SKYPE CONNECT Trunk to CUCM.
im interested in creating a sip trunk to skype connect. im new to voice but i think i can get it done.
is a cube absolutely necessary or the trunk can be created directly to cucm.
also can a regular cisco router with access to the internet work just as well as a cube ?
my current system has a voice gateway E1 to the PSTN.. is it possible to give this voicerouter access to the internet and use to connect to skype instead of the cube?
hey anas.. i have set up the skype trunk.. and calls seem to be going out my problem is incoming calls i have a skype id .. this is the matching incoming dial peer.. dial-peer voice 6 voip
description incoming skype
translation-profile incoming IncomingSkype
preference 1
destination-pattern 13478093543
session target ipv4:192.168.xxx.x (cucm)
incoming called-number 13478093543
voice-class codec 1
dtmf-relay h245-alphanumeric.
the translation profile changes the calling number to a 4 digit number
a number which matches this dial peer which is already on the working system
dial-peer voice 10 voip
description Codec Match
destination-pattern 2...
session target ipv4:192.168.xxx.x (cucm)
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad
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Skype Connect SIP trunking to Yeastar U100
I am trying to get this service running for the first time. I signed up for a SIP profile through Skype Manager and followed Yeastar's setup instructions. Both ends indicate the regestration was successful. I can receive incoming Skype calls. I followed Yeastar's example to set up and Outgoing Route and I am attempting to call the Skype Echo Test at 001760-660-4690 as they recommend. All I ever hear is a recording saying that "All Circuiits are Busy". I have tried dialing other PSTN number and I get the same failure. Yeastar Support has look at the outgoing call msgs between me and Skype and they say what I'm sending looks right but Skype Connect is saying that the number I'm calling is invalid.
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Skype Connect and Elastix for incoming and outgoin...
Hi,
I ordered Skype Connect, And i want to integrate skype connect with my Elastix server to handle incoming and outgoing calls.
I created new SIP Trunk through GUI with the following info :
Incoming Settings
[skype_in]
disallow=all
type=friend
username=sipusername
fromdomain=sip.skype.com
fromuser=sipusername
realm=sip.skype.com
host=sip.skype.com
dtmfmode=rfc2833
secret=sipuserpass
nat=yes
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qualify=yes
allow=alaw
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amaflags=default
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sendrpid=yes
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Outgoing Settings :
[Skype_out]
context=from-trunk-sip-Skype_out
Register String:
SIPUSER:[email protected]
Incoming calls are working properly, But outgoing calls not working, It keeps saying ( cannot-complete-as-dialed )
Elastix log after Dial
[Jul 17 01:01:25] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:1] ResetCDR("SIP/100-00000010", "") in new stack
[Jul 17 01:01:25] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:2] NoCDR("SIP/100-00000010", "") in new stack
[Jul 17 01:01:25] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:3] Progress("SIP/100-00000010", "") in new stack
[Jul 17 01:01:25] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:4] Wait("SIP/100-00000010", "1") in new stack
[Jul 17 01:01:26] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:5] Progress("SIP/100-00000010", "") in new stack
[Jul 17 01:01:26] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:6] Playback("SIP/100-00000010", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[Jul 17 01:01:26] VERBOSE[3501] file.c: -- <SIP/100-00000010> Playing 'silence/1.gsm' (language 'en')
[Jul 17 01:01:27] VERBOSE[3501] file.c: -- <SIP/100-00000010> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
[Jul 17 01:01:29] VERBOSE[3501] file.c: -- <SIP/100-00000010> Playing 'check-number-dial-again.gsm' (language 'en')
[Jul 17 01:01:32] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:7] Wait("SIP/100-00000010", "1") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [00201005566352@from-internal:8] Congestion("SIP/100-00000010", "20") in new stack
[Jul 17 01:01:33] WARNING[3501] channel.c: Prodding channel 'SIP/100-00000010' failed
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: == Spawn extension (from-internal, 00201005566352, exited non-zero on 'SIP/100-00000010'
