VCS-E unable to make outgoing call with search rule not found error
Hi
I deploy telepresence device registered to CUCM and connect with VCS-C and VCS-E. The issue now is I am not able to make outgoing call and my search rule show not found, request time out.
I use DNS zone to make the outgoing call and my VCS-E is able to resolve DNS SRV record. However search rule result show not found.
Any advise are appreciated.
Hi
I managed to make a call to address that I am trying to reach. But it supposed to come up with PIN number to enter instead I saw the message that no incoming video.
Can you please advise what could be the issue.
tvcs: Event="Call Connected" Service="SIP" Src-ip="10.84.83.101" Src-port="27224" Src-alias-type="SIP" Src-alias="sip:[email protected]" Dst-alias-type="SIP" Dst-alias="sip:[email protected]" Call-serial-number="24a63c27-42b4-4950-ba02-6a548b279a09" Tag="f060faef-441f-47a2-9c75-e1b0623b5e14" Protocol="TCP" Call-routed="YES" Level="1" UTCTime="2015-02-25 14:38:32,604"
Similar Messages
-
i am having bugs with the ios 5, after it got the update it is unable to make outgoing calls. in-place the incoming and outgoing texts are working please help me asap...!
Sometimes and this is network dependant if they suspect the phone to be lost or stolen as in this case with change of Sim card and provider then the origonal network can and some will block the phone untill you have rang them and proven it's not the case or if you have bough this 2nd hand then the origonal seller may have stopped paying the contract bill and thus the phone is blocked
-
HT1937 i am unable to make outgoing calls
I am unable to make outgoing calls, my carrier is a MTN and i have recently bought the phone
You are posting in a 14 month old thread.
Parampal Singh wrote:
I updated device after that it happend 2/3 days was good
See the Thread YOU started where assistance is being offered for YOUR issue.
https://discussions.apple.com/message/22414503#22414503 -
PAP2T-NA will not make outgoing calls with Asterisk
I have a PAP2T-NA connected to a Uniden cordless phone and an Asterisk server with working trunks. Inbound calls work fine - the phone rings and sound works both ways. However I am unable to make outbound calls through the PAP2T, even to other local extensions or just to voice mail. After dialing an extension, I hear a short pause and then a fast busy signal.
Dial plan is (*xx|x.). This is all I see with syslog when calling extension 1000 through sip1 (* server):
Feb 26 08:13:51 LinksysPAP [0]Off Hook
Feb 26 08:13:56 LinksysPAP Calling:1000@sip1:0
Feb 26 08:13:56 LinksysPAP [0:0]AUD ALLOC CALL (port=16440)
Feb 26 08:13:56 LinksysPAP [0:0]RTP Rx Up
Feb 26 08:13:56 LinksysPAP RSE_DEBUG: reference domain:sip1
Feb 26 08:13:57 LinksysPAP RSE_DEBUG: reference domain:sip1
Feb 26 08:13:57 LinksysPAP [0:0]AUD Rel Call
Feb 26 08:13:57 LinksysPAP CC:Failed w/ Calling
Feb 26 08:13:59 LinksysPAP [0]On Hook
This is all I see in the * log:
Feb 26 08:16:12 DEBUG[29924] acl.c: ##### Testing 192.168.1.179 with 192.168.1.0Feb 26 08:16:12 DEBUG[29924] chan_sip.c: Setting NAT on RTP to 0
Feb 26 08:16:12 DEBUG[29924] chan_sip.c: Stopping retransmission on '[email protected]' of Response 101: Match Found
Feb 26 08:16:12 DEBUG[29924] chan_sip.c: Setting NAT on RTP to 0
Feb 26 08:16:12 DEBUG[29924] chan_sip.c: Checking SIP call limits for device 100Feb 26 08:16:12 DEBUG[29924] chan_sip.c: Stopping retransmission on '[email protected]' of Response 102: Match Found
I've tried various codecs and regional settings to no effect. Setting an outbound proxy makes no change. Is there something simple I'm missing? Thanks!HI.....
