SIP UUI Question
Hi,
Could somebody please advise as how would we retrieve the UUI (User to User Info) in ICM ? The UUI is being set at the customer PBX (Nortel) and getting passed on to ICM through the PG. I`m trying to retrieve that through either through UserToUserInfo variable ,with no luck.
Call flow:
Nortel PBX ---->SIP Trunk -----> Cisco CallManager ---->CTI Route Point----->CCM PG----->ICM
Raj
Yes, same trunk will be used for both inbound and outbound calls.
Let say you create one SIP trunk between CUCM and Asterisk, same SIP trunk here will be used to deliver all calls from CUCM to Asterisk and vice versa.
When you look at SIP trunk configuration in CUCM, you will get both inbound and outbound call related parameters for appropriate configuration of outbound and incoming call routing.
Thanks
Vivek
Similar Messages
-
Hello,
I have set up 2 CUBEs with sip trunks to a provider from my CUCM 8.0.3. everything is working fine with exception of fall over to the second sip trunk in the route list. Meaning, if i bring the first CUBE down the second one never routes the outbound calls even though it is second in the route group. Does anyone know how to get this working properly? or...does this set up just not work? How can I get the call manager to notice that its first outbound choice died?
Thanks,
Ed VazquezHi,
Wrong forum. You need to move this question to IP Telephony forum...
Regards
Please rate all useful posts
Favor calificar todos las respuestas útiles.
LinkedIn Profile: do.linkedin.com/in/leosalcie
MDGDP, CCNA, CCNA Voice certified -
UC500- Outbound SIP Routing Question
We have a UC540 that is receiving its trunking from a cloud based sip server. They have two trunks on the server, one for voice traffic and one for fax traffic. Inbound this works fine but all outbound traffic goes over the 'voice' trunk. Is it possible to route outbound traffic, from the fxs ports that the fax machines are connected to, onto the 'fax' trunk of the sip server? The sip server only provides one ip address, with unique registration to each trunk.
Hey guys,
Thanks for the response. Let me clarify my question a bit. I get the translation routing part and that is very helpful but I am still stuck on how to point a dial peer to the 'fax' trunk on the sip server.
A typical dial peer on the switch looks like this:
dial-peer voice 1022 voip
corlist outgoing call-national
description **CCA*Generic Locale*Long Distance**
translation-profile outgoing SIP-Trunk-Out
preference 1
destination-pattern 91[2-9]..[2-9]......
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
and the sip-ua looks like this:
sip-ua
credentials username 7xxxxx5 password xxxxx realm xxx.xxx.xxx.236 (voice trunk/user ID on the sip server)
credentials username 7xxxxx6 password xxxxx realm xxx.xxx.xxx.236 (fax trunk/user ID on the sip server)
keepalive target ipv4:xxx.xxx.xxx.236:5060
authentication username 7xxxxx5 password xxxxxx
no remote-party-id
retry invite 2
retry register 10
timers connect 100
timers keepalive active 100
registrar ipv4:xxx.xxx.xxx.236 expires 3600
sip-server ipv4:xxx.xxx.xxx.236:5060
connection-reuse
host-registrar
Thanks,
Chad -
Hi,
We’re in strange situation where our AD domain is abc.net but we want primary sip domain to be lync.abc.com & smtp domain is abc.com so email addresses are
[email protected]
So my question is, if I publish the topology for primary sip domain as
lync.abc.com then what could be our challenges in terms of client logon and lync access …? What do I have to have in place to make this work?
Note: Our Lync deployment will be two central sites and each will handle their own specific lync traffic.You won't hit anything that would won't be able to overcome.
But what is the justification for knowingly making your SIP domain different from your SMTP name space? It's considered best practice to align the two in order to create a simplified user experience and minimise the amount of additional consideration (such
as exchange integration).
Fully appreciate you'll have your own circumstances though, and if this is a requirement - so be it =)
Kind regards
Ben
Note: If you find a post informative, please mark it so using the arrow to the left. If it answers a question you've asked, please mark the thread as answered to aid others when they're looking for solutions to similar problems or queries.
For Fun: Gecko-Studio | For Work:
Nexus Open Systems -
SIP Trunk Question - Outbound Calls Fail
Hi Folks,
I am using a Cisco 2821 as a router that will convert a SIP trunk to an E1 PRI. Si my setup is:
SIp-Trunk > 2821 Router > E1 port on 3900> CUCM
Inbound calls are working fine, but outbound are failing. I am starting to think its due to transcoding issue on the SIP-GW maybe (there is nothing configured on it for XCODE etc).
