Lync server sip trunk , IM and presence forwarded in trunk, is possible?
tank you in advance for all the help i can obtain.
I have managed to install, configure a lync server 2013 evaluation.
One domain controller and one front end server (without Edge Server)
My first goal was to configure a SIp trunk with an Open Source SIP proxy/server. (OpenSIPS)
All work properly, it was many days of suffering and hard work. The call are correcly routed in trunk in both direction.There is some possibility to forward sip message about presence (subscribe,publish, notify) inside the trunk?
Lync manage it internally but when i want to send the message cross domain, a SUBSCRIBE as example(my lync domain is lynclab.local, my OpenSIPS domain is opensips.local) i obtain this error in lync:
"Non-trusted source sent an FQDN/IP that doesn't match a .... The SIP 404 Not Found" (referring my esternal domain Opensips.local.
Probably i have deeply misudestood someting, reading for many days documents and manuals i cannot find a solution.
I need to configure an XMPP Gateway for "exchange" status message between sip domain?
thank you for all, and excuse me if the questions is tryvial or absolutely out of context.
You'll need a trusted application for this, you won't be able to forward the subscribe events through a standard sip trunk sent through the mediation role unfortunately. You'll have to find something specifically written to help the two systems interoperate,
but out of the box this would not be native functionality from the Lync side.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications
This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.
Similar Messages
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Hello Everybody
I have installed lync server 2013 in my lab . Everything is ok . but internal mobile client in my LAN can not work with lync !
I installed lync server 2013.
I create 2 DNS record to lync-server : lyncdiscoverinternal and lyncdiscover.
clients on PC (windows) work and everything is ok . when user use Anroid phone , cleint connect to lync server and list connnact list .BUT when Send IM to contact or start video call come this error :
" Failed to process the server response.Please try again . if the problem persists,contact your support team."
This is first problem. Another problem is the contact list for every user status is "Presence Unkhown" !
No online , No Offline and all of contact status "Presence Unkhown" !
If i had to config mobility in in lync server 2013 ? or mobility service built-in in lync server 2013 ?
please help me .Hi Meysam Halvaei,
As Thamara mentioned, you need to deploy reverse proxy.
Lync 2013 mobility clients are using a separate dedicated Ucwa parameter the URL is the same web services location as provided by the
External/Ucwa parameter.
You need to have a look at the following diagram.
The Lync 2013 mobile client will need to connect with the external web services. The way Lync is designed this communication is always established using HTTPS/TCP 443, so the traffic
will need to pass through a reverse proxy or similar service that will take this communication and forward it to the server hosting the external web services, now using port 4443 instead.
Best regards,
Eric -
Web based client for lync server 2013 , only for IM/presence
Hi,
for 20000 users, 4 FE servers, HA . only for IM/presence , looking for web/browser based client so that there will not be any need of thick client installation. is there any way? thanksif i integrate lync 2013 with exchange 2013 OWA, wil reverse proxy or Edge server will come into picture?
If you have users who are external to your network , then yes Edge server and reverse proxy will come into the picture.
will integration of lync 2013 with exchange 2013 provide the following
one-to-one, one-to-many chat. ?
Yes
archiving of chats at server side.
you need to install archiving server for lync .
Instant messaging should support rich text, emoticons.
Yes
Instant Messaging/ Chat should provide file transfer facility.
Yes
instant Messaging/chat should provide rich client and web client.
i guess the first question is the same thing
The mail messaging solution should have administrative control for file transfer.
Exchange server wont control file transfer. You have client policies applied through Lync server for
this.
I hope you have seen this link for information related to Lync clients.
http://technet.microsoft.com/en-us/library/gg425836(v=ocs.15).aspx -
Connected Lync Server - change between pool and FE servers
Hi There,
We have found out that when taking "Configuration Information" (you know: Ctrl + Right Click Lync icon beside of the clock) from the Lync client it is for some users one of the Lync front end servers, and for other users it could be the pool's FQDN. But
then, after some day, it could be also the FE's FQDN for those who had pool's FQDN.
When I look the tracing logs I noticed the following line:
SIP/2.0 301 Redirect request to Home Server
This happening after the client has negotiated with the pool's FQDN at first.
