-Length: 376
Content-Type: application/sdp
v = 0
o = HuaweiSoftX3000 4507886 4507886 IN IP4 (SIP_SERVER)
s = call Sip
c = IN IP4 (SIP_SERVER)
t = 0 0
m = audio RTP 11554 / AVP 8 0 18 4 2 98 98 98
a = rtpmap:8 PCMA/8000
a = rtpmap:0 PCMU/8000
a G729/8000 rtpmap:18 =
a = rtpmap:4 G723/8000
a = rtpmap:2 G726-32/8000
a = rtpmap:98 G726-40/8000
a = rtpmap:98 G726-32/8000
a = rtpmap:98 G726-24/8000
a = ptime:20
a = fmtp:18 annex b = No.
In response to GUY sends 422
Envoy:
SIP/2.0 422 Session Timer too small
Via: SIP/2.0/UDP (SIP_SERVER): 5060; Branch = z9hG4bK776928550196f0d843ca0b092
From:; tag = 6e8b9968-CC-25
Up to:; tag = 4CD1E84-2094
Date: Wednesday, January 29, 2014 22:53:19 GMT
Call ID: [email protected]/ * /.
CSeq: 1 INVITE
Allow-events: telephone-event
Min - SE: 1800
Server: Cisco-SIPGateway/IOS-15.2.4.M
Content-Length: 0
According to rfc
If the Session time-out interval is too low for a proxy (i.e., lower)
that the value Min - SE that the proxy would argue), the
Proxy denies the request with a 422 response. This response
contains a header field in Min - TO identify the minimum session
meantime, she is ready to support. The UAC will try again, this time
including the header of Min - SE in the query field. The header field
contains the largest header field Min - SE that he observed in all 422
responses received previously. In this way, the minimum timer meets the
constraints of all proxies on the way.
Response message 422
If the value of the Session header expires is too small, the UAS or proxy refuses the call with a response message 422 Session Timer too small . With 422 response, the proxy or the SAMU message includes a header of Min - SE, indicating the value of minimum session, he can accept. UAC can then try again the appeal with a higher value of session timer.
If a 422-response message is received after a GUEST query, the UAC can again INVITE him.
There is two way to fix this
1) asked the SBC (your SIP provider) the value change and the value of standards send the SIP invite session expires
(2) change the value of the Min - SE on the CME on demand
Run this Global Config on CME
voip phone service
allow sip to sip connection
SIP
90 min - to
BR,
Nadeem
Please note all the useful post.
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Forums CWMS drop after you have entered the code meeting
Hi all
We have webex 2.6.1.45.B - AE, call out of work, but a drop after you have entered the code-meeting... Work of DTMF.
may be, because: you are using a trial version of the Cisco WebEx meeting server which expires in 32 days. Buy licenses to host for your system to avoid the limitation of access host?
or someone has an idea?
Thank you very much!
Hello
This problem is caused by a bad configuration of the integration of CUCM with CWMS.
Make sure you check you SIP Trunks and SIP configuration on CUCM routing model.
Make sure that your points in routing SIP Trunk SIP of Point Application model.
Make sure that the Partition configured in routing SIP model belongs to a CSS that is configured in the LoadBalancing SIP trunk under:
1. change of itinerary calling search space
2 outside dialogue see calling search space
Also, refer to this post to understand why the model of routing SIP and Application Point SIP trunks are used: https://supportforums.cisco.com/discussion/12879351/cucm-and-cwms-call-routing
In addition, this guide to the official reference documentation: http://www.cisco.com/c/en/us/td/docs/collaboration/CWMS/2_6/Planning_Guide/Planning_Guide/Planning_Guide_chapter_0101.html
I hope this will help.
-Dejan
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SX20 installation with no CUCM
We expect use SX20 conference 2 of different city offices using a link with sufficient bandwidth and no firewall between, by the time there is no available CUCM, no VCS, no TMS, etc, and I have 2 questions about H323 and SIP configurations:
-What is the best way to configure the SX20 devices? Configure the profile H323 Gatekeeper or Direct mode? What are the recommendations on any mode?
-How do I dial to each other? By IP address?
TC6.0.0 running
Any help will be be appreciated.
Kind regards
Juan Carlos Arias
So I won't have to configure any other option on H323 menu (Authentication, Gatekeeper and H323 Alias)?? Just like that??
Yes, for a basic call you do not have to configure anything else on H323 configuration options.
So, the system information status about H323 and SIP will be always Inactive, right??
Yes, gatekeeper and SIP server status will be displayed as inactive, it means that you are not using any server to make calls. But the protocols are still capable, so you can use H323 do call by dialing the IP address normally.
Concerning
Paulo Souza
-
Problem recording movi to VCSe (VCSC work)
Hello
for some reason any Jabber client video are not able to log on to VCSe. If they are in the office and connect to the VCSC his work.