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [h@from-internal:1] Macro("SIP/100-00000010", "hangupcall") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/100-00000010", "1?endmixmoncheck") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,9)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:9] NoOp("SIP/100-00000010", "End of MIXMON check") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:10] GotoIf("SIP/100-00000010", "1?nomeetmemon") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,2
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:28] NoOp("SIP/100-00000010", "End of MEETME check") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:29] GotoIf("SIP/100-00000010", "1?noautomon") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,34)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:34] NoOp("SIP/100-00000010", "TOUCH_MONITOR_OUTPUT=") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:35] GotoIf("SIP/100-00000010", "1?noautomon2") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,41)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:41] NoOp("SIP/100-00000010", "MONITOR_FILENAME=") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:42] GotoIf("SIP/100-00000010", "1?skiprg") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,45)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:45] GotoIf("SIP/100-00000010", "1?skipblkvm") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,4
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:48] GotoIf("SIP/100-00000010", "1?theend") in new stack
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Goto (macro-hangupcall,s,50)
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: -- Executing [s@macro-hangupcall:50] Hangup("SIP/100-00000010", "") in new stack
[Jul 17 01:01:33] VERBOSE[3501] app_macro.c: == Spawn extension (macro-hangupcall, s, 50) exited non-zero on 'SIP/100-00000010' in macro 'hangupcall'
[Jul 17 01:01:33] VERBOSE[3501] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000010'
Are there any modifications should i do in Incoming and outgoing settings to work properly .?
Regards,May be the prefix is wrong. You dont have to put 00 before the country number.
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Avaya PBX versus Skype Connect
Dears,
we have Avaya PBX ver. 5.2 (upgrade 2010) with enabled SIP server. Normally we use SIP trunks as input lines and SIP extensions.
Now we want to apply Skype Connect as a new input channel to our PBX (Czech republic, Slovakia, Hungary), but I just received answer from Avaya that Skype Connect is possible to use only un USA at this moment.... In the first phase, Avaya customers in the U.S.3 will have access to Skype Connect™, providing a SIP communications channel between Avaya communications systems and Skype.
Please help me with this case.
Thank you
Tomas Husar ([email protected])Hello eximosr,
I not sure why Avaya is telling you that you cannot use SIP in your country, and don't know more specifics on the Avaya product you are using.
We have Skype for SIP users all around the world using Avaya IP Office and other Avaya systems. Half of the older systems use SIP gateway devices because they have not upgraded their systems to current releases. That may be a solution for you also. Please visit our: http://www.skype.com/intl/en/business/skype-connect/ page to see the list of vendors for gatyeway products there.
There are many ways to use Skype for SIP and if you contact you local Telecom Consultant, they will be able to provise you with more information.
I hope this helps you solve you issue.
Thanks for using Skype and Skype Community Forums.
Regards,
Victor S.
Regards,
Victor S.
Skype Enterprise Support -
Skype Connect - Long time until call is received v...
Hi Community,
I just started to use Skype Connect. I configured one of my Skype accounts for inbound calls. The user is account is displayed as ONLINE and I'm able to dial in.
My problem is that it takes 35 seconds to until I receive the SIP INVITE from sip.skype.com.
Is this normal?
Best regards
TimmiHello. I'm using Elastix (Asterisk) and REGISTER method.
Suddenly, but now everything works. Nothing changes in my config.
I think the skype servers needs time to update the record Skype->SIP.
my register string and skype trunk:
register=99999999999999:[e-mail removed for privacy and security]
[skype_trunk]
disallow=all
localnet=192.168.1.0/255.255.255.0
externip=XX.XX.XX.XX
nat=yes
defaultexpirey=30
qualify=300
externrefresh=30
username=99999999999999
type=friend
secret=xxxxxxxxxxx
insecure=port,invite
host=sip.skype.com
fromuser=99999999999999
fromdomain=sip.skype.com
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=alaw
allow=ulaw
allow=g729 -
Skype Connect times out with Elastix - Desperately...