I would like to know more about this as it seems that our Live Meeting clients are unable to handle this kind issue.
We have HLB in use because of the legacy clients.
So is it so, that when the HLB direct the connection into the correct home server or the FE where the focus service has seen the user session at last time, there are no troubles. But when it by some reason direct you into different FE of the pool, that server
redirect the session into correct server and bypass the HLB?
I have tested this with the Live Meeting client so, that I have added the FE's IP address and pool's FQDN into host file. This way I have by passed the HLB and the Live Meeting has been working fine. If I add IP from some other FE server, the LM wont working.
Also, is there way to check from the server what is user's session home server?
PetriPetri,
I encountered similar issue last week. After consulting with my good friend Kevin Peters a.k.a. "the ocsguy", turned that upon first sign-in, Lync client will cash some parameters, including Home Server. The moment first (home) server becomes available,
Lync client will attempt to sing-in to it, no matter what the real home server is.
The only way to resolve this would be to exit Lync and bomb the user's (Lync) profile found in C:\Users\USERNAME\AppData\Local\Microsoft\Communicator and sign in again.
Drago
http://www.lynclog.com -
Global SIP trunking from single provider
Our large global organisation is deploying on-premise Lync EV across 200+ sites globally.
We want to minimise the number of suppliers we use in this project and use SIP trunking where possible.
Are there any global telcos that would be able to offer me SIP trunking, with global points of presence on their network, and also 'full' SIP services for most countries.
By full SIP services I mean, competitive pricing (compared to local Telco), emergency number dialling, other special numbers, and full number portability
I have found some of our existing SIP providers can provide services in other countries, but they are not very good at it, and it is overpriced and they don't really know what they are doing.
Surely there must be a few big players who can offer me a truly global service with regional trunks and services for all our locations?Hi There
That’s a great question. The answer is, yes there are. There are a number of global carriers or at least carriers who claim to be global (and who
only serve NA/LA and Europe with bits of the pacific. However, if you use one of these guys, prepare to pay a premium, as their target market is large enterprise business who are prepared to pay for the privilege of a 'one throat too choke' relationship. I
have worked for these carriers, so know this to be fact. In most countries where this TDM telephony and now Internet, somebody has built an in country SIP business. These in country providers, tend to focus on the small to medium up to mid-market enterprise
community, for customers who want in country SIP services. A number of regional integrators have gone to these providers in the major geographies and built SIP propositions on top of their own networks and provider these to customers who are prepared to pay
for the privilege of having one supplier (again I know this to be fact, as I have worked for them).
What the industry needs is a collective agreement where Tier 1 SIP providers in each of these countries, create an alliance based network for the
handoff of traffic at International colocation points. For example, my company in the UK perhaps peering at Tele house and enabling the wholesale SIP connectivity to any carrier who likewise wants to be colocating at that point. This would enable each SIP
carrier to provider their own A-Z IPT tariff and offer in country local break out and emergency services, enabling business A in Country A to call Business B in Country B for the price of a local phone call.
It's so obvious, but finding someone who does this is a challenge. Probable today each of these in country SIP carriers have enough business by
focusing on their own country, without the need to expand their geographies and put their hands in their pockets for the required investment. Just as in the TDM industry where carriers built global A-Z traffic exchange points, this organic growth and service
evolution will take place. When, probably depends on one of two things happening. Either (a) the in country carriers seeing more opportunity from their existing customer base to develop an offer for international service provision or (b) someone visionary
and with deep pockets grabbing the opportunity.
The solution for you? Take service from your in country provider and build your own international end points if you can. If I can be of any father
help do let me know. -
Lync Server 2013 STD pool pairing - certificate questions & DNS entries
Hi guys!
We are thinking about paired pools scenario, but we have one question regarding about certificates requriments on 2nd std pool lync server.
