Initiating a SIP call via Hyperlink
So we're using lync 2013, and I would like to be able to send out a hyperlink in an email that will actually initiate a SIP video call once clicked. Just so everyone understands the use case is wanting to dial into a cloud based service video service
that interworks h.323/SIP standards based systems with Lync
I can use:
Sip:[email protected] as a hyperlink and that will bring up the presence of the SIP contact since as we're federated with there domain. Ultimately it would be nice if it initiated the call just like clicking "online meeting"
This is the code that the join online meeting uses: conf:sip:https://meet.example.com/user/7322994 and I tried replacing the URL with a SIP address with no go, because it looks like it tries to find a meeting that is supposed to happen on the AVMCU.
So ultimately if there is a way to have a hyperlink, so Lync will initiate a SIP video call that would be great.
So we're using lync 2013, and I would like to be able to send out a hyperlink in an email that will actually initiate a SIP video call. Just so everyone understands the use case is wanting to dial into a cloud based service video service that interworks
h.323/SIP standards based systems with Lync
I can use:
Sip:[email protected] as a hyperlink and that will bring up the presence of the SIP contact since as we're federated with there domain. Ultimately it would be nice if it initiated the call just like clicking "online meeting"
This is the code that the join online meeting uses: conf:sip:https://meet.example.com/user/7322994 and I tried replacing the URL with a SIP address with no go, because it looks like it tries to find a meeting that is supposed to happen on the AVMCU.
So ultimately if there is a way to have a hyperlink, so Lync will initiate a SIP video call that would be great.
Similar Messages
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Items on page not set when called via hyperlink from charts
hello anyone who can help me on this thorny issue,
i have a barchart on one page (page 39), where if you click on the bar chart, page 40 opens and variables are passed on to items on page 40. the variables depend on which bar was clicked. looks somethiing like this:
select 'F?P=102:40:'||:APP_SESSION||'::NO::P40_DATE,P40_JAHR:'||to_char(datum, 'yyyy.mm.dd')||','||to_char(datum, 'yyyy') link,
to_char(datum, 'dd.mm.yyyy') Datum,
Avg_mips
from ... some sql query....
this generally works - BUT, once in awhile, the correct page opens, but the items are NULL!
only when i press F5 to refresh the page, do the variables correctly pass to the items and the proper query is performed...
after some testing, it seems to happen when i click quite fast. quickly get back to the page 39, click a new bar and then i'm back at page 40 again.
if i do this slowly, maybe wait a couple of seconds, it generally works.
for ex., the URL would look something like this:
http://daidalos:7777/pls/apex/F?P=102:40:2118535974571692::NO::P40_DATE,P40_JAHR:2008.12.16,2008
- i know the variables are correct when i look at the URL
- yet the new graph on page 40 shows null values, and a click on the "session" button confirms that the items are null!
- once i press F5, the URL is the exact same, onle the items are now "refreshed" and the proper processing can take place.
so, can anyone tell me what is going on here and how i can avoid this?
the items are set as "hidden" (not protected). there are no computations or default settings to the items, and they are not reset to null at any time.
regards,
mikeMike:
I suspect that the chart on page 40 is being rendered by an APEX database connection that is not the same one that renders the rest of the stuff on page 40. So, it is possible that the database connection rendering the chart does not see the new values for the page items set in the URL as these may not have been committed to the database yet.
You could try this.
Change the chart query to be
select 'javascript:setStuff("' || to_char(datum, 'yyyy.mm.dd') || '","' ||to_char(datum, 'yyyy') || '")' link,
to_char(datum, 'dd.mm.yyyy') Datum,
Avg_mips
from ... some sql query...Add the JS below into the HRML page header<script>
function setStuff(p_date,p_jahr) {
$x('P40_DATE').value=p_date
$x('P40_JAHR').value=p_jahr
doSubmit('XXXX');
</script>Define an 'After Submit...' branch to page 40. Make the branch conditional on Request = 'XXXX'
varad -
ILBC calls via SIP Trunk with CUBE and CUCM7
hi there,
our SIP Provider offers the IBLC codec which promises to provide better quality compard to G.729.
I'm using this scenario:
IP-Phone(G711) --- CUCM7 --- (SIP-Trunk1) --- CUBE --- (SIP-Trunk2) --- Provider
Everything workes unless I'm configuring IBLC at the provider and on trunk2.