Is your voice provider Vonage then Vonage Supports 7-,10- and 11- digit dialing.Use 7-,10- or 11- digit dialing for calls within the same area code as your Vonage phone number. Use 10- or 11- digit dialing for calls outside of your area code.
Also forward the ports 53, 69, 5060-5061and 10000-20000 for the adapter IP address. -
I have an IPhone 4 , which is on a Vodafone network, when I try and make outgoing calls it says 'call failed' . I am still able to send/receive SMS and IM and receiving calls is also so problem, does anyone have any solutions?
You need to contact Vodafone.
-
Unable to Make a Call with Active Subscription | I...
Hi,
I Have an active subscription for India 2500 mins which starts on 16th of every month
Previous month i exhausted my 2500 min on 6th may itself, so i opted for anther 800 min subscription and made call uptill yesterday 15th may (800min also over yesterday morning)
but now the condition is like this that the payment of 2500min subscription has taken place with autorechrge on 13th may 2015 itself. but i am unable to make call to india. whenever i try it shows insufficient credit.
while the subscription page show this:
"Your subscription has been suspended as you have exceeded your allocated free minutes. Your free minutes will start again on May 16, 2015."
but today is 16 may 2015, then why the call is not being connected
Help PleaseI think this could be a prompt from the Flash Player plugin. To adjust your microphone and camera permissions for Flash, you can use the following page:
* General setting: http://www.macromedia.com/support/documentation/en/flashplayer/help/settings_manager02.html
* Site-specific settings: http://www.macromedia.com/support/documentation/en/flashplayer/help/settings_manager06.html
Note: Macromedia was the original developer of Flash before being acquired by Adobe.
Does that fix it? -
I cannot make outgoing call with uc500.
please help
i have uc500 router. i have conifured my dial peers very simply . i can receive the call from outside . but i cannot make a call. whenever i am trying to make a call.
just Ring Out is appearing with starting buy tone or some time it going telecom company. this is my simple dial peers configuration
dial-peer voice 50 pots
description long distance
preference 5
destination-pattern 9T
port 0/1/2
forward-digits all
no sip-register
dial-peer voice 51 pots
destination-pattern 9[2-9].........
port 0/1/2
forward-digits allshakirullah,
Please collect below debugs from uc500 and attach it here.
Router(config)# logging buffer 5000000
Router#debug voip ccapi inout
Router#debug vpm signal
Router#debug voip vtsp default
Router#debug voip vtsp session
Router#Clear log
Now make one test outgoing call and re-create the issue.
Router# undebug all --->turn off debugs
Router# show log----->collect the entire output of this and attach it here
Mention calling and called numbers.
-Abhi -
PAP2T unable to make outgoing calls
I recive incomming calls, no problem.
When I pick up the phone to make an out going call I hear the dial tone.
When I dial any number I am told the number is not available, or the phone sounds the engaged tone.
I use my SIP with X-Lite V3 and have no problems making outgoing calls.
Please let me know your thoughts.
Kind Regards,
Jasonreset all settings
settings-general-reset-reset all settings
now reconnect to wifi
settings- wifi- click network name- enter password - join
if issue persists back up and restore as new via iTunes
Peace, Clyde -
IPhone unable to make outgoing calls after 2.2.1 update
After updating to the 2.2.1 firmware I am unable to make calls with my iPhone. I have reset it with the backup I made and as a fresh backup and the problem has still not been resolved. The problem only began after I updated. Help?
i had exactly the same problem, i had to send it back to apple via O2 and they had to give me a new iPhone, now ive got one that works on 2.1 im not updating it till the next update comes out!
so yeah take it to your nearest O2 store and 3 weeks later everything should be sorted -
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 -
Ios 8.0.2 unable to make outgoing calls
I recently updated to ios 8.0.2 when I click on the phone button to see my recent calls, voicemail, or to dial a number the screen opens and immediately closes again. I have restarted my phone but it didn't work either. Any help?
i can't make calls on my iPhone 5 with iOS 8.0.2
I can receive, but not place calls with my phone after updating from iOS 7.1, to iOS 8.0.2. The green dial button becomes a darker shade when I press it, but then reverts back to bright green when I let go. Calling numbers or contact with Siri doesn't work either. not sure what to do... -
UC520 SIP trunk unable to make outgoing calls, incoming calls are ok
I have an new SIP trunk set on an UC520 and the incoming calls are ok, but the outgoing calls are getting an busy tone(not working).