I think my configuration is fine as I am able to recieve calls inbound. Just outbound fail.
Here are the debugs from the SIP-GW:
"Debug CCSIP calls"
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4BB4F194
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 1528xxxx
Called Number : 909
Source IP Address (Sig ): 172.29.x.xxx
Destn SIP Req Addr:Port : 10.200.7.157:5060
Destn SIP Resp Addr:Port : 10.200.7.157:5060
Destination Name : 10.200.7.157
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 172.29.5.210
Source IP Port (Media): 16786
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Nov 26 18:50:50 UTC: //929/F9E88693801B/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 28
Disconnect Cause (SIP) : 484
Can anyone shed some light on the area that I need to focus on? This is my first attempt at SIP and I am confused :)
Thanks.Hi
909 is what I meant to dial as that is the help desk for the telco.
I tried mobile numbers as well getting the same error codes. And international numbers.
If it's based on the called number being wrong then I guess I will have to play with the calling party ID and call type as well... Maybe this is causing it to fail?? -
C3-01 WiFi / WLAN & VoIP / SIP Question
Hello,
I am very interested in purchasing a C3-01, as I don't want a smart phone, but I like the advanced features this phone offers. Specifically, I want to make use of the VoIP SIP client that comes built in.
My question is this: I will be using VoIP / SIP on the WLAN (Wi-Fi) connection. In another post on this forum, I saw a user saying that Wi-Fi gets turned off automatically when the phone is locked.
So, does this mean that if I've got a VoIP account set up, and the C3-01 locks (either manually or automatically), that the WLAN will drop out and I won't be able to get incoming VoIP calls?
If this is how it works, this is a pretty major oversight, and I will look for something else.
Thanks
Solved!
Go to Solution.Yeah, Sadly Some Country still have the v05.68 as the latest firware, But the reason might be that the Firmware released in your Country may contain the fixes included in the Latest Firware for Our Country.
So thats ok, If Nokia Finds any fixes to be done then, they will release a new Firware Update for your Country. Be Rest Assured.
And Buy C3-01.
If I've helped in any way, a click upon the White star to the left would always be appreciated.
If however my answer also solves your problem clicking below " ACCEPT AS SOLUTION " it will benefit other users! -
SIMPLE QUESTION: LYNC 2010 -- AVAYA CS1000 DIRECT SIP: IS THAT POSSIBLE? PLEASE HELP!
Hello,
I would like to ask really simple question here: is that possible to establish (configure, setup, create, make etc.) a direct SIP connection between LYNC 2010 and AVAYA CS1000 PBX. On one side I have Lync 2010 Standard Edition acting as Front-End Server
with co-located Mediation Server. On another side I have AVAYA CS1000 IP PBX Version 7.0. We are trying to make this happen for 1 full year with no luck. The issue is that we do not manage our PBX (external company does).
Here is what we are basically trying to achieve:
All I want is that basically LYNC user (phone # 5678) can dial PBX user (phone # 1234) and talk. Also PBX user can call LYNC user and they can talk. I'm pretty sure that all I'm asking is very simple thing. I don't expect magic to happen (e.g. when one user
calls another user the world will change). All I want is to be able to call between LYNC and PBX users so they can talk.
Sorry for repeating again but here is my simple question: can direct SIP (or call it SIP trunk if this is the right word) be established so LYNC and PBX users can call each other and talk. Whatever protocol is (G.711, G.722 etc.) I will be happy.
If you can let me know I would very much appreciate this. I'm sure I'm not alone in this world and many companies out there using AVAYA CS1000 and deployed LYNC and probably have the same question so it may be helpful to other people as well.
Thanks!Hi Thamara for reply. I can see where you are going, but the challenge here is that I have found many threads including in this forums from people who successfully connected CS1000 PBX to Lync 2010 for two-way calling, and even dual-forking (which I cannot
even dream about at this time).