Our current (and only one) front end server (std edition) has this certificate entries:
CN=lync.domain.local
DNS Name=lync.domain.local
DNS Name=sip.domain.com
DNS Name=admin.domain.local
DNS Name=LyncdiscoverInternal.domain.com
DNS Name=lyncreverseproxy.domain.com
DNS Name=Lyncdiscover.domain.com
DNS Name=dialin.domain.local
DNS Name=meet.domain.com
which certificate entries should then have a paired std pool lync server?
bostjancAre you talking about the default cert assigned to the Lync Front Ends or the public certs?
If you are talking about the Lync Front End default cert (and you are using an internal CA) then simply get the second Lync Front End servers cert using STEP 3 of deployment wizard.
Assuming the FQDN of the 2nd std edition is lyncFE2.domain.local, it should look something like this:
CN=lyncFE2.domain.local
DNS Name=lyncFE2.domain.local - Front End #2 FQDN
DNS Name=sip.domain.com - for Lync Phone Edition (2 x DNS entries, 1 to each FE)
DNS Name=admin.domain.local - CSCP (2 x DNS entries, 1 to each FE)
DNS Name=LyncdiscoverInternal.domain.com - mobile clients (2 x DNS entries, 1 to each FE)
DNS Name=Lyncdiscover.domain.com - mobile clients
DNS Name=lyncreverseproxy.domain.com - External Lync web services
DNS Name=dialin.domain.com (assume the .local was a typo?)
DNS Name=meet.domain.com - both meet and dialin (2 x DNS entries, 1 to each FE)
Please mark posts as answers/helpful if it answers your question.
Blog
LyncSorted -
Polycom CX600 "Signing in-" "Connecting to Lync Server"
Hi,
We have a customer who's using Lync 2013 and Polycom CX600 desk phones. They are getting a strange error where UserA can sign in to the phone, sign out, then back in again without error.
If UserB tried to sign in to the that phone they would get the message “Signing in…” “Connecting to Lync Server” it never times out until you stop it!
The only way to allow UserB to sign in is to perform a hard reset of the device. It's a single AD domain with a domain integrated ADCS deployment. They have tried two different levels of firmware with the same outcome. I setup the environment for
Lync PE as per Jeff Schertz's blog post.
Anyone got ideas??
http://lync2010tips.blogspot.com - [email protected] - If this post has been useful please click the green arrow to the left or click 'Propose as answer'Have you checked the logs on the Front End Server and if nothing is there, ran logging through Lync's Centralized Logging Service? Please keep in mind that the immediacy that was OCSLogger.exe does not seem to exist based on my experiences with
CLS.
For checking on what may be occurring, I would recommend the Authentication scenario as it checks the following:
Scenario: Authentication
SipStack
UserServices
WebInfrastructure
UserPinService
CertProvisioning
Start-CsClsLogging -Scenario Authentication
Wait at least a minute, log in as the original user, then try to reproduce the issue, then wait a minute or two and stop the logging
Stop-CsClsLogging -Scenario Authentication
Give it a few minutes to compile a list then do a search of the CLS logs
Search-CsClsLogging -OutputFilePath "C:\Users\Admin\Desktop\Auth.txt"
Now, if you have the Lync 2013
Debugging Tools installed, launch Snooper (The default path is 'C:\Program Files\Microsoft Lync Server 2013\Debugging Tools') and drag the Auth.txt file into it.
With that said, you should be able to better decypher what is occurring based on what happens with the successful log on and cross referencing what happens with the failure. -
How to Integrate Microsoft Lync 2010, Asterisk, and a sip trunk.
Dear Friends.
i need you to assist me to step my new project
Objective:
Setup Asterisk
to Configure a SIP trunk between Asterisk and the SIP provider of my choice
Integrate Lync Server 2010 with Asterisk
Configure a dial plan
Configuring Voice Polices, PSTN Usage Records, and Voice Routes.
To be able to make international
local call to any mobile extension or same number range
This is a new project to me can anyone please simply assist me step by step ?
Thanks
GreenmanHi GreeMann, Which Flavor of Asterisk you are using ex: FreePBX, Elastix, AsteriskNow.
You can use any of them most of the configuration will be similar.
