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
Andy
You 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
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 -
DUALphone 3088 extension numbers problem
Hi!
We’ve bought DUALphone 3088 some days ago. We planned to use it in our virtual dial exchange (we have a virtual phone number that can be forwarded to landline phones, mobiles and SKYPE also).
The main thing is that this device must be used as main phone in our office (in Skype mode only). So when somebody of us receives call he must be able to redirect it to another employee using extension number (like ##5 in tone mode). And we were disappointed that we can’t do that! We can’t dial extension numbers when WE RECEIVE skype call. Skype on PC allows to do that. But not this device.
Nevertheless we CAN dial extension numbers when WE MAKE skype call using this device.
So this is firmware error…
Can anybody tell how ask the producer to make an update and fix this problem?same here in spain.
I am a wireless network operator, with a lot of skype customers.. Oddly those using PC's are unaffected, but myself and a few others using dualphones cannot connect.
As network manager I am able to prioritise skype, inspect packets etc and Nat across to any number of public IP's.
I can confirm that its not this network at fault nor any of my public IP's. -
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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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 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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What do calls actually cost on SKYPE connect? .80...
It says the basic charge for calls is .8 cents a minute. Then it says you can buy a 5000 minute bundle for $30. That makes no sense. .8 cents a minute is .80 EIGHTY CENTS a minute. That would mean 5000 minutes at 80 cents would be $400.
Then it gives an example of 30,000 minutes costing $171. or .57 cents per minute if you used all 30,000 minutes. Again this makes no sense what so ever. 30,000 minutes at .57 cents is $17,100.
Could it be that what you meant to say is that calls are billed at .08 cents? or .057 cents?
This would make more sense. I am certainly not signing up for something that costs 80 cents a minute. I would think that a company as large as SKYPE would know enough not to make such stupid representations as .57 cents vs. .057 cents. So which is it?
Also if I only want to use SKYPE connect on 1 line for outbound calls only, do I need an inbound phone number?
I want to use an automated dialer with a taped message. I am a dentist, and I have over 7500 patients. I need to advise all of them that I have moved my office location, and I just can't make 7500 calls by hand.
Thank you!
DocYou need to revisit your English.
If it said .8 of a cent, that would make sense. Because .8 of a CENT that would be 8/10ths of a single penny or 80% of a penny or .8. However .8 CENTS, plural is 80 CENTS!
You even said, "eight tenths of a cent per minute." You did't say, "eight tenths of a CENTS per minute" did you?
You can't call 1 PENNY, "CENTS" that is the plural of CENT indicating you are talking about more than 1 penny. You don't say, "I only have 1 cents." You don't say, "I only have 1 pennys." You say I only have 1 cent, I only have 1 penny, but tomorrow I will have 2 cents.
I have a PhD, but you don't need one for simple English.
The information is written poorly.
DocLasVegas -
In iPhoto '11 I cannot change the book settings. For example, adding page numbers. There is no "option" button as is suggested by the help item. How can I change to book settings?
The "options" button should appear at the bottom of the iPhoto window, when you view your book, then the Book Settings Button should be available and you can activate page numbers.
But remember, that the settings available will depend on the Book Theme you selected. The above shows the settings for one of the "Travel" themes. In a Picture Book Theme, e.g. you cannot add page numbers.
What theme and book size are you using?
Regards
Léonie -
How to have my dial up connection added to the network utility program?
how to have my dial up connection added to the network utility program for monitoring? though i am connected by dial-up, the connection is not monitored by the utility program. the drop down box there does not show the modem at all. ?
Huh? I don't understand what you are asking. I just pulled up the network utility and I don't see what kind of connection monitoring it does for any kind of connection? What kind of information are you looking to obtain? What problem are you interested in solving by having your question answered?
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Adding Extension field in B2B Application - From Catalog to Basket
Hi,
I am working on B2B application CRM 7.0.
My Client requirement is as follows:
User can add only one item from the catalog to the basket at a time.
Multiple products can be added to the catalog (one after the another).
While adding the product from catalog, user needs to enter some text corresponding to a line item.
This text needs to appear in the line item basket and same needs to be saved in the item order level.
