[Solved] bitlbee/skype connection trouble
Hi all, been trying to solve this problem for a while now but to no avail, there doesn't appear to be much out there that offers any hints for where to look. I'm trying to get bitlbee, irssi and Skype working together, but every time I try to connect my Skype account, I just get connection time-outs or messages telling me it's unable to connect:
22:15 -!- mode/&bitlbee [-C] by root
22:15 <@root> Password accepted, settings and accounts loaded
22:15 <@root> Trying to get all accounts connected...
22:15 <@root> skype - Logging in: Connecting
22:15 -!- mode/&bitlbee [+C] by root
22:17 <@root> skype - Login error: Connection timeout
22:17 <@root> skype - Logging in: Signing off..
22:17 <@root> skype - Logging in: Reconnecting in 5 seconds...
22:25 -!- mode/&bitlbee [-C] by root
22:25 <@root> Password accepted, settings and accounts loaded
22:25 <@root> Trying to get all accounts connected...
22:25 <@root> skype - Logging in: Connecting
22:25 <@root> skype - Login error: Could not connect to server
22:25 <@root> skype - Logging in: Signing off..
22:25 <@root> skype - Logging in: Reconnecting in 5 seconds..
22:25 -!- mode/&bitlbee [+C] by root
22:25 <@root> skype - Logging in: Connecting
22:25 <@root> skype - Login error: Could not connect to server
22:25 <@root> skype - Logging in: Signing off..
22:25 <@root> skype - Logging in: Reconnecting in 15 seconds..
22:26 <@root> skype - Logging in: Connecting
22:26 <@root> skype - Login error: Could not connect to server
22:26 <@root> skype - Logging in: Signing off..
22:26 <@root> skype - Logging in: Reconnecting in 45 seconds..
Now, I have skyped set up and running on the machine, skype4py is authorised in the Public API tab of Skype, using the default port etc. but whether I connect from a remote machine or the same one, it just times out. Strangely, when I try to run skyped -l for logging, it just shows me the help message. As far as I can tell, it has executed just fine as it gives no errors. The username is correct, as is the password I'm using. I've generated the necessary certificates, as per this guide.
$ skyped
skyped is started on port 2727, pid: 635
I can't find any log files to check for this so I'm a little bit lost as to what to do next. I can use bitlbee to connect to MSN no problem, so I don't think that's the issue. I imagine it's to do with skyped, but I don't really know how to check, so if anyone could provide any advice on the matter I would appreciate it. Even just a pointer in the right direction would help immensely. Hopefully one of my fellow Archers has this working.
Last edited by JHeaton (2012-04-01 15:16:36)
Hi pshevtsov, thank you for the reply. I checked and although skyped was starting Skype, it was not continuing to run so I tried it with --nofork as you said and I got this error message (this also appears to be the error shown in the latest comment on the AUR page for skype4py).
[joel@glitch ~]$ skyped -n
Traceback (most recent call last):
File "/usr/bin/skyped", line 120, in skype_idle_handler
skype.skype.SendCommand(c)
File "/usr/lib/python2.7/site-packages/Skype4Py/skype.py", line 778, in SendCommand
self._Api.send_command(Command)
File "/usr/lib/python2.7/site-packages/Skype4Py/api/posix_dbus.py", line 207, in send_command
result = self.skype_out.Invoke(cmd)
AttributeError: 'NoneType' object has no attribute 'Invoke'
Exiting.
[joel@glitch ~]$
I don't really know enough Python to work out how to sort this issue out and can't seem to find anything on the Sourceforge project page in terms of troubleshooting.
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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"
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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
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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
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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.
-
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.
-
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 -
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??? -
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
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