Hello All,
I'm trying to use Skype Connect with Elastix and I am having issues. I have tried everything I could.
The logs keep showing this :
Registration for ***' timed out, trying again (Attempt #3)
My registration string is [***]
Shouldn't the above line make the trunk get registered?
Please help.
Thanks
Edit: I have already tried using putty to connect to the Elastix box. I tried using telnet sip.skype.com 5060. All this works fine. I am able to ping sip.skype.com from the elastix box. I haven't found signs of connectivity issues or firewall blocking or anything like that so far. But strangely, the timedout error is being really tough on me
Current settings Screenshot (Password replaced) :Hi,
Before going further, would you please let me know the RDC version that used in the problematic computer? Was it the same with other computers?
Please temporarily disable firewall and then check if this issue can be solved. Based on your current description, it’s hard to say the root reason of this issue.
So, I suggest that you should check relevant log files and get some clues. It will help us to narrow down and solve your issue.
Hope this helps.
Best regards,
Justin Gu -
Can I use straight cable to connect trunk ports between 2 switches?
Hi,
Am I able to use straight instead of cross cable to connect trunk ports between 2 switches??
thanks!Hi Devang,
When a 10/100 Fast Ethernet interface is enabled, one end of the link must perform media dependent interface (MDI) crossover (MDIX), so that the transmitter on one end of the data link is connected to the receiver on the other end of the data link (a crossover cable is typically used).
The Auto-MDIX feature eliminates the need for crossover cabling by performing an internal crossover when a straight cable is detected during the auto-negotiation phase.
HTH, if yes please rate the post.
Ankur -
Help using the Connection Trunk command
Hello Group,
I could use a bit of assistance accomplishing the following.
I have two Cisco 1760 Routers with a WIC 1/DSU T1 and one VIC2/E/M in each router. I am new to CLI commands and could use some tips.
I would like to set up these routers (I will call them Router A and Router B) to communicate with each other via a internal IP network which spans between two seperate locations.
I would like to set up these routers to use the E&M ports to send and receive audio between locations. The E&M ports are connected to my external equipment which allows for transmit and receive audio to be delivered to the E&M port.
I need to be able to allow for the routers to use 2 wire or 4 wire audio between locations, which is matched at both ends. The routers do not dial phone numbers nor are phones in use at all. Basically the routers are going to be used to send and receive two way radio voice between the locations which the radios are interfaced to the E&M ports.
Can anyone please help me on configuring these routers to accomplish this and if I need to use the Connection Trunk command I can do so, as this is the only purpose of the routers is to relay audio between locations.
Please Help.
Thanks A bunch!!
RickThank you for taking the time to reply, However I have allready configured the routers to link the LMR sites together. I have been connecting LMR assets utilizing ROIP for many years using non-cisco equipment which is specifically designed for ROIP, I have a large inventory of 1760 routers which were originally used for Telco applications which were replaced by newer equipment. The fact that the routers have become "obsolete" does not mean that they are not usable for my application. How can you make a statement that this is "limited equipment" when the intended use of this equipment for my application has been accomplished without limitations which are specific to my operation.
I posted a request on this forum to obtain some help in the configuration of these devices from people who are familiar with the 1760 router. Your response was the first response which I had received in regards to my request for assistance, I thank you for at least attaching a link to some resources which I can utilize. Utilizing the services of a "reputable consultant" or "UC certified partner" would be acceptable in some circumstances, however utilizing the services which would be provided by such would only pose limitations on our technical skills. I do not want to have to rely on a third party consultants to acheive what is engineered and desgned in house by our staff.
I am not looking to purchase new equipment, nor am I looking for consulting help on this issue as we manage and deploy equipment in house. Enlisting the services of a consultant or certified partner for our Cisco equipment deployments would only add unnecessary costs and time to deploy equipment. Again thank you for the link to the examples.