I have the CUBE router acting as a trancoding device and also specified IBLC as codec to be handled.
SIP trunk 2 was placed in a region with IBLC as codec.
On the trunk configuration in CUCM the media ressource group with XCODE capability is configured
Transcoding workes between two IP Phones in different regions with different codecs within the intranet.
Unfortunately the CUBE router doesn't seem to use the transcoder to change internal G711u calls into IBLC codec
so calls are blocked by the CUBE device:
deb ccsip calls
for incoming call:
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4AE7AC98
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0237892992
Called Number : 036677725231
Source IP Address (Sig ): 10.100.100.50
Destn SIP Req Addr:Port : <IP SIP Provicer>
Destn SIP Resp Addr:Port : <IP SIP Provicer>:5060
Destination Name : <IP SIP Provicer>
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : ilbc
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): <IP CUBE>
Source IP Port (Media): 0
Destn IP Address (Media): <IP SIP Provicer>
Destn IP Port (Media): 22022
Orig Destn IP Address:Port (Media): [ - ]:0
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 488
(Output lookes similar to outgoing calls)
I set up ccm on cube and assigned dsp ressources without success:
Here are the relevant configuration parts:
voice class codec 1
codec preference 1 iblc
voice service voip
address-hiding
allow-connections sip to sip
allow-connections h323 to sip
allow-connections sip to h323
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
h323
sip
header-passing error-passthru
no update-callerid
midcall-signaling passthru
privacy-policy passthru
voice-card 0
dspfarm
dsp services dspfarm
dial-peer voice 40991 voip
description *** Incoming from SIP-Provider
destination-pattern 03667772523.%
session protocol sipv2
session target ipv4:<IP_of_CUCM>
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
ip qos dscp cs5 media
ip qos dscp cs5 signaling
sccp local GigabitEthernet0/0
sccp ccm 10.100.100.50 identifier 11 version 4.1
sccp
sccp ccm group 11
description *** lokaler CCM fuer Codec-Konvertierung von SIP/DUS.NET
associate ccm 11 priority 1
associate profile 21 register DE_WGT_MTP02
dspfarm profile 21 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec ilbc
maximum sessions 10
associate application SCCP
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 10
sdspfarm tag 1 DE_WGT_MTP02
max-ephones 30
max-dn 30
ip source-address 10.100.100.50 port 2000
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Mar 14 2010 02:10:34
sh sccp
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
IPv4 Address: 10.100.100.50
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.100.100.50, Port Number: 2000
Priority: N/A, Version: 4.1, Identifier: 11
Trustpoint: N/A
Call Manager: 10.1.1.55, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 10
Trustpoint: N/A
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.100.100.50, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 21
Reported Max Streams: 20, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
sh dspfarm dsp all
SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED
0 1 26.3.4 UP N/A FREE xcode 1 - - -
0 1 26.3.4 UP N/A FREE xcode 1 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
0 2 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 1 - - -
1 1 26.3.4 UP N/A FREE xcode 1 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
1 1 26.3.4 UP N/A FREE xcode 2 - - -
Thanks in advance,
DavidHi there,
Just wondering whether you ever got this resolved? I seem to have a very similiar problem.
Regards
Karen -
Hi there,
I'm having problems modifying the 'Dialed Number (DN)' text box under 'Advanced Configuration->Patterns for RNA timeout on outbound SIP calls' of the SIP tab in the Cisco Unified Customer Voice Portal 8.5(1) opsconsole. In a nut shell, I need to change the RNA timeout but some reason when typing into the Dialed Number text box, the input is not taken. The reason I want to change this settings is because my ICM Rona is not working with CVP:
https://supportforums.cisco.com/thread/2031366
Thanks in advance for any help.
Carlos A Trivino
[email protected]Hello Dale,
CVP doesn't allow you to exceed the RNA more than 60 Seconds. If you want to configure the timer for DN Patterns you should do it via OPS console, It would update the sip.properties files in correct way, the above way is incorrect.
