Forwarded calls are getting charged by ATT

I have friends call my US cell phone which I then forward to my US skype number which then forwards to a foreign land line. However, a friend has recently been charged by ATT for the call as if it was direct from her US cell phone to the foreign land line. Why would ATT be doing this?
I see on my skype history her call listed (odd because I would think I would see my cell phone and not hers since it was first forwarded to my cell). She would not have called the skype number directly because she does not know the number however even if she did that would still not be an international call but a call to my US skype.
Why did ATT charge her?

Hi Anas and Carlo, thanks a lot guys for the reply.
Will capture the debugs. But it would be difficult to capture the output for the call failure. Because, 40% of calls are not being failed. So, I cant anticipate the output for failure/success call.
In fact, I have already captured the debugs from CUBE as well as CUCM. Will capture the fresh debugs and will ping here.
Thanks,
Madhura

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No point of using Skype.. calls are getting drop f...

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Rest calls are getting 406 error with beta refresh

We are seeing that our REST API calls are getting http 406 error. We have not made any changes on the REST Service. In Console, i see following messages that might be related to this issue.
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Added this after looking at one of the forum entry.
restServiceAdapter.addRequestProperty("Accept", "application/json; charset=UTF-8");
And issue is resolved. Documentation update in sample code would help others.
Edited by: Chandresh on Sep 4, 2012 2:36 PM

Having a problem with iphone 4S. Phone calls are getting dropped and cutting in and out.

I seem to have very little or no service at all recently.
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I am extremely frustrated and annoyed that I cant even really use my phone.
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If the on/off switch does not work, that is a hardware problem. The other problems may be related to a hardware problem or faulure as well.
Make an appointment at an Apple store if there is one nearby to confirm.

Forwarded calls are not "call-parked"

Dear Lync experts,
please could You help me that the following scenario is not possible or I need to change some config parameter:
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Thanks in advance:
Norbert

Hi,
Would you please tell us why you want to forward call to another account in the same call park group?
Base on my understanding, if cancel the call forwarding, other users could pick up the call. Please check if it works after cancelling call forwarding setting. Please check if the issue still happen when cancelling call forwarding.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

How use google voice for texting so don't get charged from att???

can't find the correct settings to avoid being charged???
do I need to download google voice app on iphone ?

No, it doesn't support that. I mentioned the details in your other thread. To add to the other question, you must go to Google Voice settings on the web and make sure that your messages aren't automatically forwarded as text messages. If that is checked, you will receive a text message (which will charge) and a push notification (which will not charge). As mentioned before, you should simply use push notification to get your messages.

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Skype calls are getting failed through the application

Hi,
I keep trying to call my friend on skype and I've tried everything from reinstalling skype to power-cycling my router, but The Skype call keeps failing! If you could, help would be greatly appreciated, thanks.

Added this after looking at one of the forum entry.
restServiceAdapter.addRequestProperty("Accept", "application/json; charset=UTF-8");
And issue is resolved. Documentation update in sample code would help others.
Edited by: Chandresh on Sep 4, 2012 2:36 PM

UC520 SIP trunk unable to make outgoing calls, incoming calls are ok

I have an new SIP trunk set on an UC520 and the incoming calls are ok, but the outgoing calls are getting an busy tone(not working).
The bellow trace is showing that the cause is "No route to destination (3) ". The question is this route has to do with the firewall(ip routing) or with the voice translation rules?
001866: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x843CE50C
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 0777777777 <- main sip number
Called Number            : 0888888888 <- called number
Source IP Address (Sig ): 0.0.0.0
Destn SIP Req Addr:Port : 0.0.0.0:0
Destn SIP Resp Addr:Port : 0.0.0.0:0
Destination Name         :
001867: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 3
Disconnect Cause (SIP)   : 200

Hi Emil,
I've added bind control and media interface but outgoing calls are the same blocked, strange thing is that the cause is still no route to destination (3)
but
UC_520#show sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): ENABLED 10.10.10.5 <- fa0/0 IP address
SIP User Agent bind status(media): ENABLED 10.10.10.5 <- fa0/0 IP address
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio image
Network types supported: IN
Address types supported: IP4
Transport types supported: RTP/AVP udptl

Not all incoming calls are rejected

I've searched other posts and have not found a similar problem. I can call my Skype number from my cell phone, my daughters phone and get through every time. But it seems that calls from outside my area code cannot get through. Sometimes they report a busy signal or just a few rings then the call drops. I've checked my firewall but if my calls are getting through.... Could the problem be the recent upgrade to Windows 7 Professional SP1. ??? I am missing important calls I'm sure from customers that do not have my cell number.

Hi, Kenny1955, and welcome to the Community,
Sounds like a call routing issue. Please send a private message to me with as much detail as possible listing the numbers or exchanges from which people may have reported to you that their calls did not get through to your Skype Number as well as the carriers in question, if known (mobile or wireless network as well as fixed line).
Thank you,
Elaine
Was your question answered? Please click on the Accept as a Solution link so everyone can quickly find what works! Like a post or want to say, "Thank You" - ?? Click on the Kudos button!
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E does the person i'm calling get charged?

I'm still confused! from the uk using skype pay as you go, does the person i'm calling abroad on their mobile get charged and at what rate? i'd feel bad if I was calling at very low cost and they were receiving at an exhorbitant mobile rate. many thanks

pamp2010 wrote:
I'm still confused! from the uk using skype pay as you go, does the person i'm calling abroad on their mobile get charged and at what rate?
Hello,
All Skype calls worldwide are "local" calls. Whether or not your contact is charged by their carrier depends entirely on the cellphone Calling Plan they have. Skype makes no charge.
In any case the charge, if any, would only be a few cents as the call will be treated as "local".
TIME ZONE - US EASTERN. LOCATION - PHILADELPHIA, PA, USA.
I recommend that you always run the latest Skype version: Windows & Mac
If my advice helped to fix your issue please mark it as a solution to help others.
Please note that I generally don't respond to unsolicited Private Messages. Thank you.

