Unable to perform call transfer or call park for an outbound call via SIP Trunk (SKYPE)

We have configured the SIP Trunk & SIP profile and successfull make outbound call through SIP Trunk (SKYPE). However, we are not able to perform call transfer or call park when the call is connected.
The scenario is:
A call to an phone number via SIP trunk, when call established, A perform call-transfer to B. After the call-transfer, the call Drop and Phone B show error code "Temp Fail"
When i select "enable MTP" in SIP trunk, we are able to call transfer and call park. But it limit the number of call session to 1.

You are probably running into some sort of Codec issue. IE, your phone is G.711 and the trunk is G.729. You will need to transcode the call at somepoint.

Similar Messages

Unable to perform call transfer & call park through SIP Trunk (SKYPE)

The Scenario is:
I have set up a SIP trunk to SKYPE and we are able to make outbound call to a number via SIP Trunk.
After the call is established, when we tried to make call transfer, the call DROP and the phone at the other end shows error "Temp Fail".
I tried to "enable MTP" in SIP Trunk and We are able to perform call-transfer but it limits the call session to 1.
Anyone has facing the same issue?

MTP is needed to invoke supplementary functions like hold, transfer etc. Make sure that the MTP is checked on SIP trunk, MTP is assigned to the MRGL of the device pool on SIP trunk and has sufficient resources.
HTH
Manish

Wrap-up time for manual outbound calls (UCCX)

Is it possible to configure the wrap-up time for the manual outbound calls in UCCX? I think, this option only exists in CSQ, which is of course meant for the inbound calls. Any thoughts or any workaround to make this work?
Requirement- Once a manual outbound call is hung up, agent's state should be switched to WORK READY as per the wrap-up timer setting.
Thanks.

Dear experts,
I look forward to hear from you if you have anything to offer. Wrap-up TIME to be setup for manual outbound calls in UCCX.
Thanks,
Piyush

ILBC calls via SIP Trunk with CUBE and CUCM7

hi there,
our SIP Provider offers the IBLC codec which promises to provide better quality compard to G.729.
I'm using this scenario:
IP-Phone(G711) --- CUCM7 --- (SIP-Trunk1) --- CUBE --- (SIP-Trunk2) --- Provider
Everything workes unless I'm configuring IBLC at the provider and on trunk2.
I have the CUBE router acting as a trancoding device and also specified IBLC as codec to be handled.
SIP trunk 2 was placed in a region with IBLC as codec.
On the trunk configuration in CUCM the media ressource group with XCODE capability is configured
Transcoding workes between two IP Phones in different regions with different codecs within the intranet.
Unfortunately the CUBE router doesn't seem to use the transcoder to change internal G711u calls into IBLC codec
so calls are blocked by the CUBE device:
deb ccsip calls
for incoming call:
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4AE7AC98
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 0237892992
Called Number            : 036677725231
Source IP Address (Sig ): 10.100.100.50
Destn SIP Req Addr:Port : <IP SIP Provicer>
Destn SIP Resp Addr:Port : <IP SIP Provicer>:5060
Destination Name         : <IP SIP Provicer>
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : ilbc
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): <IP CUBE>
Source IP Port    (Media): 0
Destn IP Address (Media): <IP SIP Provicer>
Destn IP Port    (Media): 22022
Orig Destn IP Address:Port (Media): [ - ]:0
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 65
Disconnect Cause (SIP)   : 488
(Output lookes similar to outgoing calls)
I set up ccm on cube and assigned dsp ressources without success:
Here are the relevant configuration parts:
voice class codec 1
codec preference 1 iblc
voice service voip
address-hiding
allow-connections sip to sip
allow-connections h323 to sip
allow-connections sip to h323
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
h323
sip
header-passing error-passthru
no update-callerid
midcall-signaling passthru
privacy-policy passthru
voice-card 0
dspfarm
dsp services dspfarm
dial-peer voice 40991 voip
description *** Incoming from SIP-Provider
destination-pattern 03667772523.%
session protocol sipv2
session target ipv4:<IP_of_CUCM>
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
ip qos dscp cs5 media
ip qos dscp cs5 signaling
sccp local GigabitEthernet0/0
sccp ccm 10.100.100.50 identifier 11 version 4.1
sccp
sccp ccm group 11
description *** lokaler CCM fuer Codec-Konvertierung von SIP/DUS.NET
associate ccm 11 priority 1
associate profile 21 register DE_WGT_MTP02
dspfarm profile 21 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec ilbc
maximum sessions 10
associate application SCCP
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 10
sdspfarm tag 1 DE_WGT_MTP02
max-ephones 30
max-dn 30
ip source-address 10.100.100.50 port 2000
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Mar 14 2010 02:10:34
sh sccp
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
        IPv4 Address: 10.100.100.50
        Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.100.100.50, Port Number: 2000
                Priority: N/A, Version: 4.1, Identifier: 11
                Trustpoint: N/A
Call Manager: 10.1.1.55, Port Number: 2000
                Priority: N/A, Version: 7.0, Identifier: 10
                Trustpoint: N/A
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.100.100.50, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 21
Reported Max Streams: 20, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
sh dspfarm dsp all
SLOT DSP VERSION STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED
0    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
0    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
Thanks in advance,
David