To configure the SIP Trunk of service provider in asterisk check this
http://wiki.freepbx.org/display/ST/Setting+up+SIPStation+manually+in+FreePBX http://wiki.freepbx.org/display/F2/Trunk+Sample+Configurations
Here is my blog Step by step guide to Integrate asterisk ( Elastix) with Lync
http://mslyncforall.blogspot.in/2014/12/lync-2013-asterisk-pbx-integration.html
http://blogs.technet.com/b/rischwen/archive/2013/08/21/series-exchange-2013-and-lync-2013-integration-with-asterisknow-pbx-pt-1.aspx
Please let me know if you encounter any issues i am happy to help you.
Whenever you see a helpful reply, click on Vote As Helpful & click on Mark As Answer if a post answers your question. -
Problems with SNOM 7XX phones and presence of Lync Edge server
Hi to all,
we have this problem, this is the scenario (two Lync 2013 st ed. servers):
- lync 2013 FE server have internal IP address 172.21.212.XXX with internal gateway 172.21.212.254
- lync 2013 edge server have two network interface:
First INTERFACE: 3 IPs in 172.21.30.XXX (Access, web and A/V Edge) for external connection with 172.21.30.254 and internal gateway (IP NAT with public IP)
Second INTERFACE: IP 172.21.212.XXX for internal connection without gateway
- snom 7XX (50 phones) are connect to the lync server and all internal call works fine. All phones are in an internal dedicated network 172.21.218.XXX with default gateway 172.21.218.254
- when making external call with 7XX SNOM phones, the call was routed to Trunk COLT with Lync Mediation server and all works fine.
- when Lync Mediation server receive a call from our trunk COLT we have this situation:
All Lync 2013 clients work fine, audio is OK, (network 172.21.216.XXX)
Polycom CX3000 work fine audio is OK (network 172.21.218.XXX)
SNOM 710, 720, 760 FW 8.8.2.16 UC series, phones ring but NO SOUNDS from the phones and after a few seconds "Call failed due to network issues."
The only way to solve the problem is to disable the connection with Lync Edge server (remove gateway 172.21.30.254)
BUT this is not the solution because now we have no connection with INTERNET (skype, web conferencing doesen't work without edge gateway)
Why SNOM phones try to use the EDGE gateway to connect the call? Why doesn't use Lync Mediation server?
Can you help us to find a solution?
Thanks
AurelioHi,
Thanks to all for yours support.
Today, we have done some test (no employee in office today
J) and we have solved the problems.
The old implementation have had this configuration:
- the phone numbers have had a no E.164 format compliant: for all users number, the phone number have had this format TEL:012345XYZW ; EXT=XYZW with the normalization
rules:
Starting digits: 01234567
Length: At least 8 digits
Digit to remove: 0
Digit to add: nothing
Pattern to match ^(01234567\d*)$
All worked fine with this previous configuration:
Lync 2010 std with only mediation server function + Lync 2013 std front-end with all the others functions and Lync 2013 std Edge server for external connection with
Lync client Skype world, BUT we have had disabled in SNOM phones ICE function because if ICE was enabled no voice can we hear from the phones.
After dismissed Lync 2010 with only a Lync 2013 infrastructure, this configuration don’t permit to use edge server because with ICE enabled or disabled no voice from
SNOM phones.
Today we have done this operation:
Setting in Lync 2013 control panel all number for all users, in E.164 format compliant:
The phone number now have this format TEL:+39012345XYZW ; EXT=XYZW and we have deleted the previous normalization roles.
We have added this role for the EXT numbers:
Name: Routing Interno
Starting Digits: XY
Length: Exactly 4 (i.e. XYZW)
Digit to remove 0
Digit to add: +39012345
Pattern to match: ^(XY\d(2))$
Translation rule: +39012345$1
Internal extension = checked
And now all work fine.
We have solved another problem:
Lync client 2013 can't find new users:
all new Lync users are not discovered from Lync 2013 client, probably because this setting is present with Lync 2010:
PS C:\> Get-CsAddressBookConfiguration
Identity
: Global
RunTimeOfDay
: 1:30 AM
KeepDuration
: 30
SynchronizePollingInterval : 00:00:30
MaxDeltaFileSizePercentage : 20
UseNormalizationRules
: True
IgnoreGenericRules
: False
EnableFileGeneration
: True
With only Lync 2013 servers we have changed
IgnoreGenericRules to True
To set UseNormalizationRules and IgnoreGenericRules to true for Lynk 2013 infrastructure.
http://technet.microsoft.com/en-us/library/jj205160.aspx
For us all the problems are SOLVED!
