Incoming Calls Not Working

I'm using Trixbox with Skype Connect. I can make outgoing calls, but incoming calls just ring a few times, and then the call terminates.
-- When I ping sip.skype.com, I get the ip address 63.209.144.201.
-- All of my "Incoming Settings" are blank.
-- My Skype number is 4159928107
-- I am using a Linksys router, and I have forwarded UDP Ports 5060 and 8000-8004 to my Trixbox server.
Thanks for your help.

Hello Tpddkennedylaw,
I see you are having trouble receiving calls. Looking at your account, I can see you are making calls out okay, and I see that there were at least 20 attempts to your system by a 1410310xxxx number. The number did hit your PBX, but had no where to go. That tells me that you do not have a Incoming Dialing Plan assigned, or, there is not a Default Target Number assigned in the configuration of the PBX. I would advise you to read up on "Incoming Calls" in the Trixbox documentation and follow the instructions to get your Incoming Calls going, and you should be good in no time. If you still need help with that, contact the Manufacturer of the PBX for additional support.
I hope Skype has helped you with a solution to your Incoming Call problem.
Thank You for using Skype and the Skype Community Forums.
Regards,
Victor S.
Skype Enterprise Support

Similar Messages

Incoming Calls Not Working, Spotty Service

I've been having a issue with my Verizon landline (copper) service for several weeks now. Basically, outgoing calls work fine, but incoming calls are very spotty. Sometimes calls come through as normal, but most of the time the caller will hear one ring then silence, or no rings at all. In any of these circumstances, I can pick up the phone while it is "ringing" and the call connects. When there is a problem, calls do not go to voicemail.
I did the *73 business to deactive call fowarding, but that did not do anything.
I have tried every combination of having devices (DSL modem, cordless phone, corded phone) plugged in or not, but nothing seems to fix the situation. I cannot replicate the situation given a certain set of circumstances, i.e. the porblem exists no matter what combination of devices I have plugged in. It is completely random; a call from my cell phone will connect fine, but if I try 30 seconds later, it doesn't work. Even if no devices are plugged in, I still have the same issue!
I called Verizon, who ran numerous tests, and said everything was fine on their end. Naturally, they offered to send someone to my place for some insane fee; since I rent I obviously would not let them come to fix my lines -- that's my landlord's job. I can have them look at it, but I don't think it will really make a difference.
Oh, and I'm not the only one having this problem:
http://forums.verizon.com/t5/Home-Phone-Landline-or-Digital/Home-Phone-is-not-working/td-p/163581
http://forums.verizon.com/t5/Home-Phone-Landline-or-Digital/flaky-incoming-calls-no-rings-sometimes/...
http://forums.verizon.com/t5/Home-Phone-Landline-or-Digital/Short-chirp-for-in-bound-call-ring/td-p/...
http://forums.verizon.com/t5/Home-Phone-Landline-or-Digital/Phone-can-call-out-but-cannot-receive-ge...
At this point, I'm sick of dealing wtih this issue. I felt like I got the brush-off from Verizon, who does not seems to care of their copper infrastructure any longer, but is only interested in FiOS customers. I am seriously considering just dropping my landline service entirely, and getting Internet-only from either Verizon (if that's possible, they say I can get FiOS in my apartment, although I don't see how that's possible) or Comcast. I really like having a landline and would not mind paying for decent service, but this rapidly becoming not worth it.

can you please collect debug voice ccapi inout as well?
does the outbound call work through the same gateway?

Incoming call not working

Hello Guys,
I have an issue with the incoming calls. We I dial the 10 digitss number using my Cell phone, it doesn't go through. I receive a message stating "the call can't go through". But we can call from inside to outside and within the company.
Also when I try to add a dial-peer, an error message stating "Could not add peer" appears.
Could you please help me out?
The gateway is a 3640 H.323.
This is the output of the debug isdn q931. I am dialing 617-717-7982.
Oct 26 2006 02:23:56.129: ISDN Se1/0:23 Q931: RX <- SETUP pd = 8 callref = 0x00E1
Bearer Capability i = 0x8090
Standard = CCITT
Transer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xE1808386
Preferred, Interface 0, Channel 6
Net Specific Fac i = 0x00E7
Calling Party Number i = 0x2181, '6178692545'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '7982'
Plan:ISDN, Type:National
Oct 26 2006 02:23:56.169: ISDN Se1/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x80E1
Channel ID i = 0xA98386
Exclusive, Channel 6
Oct 26 2006 02:23:56.253: ISDN Se1/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x80E1
Cause i = 0x80A6 - Network out of order
Oct 26 2006 02:23:56.301: ISDN Se1/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x00E1
Oct 26 2006 02:23:56.309: ISDN Se1/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x80E1
Thanks,