Hi there,
Just wondering whether you ever got this resolved? I seem to have a very similiar problem.
Regards
Karen

To create Trunk Group with differnt T1 PRI's groups for our outbound calls

Hi All,
I would like to request all of you that I have requiremnt that we have to create "Trunk Group" of diffenent T1 controller PRI's for outbonds call, mean I need to create a trunk group with differnt T1 PRI's groups for our outbond calls, but my problem is that I don't know and no idea that how I will do this , i am also try to find some cisco doc for this still I did not find these info, so I request all of you that I will be thank full to all of you if you can help me out for my this problems.
Thanks
Rizwan

Here is a good write up, there are many others:
http://www.markholloway.com/blog/?p=452
HTH,
Chris

2901 CME: Problem with incoming call via SIP Trunk

Dear All,
I have seen some others posted similar question regarding this but mine still doesn't work by using the reference solution.
Mine is quite standard setup too -> CME setup on my 2901 router, analog phone attach to my FXS port my outgoing calls are working fine via SIP but my incoming calls are not. Caller only listen to engage tone and analog phone is not ringing at all. Attached with my config and trace log of ccsip messages. Kindly assist. Thank you so much.

Hi Carlo,
Here it is
CME_2901#show sip-ua timers
SIP UA Timer Values (millisecs unless noted)
trying 500, expires 180000, connect 500, disconnect 500
prack 500, rel1xx 500, notify 500, update 500
refer 500, register 500, info 500, options 500, hold 2880 minutes
, registrar-dns-cache 3600 seconds
tcp/udp aging 5 minutes
CME_2901#show sip-ua retry
SIP UA Retry Values
invite retry count = 6   response retry count = 6
bye retry count    = 10 cancel retry count   = 10
prack retry count = 10 update retry count    = 6
reliable 1xx count = 6   notify retry count   = 10
refer retry count = 10 register retry count = 6
info retry count   = 6   subscribe retry count = 6
options retry count = 6
CME_2901#show sip-ua min-se
SIP UA MIN-SE Value (seconds)
Min-SE: 1800

Problem with call forwarding. Calls can not be forwarded for incoming external calls

Hi Everybody, how are you?
I have a problem with call forwarding. Everything was fine but now is not working.
In the reception of an office, the receptionist activate the call forward option to an internal extension. If somebody, internal in the office, call to the reception, the call is forwarding to the extension configured. But if I call from the outside (in example, from my cellphone) the call is not forwarded to the extension configured and continue ringing in the reception phone. Why this behavior? Any idea?
If you know something please tell me.
Thanks. Best regards.
Andres Collazos.

I encounter a similar problem with 9.1.1.
My problem is link to this bug ID : CSCtq10477.
Mathieu

Unity Call Transfer with announce from IPCC

I'm trying to setup a script in IPCC for Meetme users to call into, enter in a PIN number, and be sent to the correct Meetme conference. I've got all of that working just peachy. I was hoping to also be able to use Unity's call transfer with announce functionality using a call handler. Unfortunately through testing and also verified in documentation, the call transfer functionality is ignored for forwarded calls (calls not handled by the unity auto attendant or directory handler).
Can someone enlighten me on a workaround for this? Basically I want the caller to come into IPCC, enter their PIN, IPCC shoot the caller to a Unity call handler which prompts for the user's name, then Unity announce that person's name to the conference and then finalize the call transfer to the Meetme conference.
I've tried all sorts of combinations using CTI route points in call manager, modifying direct and forwarding route patterns in Unity etc. No dice so far to get the call announce working. Many thanks in advance!
--Jason

I appreciate all the responses. I was able to get this to work, my dunce hat was on and my Unity forwarding rule was sending to the greeting of the Call Handler rather than the Attempt Transfer.
The powers that be at my company wanted to assign a different MeetMe number to each department in order to reduce the risk of multiple people trying to use the same conference line at the same time. I did not like the fact of having to setup 20 cti route points in CallManager, and then 20 call forwarders inside of Unity that then sent to 20 separate call handlers. I found a way to bypass the cti route points and call forwarders by having IPCC send the call directly to Unity.
The trick to bypassing CallManager CTI route points that simply forward to Unity, and setting up call forwarders in Unity is this:
1) In IPCC, use a consultive transfer instead of a call redirect. This allows you to first dial the main voicemail number and then dial the specific voicemail box once you are in Unity.
2) On your call handler in Unity, simply specify the extension number of the call handler (I used the same number as the MeetMe line), and also set the forwarding number.
So I've been able to change the process from Caller->IPCC Script->Callmanager CTI RP->Unity forwarder->Unity call handler->MeetMe to simply Caller->IPCC Script->Unity call handler->MeetMe
As for my IPCC script, I used arrays to handle my plethora of conference lines, it greatly cut down on the size of my script.

