WRT54G rewriting sip headers

Hi,
The router is rewriting the sip headers that contain local ip addresses with the public address which is causing problems with our session border controller.
Is there a way to turn off the sip alg?
Thanks
Peter

After bringing this up with my Cisco rep it was recommended I look at the AVS 3120 box which is supposed to support the header/url rewriting I need. Can anyone with experience with these boxes let me know if I'm being steered in the right direction? The documentation doesn't seem to specifically mention this ability or if it does it's worded in a way that doesn't make it obvious.

Similar Messages

Rewriting http headers and url for Oracle Applications

We are running Oracle financial applications and need to provide access to users on the Internet. We can do that fine except for people behind tightly controlled firewalls that only allow port 80 and 443. Oracle insist it has to run it's apps using SSL on non-standard ports like 8001 and such which means many people can't get to them. So to make sure everyone can get to the apps I need to preset it on port 443. Changing the apps/servers isn't an option for many frustrating reasons so I'm looking for something that can reverse proxy and rewrite headers and url's. The hard part is the headers and all urls in the web pages need to be changed on the fly because Oracle hard codes everything with absolute paths.
Can someone give me some advice on what Cisco product(s) may be able to help?

After bringing this up with my Cisco rep it was recommended I look at the AVS 3120 box which is supposed to support the header/url rewriting I need. Can anyone with experience with these boxes let me know if I'm being steered in the right direction? The documentation doesn't seem to specifically mention this ability or if it does it's worded in a way that doesn't make it obvious.

Can Web Cache rewrite HTTP_REFERER headers?

Can Web Cache be configured to rewrite the HTTP_REFERER header to replace the Web Cache site address:port with the dispatched application server address:port?
i.e.
If there is a Web Cache site listening on webcache.domain:80 and it is mapped to application servers: appserver1.domain:7778 and appserver2.domain:7778.
And a user requests page webcache.domain/page1, which contains a link to page2.
And then the user opens the link to page2, and Web Cache decides to send the request on to appserver2.
Can Web Cache be configured to change the HTTP_REFERER header from webcache.domain/page1 to appserver2.domain/page1?
This is important to our application, because we want to verify that our processing pages are invoked from our application validation pages (we know this method is not 100% tamper proof).
We would prefer to not have to change our application code to support Web Cache
Thanks for any info or suggestions, -Steve