Calls are not getting thru in Cisco voice GW for a particular Number

Cisco gateway is connecte to a PBX with an Qsig interface, for a particualr destination number the calls are not gettin estabilished.
the output of the Q931 debug :
Aug 16 16:17:46.145: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x7E05
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98396
                Exclusive, Channel 22
        Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
        Facility i = 0x9FAA068001008201008B0100A11D0202010002010080144E455453202
F204C4F4E472044495354414E4345
        Calling Party Number i = 0x2183, '8168911010'
                Plan:ISDN, Type:National
        Called Party Number i = 0x89, '18553808521'
                Plan:Private, Type:Unknown
        Sending Complete
Aug 16 16:17:46.149: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0xF
E05
        Channel ID i = 0xA98396
                Exclusive, Channel 22
Aug 16 16:17:55.709: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x
FE05
        Cause i = 0x80BF - Service/option not available, unspecified
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x7E0
5
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref =
0xFE05
The Qsig and dial-peer configration :
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vad

Hi Raj,
My name is Edson Pineiro, I understand that your problem description is in regards to failed incoming calls from a qsig trunk.
According to the received q931 setup message I can see the called party number is 18553808521 and as so the gateway should route the dnis based on the best match in destination-pattern. My first suggestion would be to ensure your outgoing dial-peers has a matching destination-pattern that matches the dialed number, for example:
dial-peer voice 1 voip
destination-pattern 1T
The T is a wild card for any digit any length
Or you can be very specific.
dial-peer voice 1 voip
destinaton-pattern 18553808521
The next suggestion would be to ensure that your incoming pots dial-peers contains 'direct-inward dial'. This is so that you don't receive secondary dial tone when dialing in, which I don't think is happening here.
Another suggestion would be to remove 'isdn overlap-receiving' from interface serial 0/0/0:23. Reason being is that the DNIS received is enbloc and not overlapping. You can clearly see that the complete e164 number is received within the setup and no further digits are needed.
But overall the disconnect cause code is 0x80BF the 80 portion is related to the source of the disconnect which is the router and BF "Service/option not available, unspecified" which is described as:
The network or remote equipment cannot provide the service option that the user requests, due to an unspecified reason. A subscription problem can cause this issue.
Any ways seems like the router does not support the protocol or type of message included in the Setup. After decoding one of the facility message:
        Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
decode -->
Facility IE first byte (protocol profile): 0x9f(Network Extentions), depends on Network Protocol Profile
**Note:
**0x91/0x9f both be used by older qsig spec, including:
**ISO 11582:1995, ETSI 300 239:1993/1995
**newer qsig spec use 0x9f only, including:
**ISO 11582:1995/Cor.1:1999, ECMA 165(4th), ETSI 300 239:2003
**see CSCeb58118 for CCM compatibility issue
NetworkFacilityExtension ::= {
sourceEntity: 0
destinationEntity: 0
NetworkProtocolProfile not present
APDU is a ROSE
0
DivertingLegInformation2Invoke ::= {
invokeID: 1793
operationValue: 21
argument: DivertingLegInformation2Arg ::= {
diversionCounter: 1
diversionReason: 1
originalDiversionReason: 1
divertingNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
originalCalledNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
redirectingName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
originalCalledName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
Looks like this is a redirected call (call forward or transfer), the redireted number is "50005998" and the other end of the PRI maybe attempting to do either a 2 B channel transfer or B channel optimization, which is not supported certain gateways or needs the use of a tcl scripts. Any ways is it possible to confirm if such features are enabled on the other end of the qsig trunk? and what the number 50005998 is assigned too. This may warrant a TAC case.
However please ensure your carry through the first three configuration changes before looking at the possible bad facility message.
Here are some good documents on ISDN, IOS dial-peers and call legs:
Understanding debug isdn q931 Disconnect Cause Codes
http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a008012e95f.shtml
Configuring Telephony Call-Redirect Features
Two B-Channel Transfer
http://www.cisco.com/en/US/docs/ios/voice/ivr/pre12.3_14_t/configuration/guide/ivrapp.pdf
Understanding Dial Peers and Call Legs on Cisco IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
Understanding Direct-Inward-Dial (DID) on IOS Voice Digital (T1/E1) Interfaces
http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_note09186a00801142f8.shtml
Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#prereq
Voice Translation Rules
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml
Let me know how you go.
Thanks again for asking the tuff questions.
Cheers
Edson

How can i get charged when my device is still suspended and no call is made?

Hello, Verizon
I have to complain that I was charged $106.09 for this month's statement.
But my device (removed) is still suspended, up until this week as far as I know, and I have made no call. How could I get charged for the month until April 9? I am currently not in the US territory.
It must be a mistake to charge me the groundless amount. Please double check.
Private info removed as required by the Verizon Wireless Terms of Service
Message was edited by: Admin Moderator

ashurbanipal,
Thank you for the information. We definitely want to figure out what has happened, because you should be able to suspend for up to 90 days before the service is reconnected. For review please follow my handle TonyG_VZW and send me a DM with your account info and billing password so I can review your account. I am sure we can figure out what is going on together.
Thank you,
TonyG_VZW
Follow us on Twitter @VZWSupport

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