UCCX HR Report for Outbound Calls

Hi,
We are using UCCX 9.0.
In Historical report we able to get the report for outbound call also.
The issue for the Outbound is calls which are not answered by the remote party There is also a call duration though the CUCM CDR shows zero seconds and the Agent Call Records in CAD shows the call and unanswered.
Because of this we are not able to identify the matured calls & non matured calls for the outgoing.
Is there anyway to capture unanswered outbound calls from Historical reports
Thanks,
Paul

I must say that I had similar problems with reports for outbound calls on our CC system...after some writings with TAC we concluded that reports for outbound calls in CC system are much more less informational versus inbound calls. So what I want to say is that probably your reporting is working fine and there is just no info for outbound calls which can be utilized and be usefull for you in your reports...
BR,
Dragan

Redirect SIP Trunk calls to FXO port

Hi,
This is the scenario. There are 3 branches, two of them are Cisco Call Manager Express and one of them is Elastix-based.
So, as the image explains, the three branches have SIP trunks fully operational. The branches are in different cities, so the numbers structure changes. In city A it begins with 2, in B begins with 3 and in C begins with 4. Every POTS number is a 7 digit number (2XXXXXX, 3XXXXXX, 4XXXXXX). And every user, in every branch, have a 4 digit number beginning with the city code (2XXX, 3XXX, 4XXX).
But, every time city A wants to make a call to a POTS number in city B, it goes across the A´s FXO line. So it charges a inter-city cost to the call.
The client wants that every time a city A user wants to call a POTS number in city B, goes over the SIP trunk to city B and use the FXO on the city B call manager.
I have made a pattern for city A. So, everytime the user dials 3XXXXXX, it does not use the city A´s FXO, but it goes to the branch in city B.
What do I have to do now in branch B´s Call Manager Express to redirect that call to a local FXO?
Thanks in advanced!
Regards
PS. There is a diagram of the topology. Want to do what the red line is doing.

In this situation I would do an answer-address based on ANI so you are specifically identifying your site A and then just piggy back off the local FXO out.
So assuming you are sending just 4 digits over the SIP for each site:
Dial-peer voice X voip
answer-address "blah"
protocol sipv2
...(whatever else you need to configure in these dots)
At this point your CME at site B will take the call see that it is destined for a POTs line and it should send it out whatever local dial-peer you have setup for that site when they dial out to the PSTN locally.
EDIT:
Then again, you probably already have a general incoming dial-peer, the above design would just be specific for your site A and isn't really needed.

Calls via Gamme sip trunk

Has anyone set up Lync server 2010 to use the Gamma SIP trunks, that dont require the use of a gateway?
No requirement for an additional gateway device, with direct MS Lync connectivity
The trouble is i cannot get Lync to connect to the trunks. We have purchased the SIP trunks from a gamma supplier(we didnt now they were a supplier, until recently when we asked for support and they went 'duhhhhhh me no know, we just
sell things dunow how to set things up' what a PAIN IN THE A***), and they say that the SIP trunks are pointed at our EFM IP address. which also has the DDIs assigned to it.
So, i setup a PSTN gateway on lync topology using IP of EFM, Listening port 5060 using TCP. Are these ports and protocol okay?
The VoIP phones seem to want to call, they just lack any sound, no ringing tone, no dissconnected tone. Just says calling "+44157322****" So the dial plan is changing 22**** to the correct local code and whatever the +44 thing
is.
Any advice on how i can find the problem, or how to setup the trunks up would be hugely appreciated.
P.S We initially tried to use an audiocodes mediant 1000, which was what we asked our trunk supplier about, and then they informed us about being a gamma supplier, and that the gamma trunks do not require a gateway. Followed setting
up guide for mediant 1000 with gamma trunks through audiocodes blah, to no success. I think thats because it was changing the coders, which was not needed if the trunks are directly compatable.