The bellow trace is showing that the cause is "No route to destination (3) ". The question is this route has to do with the firewall(ip routing) or with the voice translation rules?
001866: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x843CE50C
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 0777777777 <- main sip number
Called Number : 0888888888 <- called number
Source IP Address (Sig ): 0.0.0.0
Destn SIP Req Addr:Port : 0.0.0.0:0
Destn SIP Resp Addr:Port : 0.0.0.0:0
Destination Name :
001867: //3439/91242E51926F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 3
Disconnect Cause (SIP) : 200Hi Emil,
I've added bind control and media interface but outgoing calls are the same blocked, strange thing is that the cause is still no route to destination (3)
but
UC_520#show sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): ENABLED 10.10.10.5 <- fa0/0 IP address
SIP User Agent bind status(media): ENABLED 10.10.10.5 <- fa0/0 IP address
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio image
Network types supported: IN
Address types supported: IP4
Transport types supported: RTP/AVP udptl -
Changed my #, now unable to make outgoing calls or text
I recently changed my phone number (because I moved to another state) and now am unable to make calls and text. After changing my number online, I dialed *228 and option 1. It said my phone was now active, but when I try to call, I get "your mobile member is not active or is invalid".
I changed my number again via MyVerizon online (since it is free), in hopes this would correct any issues.
Turned phone off and on a bunch of times like it instructed me to do (I have an iphone 4 16GB).
I am still receiving this message. "your mobile member is not active or is invalid". What should I do. Thanks.
G-Problem fixed itself 4 hours later, go figure. Shrugs @ Verizon. Working now.
-
Content Browser (0BCT_CB_1) load fails with conversation id not found error
Hi,
I am suddenly experiencing a problem with loading data to the content browser cube using infosource 0BCT_CB_1. The load fails with an error saying: "Conversation 76635612 not found / CPIC-CALL: 'ThSA". Does anybody know how to solve this?Hi,
Were you able to get this fixed?
I am getting same issue while loading to BI 7.0 technical content.
Thanks
SA -
Failing Connecting PHP to MySQL with HTTP 404 NOT FOUND Error
I'm using Dreamweaver CC to connect php to mySQL, and read both of these articles but still not working!
1. It keep showing me HTTP 404 NOT FOUND. But I can access MySQL by web browser so I'm sure the address and login info is correct.
2. There is no MMHTTPDB.php in my project folder.
MMHTTPDB.php not creating
http://forums.adobe.com/thread/1238828?tstart=0
MySQL connection
http://forums.adobe.com/thread/1239068?tstart=0
Did I miss something?Thanks anyway but it is same as the thread I post below and above.
MySQL connection
Maybe you are looking for
-
What the 'f'low is wrong with this !
hi, i installed 1.5 from 10g CD created a DAD per instructions in the howtos to in 9iAS setup ! by the way that doc needs to be updated where it talks about wdbsvr.app (http://www.oracle.com/technology/products/database/htmldb/howtos/howto_use9ir2ohs
-
How to get info about available Toshiba updates?
I have noticed that this has taken over from the online product information button. I was wondering how do I know now when there is an update availabe for Toshiba specific software and hardware?
-
Library consolidation does not work
In itunes/edit/preferences/ advanced, I changed the location of the itunes media folder location. I then followed the instruction File/Library/consolidate library. I get an error message that a duplicate file exists. I have eliminated all duplicat
-
Hi there. I'm hoping one of the mods picks this up as from reading some other posts, it seems that it is the quickest way to get this type of problem sorted and lots of people seem to have had the same issue. My broadband activation date was 13 Dece
-
Af:query individual field(attribute) customization
Hi All, I have a question regarding af:query feature of ADF. Is there any way to customize individual field / column of af:query search form in ADF? In the source, it does not show individual columns and I want to add custom facets on one of the colu