Thanks for the reply,
My question is why are we able to make outbound call if the session-targat address is wrong. Please consider the following two Dial-peer. For a call coming in from PSTN, which Dial-peer is going to be used? Accroding to me the Dial-peer voice 1001 will be used because the call is coming from PSTN therefore it going to use the PRI. Am I right?
dial-peer voice 7000 voip
preference 1
destination-pattern 7577
session target ipv4:172.16.165.10
incoming called-number .
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 1001 pots
preference 1
destination-pattern [2-9].........
direct-inward-dial
port 1/0:23
forward-digits all
Thanks,

Outgoing calls not working on iphone 4S after upgrading to ios 6.0.1

outgoing calls not working on iphone 4S after upgrading to ios 6.0.1.
It always says call back ...
incoming is working
i tried number of sim cards from different networks but same issue
tries resetting network settings but it also didnt worked

Just counter-check the following settings.
1. Settings>General>Set Side Swith To: Rotation Lock
2. Make sure side switch is not in lock position
3. If necessary do a reset a few more times

Merge call not working in ios7 on Iphone5

Merge call not working in ios7 on Iphone5 ?

Same here. We have this call center in our company where after i have added a call to merge i have to push no. 4 and dial from the keypad. This has never been possible in ios but it have been possible to create a contact called "Merge" and add number 4 as it's phone number. Now by merging first my clients call to my collegegue and then to this "Merge" contact it has been dropping me out of line and my client and collegue are merged.
Now ios 7 wont let me call to this "Merge" contact's number "4"

Calling not working

I continuously try calling and getting called by my friends who live across the state, only to have the call not work for me. I don't get any ringing or the pop-up window, it just shows me that they are in a call. When I do try to join, it sits there for approximately 30 seconds trying to connect and then kicks me out of the call. I have reset Skype, I have checked for updates; I have the most recent one, I'm up to date. They all have no problem calling with each other but I cannot join or make calls. Oddly enough, there is only one person I can call or get calls from out of our group, but I don't receive a notification unless it's from her. I have tried calling echo and it works every time but not with my friends. Please send help or something to try to fix this problem, it is making Skype unuseable.

Glad we found a workaround. On which network and in which country are you roaming? And who is your home network operator?
This would help Nokia to investigate. And it helps those to find the workaround who find this thread via an search on the Internet.

Face time incoming call not ringing. Why? Can anyone help?

Face time incoming call not ringing. Why? Can anyone help?

Do you have anything set in restrictions that might be causing this? Settings>General>Restrictions.

Click to call not working with Firefox 31 in Windo...

Hello All,
I have a problem with Click to Call in my new computer.
Windows 8.1
Skype 6.20.0.104
Click to call 7.3.16540.9015
Firefox 31
Click to Call not working with Firefox.
With IE11 it ok.
When I directly copy the URL "skype:+48xxxxxxxxx?call" into address field of my Firefox, then automatically opening Skype with only 1 window- contacts, but not opened window with call phone, and not started to call.
When I do this same in IE11 , Skype opened 2 windows ( contacts and call phone) and start call to phone number.
I don't know why. because in my second computer ( this same Firefox, Skype, click to call) but Windows 7, Click to Call working with Firefox.
Please help me.
Maybe someone have a solution ?
Best Regards
Jarek
ps. Sorry for my english

Sounds that you have installed a Firefox Beta release and thus are on the beta update channel, see Help > About.<br />
The Beta update channel receives an update twice a week.
You need to install the current release to switch the update channel to release.
Download a fresh Firefox copy and save the file to the desktop.
*Firefox 27: http://www.mozilla.org/en-US/firefox/all.html
If possible uninstall your current Firefox version to cleanup the Windows registry and settings in security software.
*Do NOT remove personal data when you uninstall your current Firefox version, because all profile folders will be removed and you lose personal data like bookmarks and passwords from profiles of other Firefox versions.
Your bookmarks and other personal data are stored in the Firefox profile folder and won't be affected by an uninstall and (re)install, but make sure that "remove personal data" is NOT selected when you uninstall Firefox.