Hi,
Please review the SIP trunk topology.
http://technet.microsoft.com/en-us/library/gg398720.aspx
To
implement SIP trunking, you must route the connection through a Mediation Server, which proxies communications sessions between Lync Server 2010 clients and the service provider and transcodes media when necessary. Each Mediation Server has
an internal and an external network interface. The internal interface connects to the Front End Servers. The external interface is commonly called the gateway interface because it has traditionally been used to connect the Mediation Server to a PSTN gateway
or an IP-PBX. To implement a SIP trunk, you connect the external interface of the Mediation Server to the external edge component of the ITSP. The external edge component of the ITSP could be a Session Border Controller (SBC), a router, or a gateway.
Generally the gateway is not required in your organization. You need to configure Mediation Server setting. For the details about
the SIP trunk configuration of ITSP side, you need to contact Gamme Support for further assistance.
Regards,
Kent Huang
TechNet Community Support ************************************************************************************************************************ Please remember to click “Mark as Answer” on the post that helps you, and to click “Unmark as Answer” if a
marked post does not actually answer your question.

Cisco Mobile 8.1 w/iPhone 4 - Outbound Call Fails

Hello Everyone,
Using CUCM 7.1.5 and Cisco Mobile iPhone app, I set up my boss's iPhone 4 and it registers and can receive calls. I am getting a "Call Failed" when trying to make an outbound call using internal 5-digit dialing. Has anyone set this up? Is it necessary to configure these Application Dialing Rules (and install the COP file for them) that are referenced in the Cisco Mobile Admin Guide? I interpretted those rules as optional, if you want to set up things like bypassing dialing 9 for PSTN access, etc. Shouldn't the internal dialing work since the phone is registered?
Grateful for any help,
Kelly

No, you do not need Application Dial Rules for Iphone to IPPhone calls. ADR are for
like you said to add a 9, remove the + and manipluate digits.
I suggest you open a case with the TAC to help you see why you cannot call. Maybe issue
with ASA or something.
George

Calling Party No Display in SIP Trunk

Hi All,
We are integrating the CUCM & CME via SIP trunk where we facing the problem in Caller ID.
In CUCM & CME we have 4 digit extension. It will transformed to 8 digit Enterprise number while calling from/to cucm & cme. Once the call answered it displays the 4 digit extension of called party instead of 8 digit enterprise no.
Hope it's updating the Phone caller Id based on "P-Asserted Identity" in SIP messages . My requirements is it should display 8 digit no & Name.
Flow : Phone --> CUCM --> SIP Trunk --> CME --> Phone.
Any Suggestions !!!

Hi All,
I just tried with SIP Profiles on the dial peer to remove the Remote-party ID.
When call initiated from CME to CUCM , "Remote Party Id" is removed in Invite. When CUCM reply with 180 ringing it contains "Remote Party Id" but sip profile not having any effect.
This to avoid the displaying CUCM internal number series in CME instead of DID.
Can you advice how it can be achieve ?
Configuration
voice class sip-profiles 10
request ANY sip-header Remote-Party_ID remove
response ANY sip-header Remote-Party-ID remove
dial-peer voice 100 voip
destination-pattern 91T
session protocol sipv2
session target ipv4:<ip address>
voice-class sip profiles 10
Call Flow
SCCP Phone -- CME --- SIP -- CUCM --- SCCP Phone.
Thanks in Advance

Lync Monitoring Server - Attempted outbound call

Hi,
Im looking at the tables in the monitoring server as a means of monitoring outbound calls where the outgoing agent terminated the call prior to the call being picked up. Is this recorded in the monitoring server? I've checked the ErrorReport which is where
I thought this would land, as well as the sessiondetails table.
Does lync even report such transactions?
Cheers,
Dave.

Hi,
You can try to use Call Diagnostic Reports to check the attempted outbound call.
The Call Diagnostic Reports provide summary information and diagnostic data for failed peer-to-peer and conferencing sessions.
More details:
http://social.technet.microsoft.com/Forums/en-US/9d75e7c4-ba5d-4884-8ec8-ee3bd7195821/lync-monitoring-server-attempted-outbound-call?forum=ocsmonitoring
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Caller ability to toggle option for Compliance Recording on UCCX

Can one can configure UCCX and Compliance Recording to provide the option to the caller to turn off Recording for their particular call? I had the thought to present the caller with the option in the initial prompt and move the caller to another queue which isn't being recording by CR. Anyone done this before?
Thanks.

Hi
AQM supports IPC actions - that should mean that the easiest way to get this running would be:
1) In the script, allow caller to elect to be not-recorded
2) Based on that, set a extended call variable in CAD to mark it for non-recording
3) In CAD, execute a voice workflow that (based on the extended var being set to 'no-record' or whatever) sends an IPC action to delete the recording to AQM.
Details of integrating CAD to QM in this way are here: http://www.cisco.com/en/US/products/ps8293/products_programming_reference_guides_list.html
Regards
Aaron

Unable to perform call transfer or call park for an outbound call via SIP Trunk (SKYPE)

Similar Messages

Maybe you are looking for