Aurelio -
Forwarding with SIP Trunking and Retaining CID
We are using SIP trunking to get to the PSTN and CUCM 7.1. The SIP provider only permits calls originating from their own DIDs.
So how can we allow our users to forward all calls to say their cell phone. AND when a call comes in we want them to be able to see the original caller ID. Is it possible? What is the mojo? Thanks!Hi,
for the SIP trunks there are no limits from the system. Check this out: https://supportforums.cisco.com/message/3795863#3795863
If you have different voice codecs for your phones and the trunk you need DSP ressources for transcoding.
As I know the UC520 have a PVDM2-64 with 4 DSP chips. You can use the DSP calculator from cisco to find out how many DSPs you need. But keep in mind that conferencing and transcoding can't share a DSP processor.
For example 1 DSP for conferencing and 3 for transcoding.
best regards
Christian -
SIP trunking between Microsoft OCS server and Cisco Voice GW router.
Hello All,
I have a client with an existing Microsoft OCS (office communications server) environment with the OCS server in their head office. The OCS clients in the remote Office registers with the OCS server in the head office. The WAN connectivity between the remote office and the Head office is MPLS.I would like to facilitate local call (PSTN) features at the remote site through a newly proposed Voice gateway router.
Can I achieve this by doing a SIP trunk between the OCS server in the head office to the newly proposed voice GW router in the remote office through the existing MPLS link. If yes, Could any one please assist me in this regards or suggest any other best solution to achieve the same.
Thank you in advance,
Mohammed Ameen RHi David,
this is a normal behaviour. To CUCM, OCS is a remote destination (just like your mobile phone). When your mobile phone hangs up, the system will put the call on hold for 10 sec.
This is there for the mobile user to go to his desk to pick up the call and continue the conversation (part of single number reach feature)
The best practise will be for the user to ensure that the other party hangs up the call first before he hang up.
Please grade if you think it's useful =) -
Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP
Hi Cisco Community,
I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
Below is an example of a call that is connected with the current setup:
Note:
IP: 10.18.81.2 (CUBE)
IP: 10.18.81.11 (CUCM SUB)
IP: 10.111.111.254 (ITSP SBC)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
Session-Expires: 1800
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1417347869
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 301
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
s=SIP Call
c=I
PM-HO-VG-01#N IP4 10.18.81.2
t=0 0
m=audio 22256 RTP/AVP 18 0 8 101
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf9
PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,application/xml
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Type: application/sdp
Content-Length: 236
v=0
o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.80.40
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
PM-HO-VG-01#
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
PM-HO-VG-01#sh sip
PM-HO-VG-01#sh sip-ua call
PM-HO-VG-01#sh sip-ua calls
Total SIP call legs:2, User Agent Client:1, User Agent Server:1
SIP UAC CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 27218091323
Called Number : 0862000000
Bit Flags : 0xC04018 0x10000100 0x0
CC Call ID : 64511
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.111.111.254]:5060
Destn SIP Resp Addr:Port: [10.111.111.254]:5060
Destination Name : 10.111.111.254
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64511
Stream Type : voice+dtmf (0)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22256
Media Dest IP Addr:Port : [10.111.111.254]:20074
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 0218091323
Called Number : 0862000000
Bit Flags : 0xC0401E 0x10000100 0x80004
CC Call ID : 64510
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.18.81.11]:5060
Destn SIP Resp Addr:Port: [10.18.81.11]:5060
Destination Name : 10.18.81.11
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64510
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22350
Media Dest IP Addr:Port : [10.18.80.40]:21928
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1
PM-HO-VG-01#
PM-HO-VG-01#
PM-HO-VG-01#
As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22256 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 102 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 360
v=0
o=BroadWorks 316169737 2 IN IP4 10.111.111.254
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
a=inactive
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22350 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Length: 0
Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 103 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 306
v=0
o=BroadWorks 316169737 3 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 2
PM-HO-VG-01#00 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 213
v=0
o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.81.10
t=0 0
m=audio 4000 RTP/AVP 18
a=X-cisco-media:umoh
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=sendonly
Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 101 BYE
Reason: Q.850;cause=86
P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 104 BYE
Reason: Q.850;cause=65
P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 Race Condition
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
Timestamp: 1417347889
CSeq: 104 BYE
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 200
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 101 BYE
Content-Length: 0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 86
Disconnect Cause (SIP) : 200
PM-HO-VG-01#Hi Manish,
Again, excellent feedback. Much appreciated.