If you go to AVAYA WEB site, you can find official AVAYA compatibility MATRIX that is available here:
http://www.google.ca/url?sa=t&rct=j&q=avaya%20compatibility%20matrix%20lync&source=web&cd=1&ved=0CBwQFjAA&url=http%3A%2F%2Fsupport.avaya.com%2Fcss%2FP8%2Fdocuments%2F100135385&ei=v3-5TsXVKMLX0QGHpdSkCA&usg=AFQjCNG6cmW5izdOASMuW34b8iK0vWnwuw
According to this document, only CS1000 version 7.5 supports Lync 2010 and only via ACE - starting first quarter of 2011 (which is done deal by now) - see below for details. The problem is that I have a hard time to understand the meaning of the word "Supported".
I do not really need full support, all I want is to be able to call in both directions using 2 distinct numbers (e.g. 4 digits extensions) that are configured in both systems.
Another thing is this ACE. Apparently it is a big monster software that you need to buy and pay big bucks and lots of configuration just to make it working. But I'm not even sure what will work. We do have one-way calling (from Lync to PBX) - it just does
not work other way, and I have no idea why. I'm not sure if this cannot be working, or because it was not setup right. As I mentioned before we have no access and clue about IP PBX - this is one of my issue here. Withoul knowledge how it works on PBX side
I cannot be sure what is causing this.
So I was wondering if anybody in this world was able to connect CS1000 PBX to Lync and perform two-way calling using two systems.
Thanks!
Microsoft OCS 2010 Lync
Supported (via ACE) - 1Q11 -
VCSE with dual NIC / SIP H323 encrypted call / traversal and non-traversal call license questions
Dear All,
I have a few questions and I would appereciate if someone will answer my questions?
1) May I know pros and cons of using Dual NIC and single NIC with VCSE in DMZ?
2) In order to make H323 and SIP encrypted call, what configuration need to be done on Cisco endpoint, non-cisco endpoint, VCSC and VCSE (both signalling and Media need to be encrypted).
3)let say my VCSE is in DMZ- endpoint A (cisco) and endpoint B (non-cisco) are registered on VCSE. I would like VCSE to use non-traversal call license when A call to B or B call to A regardless of whether H323 or SIP call.How can I force VCSE and endpoints to use non-traversal call. I only want VCSE to handle signalling and media is EP to EP direct. As traversal license is quite expensive.
I found following information from Cisco document.
all Cisco TelePresence endpoints are traversal enabled and so a traversal license will always be needed when at least one of the endpoints involved in the call is a Cisco TelePresence endpoint.
Is it possible to disable traversal client feature of Cisco endpoint?
Thanks and Best Regards,voice register pool 4
add-->codec g711ulaw
voice register global
no create pro
create prof
and enable below debugs and send the logs
deb ccsip mess
deb voip ccapi inou
deb h225 asn1
deb h245 asn1 -
Send DTMF with SIP INFO (c2600) configuration question
I have a cisco 2600 with VIC-2FXS port as VOIP Gateway, connecting to SIP Server to receive SIP Incoming calls. I am able to receive call and the vocie has been pass through both way; and I would like the 2600 send DTMF as SIP info but was not able to do so. I have ios 12.3, and from this configuration guide http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/chapter8.html#wp1048824 , it does not require any config for SIP info. I must missing something here, please advice. Thanks.
The config is following -
version 12.3
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
hostname abc
boot-start-marker
boot-end-marker
enable secret 5 xx
enable password xx
memory-size iomem 10
no aaa new-model
ip subnet-zero
no ip routing
no ip cef
interface Ethernet0/0
ip address 192.168.1.15 255.255.255.0
no ip route-cache
full-duplex
no ip http server
ip classless
voice-port 1/0/0
voice-port 1/0/1
voice-port 1/1/0
voice-port 1/1/1
dial-peer voice 1 pots
destination-pattern 1000
port 1/1/0
dial-peer voice 2 pots
destination-pattern 1001
port 1/1/1
dial-peer voice 10 voip
destination-pattern 1.T
session protocol sipv2
session target ipv4:192.168.1.224:5061
session transport udp
codec g711ulaw
dial-peer voice 3 pots
destination-pattern 1100
port 1/0/0
dial-peer voice 4 pots
destination-pattern 1101
port 1/0/1
line con 0
line aux 0
line vty 0 4
login
endI tried to set it, and for IOS 12.3(26) - the latest for 2610 - which dose not have that option. I use dtmf-relay rtp-nte instead; but it did not send RFC2833 event. From ethereal, no OOB events. It seems that the config I have does not have OOB DTMF enable; I compare the config I have with other examples but can not found anything wrong. Any suggestion, and what debug message I should enable, that may help to identify the issue.
