Call forwarding

hello all,
I can't understand why iphone did not make the software more friendly (bussiness) talking about call forwarding.
other smartphones, have the option to forward a call depending of situation. maybe I want to forward only when I;m out of reach, or when I decline a call.
instead, aple offer us just this solution: to forward all calls.
what do you all think?
thanks,

Odd, but then again, that is a carrier provisioned service.
Call your carrier.
Message Edited by JSanders on 03-06-2009 03:41 PM
1. If any post helps you please click the below the post(s) that helped you.
2. Please resolve your thread by marking the post "Solution?" which solved it for you!
3. Install free BlackBerry Protect today for backups of contacts and data.
4. Guide to Unlocking your BlackBerry & Unlock Codes
Join our BBM Channels (Beta)
BlackBerry Support Forums Channel
PIN: C0001B7B4 Display/Scan Bar Code
Knowledge Base Updates
PIN: C0005A9AA Display/Scan Bar Code

Similar Messages

CUPC 8.6 call forward to voicemail

Hi!
I am using Cisco Personal Communicator (CUPC) 8.6 and also CUCM 8.6. I have CUPC in Deskphone mode, connected to a 6945 IP Phone. I also have Unity Connection where my voicemail box is hosted. When I want to setup call forward to voicemail button in cupc option, it is not working. CUPC will not handle the options I setup seconds before. If I manually put in a call forward to extension number of voice mail pilot call forwarding is working. also call forwarding to my mobile is working.
I checked End User settings, IP Phone is associated to my user, also CTI controll is enabled on device and line settings. user privileges are correct. I tried it on jabber client where it works fine. I also restarted CTI and Callmanager Services on the Servers.
Does anyone has an Idea if this is a general bug in CUPC or does anyone can tell me what the problem might be?
Thanks!
René

Hi,
If at least one of these phones is set to CF to VM then it will, if not, then no.
If none of your phones is set to CF to VM CUCM will not send them to VM, that is expected, if you need to ring, phone A, and if it is not answered to go to phone B, C... and so on, and send the caller to VM after you have reached all of these then use a hunt group, (the pilot can be set to CF to VM if nobody answers), if you need to ring all phones at the same time so someone can pick this up, use a hunt group with a broadcast logic.
If this is for a single user, check 'single number reach' (SNR) or mobility on CUCM.
Bottom line, there is no way to send a caller to VM if none of the phones is set to CF to VM.
HTH
Chris.

How to delete the number saved in CALL FORWARDING list?

how to delete the number saved in CALL FORWARDING list? seems like the number cannot be deleted in CALL FORWARDING list, i tried several times...
he thing is, here are three numbers in the call forwarding list, one of them were set up by mistake, and none of them can be deleted.......it brothers me all the time. i already turn it off, but the number list is still there...

Hello dennis130915 and welcome to the BlackBerry Support Community Forums.
The image you have uploaded is unable to be displayed.
Call Forwarding is a carrier controlled feature. Most likely, the number you see could be for the Voice Mail server.
It's advised you call your mobile service provider to further assist with this feature.
Thanks!
-HMthePirate
Come follow your BlackBerry Technical Team on twitter! @BlackBerryHelp
Be sure to click Kudos! for those who have helped you.Click Solution? for posts that have solved your issue(s)!

Call forwarding on iPhone4. I am sick of trying but once I input the number and go back to call forwarding, it turns off call forwarding. Any idea why

tried the simple way of call forwarding going in to settings>call forwarding> turn on call forwarding> enter number> return to call forwarding
but the status is going back to 'off' from 'on'
I also tried *61*number# but there again its failing. Can someone assist please?

Who is your carrier? Not all carriers support enabling and disabling call forwarding using the controls in Settings.

Call forward feature not working on remote app on ipad

I can not activate called forwarding of my house phone from my ipad or iphone. The pop up box does not allow the ok to work.