Thanks Jean
I've added comments to extracts from your post.
Let's say you suddenly get a request for p2, even if
it was from the history of the browser, it would still
contain a referer header of p1, and is no different
from what it would be when the user was going through
the proper steps? Are you saying the p2 generated from
your JavaScript will have a different referer from the
p2 generated from browser history? If so, how are they
differ exactly?The browser history does not include the referer. So p2 requested from the history list does not include a referer although the original link from p1 did.
Without Web Cache, the flow is:
Step 1: a user requests appserver/page1. page1 is added to the browser history. page1 is loaded and executes JavaScript to display page2 in an iFrame. page2 is not added to the browser history (just like style sheets and other embedded requests do not get included in the history). page2 is loaded and executes JavaScript to submit an html form to load page3 into a hidden iFrame and wait for it to load. page3 is added to the browser's history becuase it was submitted; not embedded. Assuming that the request for page1 came from the browser's history, favorites, a link from an e-mail (the usual for us), or was entered manually - there will be no referer for page1. The expected referer for page2 is appserver/page1, and the expected referer for page3 is appserver/page2.
Step2: while processing page3, the application compares the server name in the requested URL (appserver/page3) with the server name from the referer URL (appserver/page2), if the servers match, page3 accesses the database and sends the requested data to the browser
Step3: JavaScript in page2 detects that page3 has loaded, interprets the results from page3, updates the content displayed by page2, and returns control to the browser user
Since this works with your app, how is webcache in
front going to break it? Note that when passing a
request
to the app server, web cache does not modify the
host header in the request.without Web Cache the browser sends requests directly to the appserver, so the request/referer entries seen by the application look like:
appserver/page1 -
appserver/page2 appserver/page1
appserver/page3 appserver/page2
with Web Cache the browser sends requests to webcache, so the request/referer entries seen by the application look like:
appserver/page1 -
appserver/page2 webcache/page1
appserver/page3 webcache/page2
This is because the Web Cache changed the request URL and did not change the referer URL.
So request to webcache.domain/abc will be sent to your
app server as is.I don't think so, doesn't the requested URL change as a result of the Web Cache mapping? i.e. webcache.domain/abc --> appserver.domain/abc
And since we do not change the referer
header, either, whatever works for your app now should
still work for you with webcache in front. This is
assuming you only do comparisons on the 2 hosts to see
if they are the same. Web Cache is changing one (request URL) of the two items (request URL and referer URL) that the application server compares.
The reason has been that we are
a cache, and in principal should be as transparent
as possible, so we try to stay away from app-sever/
apache like functionality.I support that principle 100% - from the application server perspective, it should not need to differentiate between a request directly from a browser, and a request that was issued by the cache on behalf of a browser.
The cache should completely pretend that it is the originating browser. A browser sends the request and the referer URLs. In application flow, a relationship can be inferred/required between the two URLs. Web Cache is changing one without changing the other and breaking the relationship that the application server would normally expect.
But I'm still
not convinced that you need that in webcache. :)If we want to use Web Cache in front of this application, without changing the application, then we need it. :)
Of course, we could change our application, but the more I think about the principles involved, the more I believe that it is the cache that needs changing. However, I will take another look to see if there are other standard headers related to a request that are present with and without Web Cache, and that can be used to reliably compare the server in the request URL with the server in the referer URL. If there are, then I can happily change our application so that it will work with or without Web Cache.
-Steve

Rewrite Location-headers in frontend by weblogic plugin

Hi!
I've got a problem where I've got a Apache 2.2.9 acting as a proxy for a application running on BEA Weblogic.
I'm using the plugin from BEA Weblogic as the proxy.
Everything works fine except when the weblogicserver makes a redirect, i.e. sends a Location-header. The Location header looks like "Location: http://weblogic.internal.net/Client".
The servername should be rewritten to the public nam "proxy.example.com".
### START ### Weblogic configuration in apache.conf:
LoadModule weblogic_module /usr/lib/apache2/modules/mod_wl_22.so
<IfModule mod_weblogic.c>
WebLogicHost weblogic.internal.net
WebLogicPort 8202
WLProxySSL on
DebugConfigInfo on
Debug ALL
</IfModule>
<Location /Application/>
SetHandler weblogic-handler
</Location>
### END ### Weblogic configuration in apache.conf:
Any ideas on how to manage the rewrite of the Location-header?
Can weblogic plugin and mod_proxy/mod_rewrite be comined?
Is there any other configuration parameters for weblogic to state the frontend machine?
/Andreas

Hi!
I've got a problem where I've got a Apache 2.2.9 acting as a proxy for a application running on BEA Weblogic.
I'm using the plugin from BEA Weblogic as the proxy.
Everything works fine except when the weblogicserver makes a redirect, i.e. sends a Location-header. The Location header looks like "Location: http://weblogic.internal.net/Client".
The servername should be rewritten to the public nam "proxy.example.com".
### START ### Weblogic configuration in apache.conf:
LoadModule weblogic_module /usr/lib/apache2/modules/mod_wl_22.so
<IfModule mod_weblogic.c>
WebLogicHost weblogic.internal.net
WebLogicPort 8202
WLProxySSL on
DebugConfigInfo on
Debug ALL
</IfModule>
<Location /Application/>
SetHandler weblogic-handler
</Location>
### END ### Weblogic configuration in apache.conf:
Any ideas on how to manage the rewrite of the Location-header?
Can weblogic plugin and mod_proxy/mod_rewrite be comined?
Is there any other configuration parameters for weblogic to state the frontend machine?
/Andreas

TCL and SIP headers

I have a TCL script on IPIPGW that creates an outgoing sip call. I need to check some parts of the header of SIP Progress message and I'm unable to do so because I can't catch the event ev_progress (in debugs it happens but procedure is never called). Did anyone tried to check SIP Progress message on IPIPGW using TCL? How did you do it?
Regards,
Ivan

Of course, SIP progress message is incoming message since INVITE is outgoing..