Thanks.
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
interface Ethernet0/0
ip address 192.168.1.15 255.255.255.0
full-duplex
voice-port 1/0/0
voice-port 1/0/1
voice-port 1/1/0
voice-port 1/1/1
dial-peer voice 1 pots
destination-pattern 1000
port 1/1/0
dial-peer voice 2 pots
destination-pattern 1001
port 1/1/1
dial-peer voice 10 voip
description Outbound Calls
destination-pattern 1.T
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.1.250
session transport udp
dtmf-relay rtp-nte
no vad
dial-peer voice 3 pots
destination-pattern 1100
port 1/0/0
dial-peer voice 4 pots
destination-pattern 1101
port 1/0/1
dial-peer voice 100 pots
destination-pattern 8...
port 1/1/0
forward-digits 3
dial-peer voice 20 voip
description Incoming calls from PBX
incoming called-number .T
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.1.250
dtmf-relay rtp-nte
no vad -
A question about call manager traces for Sip phones.
So today I create a sip based ip communicator and pressed the new call button and heard a dial tone. I started typing my telephone number. Half way through, I heard another secondary dial tone (which indicates mis-configured route pattern somewhere) .
However, When I look at the call manager logs, I do not actually see the digits that I was typing. With SCCP, I can see the keypad button press messages in the traces, but here, I cannot see the pressed buttons in my CUCM traces. Can anyone help with telling me how I can see button presses going to call manager . All I can see are the logs below which came up as soon as I got the dial tone and the final sip invite messages. I see nothing in-between.
|SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.xx.4.xx on port 56714 index 31809 with 973 bytes:
[6387070,NET]
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 10.x.x.66:56714;branch=z9hG4bK00005b1e
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=00ffb00bc50a00340000499f-00006ab4
Call-ID: [email protected]
Date: Sat, 14 Feb 2015 14:17:40 GMT
CSeq: 19 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:[email protected]:56714;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 350
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="18" state="partial" entity="sip:[email protected]">
<dialog id="12" call-id="[email protected]" local-tag="00ffb00bc50a003300006390-00002d4f"><state>trying</state></dialog>
</dialog-info>
SIPStationD(12991) - processCommonDialogNotifyInd: Did 12 Sending Notified SIPOffHook to new CdfcHere is a more detailed explanation of how SIP calls notify cucm when they go off hook to make a call. The digit dialled here is 4080
+++++ Analysis of SIP Phone making a call +++++++++
The user picks up the phone and the IP Phone sends a NOTIFY to CUCM to indicate the start of a new dialog. This dialog begings by an offhook event
00869539.002 |14:58:13.837 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 976 bytes:
[46240,NET]
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:13 GMT
CSeq: 11 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 350
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="10" state="partial" entity="sip:[email protected]">
<dialog id="6" call-id="[email protected]" local-tag="544e42f26d0b001d00007cc9-000044a3"><state>trying</state></dialog>
</dialog-info>
++++ CUCM SIP stack processes the new connection for the phone+++++++
00869540.001 |14:58:13.837 |AppInfo |//SIP/Stack/Info/0x0/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 (SIP_NETWORK_MSG), for event 1 (SIPSPI_EV_NEW_MESSAGE)
00869540.002 |14:58:13.837 |AppInfo |//SIP/Stack/Transport/0x0/sipTransportProcessNWNewConnMsg: context=(nil)
00869540.003 |14:58:13.837 |AppInfo |//SIP/Stack/Transport/0x0/sipConnectionManagerProcessNewConnMsg: gConnTab=0xe81c0d70, addr=10.50.16.1, port=52910, connid=2748, transport=TCP
++++ Next CUCM allocates a call id for this call +++++
00869546.002 |14:58:13.838 |AppInfo |LineControl(66) - Get call instance=1 for CI=24419584
+++Next CUCM sends a 200 OK to the NOTIFY request for the new dialog ++++
00869555.007 |14:58:13.839 |AppInfo |//SIP/Stack/Transport/0x0xe7df4d48/sipTransportPostSendMessage: Posting send for msg=0xefbe9910, addr=10.50.16.1, port=52910, connId=2748 for
00869555.008 |14:58:13.839 |AppInfo |//SIP/Stack/Info/0x0/act_dialog_pending_resp_event: Changing from State: SUBSCRIBE_STATE_DIALOG_PENDING to state SUBSCRIBE_STATE_ACTIVE
00869556.000 |14:58:13.839 |SdlSig |SIPSPISignal |wait |SIPTcp(1,100,71,1) |SIPHandler(1,100,79,1) |1,100,14,31314.75^10.50.16.1^SEP00909E9D106C |*TraceFlagOverrode
00869556.001 |14:58:13.839 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46241,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
To: <sip:[email protected]>;tag=1822746380
Date: Mon, 16 Feb 2015 12:58:13 GMT
Call-ID: [email protected]
CSeq: 11 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