Hi Josie,
Fios Digital Voice is our only phone service that allows you to manage call forwarding on your Fios app.
If you do in fact have Fios Digital Voice, then please try uninstalling and reinstalling your app and let us know if that works.
~Jess

Unity 4.0 - Call Forwarding and Voice Mail

Here is the situation:
We have a DN (5301) that is not associated with a Unity mailbox but is on a 7970 phone. This extension is an "on call" number that is always forwarded to a technicans phone (local 4 digit ext or cell phone).
When a person calls 5301 I want the voicemail of the final destination to answer.
For instance if I had 5301 forwarded to 2000 - I would want 2000's voicemail to answer.
Is there a way to set this up?
Thanks in advance.
Jeff

Hi Jeff,
Sadly this cannot be changed until Unity 5.x (the ability to choose "Last Redirecting Number" in not available in any other Unity version);
Here are the Unity 5.0 release notes;
Route Forwarded Calls by the First or Last Redirecting Number
Cisco Unity supports the option of routing calls based on either the first or last redirecting number when a call is forwarded to Cisco Unity.
Note the following:
This option requires Cisco Unity-CM TSP 8.1(2) or later.
This option is not supported by integrations through PIMG units.
This option can be changed through the Advanced Settings Tool (AST), which is available in Tools Depot.
http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/release/notes/501curelnotes.html#wp507371
Call Information Exchanged by the Phone System and Cisco Unity
The phone system and Cisco Unity exchange call information to manage calls and to make the integration features possible. With each call, the following call information is typically passed between the phone system and Cisco Unity:
â¢The extension of the called party.
â¢The extension of the calling party (for internal calls) or the phone number of the calling party (if it is an external call and the phone system supports caller ID).
â¢The reason for the forward (the extension is busy, does not answer, or is set to forward all calls). There is also a reason code for Direct Calls.
Cisco Unified Communications Manager SCCP and SIP trunk integrations can also provide the following call information (the choice of first and last redirecting number is set in the Advanced Settings Tool, which is available in Tools Depot):
â¢Called number
â¢First redirecting number
â¢Last redirecting number
Note Cisco Unity can use either the first redirecting number or last redirecting number, depending on the setting in the Advanced Settings Tool, which is available in Tools Depot.
http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/design/guide/5xcudg060.html#wp1040786
If this was just a one-up type of setup you can configure a Voicemail profile (in CCM) for 2000 and apply it to 5301 that will allow this type of Call Forward to 2000's mailbox. The fact that you need this for Multiple Tech's will not work. Is there any way the Techs could use a Shared Line? Then these solutions could be adapoted.
Or in Unity set up Alternate Extensions so that User A is an Alternate Extension for User B etc. Sharing a Cisco Unity Voice Mail Box between Two or More IP Phones
Configure Alternate Extensions
Open the Unity System Administrator web page.
Navigate to the subscriber's profile. Select Subscribers > Find and Select a Subscriber > Enter Subscriber Information then click Find and click the Subscriber's name for the subscriber that owns the primary phone (2000).
When the subscriber page comes up, select the Alternate Extensions option and click Add.
Enter the alternate extension number (in this case 5301) and click the Save icon.
From this good Unity doc;
http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_configuration_example09186a008015ceec.shtml#steps
Hope this helps!
Rob

CME, Call forward to CUE from CCM IP phone

I want to call forward the call from CCM IP phone to CME ephone's voicemail which setup in CUE. works okay between CME ephones. configured voice service as follows but no luck. what did I missing to implement?
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
-CCM4.1.3, configured H225 trunk. leave uncheck the MTP on the trunk device
-gatekeeper to connect between CCM and CME
-CME3.3, h323 to gatekeeper and sip to CUE
-CUE2.1
Thanks in advance,