Regards,
Senthil -
Incoming sip calls are not working but outgoing is working with cme
I have CME setup with voip.ms on my 2800 router, my outgoing calls are working but my incoming calls are not. Below is my config, please let me know if it is something with my config:
voice translation-rule 3
rule 1 /^9142281\(...\)$/ /\1/
voice translation-profile INCOMING_CALL_1
translate called 3
dial-peer voice 1 voip
translation-profile incoming INCOMING_CALL_1
session protocol sipv2
session target sip-server
incoming called-number .%
voice-class codec 1
dtmf-relay rtp-nte
no vadI made the change, but I am getting no output from debug voip ccapi inout. What does concern me from debug ccsip messages is:
Aug 31 12:42:04.195: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid Host'
Via: SIP/2.0/UDP 107.6.67.238:5060;branch=z9hG4bK000d3c36;rport
From: "+19144410197" <sip:[email protected]>;tag=as7439b9c1
To: <sip:[email protected]:1061>;tag=829C8-2532
Date: Sun, 31 Aug 2014 12:42:04 GMT
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=100
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
I also am getting this:
voicertr2#debug ccsip error
SIP Call error tracing is enabled
voicertr2#
Aug 31 12:45:07.359: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr
Aug 31 12:45:07.359: //-1/78AE76E98009/SIP/Error/sact_idle_new_message_invite: Invalid URL in incoming INVITE -
Hold doesn't work with calls via Gatekeepr
Hi,
I have a CME and CM via a gatekeeper.
Both can call each other via gatekeepr or PSTN.
If PSTN call was initiated first, I can still make another call via GK while put PSTN call on-hold.
But, if call via GK was established first, can't put the call on hold to mak/answer new call.
This only happen with phones on CM.
Any idea?
Thanks,Hi Brandon,
Box was checked because the instruction I used suggested as well.
Just out of curiosity, I did unchecked and now it works fine.
Not sure what is going on.
I have CUCM 7 and CME 7.
Thanks, -
EEM - puts action fails in EEM applet/script when called via HTTP
I have an EEM script which produces some diagnostic output. The script is written to be initiated by "event manager run" and is running in sync mode. Script uses "puts" into stdout to produce the output. Everything is working fine, except when script is initiated via HTTP/HTTPS I get no output.
To reproduce the issue I have created the following applet:
event manager applet TEST1
event none sync yes
action 1 puts nonewline "SOMETEXT"
Here is output from command line (tested from console and telnet):
router#event manager run TEST1
SOMETEXT
router#
When applet is called via HTTP using http://router/level/15/exec/-/event/manager/run/TEST1/CR URL the output is empty. The test message goes to the console versus to HTTP reply.
So it looks like the stdout is not redirected to HTTP session successfully.
Any advice on why it is happening, or better yet how to fix it would be apprecaited.Thank you. This is also what I was thinking. As you probably already know I have opened a TAC case on the matter.
-
SIP Calls Drop. Receive Bye From Cube 15min,30min, 45min
Hello,
Running into an odd issue. I've seen several others having this problem with calls dropping after 15min duration. But this is a bit different. Sometimes long duration calls drop at 15min. Some at 30min, others at 45min. And sometimes not at all. Call flow is such.
8831-sip--CUCM--sip--Cube--ITSP
I'm convinced this is likely a problem with the refresh timer. But I can't explain why it wouldn't just fail only at 15min. It's also interesting to note I've only seen this on the 8831. I tried getting the issue with debugs from the cube but of course it didn't happen once I turned on ccsip message.
From the callmanager traces I see the bye arrive from cube with Reason Q.850 cause=102.
The CUCM version is 9.1.2 and cube is 15.2(4)M1. I did see some odd defect in 15.1 related to this where the refresh on the cube would send out 3 invites to the ITSP on an update. I guess it would have only 33% chance of getting it right. Any help someone could provide I'd appreciate it.Thanks for the replies.
So was able to capture it while had debugs running. This time it disconnected after an hour. Same cause=102.
Now here is where it gets interesting in the debugs. I see an invite is sent 3 seconds from callmanager. I assume this is a refresher with the same call-id. Cube receives it and sends out to ITSP. With a new call-id. We then receive a bye from ITSP cause=86. Which then of course is sent to callmanager. Here are the relevent sections of debugs.