I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
I will be doing some intensive test again later on this week and will send the logs.
Here is my question to both of you:
Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
Thanks again for your support fellows. -
Cucm , Cube via Sip and Sip Trunk to ISP , Outgoing calls not working
Hi
We have issue with the outgoing calls to sip trunk
Below is the config and the debugs
It will be great if you give your thoughts since we have stuck here
My thoughts are:
i see that for unknown reason the called number is going with 4 digits instead of 8 digits
i dont see any sip message comming from ISP
Maybe the call not going there ? to isp trunk? From the trace the call hit the correct dialpeer 888 but i see 4 digits as a called number , but i dodnt understant the reason to translated in 4 digits the called number.Not apply a translation rule for that
confused!!!
Calling Numbner:22324086
Called Number: 23823690
CUCM:192.168.1.241 and 242
CUBE:192.168.1.10
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*\(....\)/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^\(9\)\(.......$\)/ /99\2/
rule 3 /^\(2\)\(.......$\)/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*\(....\)/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^\(9\)\(.......$\)/ /99\2/
rule 3 /^\(2\)\(.......$\)/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vadHi Aok
I change the default value for IPVMS from g711ulaw to g711alaw but the results remained the same
Also i have restarted the IPVMS
SIP-GW#
SIP-GW#
*Mar 5 14:19:57.854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 38874 2 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24784 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:57.878: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1362493197
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3807 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 101
c=IN IP4 10.249.13.130
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.878: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:57.926: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 102 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 213
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 2 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=inactive
*Mar 5 14:19:57.942: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5774 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 101
c=IN IP4 192.168.1.10
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.946: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3562A4
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:57.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK798246ab3597
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
*Mar 5 14:19:58.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Length: 0
*Mar 5 14:19:58.158: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1362493198
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:58.158: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:58.218: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 103 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 216
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 3 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 18 96 99
a=rtpmap:96 AMR/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
a=sendrecv
*Mar 5 14:19:58.234: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 283
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5775 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 18 101
c=IN IP4 192.168.1.10
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:58.242: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7985648033f2
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 192
v=0
o=CiscoSystemsCCM-SIP 38874 3 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 192.168.1.241
t=0 0
m=audio 4000 RTP/AVP 8
a=X-cisco-media:umoh
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly
*Mar 5 14:19:58.262: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK358582
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3808 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 99
c=IN IP4 10.249.13.130
a=sendonly
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
SIP-GW#
SIP-GW#sh voip rtp connections
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 716 717 19314 4000 192.168.1.10 192.168.1.241
2 717 716 19234 54932 10.249.13.130 10.224.42.72
Found 2 active RTP connections -
PROBLEM WITH FORWARDING ALL - SIP TRUNK
Hello,
I'm experiencing the following problem:
I have this scenario: PSTN - SIP GW - CUCM6.1 - SIP TRUNK - CUM8.6
Phone A (extension 33476761834) is registered on CUCM8.6, with external forwarding to the PSTN.
If someone from CUCM8.6 calls - it works.
If someone from CUCM6.1 or from PSTN calls - fast busy. Error in CUCM traces:
19:20:47.056 |//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 145.245.235.201 on port 36615 index 3295
[1925279,NET]
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/TCP 145.245.235.201:5100;branch=z9hG4bK3744a3159aed8f
From: <sip:[email protected]>;tag=e1d37fe6-cb7b-46e7-a868-6fe81d6bb391-40319184
To: <sip:[email protected]>;tag=907605~2b367b5a-23ed-4193-a18b-e8c2f777615e-62999964
Date: Thu, 26 Sep 2013 17:20:46 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: presence
Reason: Q.850;cause=100
Content-Length: 0
|2,100,63,1.580442^145.245.235.232^*
Any idea? Thanks a lot!