++++ The IP Phone sends its connection ID to CUCM, its ip address and its port number+++++++++
00869541.001 |14:58:13.838 |AppInfo |SIPStationInit: connID=2748, SEP00909E9D106C, 10.50.16.1:52910, Routed signal by connection index to (1,100,73,66)
++++ Next CUCM informs us that the NOTIFY message is for an offhook event ++++++
00869542.003 |14:58:13.838 |AppInfo |SIPStationD(66) - processCommonDialogNotifyInd: Notified Dialogs - Did 6 State trying
00869542.004 |14:58:13.838 |AppInfo |SIPStationD(66) - processCommonDialogNotifyInd: Did 6 Sending Notified SIPOffHook to new Cdfc
00869542.010 |14:58:13.838 |AppInfo |SIPStationD(66) - processSIPOffHook Primary Call Not-Found
00869543.000 |14:58:13.838 |SdlSig |SIPOffHookInd
+++ The next thing is the USER dials a digit on the phone ++++++
This is where it gets a little complicated. So lets examine this. The first digit that is dialled generates an INVITE to CUCM like this:
In this example the user dialled "4" first so we see an "INVITE sip:4@host-IP"
00869559.002 |14:58:14.064 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 1445 bytes:
[46242,NET]
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
Max-Forwards: 70
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 101 INVITE
User-Agent: Cisco-SIPIPCommunicator/9.1.1
Contact: <sip:[email protected]:52910;transport=tcp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Emre ESEN" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 373
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 21020 0 IN IP4 10.50.16.1
s=SIP Call
t=0 0
m=audio 20250 RTP/AVP 0 8 18 9 116 124 101
c=IN IP4 10.50.16.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:124 ISAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
+++++ NEXT CUCM sends a trying for the INVITE it received +++++++++++
00869562.001 |14:58:14.065 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46243,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
To: <sip:[email protected];user=phone>
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0
++++NOW CUCM evaluates the DTMF supported by the phone to determine how to inform the phones to send the remaining dtmf digits++++
From the INVITE cucm concludes that KPML and rtp-nte is supported
00869566.009 |14:58:14.066 |AppInfo |setEndpointsDtmfCaps: KPML Supported.
00869566.010 |14:58:14.066 |AppInfo |setEndpointsDtmfCaps: Detected inband DTMF support
Next CUCM generates kpml event pkg which is going to be used to receive the remaining digits from the phone
00869590.001 |14:58:14.067 |AppInfo |SIPEventPkg::SIPEventPkg 0xe4a1d1e0 scbId[16725], event name[kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3], id[]
+++ Next CUCM sends a SUBSCRIBE to the IP phone for kpml event +++++
00869594.001 |14:58:14.068 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46244,NET]
SUBSCRIBE sip:[email protected]:52910 SIP/2.0
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
From: <sip:[email protected]>;tag=480227084
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 SUBSCRIBE
Date: Mon, 16 Feb 2015 12:58:14 GMT
User-Agent: Cisco-CUCM10.5
Event: kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3
Expires: 7200
Contact: <sip:[email protected]:5060;transport=tcp>
Accept: application/kpml-response+xml
Max-Forwards: 70
Content-Type: application/kpml-request+xml
Content-Length: 424
<?xml version="1.0" encoding="UTF-8" ?>
<kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-request.xsd" version="1.0">
<pattern criticaldigittimer="1000" extradigittimer="500" interdigittimer="15000" persist="persist">
<regex tag="Backspace OK">[x#*+]|bs</regex>
</pattern>
</kpml-request>
+++ Next we get a 200 OK to the SUBSCRIBE from the ip phone ++++
00869595.002 |14:58:14.118 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 459 bytes:
[46245,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
From: <sip:[email protected]>;tag=480227084
To: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 101 SUBSCRIBE
Server: Cisco-SIPIPCommunicator/9.1.1
Contact: <sip:[email protected]:52910;transport=TCP>
Expires: 7200
Content-Length: 0
+++ NEXT the IP phones sends the remaining digit dialled on the phone to CUCM +++
00869603.002 |14:58:14.183 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 573 bytes:
[46247,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1000 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 0
00869608.001 |14:58:14.183 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46248,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1000 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++Next the IP phone sends the next digit. Here its important to note that the NOTIFY doesnt contain the next digit,
the NOTIFY is still the same as the first digit but the next digit is carried in the xml document attached to the NOTIFY.