Ask the Cisco VIP: Troubleshooting SIP in Cisco Unified communications

Troubleshooting SIP in Cisco Unified communications deployments with Cisco VIP Ayodeji Okanlawon
This is a Q&A Ask the Expert Session continuation from the Live Webcast
Ask your questions on Session Initiation Protocol (SIP) and how it is redefining our UC world.The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
Featured Expert
Ayodeji Okanlawon, a Cisco Designated VIP, is the Lead Consultant Engineer for Global Solutions Design and Engineering at Verizon Business. In his past, he has worked at Intact IS, NCS Global, and Schlumberger Information Solutions. His experience includes development of design and deployment of large scale IP telephony projects on Cisco Call Manager platforms, Cisco Voice gateways, Cisco Jabber cloud and on premise solution. His expertise includes SIP solutions, CUBE design and Deployment, Troubleshooting: Voice gateways, CUCM, Unity connection, CUPS. Deji has been awarded the Cisco Designated VIP in 2013 and 2014. Deji holds a Bachelor of Science (BS), Electrical and Electronics Engineering, Second Class Upper from Obafemi Awolowo University.
According to Deji, “If you want to advance your career, if you’re serious about your skill sets, you’ve got to be in the forums.” (Read the Interview >>)
We look forward to your participation. This event is open to all, including partners.
* * Remember to use the rating system to let Deji know if you have received an adequate response. * *
Deji might not be able to answer each question due to the high volume expected during this event. This event runs January 13 through January 23, 2015. Visit this forum often to view responses to your questions and the questions of other community members.

Derrick,
RFC 3261defines ways to provide increased security for a SIP session.
The following describes areas in SIP that provides security for the protocol
1. Authenticating users.
We need to authenticate a user to ensure that the sender of the message is who he claims to be.
To achieve this SIP uses digest authentication between a UAC, proxy and a UAS. This provides the most basic level of authentication challenge between a client, proxy and a server.
2. Secure SIP signalling
The next area we can secure is SIP signalling itself. For this we use SSL/TLS. This is similar to using https in web browsers. With TLS before our any signalling is exchange X.509 certificates are used create a secure TLS channel. All our SIP messages are then transported within the secure channel.
NB: The digest authentication mentioned above for authenticating a user agent is just authentication. The messages are not protected from reading or modification hence it is recommended that these messages are carried inside a secure TLS channel for better security.
3. Privacy and Identification
Additional security features in SIP provides means where any user can choose to either reveal or conceal his identity.
4.Secure RTP
SIP also provides the ability to secure the media channel. It is not enough to secure signalling while anyone can listen to the media. RFC3830 discusses how the encryption should be done.
5. S/MIME
S/MIME encapsulation is used to protect sip headers making it impossible for any one in between the sender and receiver to modify the sip headers
Regards

CSS11506 http header rewrite question

Hi
I read the ACE doc, and it said that cisco ACE supports the capability to rewrite http headers in both client requests and server responses. Is CSS11506 can do it?
I have a lot of problems that application on the local server redirect https to http. Because the way they do installation which standard way and it can not fix or hardly to fix.
I would like to get a tip to let css11506 rewrite the server's rewrite. is it possible?
Any comments will be appropriated
Thanks in advice
julxu

Hello Julxu,
If I understand your question correctly, you are looking for the CSS rewrite the URL from http to https when the server sends a redirect to the client. If I'm correct, then you can find out how to accomplish this in the Specifying Secure URL Rewrite sectioin of the CSS configuration guides.
Hope this helps,
Sean