(Attaching the complete trace).Hi,
Making some tests regarding the problem we saw this:
Oct 1 23:02:22: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Date: Tue, 01 Oct 2013 23:02:22 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, PUBLISH
From: <sip:[email protected]>;tag=e1d37fe6-cb7b-46e7-a868-6fe81d6bb391-40468009
Allow-Events: presence
Supported: timer,replaces
Min-SE: 1800
Diversion: " Elodie" " <sip:[email protected]>;reason=unconditional;privacy=off;screen=yes
Remote-Party-ID: <sip:[email protected]>;party=calling;screen=yes;privacy=off
Content-Length: 0
User-Agent: Cisco-CUCM6.1
To: <sip:[email protected]>
Contact:
Expires: 180
Call-ID: [email protected]
Via: SIP/2.0/TCP 145.245.235.201:5060;branch=z9hG4bK3863af2fe80489
CSeq: 101 INVITE
Session-Expires: 1800
Max-Forwards: 67
Oct 1 23:02:22: //350172/5D95E6C38917/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Malformed CC-Diversion/Diversion/CC-Redirect Header'
Via: SIP/2.0/TCP 145.245.235.201:5060;branch=z9hG4bK3863af2fe80489
From: <sip:[email protected]>;tag=e1d37fe6-cb7b-46e7-a868-6fe81d6bb391-40468009
To: <sip:[email protected]>;tag=DA70FE78-21AE
Call-ID: [email protected]
CSeq: 101 INVITE
Reason: Q.850;cause=100
Content-Length: 0
The problem is because in the redirect header, the comma "," is not a valid parameter so in the alerting name I removed the comma after the surname (i.e. "Elodie, Mary" to "Elodie Mary" and now is working.
We will change the alerting name by the moment and I will also investigate if there is a parameter to just not divert this name because is not needed this info in forwarded calls.
Anyway the problem is solved -
Configuring Level3 SIP trunk with Lync 2013
Hi, I ran into some issues trying to configure SIP trunk from Level 3 and I was hoping someone here can help. We have our mediation server collocated with FE and SIP traffic goes from public IP, port 5060 via NAT, to local IP on FE, port 5060.
Level 3 provided us with one signaling IP and two RTP IPs.
I tried multiple trunk configuration settings and I can see that when I'm placing a call from Lync to an outside number I'm getting INVITE from Level 3 signaling IP, the session is established, phone rings, but there is no audio on either side. There's also
a METHOD NOT ALLOWED message coming from them, which doesn't tell me much about what's happening.
If I call to a Level 3 DID (assigned to my Lync user account) there's also INVITE from their side, but later I receive a CANCEL from them due to idle session. The phone never rings.
Questions:
1) Does anyone have Level 3 SIP trunks configured and can share their Get-TrunkConfiguration settings? What settings should I have for encryption, refer, sessionTimer / RTCP, and others? Level 3 refuses to provide any additional information besides IPs.
2) Do I understand this correctly that when configuring PSTN gateways in topology, one of the RTP IPs should be entered in the "alternate media IP" field? We have SIP trunks from another provider (which work fine), and they only use one IP
for everything, so I don't have any experience configuring separate SIP and media IPs with Lync.
Thanks, and let me know if I should provide additional info.Hi,
On Lync topology PSTN gateways interface, please check if you enter gateway listening port 5060 and enable TCP option.
Please also check if you enable refer support on Lync Server Control Panel, if you enable it please uncheck it.
You can compare the trunk configuration for Level 3 in the part “Sample Trunk Configuration for Level 3” in the link below with yours’, it is for Lync server 2010 but similar for Lync server 2013:
http://blogs.technet.com/b/nexthop/archive/2013/04/10/configuring-lync-2010-server-to-work-with-level-3-sip-trunking-services.aspx
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support
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