It works by restart the CME router and have a question the sip-ua output. I have two media streams but the 2nd shows "STREAM_IDLE". I think this is for g729 connected to CCM via h323 gk. Can I get an explanation why?
CME#sh sip-ua calls
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 : 4083132006
Called Number : 4211
Bit Flags : 0x101A0030 0x100000 0x500
CC Call ID : 95
Source IP Address (Sig ): 10.253.66.254
Destn SIP Req Addr:Port : 10.253.66.2:5060
Destn SIP Resp Addr:Port: 10.253.66.2:5060
Destination Name :
Number of Media Streams : 2
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 95
Stream Type : voice-only (0)
Negotiated Codec : g711ulaw (160 bytes)
Codec Payload Type : 0
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: 10.253.66.254:16998
Media Dest IP Addr:Port : 10.253.66.2:16904
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Media Stream 2
State of the stream : STREAM_IDLE
Stream Call ID : -1
Stream Type : voice+dtmf (1)
Negotiated Codec : No Codec (0 bytes)
Codec Payload Type : 255 (None)
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: 10.253.66.254:17120
Media Dest IP Addr:Port : 0.0.0.0:0
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Number of SIP User Agent Server(UAS) calls: 0
CME#sh sccp connections
sess_id conn_id stype mode codec ripaddr rport sport
1 2 xcode sendrecv g711u 10.253.66.254 2000 16518
1 1 xcode sendrecv g729 10.253.66.254 2000 17620
Total number of active session(s) 1, and connection(s) 2

IP Phone call forward to Attendant Console - reach Attendant Console Voicemail

Hello,
I have an old Attendant Console based on CUCM7.1(5) with CUC 9.
Most of my user have voice mail box.
My issue is following: Users forward there phone to AC pilot point during lunch time or meetings.
When a call is then transfered to Attendant Console, if it is not answered, it is redirected to Unity connection and reach User Voice Mail box.
This is the good way of working for all users (to reach first redirect number mailbox), but my customer wants in the particular case of calls forwarded to standard, to reach standard voice mail box.
How can we achieve this?
If I setup a Translation Pattern on CUCM to translate a dummy number into attendant console Pilot Point number, and forward user to this dummy number, does Translation Pattern replace the first redirect number?
Thanks for your help.
Thibaut

It is likely that they are transferring a call to the operator on a line they are not logged into instead of the queue.
All calls should be routed through the queue in order to receive full CTI control of that call, I would suggest you create a speed dial with the DN of the CTI Route Point for the operators queue and then transfer the call to that instead of the operator directly, this way the call sits in the queue with any other callers.

Call forward to external number(mobile)

Dears please help me on this
voice translation-rule 1
rule 1 /2837599/ /599/
rule 6 /2837596/ /596/
rule 7 /.*2837555/ /123/
2837... are my SIP DID nos
123 is my AA extn
596 and 599 is an ip phone exten
i need to transfer directly to an external no (mobile no) when i call 2837596 from outside without extension
what is the config to be done

dears , i tried it but call not forwarding please need our help
voice translation-rule 1
rule 13 /.*2837499/ /499/
ephone-dn 499 dual-line
number 499
label website
description 499
call-forward all 90504495705
corlist incoming user-international
ephone 37
device-security-mode none
video
mac-address 001E.F727.F567
ephone-template 16
username "700" password 700
type 7911
button 1:499
pin 1700

Call forward to external number which has auto attendant

Hi
I am a voice administrator in my company
I want to forward all of my calls to my Other location's number.
Other location has only POTS line and to reach my extension user needs to go through Auto attendant menu.
Is it possible to enter a digit pattern in call forward destination in CUCM so that it can take care of Auto attendant menu of my Other location and land on my number?
We have CUCM 8 running.
Please help!!
Ashwin

Hello Ashwin,
"Other location has only POTS line and to reach my extension user needs to go through Auto attendant menu."
What type of phone system and voice mail is providing the auto attendant?
How are the POTS lines(analog only correct?) terminated into the phone system?

Call Forward to PSTN numbers not working

We have a CUCM version 8.6. The call farward to internal extensions are working fine. But, to the PSTN numbers it is now working. We are able to call the PSTN numbers without any issue. Can somebody help us on this?