Incoming calls not using Bluetooth / Music tracks repeating

I've been having a problematic issue with Bluetooth since upgrading to iOS 7. I have my iPhone 5 paired with my 2013 Hyundai Elantra's Bluetooth system.
Outbound calls use Bluetooth just fine, as does Siri, but any incoming calls all use my iPhone's headset by default. I have to manually switch the phone's output back to my car's bluetooth in order to talk using my car's hands-free system... which really defeats the purpose of hands-free.
I'll note that this is happening both when my iPhone is plugged in via USB and listening to music on the car's stereo, as well as when the iPhone is simply idle in my pocket. I have not tested what happens when the car is playing my iPhone's music via Bluetooth (the audio quality is sub-par to a direct connection and doesn't display the artist/track/album on the console when playing via Bluetooth - ergo I rarely use it)
Secondly, I am having another issue, this time with playing music. I don't believe it's related to the above Bluetooth issue, but it may be related to my car.
Like I said above, I often use my Elanta's USB port to directly plug in my iPhone so I can a) listen to music on the car's stereo and b) skip tracks with the wheel-mounted controls. This all worked flawlessly with iOS 6.
Now, if I skip a track with my car's controls, the Music app will repeat that next track... over and over. When the song is completed, it will attempt to move on to the next song, then immediately backtrack and start playing the previous song again (you can tell so by watching the Music app. The album art will swipe over, then quickly swipe back). I have to manually skip forward to play a new song... Highly irritating.
I'll also note that skipping songs via the wheel-mounted controls is considerably slower than it was with iOS 6. It's definitely a couple seconds after I press a button before the next song starts to play. It used to be near-immediate.
Anyone else having either of these issues? Hopefully it's something Apple picks up on and fixes soon!

Don't know if you got this yet, but I came here looking for answer. Since I just found one, figured I'd pass it on for others:
You have to set your phone's incoming calls to "headset" under accessibility. (Silly, I know, considering, if you hook up a bluetooth, it should automatically default there, but no ...) This walks you through:
http://www.imore.com/how-set-default-call-answer-mode-iphone

Call merging (conference calling) not working on iOS 8.1 & iPhone 6 (VZW)

Hello Everyone. I've discovered a problem with call merging on my new Verizon iPhone 6 (iOS 8.1). When I add a caller and then hit the merge call button after the second caller answers, the screen says that the calls are now merged. However, the original caller is still on hold and cannot join the call. This problem seems to exist regardless of whether any of the callers in question have CallerID blocking or not. Also, when placing a call on hold to answer an incoming call, I have access to switch calls button, but not the merge calls button.
Has anyone else experienced this glitch? Is there a solution or work-around?
Thanks,
Jamie

I have the same problem with T-Mobile, but this must be an Apple thing! Why do they feel the need to fix ring tones but screw up more important things like Bluetooth and Call Merging???? Seriously, their rein at the top won't last with this BS!

Merge call not working

I tried to initiate a 3 way call using an Apple 5s, and it did not work. I am traveling in Toronto, could that have something to do with it?

@DanPetrie
Let's see what 3 way / Conference calling options are available to you. Please share more details of what you experienced. Were you on an existing call, trying to place another call to conference or were you attempting to accept a second incoming call to merge? Please see more details about Call Conferencing in the Device Manual here: http://bit.ly/1ruLn6H on page 52. Thanks!
AnthonyTa_VZW
Follow us on Twitter @VZWSupport

Caller ID for incoming calls NOT displaying name from contacts (only displays phone # on BB Z10)

on the Z10 Whenever I receive an incoming call, the caller ID only shows the phone #, even when I have that number matched to a contact in the phone, even the call log it show the number and not the name. How do I get the caller ID to recognize and display the appropriate contact name with the phone fro an incoming call? Any help would be greatly appreciated! Thanks, Akhanbilal