SPA303 - Reordering the position of Content-Length header in SIP INVITE

Hi,
I have SPA303 IP Phone connected behind a SIP ALG router but have been facing issues with media setup for incoming and outgoing calls.
Further investigation using SIPp script helped me out to understand the root cause of the issue which is as follows:
If the SIP INVITE or 200 OK for SIP INVITE has Content-Length header ahead of the Content-Type header, the SIP ALG router is not able to handle the RTP traffic for the calls. Cisco SPA303 IP phone exhibits this behaviour and hence couldn't successfully establish call with the SIP ALG that I use.
Can you please confirm if it is configurable to reposition or re-order the Content-Length header to resolve this issue?
Thanks in advance.
Regards,
Anand Krishnan

As far as I know it's not configurable. According SIP protocol, the order of SIP headers is not meaningfull.
Your router need to accept both orders as both are corrrect and have same meaning. Ask the vendor of router for updated firmware ...

.htacess rewrite question

I am trying to perfect my .htaccess file and I have it all working except one of the re-write sections:
Simply I am trying to re-write my domain to hide a section of it in the browser address bar as such . . .
Currently it reads:
https://www.hoodfellaent.secure.omnis.com
but I would like it to always read:
https://www.hoodfellaent.com/
Basically I need it to mask but not get rid of the part that reads ".secure.omnis"
Can someone please help me figure this out I have Googled this and altered a few different code snippets but I have not been able to make it work when I visit the website - I either get an error page or the address looks unchanged in the browser address bar.
I need this to work before we upload our new web design!
Thanks a bunch to anyone that can help!

I don't think you can hide parent level URL paths from inside your virtual directory. That would have to be done from the top level by establishing a DNS Alias there or using the root level .htacess rules to rewrite the headers.
Mylenium

SIP client and delayed SDP offer-answer

Hi,
I'm trying to connect the E51 SIP client to the company PBX.
Calling out works fine, but I can't receive calls from the PBX.
It seems that the source of the problem is that the PBX is sending INVITEs without an SDP offer.
Do you know if the E51 SIP client supports SDP delayed offer-answer?
Thankes,
Dan

Well,
The reason is not the delayed offer-answer. The E51 accepts SIP calls from X-Lite. But, not from the PBX (even when SDP is present in initial INVITE). For the PBX it responds with: 480 Temporarily Not Available (see below)
Which is quite interesting...
Maybe it is because all those additional SIP headers.
Any thoughts?
Dan
Oct 29 17:53:58 2008 : [Send Request ]
{connection: host=135.64.17.239 port=5060 protocol=UDP}
INVITE sip:[email protected];transport=UDP SIP/2.0
Call-ID: 012bdb8aba6dd1dd349c602800
CSeq: 1 INVITE
From: "Dan Gluskin" ;tag=012bdb8aba6dd1dc349c602800
Record-Route: ,
To: "37400"
Via: SIP/2.0/UDP 149.49.130.131:5060;branch=z9hG4bK0303031313132323235f4.0,SIP/2.0/TLS 149.49.130.131:6001;psrrposn=2;received=149.49.130.131;branch=z9hG4bK012bdb8aba6dd1de349c602800
Content-Length: 304
Content-Type: application/sdp
Contact: "Dan Gluskin"
Max-Forwards: 69
User-Agent: Avaya CM/R013w.01.2.024.0
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS
History-Info: ;index=1
History-Info: "37400" ;index=1.1
Accept-Contact: *;+avaya-cm-line=1
Supported: 100rel,timer,replaces,join,histinfo
Alert-Info: ;avaya-cm-alert-type=internal
Min-SE: 1200
Session-Expires: 1200;refresher=uac
P-Asserted-Identity: "Dan Gluskin"
v=0
o=- 1 1 IN IP4 149.49.130.131
s=-
c=IN IP4 149.49.130.132
t=0 0
m=audio 2376 RTP/AVP 0 8 18 110 4 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 G726-32/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:127 telephone-event/8000
Oct 29 17:53:58 2008 : [Recv Response ]
{connection: host=135.64.17.239 port=5060 protocol=UDP}
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 149.49.130.131:5060;branch=z9hG4bK0303031313132323235f4.0,SIP/2.0/TLS 149.49.130.131:6001;psrrposn=2;received=149.49.130.131;branch=z9hG4bK012bdb8aba6dd1de349c602800
To: "37400"
From: "Dan Gluskin" ;tag=012bdb8aba6dd1dc349c602800
Call-ID: 012bdb8aba6dd1dd349c602800
CSeq: 1 INVITE
Content-Length: 0
Oct 29 17:53:58 2008 : [Recv Response ]
{connection: host=135.64.17.239 port=5060 protocol=UDP}
SIP/2.0 480 Temporarily Not Available
Via: SIP/2.0/UDP 149.49.130.131:5060;branch=z9hG4bK0303031313132323235f4.0,SIP/2.0/TLS 149.49.130.131:6001;psrrposn=2;received=149.49.130.131;branch=z9hG4bK012bdb8aba6dd1de349c602800
To: "37400" ;tag=f9srkqi8iphc6t5c9eru
From: "Dan Gluskin" ;tag=012bdb8aba6dd1dc349c602800
Call-ID: 012bdb8aba6dd1dd349c602800
CSeq: 1 INVITE
Content-Length: 0