Deployment:
VCS: 7.2.2
TMSPE: 14.2
VCSC is joined to the domain, so people should be able to login with LDAP credentials. Since a few days we have changed an OU for users in AD. I resynced the TMS users. In VCSC my thought is that I have nothing to do, right? I do not know its function, change OR cause the customer movi is not som uch in use.
I started track of debugging on VCSe registration go external with movi (IP addresses and names have been changed):
SIPMSG:
| SIP SUBSCRIBE:
@domain.comSIP/2.0 Via: SIP/2.0/TLS 192.168.100.10:3157; branch = z9hG4bK06fbfdf1d928fdea44551eb01444f279.1; received = 78.51.4.182; = 45938 rport
Call ID: [email protected]/ * /.
CSeq: 201 SUBSCRIBE
Contact: <>
@192.168.100.10:3157; transport = tls > From: <>
@domain.com >; tag = ff9e0e9142437b34 To: <> [email protected] / * />
Max-Forwards: 70
Directions:
User-Agent: TANDBERG/774 (4.5.7.16762 PCS) - Windows
Expires: 300
Event: ua-profile;model=movi;vendor=tandberg.com;profile-type=user;version=4.5.7.16762;clientid="S-1-5-21-176747291-1711663684-1688638919";connectivity=1
Accept: application/pidf + xml
Content-Length: 0
SIPMSG:
| SIP/2.0 404 not found
Via: SIP/2.0/TLS 192.168.100.10:3157; branch = z9hG4bK06fbfdf1d928fdea44551eb01444f279.1; received = 78.51.4.182; rport = 45938; DefaultZone = ingress-box
Call ID: [email protected]/ * /.
CSeq: 201 SUBSCRIBE
From: <>
@domain.com >; tag = ff9e0e9142437b34 To: <> [email protected] / * />; tag = a9bdf7d526c0cb56
Server: TANDBERG/4120 (X7.2.2)
WARNING: 79.34.34.123:5061 399 'political response '.
Content-Length: 0
So we have a 404 not found error, but why and what is not found, the user who tried to connect? For some reason, it seems that demand is not going through VCSe VCSC, it is my feeling?
Auth area is as follows:
VCSe:
DZ: do not check the credentials
TZ: do not check the credentials, accept records submitted by proxy: refuse
DSZ: do not check the credentials, registration: refuse
VCSC:
DZ: verify the credentials
TZ: verify the credentials, accept records sent by proxy: enable
DSZ: verify the credentials, registration: enable
I also implemented a CPL on VCSe (not on the VCSC) but I have disabled also the policy tried it unsuccessfully with so this should not be the problem.
I tested a few other constelattion without success. :-( Maybe someone has an idea of what I can test. What is a firewall issue?
Concerning
Thorsten
Hello
Could check you if the 'Record SIP proxy Mode' setting is on "Known Proxy" only on Hwy VCS? You will find this SIP configuration setting.
This is a required parameter for allow VCSe work as a proxy SIP for registration. It is not enough to define 'Recording proxy Acept' control VCS. In general, you should have this record transmitted by proxy configuration:
If you don't want to use the proxy record but jabbers on the internet for you sign up directly for VCSe, you must disable the Save proxy functionality and create the SIP domain in VCS Expressway. It would be something like this:
Try this.
Paulo Souza
Please note the answers and mark it as "answered" as appropriate.
-
Hello
We acquired 3 Polycom HDX7000 via a merger. I tried to discover with TMS, but TMS learns only H323 configurations. He doesn't learn the SIP configuration. How can I fix this?
SIP is not supported for these types of devices in the TMS - only H.323 and H.320 for devices of Polycom.
-
Need help on the Configuration of the SX10...
I'm tryng do SX10 to work in stand-alone mode. So far, I did:
-Give SX10 IP internal
-Assign an external IP (NAT) on the firewall
-Open SIP port 5060-5061 and SIP Media 2326-2486 firewall port
But when I run the diagnostic tests, it still gives me the error:
Is someone can shine some light on this error? What to do to make this device work? Thank you!
========================================================
ERROR: Dial the record Protocol
The system is not on SIP. It will not be able to place or receive calls until this as correctly configured.Make sure that SIP Listenport is set on it.
/ Magnus
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SPA3102 - Can EU - I configure two different VOIP - SIP providers?
Hello.
In Belgium, I can configure to use a SIP account I have VOIP geo number to receive calls and another VOIP provider only for outgoing calls?
If so, how do I do this?
Thank you for your response.
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Configure (fxs) analog phones with Sip trunk
Hi all
IS - this configuration FXS possiple to knit on SIP TRUNK? I HAVE 16 FXS ports and voip gateway 2921 cisco.
Configuring fxs to work with her?
any help thanks.
Is there a PBX at stake here, i.e. CUCM or CME?
In both cases, you have set good dial-peers to point to the FXS ports to match the destination of the SIP trunk, which is pretty simple.
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