At this point I will insert a paragraph from the RFC 4730 for SIP KPML
+++++++++++++
The event package uses SUBSCRIBE
messages and allows for XML documents that define and describe filter
specifications for capturing key presses (DTMF Tones) entered at a
presentation-free User Interface SIP User Agent (UA). The event
package uses NOTIFY messages and allows for XML documents to report
the captured key presses (DTMF tones), consistent with the filter
specifications, to an Application Server +++++++++++++++++++++++++++
00869609.002 |14:58:14.209 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46249,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1001 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
00869622.001 |14:58:14.210 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46250,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1001 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++ Again we get the next digit ++++
00869624.002 |14:58:14.262 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46251,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1002 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="8" tag="Backspace OK"/>
00869637.001 |14:58:14.263 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46252,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1002 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++ Finally we get the last digit ++++
00869638.002 |14:58:14.390 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46253,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00006c1c
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1003 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
Once digit collection is completed CUCM proceeds to finalise its digit analysis process.
Note that digit analysis is carried out for each digit that is recieved. I have only included the final DA here
00869648.003 |14:58:14.391 |AppInfo |Digit Analysis: star_DaReq: Matching SIP URL, Numeric User, user=4080
00869648.004 |14:58:14.391 |AppInfo |Digit Analysis: getDaRes data: daRes.ssType=[0] Intercept DAMR.sstype=[0], TPcount=[0], DAMR.NotifyCount=[0], DaRes.NotifyCount=[0]
00869648.005 |14:58:14.391 |AppInfo |Digit Analysis: getDaRes - Remote Destination [4080] isURI[0]
00869648.012 |14:58:14.391 |AppInfo |Digit analysis: match(pi="2", fqcn="9106", cn="9106",plv="5", pss="", TodFilteredPss="", dd="4080",dac="0")
00869648.013 |14:58:14.391 |AppInfo |Digit analysis: analysis results
00869648.014 |14:58:14.391 |AppInfo ||PretransformCallingPartyNumber=9106
|CallingPartyNumber=9106
|DialingPartition=
|DialingPattern=4XXX
|FullyQualifiedCalledPartyNumber=4080
|DialingPatternRegularExpression=(4[0-9][0-9][0-9])
|DialingWhere=
+++++Once this is done CUCM then proceeds to send the call out to to the intended destination as configured in the RL ++++
00869701.001 |14:58:14.435 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.250.0.13 on port 5060 index 2754
[46256,NET]
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce931ee3d74
From: "Emre ESEN" <sip:[email protected]>;tag=16726~813ee89e-33db-4d58-9f6a-61542cc840ee-24419585
To: <sip:[email protected]>
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces -
SIP Trunk Recommendations, PoE question
Thanks for the good info. Most likely one PBX on a ESXi host at our HQ. Our structure is fairly simple - one HQ, a secondary "HQ", and 22 branch locations. Each branch locations consists of 6-10 users. The second HQ has 35 users.
HQ: 25
Second HQ: 35
Branch offices: ~150
Total: ~210
Looking at our usage, we rarely see more than 20 active calls company-wide.
I think we could reduce our DID count. I need to survey our users. If I were to guess, about 50 of our DIDs are only used for extension-to-extension internally - they never advertise their DID to external callers.Regarding providers, I think most decent SIP providers will be able to port the numbers to their service (cannot always be guaranteed though) and handle 300 DIDs in your account without an issue. I would be most concerned about the cost per month per DID. There are varying charges depending on the provider you choose. I've used NexVortex and Fusion (formerly Broadvox) with success. I would consider FlowRoute and Vitelity as well. You will want to know how many concurrent calls you need as many providers charge you based on that need. Nexvortex will give you unlimited concurrent calls and a certain number of minutes per month plus a few per DID. Fusion charges per concurrent call you need per month, some plans have free local or LD calling, and then there is a very small fee per DID.Consider other little fees like caller ID (inbound...