SIP Trunk Configuration

Hi
We are migrating from Analogue to IP Telephony. I have recieved the following guidlines to configure the SIP Trunk:
*For signaling: use IP : x.x.211.70   ( SIP ) on PORT 5060
*Regarding Numbering Format, use the following:
•             For outgoing Calls :
                The originating Number (A#), should be 96611510XXXX format.
               The Destination Number should be 0NXXXXXX (N area code) or 00XXXXXXXXX (for international)
•             For incoming Calls:
                The Destination Number (B#), should 011510XXXX Format.
                The originating Number (A#), will be 0NXXXXXXX or 00XXXXXXXXXXX Format
*Use Audio Codec's G711-aLaw ; G711-uLaw & G729
*Use T.38 For FAX
*set DTMF to RFC2833
*Make sure to reply with 200Ok for our OPTIONS messages ( ping messages for the SIP)
* configure the following SIP Timers:    “Min-SE=1800 “’ & “Expires=300”
For connectivity consider the following:
SIP CE: 10.65.13.110 (it might be needed to translate this IP to the PBX local IP).
SIP GW: 10.65.13.109
Subnet mask: /30
SIP VLAN: 1191
Notes:
Kindly make sure to have GO SIP GW (x.x.211.70) routed to SIP GW (10.65.13.109) as next-hop.
Kindly make sure to have SIP CE IP addresses are in VLAN 1191.
Can please anyone explain what have to done?
Regards