Thank you Arsen but, as with everyone else : (1) When you get an in-coming call, it has no call log on the phone app call log that allows you do a thing under the sun except to touch the number of the last call, and when you do, it CALLS THEM BACK. (2) On every other BlackBerry I have owned, about 10 to-date, I simply selected the last in-coming call and had a button at the bottom of the screen to allow me to ADD TO CONTACTS. (3) This does NOT exist on the Z10, and there certainly is no "Click on number, little slider pops on the right, select add to contact." Instead it dials them. Frustrating other uses have described Arsen. Further, while on the topic : (4) None of this has anything to do with 10-digit national dialing plan or anything else on swipe finger down from top in Phone App, Arsen. (5) When you swipe Phone App down, you see SETTINGS. Then, there are 13 items. I have looked at all 13 items and see nothing as you describe to select, nor any instructions on what to set the SMART DIALING to when there are 3 area codes local for my city and country code shows +1. Further, while on topic here : (6) Visual Voice Mail has been turned-on and whatever visual voice mail is, never supported on any previous BlackBerry including my 9780 before this, I have it with T-Mobile and have it turned on on the BlackBerry Z10 and it too does not work. On the positive side, I do have a work-around for BlackBerry Z10 to 2013 Toyota entune. (7) I first took a vm msg .wav file of just hello (blank will not send) and saved it on BlackBerry Z10. The Toyota entune is LOOKING for a music and will LOCK-UP if none are found. By giving it this, it worked on my 2013 Toyota entune for the 1st time ever. Yeah ! It is a little frustrating every few minutes listening to myself say HELLO. But, hey no one really notices it with all the other stuff going on NAVIGATION, radio, cell phone calls, and all the computerized alerts on 2013 Toyota even without entune. (8) What I did now was to go into Navigation and remove SYNC with MUSIC. I accomplished that after pairing to Z10 to 2013 Toyota, by CANCELLING ALL CONNECTIONS leaving only the pairing itself. Then, you must choose some connections in the pairing section of SET-UP within Navigation itself. If you cannot even get to that point, try depressing on the Navigation Screen, INTERNET at which point it will allow you access to SET-UP when you see the message NO ENTUNE APP FOUND ON YOUR BlackBerry Z10. Red-headed step-child is what we all are. Have to help each other. Arsen I appeciate you helping all of us. Seems the list is about 20 who have voiced it here in this one thread. We voice it for the 200 million BlackBerry users worldwide, all left out in the cold despite a pretty cool Z10 product. Far better than iPhone 5. I chose phone and internet. Internet comes back no BLACKBERRY Z10 TOYOTA ENTUNE APP. But, at least now I don't have to hear the music HELLO and do not also have the BlackBerry Z10 LOCKED-UP with pop up message you have to say OK to 10 million times to clear and never can actually do a thing on the BlackBerry Z10 as a result. Cannot add an in-coming call to Contacts Z10. Cannot see name and number calling me, just the bloody number, no clue who is calling ridiculous. Cannot do diddle-e-squat on BlackBerry Z10 on Toyota entune. No App. Horrendous. TAGS : Z10 no caller-id name, Z10 no app Toyota 2013 entune, Z10 add call to Contacts