Received from cucm to cube:
820421: May 6 09:00:42.976: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:43 2365082: Received:
2014-05-06 09:00:43 2365083: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
2014-05-06 09:00:43 2365084: Via: SIP/2.0/TCP 10.38.246.136:5060;branch=z9hG4bK28bab16dbd5664
2014-05-06 09:00:43 2365085: From: "Marcos Vazquez" <sip:[email protected]>;tag=3831180~dfbf10b3-6c69-4443-852f-cbf609935a6f-35009402
2014-05-06 09:00:43 2365086: To: <sip:[email protected]>;tag=5EBA2282-19C8
2014-05-06 09:00:43 2365087: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:43 2365088: Call-ID: [email protected]
2014-05-06 09:00:43 2365089: Supported: 100rel,timer,resource-priority,replaces
2014-05-06 09:00:43 2365090: Min-SE: 1800
2014-05-06 09:00:43 2365091: User-Agent: Cisco-CUCM9.1
2014-05-06 09:00:43 2365092: Allow: INVITE, OPTIONS, I
2014-05-06 09:00:43 2365093: NFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
2014-05-06 09:00:43 2365094: CSeq: 106 INVITE
2014-05-06 09:00:43 2365095: Max-Forwards: 70
2014-05-06 09:00:43 2365096: Expires: 300
2014-05-06 09:00:43 2365097: Allow-Events: presence, kpml
2014-05-06 09:00:43 2365098: Supported: X-cisco-srtp-fallback
2014-05-06 09:00:43 2365099: Supported: Geolocation
2014-05-06 09:00:43 2365100: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:43 2365101: Remote-Party-ID: "Marcos Vazquez" <sip:[email protected]>;party=calling;screen=yes;privacy=off
2014-05-06 09:00:43 2365102: Contact: <sip:[email protected]:5060;transport=tcp>
2014-05-06 09:00:43 2365103: Content-Type: application/sdp
2014-05-06 09:00:43 2365104: Content-Length: 371
2014-05-06 09:00:43 2365105:
2014-05-06 09:00:43 2365106: v=0
2014-05-06 09:00:43 2365107: o=CiscoSystemsCCM-
2014-05-06 09:00:43 2365108: SIP 3831180 1 IN IP4 10.38.246.136
2014-05-06 09:00:43 2365109: s=SIP Call
2014-05-06 09:00:43 2365110: c=IN IP4 10.96.5.28
2014-05-06 09:00:43 2365111: b=TIAS:64000
2014-05-06 09:00:43 2365112: b=AS:64
2014-05-06 09:00:43 2365113: t=0 0
2014-05-06 09:00:43 2365114: m=audio 31146 RTP/AVP 18 0 116 101
2014-05-06 09:00:43 2365115: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:43 2365116: a=ptime:20
2014-05-06 09:00:43 2365117: a=rtpmap:116 iLBC/8000
2014-05-06 09:00:43 2365118: a=ptime:20
2014-05-06 09:00:43 2365119: a=maxptime:60
2014-05-06 09:00:43 2365120: a=fmtp:116 mode=20
2014-05-06 09:00:43 2365121: a=rtpmap:18 G729/8000
2014-05-06 09:00:43 2365122: a=ptime:20
2014-05-06 09:00:43 2365123: a=fmtp:18 annexb=no
2014-05-06 09:00:43 2365124: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:43 2365125: a=fmtp:101 0-15
2014-05-06 09:00:43 2365126: 5820422: May 6 09:00:42.978: //3024943/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
Sent to ITSP:
Sent: Which looks like 3 are sent.