Hi Siva,
The most likely cause for this type of issue is the CSS that is applied @ the Call Forward All level on
the DN config page. Check out the CFWDALL CSS to make sure they are set with a level with
access to PSTN numbers
Cheers!
Rob
"Your life is worth much more than gold."
- Bob Marley

Call forward to unity connection call handler

have the following setup:
cucm 8.6
cuc 8.6
a cti route point (DN 1000 )with forward all to unity connection call handler.
phone users who choose to have some quiet time can forward all their calls to the cti rp.
1. i need to allow only specfic DNs the ability to forward to the cti rp.
2. at times that the allowed DNs didnt choose to forward to cti rp then normal behaviour should occur - busy or unanswerd calls should go to DNs personal voice mail.
thanks

You can assign a specific partition to cti-rp and assign it to a CSS and this CSS should be assigned to those users (at call forward settings in line) only to whom you are trying to give this facility and for other users you can
or you can create a TP for it with specific partition and set the called number as per voice mail.
Suresh

Call forward to PSTN on cme

Hi,
unable to set up call forward to PSTN.
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Table Normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:"Times New Roman";
mso-fareast-theme-font:minor-fareast;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;}
I have tried activating the Call forward via the phone or manually via the config, but when I attempt a call to IP Communicator from PSTN or via extn I am not seeing re-INVITE which should be generated for the forwarded call. Am i missing something?
PSTN / IP phone ------> Calling extn on CME (which is call forwarded to another PSTN number)
config below:
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
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 sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 250 min 200
asserted-id pai
localhost dns:XXXXX
outbound-proxy dns:XXXXX
dial-peer voice 100 voip
description ** Incoming call from SIP trunk **
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 101 voip
description ** Outgoinging call to SIP trunk **
translation-profile outgoing SIPOUT
destination-pattern 1T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 101
dtmf-relay rtp-nte
no vad
dial-peer voice 102 voip
description ** Outgoinging call to SIP trunk **
destination-pattern 0[2-9].T
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
telephony-service
max-ephones 4
max-dn 12
ip source-address 192.168.100.2 port 2000
calling-number initiator
timeouts interdigit 5
load 7960-7940 P00308010200
date-format dd-mm-yy
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
transfer-system full-consult dss
transfer-pattern .T
transfer-pattern 0.T
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 4961 secondary 99474961 no-reg both
label 4961
name 4961
call-forward all 021605547

/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Table Normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:"Times New Roman";
mso-fareast-theme-font:minor-fareast;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;}
Does a direct call (without forwarding) work through this dial-peer? YES
The session target of dial-peer 101 is the "sip-server". In wich way is configured? Is it an IP address or a name? FQDN
Can you ping it from the CME? YES
The CME can resolve the name via DNS? Resolved on the CME Can you post the sip-ua config?
sip-ua
credentials number 99474960 username 99474960 password 7 XXXXXXXXX realm as-test.xys.net
authentication username 99474960 password 7 XXXXXXX
calling-info pstn-to-sip asserted-id number set 99474960
no remote-party-id
disable-early-media 180
retry invite 2
retry register 3
timers connect 100
registrar dns:as-test.xys.net expires 60 sip-server dns:as-test.xys.net
host-registrar

Call forward to VM not working

Hi All;
we have switched from TDM to SIP trunk. after that call forward to voicemail and also automated attendant don't work. then I changed VM pilot number to full number instead of 6000. still when inbound call is received, and called party doesn't answer the call, it is supposed to forward to VM but call is failed. also when somebody calls to main number, it is supposed forward to AA, but call is failed. very appreciate, if there is any suggestion.
thanks
Alex

The problem is that Unity sees the forward call as "forward" actually is not a problem is the normal behavior. What we need to do is to Unity see the call as "Direct" configuring Routing Rules resolve the problem
Try:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00800a6a14.shtml

Call forward to another users voicemail

Here is the scenario, i cannot find a way to accomplish this.
i am using CUCM 8.5 and Unity Connection 8.5
here is the requirement
1. User1 call forwards her phone to User2
2. A call comes in to User1, the call is forwarded to User2
3. If User2 is unavailable, the call is redirected to User2 voicemail.
The default behaviour is that if User2 does not answer, the call is redirected to the original called number (User1)
I have created forwarded routing rules in Unity connection and i can get the call to end up at User2 voicemailbox however, the user requires that this happens only when the call is forwarded manually. For example, if i call User1 and let it go to Voicemail, it will still go to User2
They want that it goes only to the User2 voicemail only in the specific circumstance that the phone is in call forward mode, not by letting it go to voicemail
Does it make sense?
I think my users are asking too much