Lync 2013 PSTN calling not working with Sonus SBC 1000 over TLS and SRTP

Dear All,
We have recently installed Lync 2013 Enterprise Edition with a Pool of 3 FE Servers (MEDIATION COLLOCATED).
We need to implement TLS and SRTP with Sonus SBC 1000. However calls are not routing b/w SBC and Lync.
We are using wild card certificate with multiple SIP Domains as SAN(s), for internal FE servers as well SBC.
Also i would like to mentioned here that inbound and outbound calls are routing properly when we tested it over TCP.
When I move to TLS Only calls from Lync to SBC (outgoing) are working without encryption.
Here are the OCS Logger traces for incoming calls which are not landing on lync:
TL_INFO(TF_PROTOCOL) [1]2C5C.0D30::04/30/2014-14:35:18.020.00026518.020.00026518.020.00026518.020.00026518.020.00026518.020.00026518.020.00026518.020.000265d2
(S4,SipMessage.DataLoggingHelper:sipmessage.cs(774))[3491463749]
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_AE0419>], 10.10.0.11:5067->10.10.7.50:25678
SIP/2.0 400 Bad Request
FROM: "3158222726"<sip:[email protected]>;tag=ac3201ce-4d7
TO: <sip:[email protected]:5067>;epid=D2091CF753;tag=f373543c
CSEQ: 2 INVITE
CALL-ID: [email protected]
VIA: SIP/2.0/TLS 10.10.7.50:5067;branch=z9hG4bK-UX-ac32-01ce-0b14
CONTENT-LENGTH: 0
SERVER: RTCC/5.0.0.0 MediationServer
------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [1]2C5C.0D30::04/30/2014-14:35:18.027.00026518.027.00026518.027.00026518.027.00026518.027.00026518.027.00026518.027.00026518.027.000265d7
(S4,SipMessage.DataLoggingHelper:sipmessage.cs(774))[2666394843]
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_370F030>], 10.10.0.11:58059->10.10.0.13:5061
SERVICE sip:2138797082;[email protected];user=phone SIP/2.0
FROM: <sip:2138797082;[email protected];user=phone>;epid=DCFDB95F4C;tag=17d286a93
TO: <sip:2138797082;[email protected];user=phone>
CSEQ: 3 SERVICE
CALL-ID: de750f98bdd94e908be5f2f975228ff7
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 10.10.0.11:58059;branch=z9hG4bKd47f1d3c
CONTACT: <sip:[email protected];gruu;opaque=srvr:MediationServer:CiGdW3iH5FiI3Qvr3PIKGQAA>
CONTENT-LENGTH: 630
SUPPORTED: gruu-10
USER-AGENT: RTCC/5.0.0.0 MediationServer
CONTENT-TYPE: application/msrtc-reporterror+xml
<?xml version="1.0" encoding="us-ascii"?>
<reportError xmlns="http://schemas.microsoft.com/2006/09/sip/error-reporting">
<error callId="[email protected]" fromUri="sip:3158222726;[email protected];user=phone" toUri="sip:2138797082;[email protected];user=phone" fromTag="ac3201ce-4d7"
toTag="" requestType="INVITE" contentType="application/sdp;call-type=audio" responseCode="400"><diagHeader>10013;reason="Gateway peer in inbound call is not found in topology document or does not depend
on this Mediation Server"</diagHeader><progressReports /></error></reportError>------------EndOfOutgoing SipMessage
Call
Send SMS
Add to Skype
You'll need Skype CreditFree via Skype