2014-05-06 09:00:43 2365128: INVITE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:43 2365129: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D699C1126
2014-05-06 09:00:43 2365130: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:43 2365131: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:43 2365132: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:43 2365133: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:43 2365134: Call-ID: [email protected]
2014-05-06 09:00:43 2365135: Supported: 100rel,timer,resource-priority,replaces,sdp-an
2014-05-06 09:00:43 2365136: at
2014-05-06 09:00:43 2365137: Min-SE: 1800
2014-05-06 09:00:43 2365138: Cisco-Guid: 3422833792-0000065536-0000746374-2297832970
2014-05-06 09:00:43 2365139: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:43 2365140: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
2014-05-06 09:00:43 2365141: CSeq: 105 INVITE
2014-05-06 09:00:43 2365142: Max-Forwards: 70
2014-05-06 09:00:43 2365143: Timestamp: 1399388442
2014-05-06 09:00:43 2365144: Contact: <sip:[email protected]:5060>
2014-05-06 09:00:43 2365145: Expires: 60
2014-05-06 09:00:43 2365146: Allow-Events: telephone-event
2014-05-06 09:00:43 2365147: Content-Type: application/sdp
2014-05-06 09:00:43 2365148: Content-Length: 334
2014-05-06 09:00:43 2365149:
2014-05-06 09:00:43 2365150: v=0
2014-05-06 09:00:43 2365151: o=CiscoSystemsSIP-GW-UserAgent 8182 4488 IN IP4 12.17.223.243
2014-05-06 09:00:43 2365152: s=SIP Call
2014-05-06 09:00:43 2365153: c=IN IP4 1
2014-05-06 09:00:43 2365154: 2.17.223.243
2014-05-06 09:00:43 2365155: t=0 0
2014-05-06 09:00:43 2365156: m=audio 18760 RTP/AVP 18 0 100 101
2014-05-06 09:00:43 2365157: c=IN IP4 12.17.223.243
2014-05-06 09:00:43 2365158: a=rtpmap:18 G729/8000
2014-05-06 09:00:43 2365159: a=fmtp:18 annexb=no
2014-05-06 09:00:43 2365160: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:43 2365161: a=rtpmap:100 X-NSE/8000
2014-05-06 09:00:43 2365162: a=fmtp:100 192-194
2014-05-06 09:00:43 2365163: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:43 2365164: a=fmtp:101 0-15
2014-05-06 09:00:43 2365165: 5820423: May 6 09:00:42.978: //3024942/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:43 2365166: Sent:
2014-05-06 09:00:43 2365167: SIP/2.0 100 Trying
2014-05-06 09:00:43 2365168: Via: SIP/2.0/TCP 10.38.246.136:5060;branch=z9hG4bK28bab16dbd5664
2014-05-06 09:00:43 2365169: From: "Marcos Vazquez" <sip:[email protected]>;tag=3831180~dfbf10b3-6c69-4443-852f-cbf609935a6f-35009402
2014-05-06 09:00:43 2365170: To: <sip:[email protected]>;tag=5EBA2282-19C8
2014-05-06 09:00:43 2365171: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:43 2365172: Call-ID: [email protected]
2014-05-06 09:00:43 2365173: CSeq: 106 INVITE
2014-05-06 09:00:43 2365174: Allow-Events: telephone-event
2014-05-06 09:00:43 2365175: Server: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:43 2365176: Content-Length: 0
2014-05-06 09:00:43 2365177:
2014-05-06 09:00:43 2365178: 5820424: May 6 09:00:43.479: //3024943/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:43 2365179: Sent:
2014-05-06 09:00:43 2365180: INVITE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:43 2365181: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D699C1126
2014-05-06 09:00:43 2365182: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:43 2365183: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:43 2365184: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:43 2365185: Date: Tue, 06 May 2014 15:00:43 GMT
2014-05-06 09:00:43 2365186: Call-ID: [email protected]
2014-05-06 09:00:43 2365187: Supported: 100rel,timer,resource-priority,replaces,sdp-an
2014-05-06 09:00:43 2365188: at
2014-05-06 09:00:43 2365189: Min-SE: 1800
2014-05-06 09:00:43 2365190: Cisco-Guid: 3422833792-0000065536-0000746374-2297832970
2014-05-06 09:00:43 2365191: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:43 2365192: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
2014-05-06 09:00:43 2365193: CSeq: 105 INVITE
2014-05-06 09:00:43 2365194: Max-Forwards: 70
2014-05-06 09:00:43 2365195: Timestamp: 1399388443
2014-05-06 09:00:43 2365196: Contact: <sip:[email protected]:5060>