Ahmed,
Wao..Where do I start...This information is required for configuration on your CUBE..which will be your 2921 router...
Ahmed, here are some pointers I wrote a while ago..
In addition to these points, you will need to configure your cube to be able to route traffic to your ITSP using all the information given to you
1. Configure CUBE for media flow through. In this Mode CUBE acts as a true B2BUA. Advantages you get include address hiding and security becaue CUBE terminates and re-originate both signalling and Media. In this mode CUBE becomes a point of demarcation from th external world.
2. Configure CUCM to use Delayed Offer and CUBE to convert delayed offer to ealry offer...This prevents the need for you to use MTP to send Early offer on CUCM
voice service voip
early-offer forced
3. Configure DTMF signalling method on sip trunk to "No preference" This setting allows Unified CM to make an optimal decision for DTMF and to minimize MTP allocation.
4. Configure your CUBE to meet the requirements of your ITSP. Ask if they have configuration templates or any specific configuration they like you to use. This will save you time troubleshooting. Most of them dont use the default port 5060 because of security, confirm with your proivider what ports they use.
voice service voip
allow-connections sip to sip
sip
early-offer forced
header-passing
error-passthru
5. Use SIP to SIP...Use end to end sip. CUCM---sip---CUBE--sip----ITSP
6. Create a Trusted list of IP addresses on your CUBE is your CUBE IOS is 15.1 .2(T) and above.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
This is imprtant because sometimes your ITSP will send you a single ip address for signalling and will then send media on a different IP adress. So get all the IP address your ITSP is using and add them to the trust list as shown above
7. Configure your inbound and outbound dial-peer approriately
Inbound Dial-Peer for calls from CUCM to CUBE (CUCM sending 9 +all digits dialled to CUBE)
dial-peer voice 100 voip
description *** Inbound LAN side dial-peer ***
incoming called-number 9T
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to CUCM (SP will be sending 10 digits inbound)
dial-peer voice 200 voip
description *** Outbound LAN side dial-peer ***
destination-pattern [2-9].........
session protocol sipv2
session target ipv4:
codec g711ulaw
dtmf-relay rtp-nte
Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM redundancy/load balancing
Inbound Dial-Peer for calls from SP to CUBE
dial-peer voice 100 voip
description *** Inbound WAN side dial-peer ***------------------(catch-all for all inbound PSTN calls)
incoming called-number [2-9].........
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to SP
dial-peer voice 200 voip
description *** Outbound WAN side dial-peer ***
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4::XXXX (where XXXX is the port number your provider is using if different from 5060)
codec g711ulaw
dtmf-relay rtp-nte
8. SIP Normalization:
You may need to configure sip normnalization to modify sip headers, CLI etc. A good example is during call forwarding. You may need to change your diversion headers to match the CLI your provider is expecting.. if your call forwarding is failing during testing this may be the reason..We can help you with this.
9. Media Resources
Plan your solution properly. Consider if you will need Xcoders, MTP, Conference bridge etc. You may avoid the need for xcoders if you confure your regions properly and use voice class codecs on your sip profiles. It is important to know if there are any endpoints in your network that do not support dtmf relay rtp-nte. You can avoid the use of MTP if you configure your dial-peers to have multiple dtmf types for thos phones that do not support rtp-nte
e.g
dial-peer voice 1 voip
session protocol sipv2
dtmf-relay rtp-nte digit-drop sip-kpml (if your phones support kpml..then this will be used)
If in your environment you will need to do xcoding or CFB then ensure you have PVDMS
.10.FAX
If you have FAX in your network, determine what fax protocol your sip provider supports. Dont assume. Ask them and confirm in writing what they support. I have seen legal cases because of fax failures over sip trunks
Configure your FAX devices in seperate device pools and use porefix to route calls using G711 only. Even if you are using T38, ensure your fax use G711 to establish the voice calls
Finally
11. Have a detailed and carefully planned TEST Plan. Test the FF:
Inbound and outbound Local, Long distance, International calls for G711 & G729 codecs (if supported by provider)
Outbound calls to information and emergency services
Caller ID and Calling Name Presentation
Supplementary services like Call Hold, Resume, Call Forward & Transfer
DTMF Tests
Fax calls – T.38, modem pass-through--whichever one you decide to use
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"

Passing Information from UCCX call variables through trunk SIP to Astersik

Hi All,
We need to pass some informations from our UCCX 8.5SU3/CUCM 8.6.2a to our Asterisk Server.
This two PBX are connected by a trunk sip.
Is it possible to do it?I've read about sip header,but i've never work on it.
Is it possible with a javascript?
Could you please help us?
Thanks
Stefano

No. CCX uses JTAPI (CTI/QBE) to integrate with CUCM, not SIP. As such there is no mechanism for it to manipulate or add extra SIP headers. You would need to use one of the native scripting options (e.g. ODBC, HTTP GET/POST, SMTP) or write a custom Java class that can interface natively with the other application. Examples of this exist such as the excellent documents on SFTP, CIFS, and LDAP.
Please remember to rate helpful responses and identify helpful or correct answers.