Respectfully,
Richard Wiglesworth -
Can't make outgoing call with Skype Connect
I have my Asterisk PBX configured with Skype Connect using SIP with TLS and SRTP. Most of my outgoing calls go through, but sometimes I can't get call out. I was able to leave asterisk console up and collect verbose and sip debug data. Can somebody help me diagnose why my calls aren't going through?
I've changed my external IP (I'm behind a NAT'd firewall) to 1.2.3.4 and my SIP profile's user ID to 11111111111111. and my domain name to test.com. If someone working for Skype needs that information they can email me and I'll send it privately.
My config:
[general]
context=default_context
allowguest=no
alwaysauthreject=yes
allowoverlap=no
udpbindaddr=0.0.0.0
tlsenable=yes
tlsbinddir=0.0.0.0
tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,tcp,tls
srvlookup=yes
dynamic_exclude_static = yes
buggymwi=yes
contactpermit=192.168.1.0/24
register => tls://111111111111111:[email protected]
[skype]
type=friend
context=from-skype
dtmfmode=rfc2833
host=sip.skype.com
username=11111111111111
fromuser=11111111111111
secret=abcd12345
disallow=all
allow=ulaw
allow=alaw
nat=yes
fromdomain=sip.skype.com
insecure=port,invite
transport=tls
srtpcapable=yes
encryption=yes
SIP Debugging enabled
[2012-08-23 19:22:33] NOTICE[16552]: chan_sip.c:13465 sip_reregister: -- Re-registration for [email protected]
> doing dnsmgr_lookup for 'sip.skype.com'
> ast_get_srv: SRV lookup for '_sips._tcp.sip.skype.com' mapped to host 1.sip.skype.com, port 5061
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to 63.209.144.201:5061:
REGISTER sip:sip.skype.com:5061 SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as6edf93cf
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 32495 REGISTER
User-Agent: Asterisk PBX 10.5.2
Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:sip.skype.com:5061", nonce="5036b5770000182c78c1d1909cfd5c74e33f033c952d240d", response="81001ceacd91b16ebb956d3c55991471"
Expires: 120
Contact: <sip:[email protected]:5061;transport=TLS>
Content-Length: 0
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 200 OK
From: <sip:[email protected]>;tag=as6edf93cf
To: <sip:[email protected]>;tag=c990d13f-90f7a10d-0-55cb59a8-0
Call-ID: [email protected]
CSeq: 32495 REGISTER
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK2726fcb8;rport=50541;received=1.2.3.4
Expires: 45
Contact: <sip:[email protected]:5061;transport=tls>;expires=45
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[2012-08-23 19:22:33] NOTICE[17932]: chan_sip.c:21399 handle_response_register: Outbound Registration: Expiry for sip.skype.com is 45 sec (Scheduling reregistration in 30 s)
<--- SIP read from UDP:192.168.1.16:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Scott's Office" <sip:[email protected]:5060>
Expires: 240
User-Agent: Cisco/SPA504G-7.5.2b
Content-Length: 234
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp
v=0
o=- 88651316 88651316 IN IP4 192.168.1.16
s=-
c=IN IP4 192.168.1.16
t=0 0
m=audio 16484 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.1.16:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer 'scott_office' for 'scott_office' from 192.168.1.16:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.16:16484
Looking for 19739928881 in home (domain asterisk.test.com)
list_route: hop: <sip:[email protected]:5060>
<--- Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Executing [19739928881@home:1] Dial("SIP/scott_office-000000b0", "SIP/skype/+19739928881") in new stack
== Using SIP RTP CoS mark 5
Audio is at 9302
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 63.209.144.201:5061:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.5.2
Date: Thu, 23 Aug 2012 23:22:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 370
v=0
o=root 1671301052 1671301052 IN IP4 192.168.1.15
s=Asterisk PBX 10.5.2
c=IN IP4 192.168.1.15
t=0 0
m=audio 9302 RTP/SAVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
-- Called SIP/skype/+19739928881
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 100 Trying
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 407 Proxy Authentication Required
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip.skype.com", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", algorithm=MD5
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport=50541;received=1.2.3.4