@Paul, Thanks for you response.
All ports / IP Add / DNS are defined properly. Telenet on listening port is working.
We are using Public Certificate for 3 Domains (wild card) and same is loaded and verified in SBC
I've not reviewed the OCS logs properly posted above.
What i've found or seems to me is that in a TLS Calls:
After receiving SIP Invite from SBC, mediation server started TLS Negotiation Process b/w Lync 2013 Server Pool and it fails.
SIP Domains:
contoso.com (default)
fabrikam.com
Lync FE Pool (lync.contoso.com
Here are the some more logs.
TL_INFO(TF_PROTOCOL) [0]2DF8.2930::05/01/2014-11:50:31.612.00025e49 (S4,SipMessage.DataLoggingHelper:sipmessage.cs(774))[2716989131]
<<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_103DFE0>], 10.10.0.11:5067<-10.10.7.50:24591
INVITE sip:[email protected]:5067 SIP/2.0
FROM: "3158222726" <sip:[email protected]>;tag=ac3201ce-ae
TO: <sip:[email protected]:5067>
CSEQ: 2 INVITE
CALL-ID: [email protected]
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 10.10.7.50:5067;branch=z9hG4bK-UX-ac32-01ce-010c
CONTACT: <sip:[email protected]:5067;transport=TLS>
CONTENT-LENGTH: 406
SUPPORTED: replaces,update,100rel
USER-AGENT: SONUS SBC1000 3.1.2v293 Sonus SBC
CONTENT-TYPE: application/sdp
ALLOW: INVITE, ACK, CANCEL, BYE, NOTIFY, OPTIONS, REFER, REGISTER, UPDATE, PRACK
P-ASSERTED-IDENTITY: "3158222726" <sip:[email protected]>
v=0
o=SBC 9 1001 IN IP4 10.10.7.50
s=VoipCall
c=IN IP4 10.10.7.50
t=0 0
m=audio 16418 RTP/AVP 8 0 101 13
c=IN IP4 10.10.7.50
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:13 CN/8000
a=ptime:20
a=tcap:1 RTP/SAVP
a=pcfg:1 t=1
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pqL6Tke8pVmXPuplJ1G3+Sr9jM97H8R7iBagWzzh|2^31|1:1
a=sendrecv
------------EndOfIncoming SipMessage
TL_INFO(TF_PROTOCOL) [1]2DF8.0E04::05/01/2014-11:50:31.665.00025e8e (S4,SipMessage.DataLoggingHelper:sipmessage.cs(774))[2716989131]
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_103DFE0>], 10.10.0.11:5067->10.10.7.50:24591
SIP/2.0 100 Trying
FROM: "3158222726"<sip:[email protected]>;tag=ac3201ce-ae
TO: <sip:[email protected]:5067>
CSEQ: 2 INVITE
CALL-ID: [email protected]
VIA: SIP/2.0/TLS 10.10.7.50:5067;branch=z9hG4bK-UX-ac32-01ce-010c
CONTENT-LENGTH: 0
------------EndOfOutgoing SipMessage
TL_INFO(TF_CONNECTION) [1]184C.0EFC::05/01/2014-11:50:32.652.00025f32 (SIPStack,SIPAdminLog::WriteConnectionEvent:SIPAdminLog.cpp(454))[946832530] $$begin_record
Severity: information
Text: TLS negotiation started
Local-IP: 10.10.0.11:5061
Peer-IP: 10.10.0.11:52529
Connection-ID: 0x10BE00
Transport: TLS
$$end_record
TL_INFO(TF_PROTOCOL) [1]184C.0EFC::05/01/2014-11:50:32.669.00026236 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[1853494582] $$begin_record
Trace-Correlation-Id: 1853494582
Instance-Id: 425D
Direction: incoming
Peer: 10.10.0.11:52529
Message-Type: request
Start-Line: NEGOTIATE sip:127.0.0.1:5061 SIP/2.0
FROM: <sip:contoso.com>;ms-fe=LYNCFE1.fabrikam.com
TO: <sip:contoso.com>
CALL-ID: aa53739ef9b34b93ba9c97d3ee56cb99
CSEQ: 1 NEGOTIATE
VIA: SIP/2.0/TLS 10.10.0.11:52529
MAX-FORWARDS: 0
CONTENT-LENGTH: 0
SUPPORTED: NewNegotiate
SUPPORTED: ECC
REQUIRE: ms-feature-info
SERVER: RTC/5.0
$$end_record
TL_INFO(TF_CONNECTION) [1]184C.0EFC::05/01/2014-11:50:32.669.0002636e (SIPStack,SIPAdminLog::WriteConnectionEvent:SIPAdminLog.cpp(383))[946832530] $$begin_record