-
Hi,
I have a requirement to move from a h323 environment to a SIP environment. I am looking for best practises especially around security. I have 2 CUCM servers (8.5) located in separate cities for redundancy. I also have 2 voice gateways which at the moment are h323 to the PSTN, each located at different cities.
My requirements are:
1. Creat a sip trunk to the provider instaed of using PRI.
2. If the Wan link fails on one gateway to provider, the alternate router in the other location should be able to receive the setup messages and if a user logs on via extension mobility, should be able to answer the call.
Are there any simplified design docos about for this? I am hesitant to create a SIP trunk straight to the provider for security, so thinking of terminating the call on the voice routers with CUBE. I'm pretty sure this is run of the mill and would appreciate some input.
Cheers!
Pieter+5 to Chris..Always use CUBE...
Here are more ideas..
1. Create two sip trunks, first one to Cube 1, second to CUBE 2
on your sip trunk set DTMF as no preference
2. Assign the trunks to CUCM group with your two servers in it
3. Configure route groups with Circular algorithm distribution (this way you have load balancing on your two cubes)
4. Configure dial-peers on your CUBE gateway to point in preferential order to your cucm servers
5. Use dtmf-relay rtp-nte on your dial-peer (ensure you have sip as the protocol on your dial-peers)
6. configure your codec selection properly on your dial-peers
7. configure your region settings properly between your sip trunk and phones.
8. Evaluate if you need xcoders, provide one if you do and ensure you set your region correctly between your xcoder and CUBE
9. Is there voicemail involved? Ensure you set your region settings between cube and voicemail correctly, otherwise calls to voicemail may invoke xcoder (from experience)
10. Provision adequate bandwidth for your calls. Once you move to sip, you loose the luxury of e1 channels. You are solely relying on Bandwidth. Ensure you have adequate bandwidth for your concurrent calls
11. Provision QoS for your calls
12. Is there fax involved? You need to think carefully on this one. What fax method do you want to use. T.38/pass through. Does your provider support T.38?
13. Have a thorough test plan. Test call transfers, call on hold, call forward etc
Just a few pointers..Careful planning and implementation is required for a successful SIP implementation -
Question regarding the use of Built-in Bridge SIP call setup to recording device
We have an application that uses Built-in Bridge (BiB) to setup a SIP call to our recording application. Recently the topology of our network was changed and for some reason the RTP for the call is not being sent. What appears to be happenning is that the SIP messages are sent from the UCCM and once the call is established we get an end call event.
We are not sure how to troubleshoot our network to determine the root cause. Any pointers will be appreciated.
Thanks.
-ArunHi
Did you ever figure this out? I am having an issue with recording calls to the PSTN from 9971 SIP phones. Station to station calls seem fine, and 7945 phones running SCCP seem fine to the PSTN.
Thanks,
Aaron -
Questions in Microsoft sip processing language scripts?
Hi Team,
Just a quick query.
Is SBS / SBA able to support MSPL scripts or a software agent that uses mspl script ?
Please advise.Hi,
Agree with Holger.
The Microsoft sip processing language scripts is a server application which can't be installed in SBA or SBS.
Generally, The MSPL script language can run on FE Server and Edge Server.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support -
Hi all,
just a quick question (not for me as I have lack od knowledge on UCCX)
We have to redirect a call from UCCX to an external resource, via ISDN. As the ISDN is the only connection provided, we have to transfer some information in the UsertoUserInformation field (is. call identification or key, ANI,...).
Is it possible to populate the UUI and to send the information through the network untill the ISDN target (or ISDN voice gateway)?
Regards
MarioHi,
thank you for your response.
About your answer.
Using CCM I am not sure that is possible to define and set a global key, thatis used to correlate the calls sent to ISDN external devices
We cannot use ip as the requirement is to have a ISDN between the UCCX an external devices (outsourcer not integrated for instance). UUI is used to send some information and a correlation key generated inside UCCX.
I was thinking to use IVR script. Do you know if it is possible to set UUI field and if it content is mantained in the SIP header for the outgoing call?
Thanks
Mario
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