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 63.209.144.201:5060
Transmitting (NAT) to 63.209.144.201:5061:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1c3f16ee;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.5.2
Content-Length: 0
Audio is at 9302
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 63.209.144.201:5061:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.5.2
Proxy-Authorization: Digest username="11111111111111", realm="sip.skype.com", algorithm=MD5, uri="sip:[email protected]", nonce="5036bb5800012cdd3d20e5090cc200805f7d0bbd58318e9e", response="6efb0e37178bae868f0a1e0ddf110e3c"
Date: Thu, 23 Aug 2012 23:22:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 370
v=0
o=root 1671301052 1671301053 IN IP4 192.168.1.15
s=Asterisk PBX 10.5.2
c=IN IP4 192.168.1.15
t=0 0
m=audio 9302 RTP/SAVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:TRh/HeKozlBO/mmYHNTiS5KMnefVI0aRicLoDNjb
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 100 Trying
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: REGISTER
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 180 Ringing
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: SipGW 8
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Contact: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
list_route: hop: <sip:[email protected]:5061;maddr=63.209.144.201;transport=tls>
-- SIP/skype-000000b1 is ringing
<--- Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
<--- SIP read from TLS:63.209.144.201:5061 --->
SIP/2.0 408 Request Timeout
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport=50541;received=1.2.3.4
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
[2012-08-23 19:22:45] WARNING[17932]: chan_sip.c:20947 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '[email protected]'. Giving up.
set_destination: Parsing <sip:[email protected]> for address/port to send to
set_destination: set destination to 63.209.144.201:5060
Transmitting (NAT) to 63.209.144.201:5061:
ACK sip:[email protected]:5061;maddr=63.209.144.201;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.1.15:5061;branch=z9hG4bK1b988ba5;rport
Max-Forwards: 70
From: "Scott's Office" <sip:[email protected]:5060>;tag=as049830bd
To: <sip:[email protected]>;tag=c990d13f-13c4-5036bb3b-714198b9-32d150b4
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.5.2
Content-Length: 0
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [19739928881@home:2] Hangup("SIP/scott_office-000000b0", "") in new stack
== Spawn extension (home, 19739928881, 2) exited non-zero on 'SIP/scott_office-000000b0'
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
<--- Reliably Transmitting (NAT) to 192.168.1.16:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62;received=192.168.1.16;rport=5060
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 10.5.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.1.16:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.16:5060;branch=z9hG4bK-819ce62
From: "Scott's Office" <sip:[email protected]>;tag=9961686dacab532ao0
To: <sip:[email protected]>;tag=as3f27fa61
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Scott's Office" <sip:[email protected]:5060>
User-Agent: Cisco/SPA504G-7.5.2b
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
Really destroying SIP dialog '[email protected]' Method: ACKI wound up calling skype support. This is the final sip.conf looks like. Hope it helps. Good luck.
Scott
[general]
context=default_context
allowguest=no
alwaysauthreject=yes
allowoverlap=no
udpbindaddr=0.0.0.0
tlsenable=yes
tlsbinddir=0.0.0.0
tlscertfile=/usr/local/asterisk/etc/asterisk/keys/asterisk.pem
tlscafile=/usr/local/asterisk/etc/asterisk/keys/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp,tcp,tls
srvlookup=yes
dynamic_exclude_static = yes
buggymwi=yes
contactpermit=192.168.1.0/24
register => tls://[email protected]
[skype]
type=friend
context=from-skype
dtmfmode=rfc2833
host=sip.skype.com
username=user
fromuser=user
secret=pass
disallow=all
allow=ulaw
allow=alaw
nat=yes
fromdomain=sip.skype.com
insecure=port,invite
transport=tls
srtpcapable=yes
encryption=yes -
Third Party Phone over SIP Trunk with CUCM 9.x
Hi all,
I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
Cisco Phone: INVITE sip.60xxxx%23@ipadress
Third Party SIP Phone: INVITE sip:[email protected]
It seems the Cisco phones gets some extra configured the Third Party ones dont...