Severity: information
Text: Connection established
Peer-IP: 10.10.0.11:52529
Peer: lync.contoso.com:52529;ms-fe=LYNCFE1.fabrikam.com
Peer-Cert: contoso.com(LYNCFE1.fabrikam.com)
Transport: M-TLS
Data: alertable="yes"
$$end_record
TL_WARN(TF_CONNECTION) [1]184C.0EFC::05/01/2014-11:50:32.669.00026387 (SIPStack,SIPAdminLog::WriteConnectionEvent:SIPAdminLog.cpp(386))[946832530] $$begin_record
Severity: warning
Text: The pool FQDN provided by the peer in its NEGOTIATE feature information does not match the pool configured in CMS for the server FQDN that it provided
Peer-IP: 10.10.0.11:52529
Peer: lync.contoso.com:52529;ms-fe=LYNCFE1.fabrikam.com
Peer-Cert: contoso.com(LYNCFE1.fabrikam.com)
Transport: M-TLS
Data: fqdn="LYNCFE1.fabrikam.com";pool="contoso.com";expected-fqdn="lync.contoso.com";info="Possible server configuration issue"
$$end_record
TL_INFO(TF_DIAG) [1]184C.0EFC::05/01/2014-11:50:32.670.000265be (SIPStack,SIPAdminLog::WriteDiagnosticEvent:SIPAdminLog.cpp(802))[1853494582] $$begin_record
Severity: information
Text: Routed a locally generated response
SIP-Start-Line: SIP/2.0 200 OK
SIP-Call-ID: aa53739ef9b34b93ba9c97d3ee56cb99
SIP-CSeq: 1 NEGOTIATE
Peer: lync.contoso.com:52529;ms-fe=LYNCFE1.fabrikam.com
$$end_record
TL_INFO(TF_PROTOCOL) [1]184C.0EFC::05/01/2014-11:50:32.670.00026615 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[1853494582] $$begin_record
Trace-Correlation-Id: 1853494582
Instance-Id: 425E
Direction: outgoing;source="local"
Peer: lync.contoso.com:52529;ms-fe=LYNCFE1.fabrikam.com
Message-Type: response
Start-Line: SIP/2.0 200 OK
FROM: <sip:contoso.com>;ms-fe=LYNCFE1.fabrikam.com
To: <sip:contoso.com>;tag=C3A751556F332F7265E9BA2517C878D4
CALL-ID: aa53739ef9b34b93ba9c97d3ee56cb99
CSEQ: 1 NEGOTIATE
Via: SIP/2.0/TLS 10.10.0.11:52529;ms-received-port=52529;ms-received-cid=10BE00
Content-Length: 0
Require: ms-feature-info
Supported: NewNegotiate,OCSNative,ECC,IPv6,TlsRecordSplit
Server: RTC/5.0
$$end_record
TL_INFO(TF_PROTOCOL) [1]2DF8.1078::05/01/2014-11:50:32.671.000266da (S4,SipMessage.DataLoggingHelper:sipmessage.cs(774))[720988281]
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_F8A09B>], 10.10.0.11:52529->10.10.0.11:5061
SERVICE sip:2138797082;[email protected];user=phone SIP/2.0
FROM: <sip:2138797082;[email protected];user=phone>;epid=16FEE4A02E;tag=22fd877f3a
TO: <sip:2138797082;[email protected];user=phone>
CSEQ: 3 SERVICE
CALL-ID: ac0f7bc4cdc94c1dbd0bb51c7c02c890
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 10.10.0.11:52529;branch=z9hG4bK67a4c9d1
CONTACT: <sip:[email protected];gruu;opaque=srvr:MediationServer:CiGdW3iH5FiI3Qvr3PIKGQAA>
CONTENT-LENGTH: 628
SUPPORTED: gruu-10
USER-AGENT: RTCC/5.0.0.0 MediationServer
CONTENT-TYPE: application/msrtc-reporterror+xml
- <reportError xmlns="http://schemas.microsoft.com/2006/09/sip/error-reporting">
- <error callId="[email protected]"
fromUri="sip:3158222726;[email protected];user=phone"
toUri="sip:2138797082;[email protected];user=phone"
fromTag="ac3201ce-ae"
toTag=""
requestType="INVITE"
contentType="application/sdp;call-type=audio"
responseCode="400">
<diagHeader>10013;reason="Gateway peer in inbound call is not found in topology document or does not depend on this Mediation Server"</diagHeader>
<progressReports/>
- </error>
------------EndOfOutgoing SipMessage