2014-05-06 09:00:43 2365197: Expires: 60
2014-05-06 09:00:43 2365198: Allow-Events: telephone-event
2014-05-06 09:00:43 2365199: Content-Type: application/sdp
2014-05-06 09:00:43 2365200: Content-Length: 334
2014-05-06 09:00:43 2365201:
2014-05-06 09:00:43 2365202: v=0
2014-05-06 09:00:43 2365203: o=CiscoSystemsSIP-GW-UserAgent 8182 4488 IN IP4 12.17.223.243
2014-05-06 09:00:43 2365204: s=SIP Call
2014-05-06 09:00:43 2365205: c=IN IP4 1
2014-05-06 09:00:44 2365206: 2.17.223.243
2014-05-06 09:00:44 2365207: t=0 0
2014-05-06 09:00:44 2365208: m=audio 18760 RTP/AVP 18 0 100 101
2014-05-06 09:00:44 2365209: c=IN IP4 12.17.223.243
2014-05-06 09:00:44 2365210: a=rtpmap:18 G729/8000
2014-05-06 09:00:44 2365211: a=fmtp:18 annexb=no
2014-05-06 09:00:44 2365212: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:44 2365213: a=rtpmap:100 X-NSE/8000
2014-05-06 09:00:44 2365214: a=fmtp:100 192-194
2014-05-06 09:00:44 2365215: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:44 2365216: a=fmtp:101 0-15
2014-05-06 09:00:44 2365217: 5820425: May 6 09:00:44.479: //3024943/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:44 2365218: Sent:
2014-05-06 09:00:44 2365219: INVITE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:44 2365220: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D699C1126
2014-05-06 09:00:44 2365221: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:44 2365222: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:44 2365223: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:44 2365224: Date: Tue, 06 May 2014 15:00:44 GMT
2014-05-06 09:00:44 2365225: Call-ID: [email protected]
2014-05-06 09:00:44 2365226: Supported: 100rel,timer,resource-priority,replaces,sdp-an
2014-05-06 09:00:44 2365227: at
2014-05-06 09:00:44 2365228: Min-SE: 1800
2014-05-06 09:00:44 2365229: Cisco-Guid: 3422833792-0000065536-0000746374-2297832970
2014-05-06 09:00:44 2365230: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:44 2365231: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
2014-05-06 09:00:44 2365232: CSeq: 105 INVITE
2014-05-06 09:00:44 2365233: Max-Forwards: 70
2014-05-06 09:00:44 2365234: Timestamp: 1399388444
2014-05-06 09:00:44 2365235: Contact: <sip:[email protected]:5060>
2014-05-06 09:00:44 2365236: Expires: 60
2014-05-06 09:00:44 2365237: Allow-Events: telephone-event
2014-05-06 09:00:44 2365238: Content-Type: application/sdp
2014-05-06 09:00:44 2365239: Content-Length: 334
2014-05-06 09:00:44 2365240:
2014-05-06 09:00:44 2365241: v=0
2014-05-06 09:00:44 2365242: o=CiscoSystemsSIP-GW-UserAgent 8182 4488 IN IP4 12.17.223.243
2014-05-06 09:00:44 2365243: s=SIP Call
2014-05-06 09:00:44 2365244: c=IN IP4 1
2014-05-06 09:00:44 2365245: 2.17.223.243
2014-05-06 09:00:44 2365246: t=0 0
2014-05-06 09:00:44 2365247: m=audio 18760 RTP/AVP 18 0 100 101
2014-05-06 09:00:44 2365248: c=IN IP4 12.17.223.243
2014-05-06 09:00:44 2365249: a=rtpmap:18 G729/8000
2014-05-06 09:00:44 2365250: a=fmtp:18 annexb=no
2014-05-06 09:00:44 2365251: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:44 2365252: a=rtpmap:100 X-NSE/8000
2014-05-06 09:00:44 2365253: a=fmtp:100 192-194
2014-05-06 09:00:44 2365254: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:44 2365255: a=fmtp:101 0-15
2014-05-06 09:00:45 2365256: 5820426: May 6 09:00:45.147: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:45 2365257: Received:
And then I don't see a response then send out a bye:
Sent:
2014-05-06 09:00:46 2365897: BYE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:46 2365898: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D69A54BC
2014-05-06 09:00:46 2365899: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:46 2365900: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:46 2365901: Date: Tue, 06 May 2014 15:00:44 GMT
2014-05-06 09:00:46 2365902: Call-ID: [email protected]
2014-05-06 09:00:46 2365903: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:46 2365904: Max-Forwards: 70
2014-05-06 09:00:46 2365905: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:46 2365906: Timestamp: 1399388446
2014-05-06 09:00:46 2365907: CSeq: 106 BYE
2014-05-06 09:00:46 2365908: Reason: Q.850;cause=86