Header manipulation problem

I have a SIP trunk provider that requires me
1) to always use my account name in the "From" header, so I have to use header header rewriting to achieve this
2) to set the "P-Preferred-Identity" header to change the caller id (as From cannot be used due to point 1)
Now the header rewriting for 1) works fine, but 2) is bit more complicated, as I need to fetch the source number (ideally before it has been modified) and then use it in a new header line. (aka "copy contents from one header to another header in an outgoing SIP message")
From the "Conditional Header Manipulation of SIP Headers" document I would have assumed that the following works:
voice class sip-profiles 101
request INVITE peer-header sip From copy ".*<sip:(.*)@" u01
request INVITE sip-header P-Preferred-Identity add "P-Preferred-Identity: <sip:555123\[email protected]>"
But unfortunately it doesn't, "\u01" is being sent literally instead of the content:
P-Preferred-Identity: <sip:555123\[email protected]>
Any ideas? I'm running CUBE on a 2901ISR with IOS version 15.3(2)T
Edit: Not sure what the difference between sip-header and peer-header is in this context, but using
request INVITE sip-header From copy ".*<sip:(.*)@" u01 (instead of the peer-header line)
doesn't work either.

If the profile you configured is passing the "vairable" instead of the actual content, there is something wrong, please look at the following docs for a better understanding of how the feature works:
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip.html
http://www.cisco.com/en/US/docs/ios-xml/ios/voice/cube_sip/configuration/12-4t/voi-condl-header.pdf
http://www.cisco.com/en/US/docs/ios-xml/ios/voice/cube_sip/configuration/15-2mt/voi-condl-header.html
HTH
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers.

Upload failed your changes were saved but could not be uploaded because of an error. you may be able to upload this file using server web page. save a copy

Hi All,
upload failed your changes were saved but could not be uploaded because of an error. you may be able to upload this file using server web page. save a copy button.
This is the issue which I am facing while working with SharePoint 2010. In a sharePoint 2010 document library I am having an excel file and I am trying to open it from Windows 7 and is office 2010.
I cam e across few suggestion as mentined below but unable to find the location where to do
Go to Resource Policies > Web >
Rewriting > Custom Headers > (if 'Custom Headers' is not visible, click
Customize on the right top to enable the view).
Create a new policy with the Resource as <fully qualified domain name of the SharePoint server:*/*> (for example https://sharepoint.juniper.net:*/* ).
Create the action as Allow Custom Headers.
Apply the settings to the required roles.
Please suggest.

Hi rkarteek
All things you have to do is as follows:
1. Open regedit.exe
2. Naviagate to following key:
[HKEY_CURRENT_USER\SOFTWARE\Microsoft\Office\14.0\Common\Internet]
3. Click Edit Menu -> New -> DWORD with name of "FSSHTTPOff"
(without quotes)
4. Click on "FSSHTTPOff" and enter value of 1
5.
Close any Office Applications and browser sessions
6. Try to reopen your document (no more read only or failure to upload)
have a nice day!