Thanks in advance for any help.
//PerThanks for the answer
Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty. The termination Cause Code is that the number requested is Unallocated/Unassigned..
In other words something is missing to get CUCM to strip 60 from the Polycom phones dialed number and send it towards the SIP trunk like it does when the Cisco phones call it.
Unfortunatley i dont have the meens to attach the trace...
Thanks again for any help/advice
With regards, Per. -
Mitel System With Trunk To CUCM DTMF Not Recognized.
Hello,
Not sure if this is correct forum but maybe someone has seen this. Overview is that DTMF is not being recognized by CVP from the greeting. Call flow is such.
Mitel 5000 --siptrunk---CUCM9---siptrunk---CVP9
Currently the DTMF type on all trunks is RFC2833. And verified the Mitel is RFC as well. I think the problem is that for payload Mitel is sending 96. But on the Cisco CUCM the trunks are using the default of 101 for both the trunk to Mitel and CVP. Here is the invite in from Mitel you can see it sends payload of 96.
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.254.20.32:5060;branch=z9hG4bK2312043656-10065
Max-Forwards: 70
Allow: NOTIFY,REGISTER,REFER,SUBSCRIBE,INVITE,ACK,OPTIONS,CANCEL,BYE
User-Agent: Mitel-5000-ICP-5.1.0.56
P-Asserted-Identity: "Laura B." <sip:[email protected]>
From: "Laura B." <sip:[email protected]:5060>;tag=Mitel-5000_2312043669-10065
To: 72799 <sip:[email protected]:5060>
Call-ID: 2312043591-10065
CSeq: 1 INVITE
Contact: "Laura B." <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 266
v=0
o=Mitel-5000-ICP 169457268 1404250460 IN IP4 10.254.20.31
s=SIP Call
c=IN IP4 10.254.20.31
t=0 0
m=audio 6770 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=ptime:20
a=maxptime:30
a=cdsc:1 image udptl t38
5:16 PM
here is 200 ok
5:16 PM
5:16 PM
Jay Schulze:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.254.20.32:5060;branch=z9hG4bK2312043656-10065
From: "." <sip:[email protected]:5060>;tag=Mitel-5000_2312043669-10065
To: 72799 <sip:[email protected]:5060>;tag=10585346~dfbf10b3-6c69-4443-852f-cbf609935a6f-42551743
Date: Tue, 01 Jul 2014 21:34:24 GMT
Call-ID: 2312043591-10065
CSeq: 1 INVITE
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Allow-Events: presence, kpml
Supported: replaces
Supported: X-cisco-srtp-fallback
Supported: Geolocation
P-Preferred-Identity: <sip:[email protected]>
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 238
v=0
o=CiscoSystemsCCM-SIP 10585346 1 IN IP4 10.38.246.136
s=SIP Call
c=IN IP4 10.38.246.166
b=TIAS:64000
b=AS:64
t=0 0
m=audio 23956 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
This is the invite over to CVP. Still sending 96. I'm pretty sure I can fix this by setting the payload type to 96 on both trunks on CUCM. I was able to test with it afterhours. After reading through Mitel docs there is no way to change payload type on their side. The question is really would there be anything else this could maybe break on the CVP side?Well that did not work. What I found was it was going through 96 all the way to CVP. However when CVP returned a label to the VXML. The VXML sent back the 200 OK with 101.
If I changed the the rtp payload-type nte 96 on the VXML dial-peer it did work. However it broke any other call flow to CVP from being able to recognize DTMF.