2014-05-06 09:00:46 2365909: P-RTP-Stat: PS=180295,OS=3604444,PR=180354,OR=3607080,PL=0,JI=0,LA=0,DU=3603
2014-05-06 09:00:46 2365910: Content-Length: 0
2014-05-06 09:00:46 2365911:
2014-05-06 09:00:46 2365912: 5820458: May 6 09:00:46.479: //3024942/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:46 2365913: Sent:
2014-05-06 09:00:46 2365914: BYE sip:[email protected]:5060;transport=tcp SIP/2.0
2014-05-06 09:00:46 2365915: Via: SIP/2.0/TCP 10.38.246.166:5060;branch=z9hG4bK2D69A6E75
2014-05-06 09:00:46 2365916: From: <sip:[email protected]>;tag=5EBA2282-19C8
2014-05-06 09:00:46 2365917: To: "Marcos Vazquez" <sip:[email protected]>;tag=3831180~dfbf10b3-6c69-4443-852f-cbf609935a6f-35009402
2014-05-06 09:00:46 2365918: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:46 2365919: Call-ID: [email protected]
2014-05-06 09:00:46 2365920: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:46 2365921: Max-Forwards: 70
2014-05-06 09:00:46 2365922: Timestamp: 1399388446
2014-05-06 09:00:46 2365923: CSeq: 101 BYE
2014-05-06 09:00:46 2365924: Reason: Q.850;cause=102
2014-05-06 09:00:46 2365925: P-R
2014-05-06 09:00:46 2365926: TP-Stat: PS=180239,OS=3604780,PR=180295,OR=3604444,PL=0,JI=0,LA=0,DU=3603
2014-05-06 09:00:46 2365927: Content-Length: 0
2014-05-06 09:00:46 2365928: -
SIP to SIP Call Failures on CME to CME - sip-ua conflict/issue?
Hi,
I have two existing CME systems which I wish to allow internal calls between. These calls will go over an IPSec VPN. However the calls are failing.
Phones DN22xx - London CME 2801 - PIX505 --- Internet ---ASA5505 - India CME 2801 - Phones DN400x
I have configured dial peers on both CME's and the IPSec VPN. I can ping between both systems. The VPN allows traffic between the interface IP's of the CME systems only.
London CME (local SCCP phones 22xx):
interface FastEthernet0/0.100
encapsulation dot1Q 100 native
ip address 10.0.10.250 255.255.255.0
voice class codec 101
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 25 voip
description *** SIP Peer to India ***
answer-address 400.
destination-pattern 400.
voice-class codec 101
session protocol sipv2
session target ipv4:192.168.15.10
incoming called-number 400.
no vad
India CME (Local SSCP phones 400x):
interface FastEthernet0/0
ip address 192.168.15.10 255.255.255.0
voice class codec 100
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 10 voip
description *** SIP Peer to London UK ***
answer-address 22..
destination-pattern 22..
voice-class codec 100
session protocol sipv2
session target ipv4:10.0.10.250
incoming called-number 22..
no vad
The CME system at India also has an existing SIP dial peer to a service provider and has sip-ua configured (username, password, realm and registrar).
A call from India (4005) to London (DN2207) fails, the ccsip debug attached. I'm assuming its because the sip-ua configuration is being used for these calls to when I don't want it to be. The from field shows âFrom: <sip:[email protected]â when I need this to be the internal IP 192.168.15.10.
Can anyone offer any assistance with this?
Regards,
ChrisHi,
thanks for your input however thats not the problem. 201.196.128.56 isn't an address on the router, it only has one IP and its 192.168.15.10.
The 201.196.128.56 address is the NAT'd address on the firewall. So that when a SIP call is made to the internet with sip-ua the from address is the public IP.
Chris -
Is there a software in which I can hook up my iPhone 4s to my computer and send and receive text messages and calls (via headset) through my computer? Whether it's free or cost money, can someone please give me a name of a program or software that allows me to do this? I can't seem to find anything like this for the iPhone.
No.
-
JNI_CreateJavaVM hangs in DLL called via Adobe plug-in
We have a plugin that works in Adobe Acrobat and Adobe Reader. The plugin is written in VS2008 VC++ and calls into a DLL that we have written.
The DLL in turn uses JNI and calls into some Java code. We've tested the DLL and calls into Java separately and that works OK.
But when it is called via the plugin in Acrobat it is failing. It seems to die immediately during the JNI initialization in that case. There is a method called JNI_CreateJavaVM where it hangs.
