Skype Connection
I have tried installing Skype through every means possible - synaptic, website, old versions etc. but Skype will not connect.
When I try the normal method of contacting it states that the Chat Operator has opened in a new window but THERE IS NO NEW WINDOW.
Maybe all the Linux Users should band together and start a Class action against Microsoft for deliberately excluding Linux Users from Skype
I don't use iSkoot, but it sounds like a permissions problem to me. Try changing the application's settings (Options->Advanced options->Applications) to something moree open.
Under "Connections" it might say "Custom"'. You probably need to change to "Allow"
Similar Messages
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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 -
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
insecure=invite
qualify=yes
allow=alaw
allow=ulaw
amaflags=default
trustrpid=no
sendrpid=yes
context=from-trunk-sip-Skype_out
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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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.
I have been in 3 separate Chats with Skype Business support and they have not be able to give me any help as to what is going on.Each channel carries one phone call. So if you have 4 channels, you can carry 4 simultaneous phone calls, in any combination of incoming or outgoing.
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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 -
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 -
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.
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Hi,
I'm new to skype so I don't understand completely how it works, me and my son who lives in Northern Virginia and I live in VA. Beach, we are tying to set-up skype with video on the computer, well I am able to see my self on my camera, but when I try to connect he gets a message that says person can only receive IM messages.Hi,
I have a problem with skype connection, for more than a week now. Every time when I sign in to my account I get a warning message "Skype home is unavailable at the moment. Check back later to see you news and alerts" I can't make any kind of calls and still my credits are enought what can i do? It like Iam always of line why??? -
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 -
How to configure Lync-Skype connectivity
i have configured Lync edge with single public IP without reverse proxy and my external users can connect it.. now my next task is to provide the lync-skype connectivity for which i need to configure federation services.. i have an extra Public IP and
can create a public fed SIP SRV record and map to my new Public IP. what other changes are required on server end for implementation of federation services ?Hi babarmunir,
Is there any update on the federation issue?
Just for your reference, here is another case explains the SRV record for Lync federation:
http://social.technet.microsoft.com/Forums/lync/en-US/7a6ce1c5-5c0b-45c2-9c1d-732446743fee/federation-srv-records
Kent Huang
TechNet Community Support -
Belgacom BBox 2 and Skype Connect
Hello to all,
I need some help
1st my configuration
1 Belgacom BBox2 (with 2 pst connextion linked to sip account)
1 Belgacom Forum 3000 (with 1 pstn incomming line)
2nd what i want to do
I want to configure a skype connect account into my bbox2 and connecte the bbox 2 pstn port to the incoming port of the Forum3000 so i can recive incomming calls by skype and callers can skype us for free.
3rd my question
Can somone help me to configure the skype connect sip account into the bbox?
Tks to all for helpNevermind. I found a solution on a Forum. I tried downloading Skype for Android on my computer. I was asked for my phone number and a link to download Skype was sent on my phone. I accessed it and Skype start installing (have no idea what was installed, anyway). I start Skype and it is working fine.
The most interesting thing is that I still have the same Skype version (5.1.0.57240) but it is localized in Romanian.
Best regards,
Sebastian -
Hi,
Could you please any give me more information about VoIP gateway, and how to connect it with PBX and how to buy it and what is the price?
Thanks,
Mohamed ElDiebHello ToshibaJoe,
If your PBX doesn’t support SIP check with your manufacturer to see whether a SIP gateway, module or upgrade is available for your existing PBX, that would enable you to use Skype Connect/SIP.
Also, it is highly recommended that a Skype Connect certified device is used, to gaurantee all features and functions work. Following is a link to that lists all Skype Connect certified products:
http://www.tekvizionlabs.com/3rdpartyprograms/skype/skype_verified_products.php
Please let us know if you have further questions.
Thank you,
MariaA
Skype Enterprise Support -
Skype connect 3cx phonesystem - dtmf problem
Dear community members,
i've run into a brick wall. im using 3cx phonesystem, with skype connect mapped on a single channel. My problem is on dialing the skype in number 3cx phone system detects the call and forwards to the one of the extensions, but the problem is the dtmf doesnt gets transmitted, im using an ivr application and the same application is working fine if the call is forwarded from an isdn gateway.
As per the technical details, i think that skype connects forwards dtmf tones in the standard rfc format.
Please help as this is a urgent matter for us and we need to get it working.
P.S. : My server hardware doesnt have a sound card installed.I am sorry to say I have no answer for you, but it sounds like you are further than I am in the 3cx configuration. Would you mind posting some of the steps you had to do to get 3cx to receive calls from Skype? I
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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 - adding extension numbers to large ...
Hi
We are looking at Skype Connect to provide access to all our workers via their desk extensions. we have sucessfully trialled a number of users by manually adding account details and extension numbers.
The problem is the next step of rolling out to all users. i can see i can create the business accounts via CSV file, but i cannot see a method of adding the extension numbers to specific users.
do i have to do this task manually or am i missing a trick.
Any advice out there?
Many Thanks
AndyYou should look at this site:
Raising
Gaimee
The variables rdinfoCurrentSlide (current slide) and
rdinfoSlideCount (total slides) are what you're looking for, and by
downloading the FLA-file from the link above, you'll get this.
Export the FLA to SWF, import it as an animation to be visible
throughout the CP-presentation.
I think that's what you're looking for. ;)
*EDIT* Damn you, excorpy. :P
Maybe you are looking for
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Could not lookup message because there is no database connection
I am getting this error message when I try to run test_fwktutorial.jsp page. This happens for my custom Jdeveloper project also in which I am trying to build a brand new page. I did define the Run Time connection properly as following: DBC file: D:\o
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"Can't connect to local MySQL server through socket '/tmp/mysql.sock'"
Data Services=3.1 Repository=12.2.2.0000 Red Hat Enterprise 5 Designer,Job Server,Job Engine=12.2.2.3 After an unscheduled server reboot with DS up and running when trying to start a job in either Data Services Management Console or DS Designer getti
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Dear All, Can we remove the message-id from the auto-reply? in an auto-reply the message bounces back to the user including in the body part the following: Message-id: <[email protected]> Date: Thu, 31 Jul 2008 21:36:08 +0300 From: [email protected]
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Creating Photo Albums on iPhone
Can anyone explain how I create photo albums on the iPhone? And then how to move pics to different albums? Thanks in advance!
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How do I restore keywords to the keyword HUD? I deleted all keywords from the HUD, edited the keyword list and imported the edited list. I assumed that I could repopulate the keywords by updating the metadata from master (these are referenced files),