I wonder if this is some type of bug on VXML. Because it was my understanding by having the command 'asymmetric payload full' under SIP. It would pass the payload type from one call leg to the other. -
Skype connect, outbound failing
I recently setup skype connect for product testing on an NEC SV8100.
I can recieve call ok.
When i dial out the other end rings, but no RTP audio, then it gives engaged.. I still get charged for each call !!
Done a wireshark trace :
No. Time Source Destination Protocol Length Info
1 0.000000 172.10.0.1 63.209.144.201 SIP/SDP 934 Request: INVITE sip:[email protected], with session description
Frame 1: 934 bytes on wire (7472 bits), 934 bytes captured (7472 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No. Time Source Destination Protocol Length Info
2 0.261931 63.209.144.201 172.10.0.1 SIP 521 Status: 407 Proxy Authentication Required
Frame 2: 521 bytes on wire (4168 bits), 521 bytes captured (4168 bits)
Ethernet II, Src: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f), Dst: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff)
Internet Protocol Version 4, Src: 63.209.144.201 (63.209.144.201), Dst: 172.10.0.1 (172.10.0.1)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No. Time Source Destination Protocol Length Info
3 0.328555 172.10.0.1 63.209.144.201 SIP 455 Request: ACK sip:[email protected]
Frame 3: 455 bytes on wire (3640 bits), 455 bytes captured (3640 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No. Time Source Destination Protocol Length Info
4 0.457122 172.10.0.1 63.209.144.201 SIP/SDP 1175 Request: INVITE sip:[email protected], with session description
Frame 4: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No. Time Source Destination Protocol Length Info
5 0.999458 172.10.0.1 63.209.144.201 SIP/SDP 1175 Request: INVITE sip:[email protected], with session description
Frame 5: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No. Time Source Destination Protocol Length Info
6 2.017736 172.10.0.1 63.209.144.201 SIP/SDP 1175 Request: INVITE sip:[email protected], with session description
Frame 6: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No. Time Source Destination Protocol Length Info
7 4.000270 172.10.0.1 63.209.144.201 SIP/SDP 1175 Request: INVITE sip:[email protected], with session description
Frame 7: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No. Time Source Destination Protocol Length Info
8 8.019283 172.10.0.1 63.209.144.201 SIP/SDP 1175 Request: INVITE sip:[email protected], with session description
Frame 8: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No. Time Source Destination Protocol Length Info
9 11.947857 63.209.144.201 172.10.0.1 UDP 214 Source port: 26998 Destination port: 10020
Frame 9: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits)
Ethernet II, Src: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f), Dst: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff)
Internet Protocol Version 4, Src: 63.209.144.201 (63.209.144.201), Dst: 172.10.0.1 (172.10.0.1)
User Datagram Protocol, Src Port: 26998 (2699, Dst Port: 10020 (10020)
Data (172 bytes)
No. Time Source Destination Protocol Length Info
10 11.948226 172.10.0.1 63.209.144.201 ICMP 70 Destination unreachable (Port unreachable)
Frame 10: 70 bytes on wire (560 bits), 70 bytes captured (560 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
Internet Control Message Protocol
No. Time Source Destination Protocol Length Info
11 11.967964 63.209.144.201 172.10.0.1 UDP 214 Source port: 26998 Destination port: 10020
Frame 11: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits)
Ethernet II, Src: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f), Dst: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff)
Internet Protocol Version 4, Src: 63.209.144.201 (63.209.144.201), Dst: 172.10.0.1 (172.10.0.1)
User Datagram Protocol, Src Port: 26998 (2699, Dst Port: 10020 (10020)
Data (172 bytes)Try to reset all Skype settings.
Quit Skype or use Windows Task Manager to kill any Skype.exe process. Go to Windows Start and in the Search/Run box type %appdata% and then press Enter or click the OK button. The Windows File Explorer will pop up. There locate a folder named “Skype”. Rename this folder to something different, e.g. Skype_old.