Incoming Email not working for one web app, but IS working for others in same farm.

I have enabled incoming email in a large multi-web application farm running SharePoint 2013 SP1 and the Feb 2015/March 2015 CUs that were pushed down in Windows Updates. It works just fine for two web applications, but it will not work for one particular
web application. I see the email land in the /drop folder. If I remove the email address from the library in http://nonworkingapp web
application and use it on a library in a http://working web application, then the email is processed successfully. Conversely,
if I take an address from a library in http://working and move it to a library inhttp://nonworkingapp it
does not work. I've tried multiple site collections on http://nonworkingapp. This rules out any SMTP issues, etc. and means
the issue is bound to this web application. Permissions are the same between the "non-working" and "working" locations: domain users are members on the site, and the library inherits permissions from the site. The App Pool and Timer
Service accounts have rights to the folder and to the library.
The error in the ULS log is the typical error processing message, but it gives no additional information even though I have enabled Verbose logging.
The Incoming E-Mail service has completed a batch. The elapsed time was 00:00: 00.0156294. The service processed 2 message(s) in total. Errors occurred processing 2 message(s): Message ID: Message
ID:
Typically, you see "alias not found" or some permissions or quota issue here, but this issue is NOT the same as those. In this case NO additional information is provided in ULS.
Edit: running stsadm.exe -o refreshdms -url http://one/failing/site fixed some similar issues where "Unknown alias" was
reported even though the list did have an alias (see here:http://blogs.technet.com/b/vinitt/archive/2009/07/15/e-mail-enabled-list-alias-information-is-not-synchronized-between-configuration-database-and-content-database.aspx).
This did not fix the issue for the case mentioned above.
What could possibly be the issue preventing incoming email from working on this one web application?

Thanks for your reply. Incoming E-Mail was running on all 3 WFE, however I do not think this was the cause. As noted, _other_ emails send to http://workingapp are working just fine. I can send two emails - one to a library on http://workingapp
and another on http://notworkingapp. The email to http://notworkingapp will sit in the drop folder, while the one to http://workingapp will be processed correctly. In addition, the ULS log on WFE1 indicates an attempt to process the file fails.
So clearly, the timer job on WFE1 is running and attempting to process the file. For kicks, I disabled the service on WFE2 and WFE3, but with no change in the behavior.
WRT to #1, I have done this already (as noted in the original post). I can remove an alias from http://workingapp and place it on http://nonworkingapp and it is not processed. If I return it to http://workingapp it is processed. In other
words, I go into Library Settings -> Incoming Email for a working library, and copy the alias, then disable incoming email for that library. I then go to Library Settings -> Incoming Email for a non-working library, enable incoming email, and paste
the alias. Once done, emails sent to the alias are not processed. I then do the reverse, and it works again.
On #2, I have already tried multiple libraries on http://nonworkingapp (also as noted in the original post).
In my environment, incoming email does not work for any library on http://nonworkingapp, but does work for any library on http://workingapp. Troubleshooting listed above has ruled out any problem with basic mail server configuration or library settings
- the problem is bound to http://nonworkingapp
I'm not sure what you mean by "steps to reproduce the issue". If I knew that, I would probably be able to solve the issue. The best I could come up with to describe the situation would be:
Set up SP 2010 in a three-node configuration with multiple web apps
Enable incoming email and confirm it works
Migrate to a SP2013 SP1 farm in a three node configuration.
Enable a new email alias for a library
Observe that it doesn't work for one web application.
Obviously, it's pretty unlikely that attempting that in a lab would actually have the same result.

Mac 10.6.8's video calls not working! Please help!

Reposting the text below because there was no reply to the last message. Ack! Please help. Skype, is there a solution yet? Or even just an explanation and a time frame so we know when we can expect video calls to work again? Thanks! ---------------I have a Mac laptop, version 10.6.8. I have the most updated version of Skype available for my computer, Skype version 6.15. For the past few months, I've been unable to use video chat or file sharing. I can receive some files, but cannot send files. Other people can video chat with me and I can see them, but they can't see me. My built-in camera works just fine for other purposes, so I know it's not an issue with my camera. I've also noticed that a lot of Mac users seem to be having trouble with Skype lately. I've checked my audio/video preferences for Skype and my computer's camera preferences, and everything is as it should be. Does anyone know how to fix this?
Thanks in advance for any help!

See: How to perform a "clean install" of Flash Player in Mac OS X
In addition to these steps, I recommend, BEFORE emptying the trash:
Go to: [User]/Library/Preferences (hold the "Option" key when clicking "Go" from the Finder menu to reveal the hidden Library folder)
Trash the ENTIRE Macromedia folder from there
Aside from that additional step, follow the rest of the instructions to the letter and you should be back up and running.

Incoming Calls Not Working

Similar Messages

Maybe you are looking for