Do you have any idea why that might happen? Is there something protected that would not permit a dll within the scope of Acrobat to call into Java code?Hi Leonard,
Can you please let me know the URL and any other contact information for opening a formal case with Adobe developer support group that can help with this issue?
I appreciate your time and consideration.
Thanks,
Snehal.
Snehal Sao | Senior Software Engineer | Formtek | 2190 Meridian Park Blvd | Suite G | Concord, CA 94520
p. 925.459.0490 | f. 925.459.0487 | e. [email protected] | url: http://www.formtek.com/
This electronic message transmission contains information, which may be confidential. The information is intended for the use of the individual or entity named above. If you are not the intended recipient, and have received this electronic transmission in error, please notify sender then delete immediately. -
I have a document made up of separate PDF files which reside in a folder and are linked to each other via hyperlinks. Each pdf file is set to open with bookmarks displayed, however if I link from one PDF file to another and use the "Previous View" button to navigate back to my starting point the bookmarks are replaced by "page thumbnails". Is there anyway to stop this from happening?
Hi Pusman,
While setting up the links, if you choose to open the file in a new window then you won't face this issue, then you can simply switch to the previous file and bookmark view will remain as it is.
Does that helps with your query?
Regards,
Rahul -
Adobe PDF files will not display when called via a link in page (FF 4.01)
adobe PDF files will not display when called via a link in page.
This happens when testing new web page from DWeaver CS5 - all other browser tests are OK (IE9, Safari5.05, Chrome etc.)[moved to the Adobe Reader (desktop) forum]
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How to set a cookie in the browser from an html page called via an Iview
How to set a cookie in the browser from an html page called via an Iview
Hello all,
I have an issue which is causing problems. I have a snap survey (html form with submit and cookie setting) which is embedded in a url iview.
Although the submit and the form work fine, the portal will not allow the cookie to be set it seems.
Is there a way to allow cookies to be set from an embedded page in a url iview??
You will make my day if you know!
System: EP7 SP13
Kind regards
AlexHi,
Check this:
http://www.oracle.com/technology/products/ias/portal/html/same_cookie_domain_with_pdkv2.html
Cookie Basics
Web browsers have built in rules for receiving and sending cookies. When a browser makes a request to a web server and the web server returns cookies with the response, the browser will only accept a cookie if the domain associated with the cookie matches that of the original request. Similarly, when a browser makes a subsequent request, it will only send those cookies whose domain matches that of the target web server.
These rules are designed to ensure that information encoded in cookies is only "seen" by the web server(s) that the originator of the cookie intended. These rules also ensure that the cookie cannot be corrupted or imitated by another server. By default, the domain associated with a cookie exactly matches that of the server that created it. However, it is possible to modify the domain at the time the cookie is created. Relaxing the cookie domain increases the scope of the cookie's visibility making it available to a wider "audience" of web servers.
For example, if a cookie is created by a.us.oracle.com, it's domain will usually be set to a.us.oracle.com. This means that the browser will only send the cookie to a.us.oracle.com. It will never send it to any other servers. However, if at the time of creation, the domain of the cookie is set to .us.oracle.com, the browser will send the cookie to any server whose domain falls within .us.oracle.com. such as portal.us.oracle.com, provider.us.oracle.com, app.us.oracle.com etc
Regards,
Praveen Gudapati -
My BB Torch can't make a call via VIBER
Dear Team,
Kindly assist me, when did i try to make call (Viber to Viber) via my BB device I can't able to do this. Majority says BB isn't support voice call on VIBER. Is there any future upgradation are coming in BB IOS or any other version is required? which's allow to make voice call in BB or if any setting is required then please let me know.
Note: I've new version of Viber, and have Black Berry Torch 9800 IOS: 6
Regards,
Ahsan SyedHello Ahsanjafri.
Viber works if calling via WiFi or 3G considering that the two must have installed the same application on devices..
To this I add other application that Viber for BlackBerry is in its development phase still not as automated as IOS.
Regards...
Kudos **Do not forget to give those people who help and advise you regarding your questions, as well give the answer like**
@ gutijose14
BBM Channels PIN: C0007093A
Do not forget to give LIKE Those people who help you and advise you about your doubts. if the review has been SOLVED** # 4LL #ÉliteRoad Make a backup of your BlackBerry
BlackBerry Protect and BlackBerry Link constantly. #ichooseBlackBerry10 Gutijose14 Forums Veteran I
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