If you are on the latest Skype 6.5/6.6 version, then do also this:
Go to Windows Start and in the Search/Run box type %temp%\skype and then press Enter or click the OK button. Delete the DbTemp folder.
Restart Skype.
N.B. If needed, you will still be able to re-establish your call and chat history. All data is still saved in the Skype_old folder. -
Skype connectivity with lync online plan1
hi ,recently I purchased lync online plan1 to enable skype connectivity with lync but not able to add any contact from lync client as I am using lync 2010 client at windows pc and lync 2011 client at mac PC
and I already enabled external communication setting for public messenger like skype from lync admin center(office365 based).Hi skype user,
As Edwin mentioned, you could try to update your client to the latest version.
For Windows, you could install the Lync 2013 Basic.
http://www.microsoft.com/en-us/download/details.aspx?id=35451
For Mac OS , you could install the latest update “October 2014 update for Lync for Mac 2011 14.0.10”
http://www.microsoft.com/en-us/download/details.aspx?id=36517
If it still does not work, you could post the question on Office365 forum for assistance. Thank you for your understanding.
http://community.office365.com/en-us/f/166.aspx
Best regards,
Eric -
How to create multiple sip trunks between cucm and cisco unified sip proxy
Dear Expert,
Is there a way to create multiple sip trunks between CUCM and Cisco Unified SIP Proxy (CUSP)? How to achieve it without creating multiple IP interfaces on the CUSP module.
CUCM: 8.5.1.10000-9
CUSP: 8.5.2
Thank you,
.wanHello Michael,
This SIP trunk is part of UCCE solution, which used between CVP, CUSP, and CUCM.
The requirements:
1) To have different codecs for different type of calls, as the phones are at few countries
2) To pass different number of digits from CUSP to CUCM for different call treatments
.wan -
Skype connect and bye more credit
I already paid 2 times 10 Euro each time through paypal. And in member features - I have 20 Euro Skype credit. But, when I attempt to do anything with Skype connect - my credit is 0, and I am again prompted to by more credit. But I do not need any skype services. I just need to see how Skype connect can be setup.
It is simple. For me, my country etc - there are many much better much cheaper services, and Skype can't offer nothing to beat them. But my client wants to use Skype Connect, and I have to solve his problems.
Unfortunately up to this moment, I see - Bye more credit, Bye more credit, and even I bought 2 times 10 Euro more credit, in Skype connect I have 0 credit and Bye more credit.
Unfortunately I can't see how I can send direct question to support, with all data of "bought" credit etc...
I was ready to pay something to setup Skype Connect, but it seems I am just paying to get nothing.This is a killer for professional users. I postponed my plans to get / promote a business account until Microsoft gets its act together.
Maybe you are looking for
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After upgrade to Maverick bootcamp doesn't work
Hi I bought my computer on january 2011 it's a macbook pro and the version was 10.6.8 for all this time, yesterday I decided to upgrade to maverick because my computer was very slow and stoping all the time. So far is very good but I'm trying to put
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I had a warning while opening photoshop it stated I had to remove the program and reinstall it would
not open with out reinstalling. so I did it and now the program will not download at all...I am without the whole adobe creative suite 3 master collection. I called the help line and they are saying no info on the problem since it is old. Does anyon
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How to display XML content in a JSP
Hi, can anyone help me in displaying xml content in a JSP?
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0valstckval - not in update rules
I inserted 0VALSTCKVAL in a communication structure / transfer structure; however, when I link this to an infocube (also containing this InfoObject) then 0VALSTCKVAL is not present in the update rules. Can anyone explain me why not?
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Hi, I am still using Flash 2004, and I just wondered what effect it would have if I upgraded to the new Flash Player 8? Would it cause any problems in running my .swf movies, or would it improve the playback quality? Thx.