As per the Dev Extension guide, extra fields can be saved using the "item.addExtensiondata(obj,obj)" method at the object level. I am adding this in customerExitCreateBasketTransferItem() method corresponds to "com.sap.isa.catalog.actions.AddToBasketAction" .
But, when i go to basket(order.jsp) "item.getExtensionData(obj)" returning the null value.
Please let me know where i need to add this extension.
-Chandra.There are two possibilities:
After adding extension data to any standard ISA object, if there is a backend roundtrip - that is, if that happens to be used in the ISA BAdI (the header and item) BadIs, you have to watch for your extensions carefully. They have to be read in the backend set method in the BAdI, persist them and send them back in the get methods to the Java layer.
Another reason could be, the you are adding the extension to one object and trying to read it in another. -
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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Does anyone have any experience with iphone when incorporating extension numbers in work phone numbers?
Do you mean to autodial it from contacts? When putting in a number for a contact, if you hit the bottom, left key (the + * # key) you change the keyboard and have the option of entering a pause (~2 seconds long) or a hard wait (in which case it will pause until you hit a key to continue). Just put the extension after the pause or wait. If using pauses to have it autodial, if one pause turns out to not be long enough, just use as many as you need.
If dialing in phone app, press and hold the "*' for a soft pause, or press and hold the "#" for a hard pause (ie. a wait).
P.S. The "wait" option might be new to iOS 5. I know it was not there in previous versions, but not sure when it showed up, in 4.3.5 or only with iOS 5? If you are running something older than iOS 5 though, you may only have the option of the ~2s pause. -
ADDING WEEK NUMBERS IN THE CALENDAR OF IPHONE
Hi,
haven't found option of adding week numbers on my iphone 3gs calendar. in working life week numbers are pretty important and would need the option on daily basis. i syncronise my calendar from office outlook / exchange. office supports the feature, but week numbers dont follow.
any ideas anyone?!I use Google calendar and sync it to my iPhone calendar over the air via Google Sync. There are a number of "gadgets" or sub-calendars you can add to your Google calendar (i.e. Moon phases, contacts birthdays, etc.) that will sync over to iPhone. I searched for week numbers and sure enough there is a "gadget" that will add this functionality with an option of weeks starting on Sundays or Mondays. Though I've never used them, I now have week numbers on my iPhone calendar.
You would have to set up a free gmail account to create a google calendar but you wouldn't need to use it (mail or calendar) except to sync week numbers if you don't want to. If you want to go through the (relatively) small amount of trouble to do this just start a gmail account and then follow these instructions: http://www.google.com/support/mobile/bin/answer.py?answer=138740&topic=14252
Except of course you will only turn on calendar. Then (on your MAC) go to your gmail and click on 'calendar' in the upper left. From there click on 'settings' in the upper right, then on the 'labs' tab. Enable the option to 'Add any gadget by URL.' Save this setting then go back to the calender itself and on the right where it says 'Gadget URL' paste http://sites.google.com/site/gcalweeknumbers/Home/calendar-gadget.xml
Then, on your phone's web browser go to http://m.google.com/sync and sign in with your gmail info. Bookmark this page. Select your device and check the boxes for 'My Calendars' and 'Week Numbers'. Now when you open the calendar app on iPhone tap 'calendars' on the upper left and you should see (yourname)@gmail.com (Exchange) with 'Week Numbers' listed below it. Make sure you are viewing 'All Calendars' within the app and week numbers will show up on the Sunday or Monday of each week.
I apologize if these directions are not clear enough or too wordy. I tried to describe it for someone without much experience with this stuff (meaning myself). It may seem like a lot of trouble, but if you really want week numbers you can do it! -
Extension numbers not synchronizing with CUCM
Hello everybody.
I'm having the following issue with synchronism as you can see on the tittle above. The client also says that the extension numbers are not updating according to the Active Directory.
I went to "System/LDAP/LDAP System" in the CUCM and the "Enabel synchronizing from LDAP Server" box is checked.
What could be the problem? What should I do?
Any help is really appreciated.
Regards.
PatricioHay un foro para español, si deseas preguntar y que te respondan en español, te recomiendo que postees ahi.
El idioma principal aqui es ingles, asi todo mundo entiende las preguntas y respuestas.
Estas sincronizando el campo que contiene esa informacion en CUCM???
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk -
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.
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