Hi Steven,
I'll just add a note to the great tips from Hailey & Roger (+5 each!)
I thought this was an interesting question, so I tried a number of ways
to see if this could be done
The problem, as you've discovered, is that the original Forwarded CLID
is so "sticky" as to render most of the standard routing methods moot.
In my tests I was trying to route calls that come into 7001 and are CFWDALL
to 5126 to route to the 5126 mailbox without changing the Originally dialed
to last redirecting setting (as nicely noted by Hailey).
I had to put in a mid-point number that was set to CFWDALL (via PSTN)
back to the second number.
So I setup a CTI-RP on a phantom (non-DID) DN of 4241 and set the CFWDALL
to 5126. I could then CFWD from 7001 to 4241 which then routed to 5126 with
the CLID stripped off and would then give us the desired results. No other method
that I could come up with would work.
Cheers!
Rob
"Why not help one another on the way" - Bob Marley

CUCM 8.6 Call Forwarding to External Number Issue

Hello,
Call forwarding worked without problems, we could forward our phones to external numbers and everything was ok, when somebody called to my phone, I could got the call to my cell phone.
But now when I forward my phone to external number and try to call to my phone I get busy trigger.
We didn't change configuration or install any update.
I think its my ISP-s problem, to whom we have SIP Trunk.
I don't understand log file, so can you tell what is the problem?
Here is log:
057729XXXX is called party, cell phone number
original calling party number is 240XXXXX, but it is forwarded to 2484XXX
INVITE sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191815-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5060>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052091 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29790 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 2|2,100,56,1.173711429^MY-CUCM-IP^MTP_3
17:34:18.526 |EnvProcessUdpHandler::fireSignal - SEND: index = 2, handler = 0xb2d59c98|*^*^*
17:34:18.526 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP-IP:5060|*^*^*
17:34:18.526 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1172, ISP-IP:5060)|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.536 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.536 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 358 from ISP-IP:[5060]:
[12623361,NET]
SIP/2.0 100 Trying
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/ccsip_spi_get_msg_type returned: 2 for event 1|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/context=(nil)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Transport/0x0/gConnTab=0xf484290, addr=ISP-IP, port=5060, connid=2, transport=UDP|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Return existing connection for port 5060 connId 2|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0x0/Checking Invite Dialog|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/Info/0xb1b50c90/INVITE response with no RSEQ - disable IS_REL1XX|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_TRYING value=500 retries=3|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/Stack/States/0xb1b50c90/0xb1b50c90 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_stop_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.536 |//SIP/SIPHandler/ccbId=0/scbId=0/sip_start_timer: type=SIP_TIMER_EXPIRES value=180000 retries=0|2,100,230,1.4901096^ISP-IP^*
17:34:18.561 |EnvProcessUdpHandler::handle_input - handle = 334|*^*^*
17:34:18.561 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 2|*^*^*
17:34:18.561 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 396 from ISP-IP:[5060]:
[12623362,NET]
SIP/2.0 403 Forbidden
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
Server: CISCO-SBC/2.x
Content-Length: 0
Contact: <sip:ISP-IP:5060>
[12623363,NET]
ACK sip:2484XXX@ISP-IP:5060 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5060;branch=z9hG4bK1003a84126249
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052091~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP-IP>;tag=sip+1+b3a00013+867def6a
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e7-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0
INVITE sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Cisco-Guid: 0383266432-0000065536-0000191816-2219117834
Session-Expires: 1800
P-Asserted-Identity: <sip:057729XXXX@MY-CUCM-IP>
Remote-Party-ID: <sip:057729XXXX@MY-CUCM-IP>;party=calling;screen=yes;privacy=off
Contact: <sip:057729XXXX@MY-CUCM-IP:5062>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 215
v=0
o=CiscoSystemsCCM-SIP 4052092 1 IN IP4 MY-CUCM-IP
s=SIP Call
c=IN IP4 MY-CUCM-IP
t=0 0
m=audio 29792 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|2,100,56,1.173711431^MY-CUCM-IP^MTP_3
17:34:18.567 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xa6b4d7c0|*^*^*
17:34:18.567 |EnvProcessUdpPort::fireSignal - SEND, destination = ISP's-Other-IP:5062|*^*^*
17:34:18.567 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 1177, ISP's-Other-IP:5062)|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input - handle = 335|*^*^*
17:34:18.569 |EnvProcessUdpHandler::handle_input   Status: 0, Id: 0|*^*^*
17:34:18.569 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 394 from ISP's-Other-IP:[5062]:
[12623365,NET]
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
CSeq: 101 INVITE
Server: kamailio (3.3.1 (x86_64/linux))
Content-Length: 0
17:34:18.587 |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 375 from ISP's-Other-IP:[5062]:
[12623366,NET]
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900;rport=5062
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
CSeq: 101 INVITE
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0
|2,100,230,1.4901099^ISP's-Other-IP^*
[12623367,NET]
ACK sip:2484XXX@ISP's-Other-IP:5062 SIP/2.0
Via: SIP/2.0/UDP MY-CUCM-IP:5062;branch=z9hG4bK1003a95b8f3900
From: <sip:057729XXXX@MY-CUCM-IP>;tag=4052092~294be736-ce3b-450f-a7f1-c801f3cc9a7e-27746002
To: <sip:2484XXX@ISP's-Other-IP>;tag=dc6a4ae7
Date: Wed, 18 Dec 2013 13:34:18 GMT
Call-ID: 16d82e80-2b11a45a-c43e8-84450d0a@MY-CUCM-IP
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0

SIP/2.0 403 Forbidden error
If your router is sending a SIP/2.0 403 Forbidden error to the SIP server you are registered to, there is a good chance your router is blocking the incoming call due to the toll-faud prevention feature that was added to IOS version 15.1(2)T.
How to Identify if TOLLFRAUD_APP is Blocking Your Call
If the TOLLFRAUD_APP is rejecting the call, it generates a Q.850       disconnect cause value of 21, which represents ‘Call Rejected’. The       debug voip ccapi inout command can be run to       identify the cause value.
Additionally, voice iec syslog can be       enabled to further verify if the call failure is a result of the toll-fraud       prevention. This configuration, which is often handy to troubleshoot the origin       of failure from a gateway perspective, will print out that the call is being       rejected due to toll call fraud. The CCAPI and Voice IEC output is demonstrated       in this debug output:
%VOICE_IEC-3-GW: Application Framework Core: Internal Error (Toll fraud call rejected):
IEC=1.1.228.3.31.0 on callID 3 GUID=F146D6B0539C11DF800CA596C4C2D7EF
000183: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallSetContext:
   Context=0x49EC9978
000184: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 3 with tag 1002 to app "_ManagedAppProcess_TOLLFRAUD_APP"
000185: *Apr 30 14:38:57.251: //3/F146D6B0800C/CCAPI/ccCallDisconnect:
   Cause Value=21, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
The Q.850 disconnect value that is returned for blocked calls can also       be changed from the default of 21 with this command:
voice service voip
ip address trusted call-block cause
How to Return to Pre-15.1(2)T Behavior
Source IP Address Trust List
There are three ways to return to the previous behavior of voice       gateways before this trusted address toll-fraud prevention feature was       implemented. All of these configurations require that you are already running       15.1(2)T in order for you to make the configuration change.
Explicitly enable those source IP addresses from which you would like           to add to the trusted list for legitimate VoIP calls. Up to 100 entries can be           defined. This below configuration accepts calls from those host           203.0.113.100/32, as well as from the network 192.0.2.0/24. Call setups from           all other hosts are rejected. This is the recommended method from a voice           security perspective.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
Configure the router to accept incoming call setups from all source           IP addresses.
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
Disable the toll-fraud prevention application completely.
voice service voip
no ip address trusted authenticate
Two-Stage Dialing
If two-stage dialing is required, the following can be configured to       return behavior to match previous releases.
For inbound ISDN calls:
voice service pots
no direct-inward-dial isdn
For inbound FXO calls:
voice-port
secondary dialtone

Call forwarding

Similar Messages

Maybe you are looking for