Problem compiling apache (2.4.4) with openssl 1.0.1e

Hi Experts
i am struggling since last 2 weeks to compile apache2.4.4 with openssl 1.0.1e on sun spark server.
System Details: SunOS hostname 5.10 Generic_147440-19 sun4v sparc SUNW,Sun-Blade-T6320
bash-3.00# cat /etc/release
Oracle Solaris 10 8/11 s10s_u10wos_17b SPARC
Copyright (c) 1983, 2011, Oracle and/or its affiliates. All rights reserved.
Assembled 23 August 2011
We already have apache 2.2 running on this server, now we need 2.4.4 with latest openssl.
i donwloaded apr, apr-utils, pcre, openssl and httpd2.4.4 from their respective websites.
then i comiled these prerequisites into the path /local/apache/2.44/apr, /local/apache/2.44/apr-util, /local/apache/2.44/pcre, /local/apache/2.44/openssl
and finally i am trying to build httpd2.4.4 with above compiled modules.
so i used this command:
./configure prefix=/local/apache/2.44 with-apr=/local/apache/2.44 with-apr-util=/local/apache/2.44/apr-utils with-pcre=/local/apache/2.44/pcre enable-proxy enable-proxy-http enable-so enable-auth-digest enable-mime enable-deflate enable-rewrite enable-headers enable-ssl with-ssl=/local/apache/2.44/openssl
when i run make, i received this error:
dlopen 0x44 /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
dlopen 0x748 /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
dlclose 0xb4 /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
dlclose 0x1a0 /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
dlclose 0x770 /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
dlsym 0x268 /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
dlsym 0x388 /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
dlsym 0x764 /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
dlerror 0xdc /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
dlerror 0x2cc /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
dlerror 0x3dc /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
dlerror 0x714 /local/apache/2.44/openssl/lib/libcrypto.a(dso_dlfcn.o)
__umoddi3 0x64 /local/apache/2.44/openssl/lib/libcrypto.a(bn_word.o)
__umoddi3 0xac /local/apache/2.44/openssl/lib/libcrypto.a(b_print.o)
__udivdi3 0xe8 /local/apache/2.44/openssl/lib/libcrypto.a(b_print.o)
__udivdi3 0x394 /local/apache/2.44/openssl/lib/libcrypto.a(bn_div.o)
ld: fatal: relocations remain against allocatable but non-writable sections
collect2: ld returned 1 exit status
*** Error code 1
make: Fatal error: Command failed for target `mod_ssl.la'
Current working directory /local/apache244pkg/httpd-2.4.4/modules/ssl
*** Error code 1
The following command caused the error:
if test `pwd` = "/local/apache244pkg/httpd-2.4.4"; then \
echo "" ; \
fi; \
list=''; for i in $list; do \
target="shared-build"; \
if test "$i" = "."; then \
made_local=yes; \
target="local-shared-build"; \
fi; \
if test "$i" != "srclib"; then \
(cd $i && make $target) || exit 1; \
fi; \
done; \
if test -f 'modules.mk'; then \
if test -n 'mod_ssl.la'; then \
echo "Building shared: mod_ssl.la"; \
if test "$made_local" != "yes"; then \
make "local-shared-build" || exit 1; \
fi; \
fi; \
fi; \
if test `pwd` = "/local/apache244pkg/httpd-2.4.4"; then \
echo "" ; \
fi
make: Fatal error: Command failed for target `shared-build-recursive'
Current working directory /local/apache244pkg/httpd-2.4.4/modules/ssl
*** Error code 1
The following command caused the error:
if test `pwd` = "/local/apache244pkg/httpd-2.4.4"; then \
echo "" ; \
fi; \
list='aaa cache core database debugging filters http loggers metadata proxy session slotmem ssl proxy/balan cers arch/unix dav/main generators dav/fs mappers'; for i in $list; do \
target="shared-build"; \
if test "$i" = "."; then \
made_local=yes; \
target="local-shared-build"; \
fi; \
if test "$i" != "srclib"; then \
(cd $i && make $target) || exit 1; \
fi; \
done; \
if test -f 'modules.mk'; then \
if test -n ''; then \
echo "Building shared: "; \
if test "$made_local" != "yes"; then \
make "local-shared-build" || exit 1; \
fi; \
fi; \
fi; \
if test `pwd` = "/local/apache244pkg/httpd-2.4.4"; then \
echo "" ; \
fi
make: Fatal error: Command failed for target `shared-build-recursive'
Current working directory /local/apache244pkg/httpd-2.4.4/modules
*** Error code 1
The following command caused the error:
if test `pwd` = "/local/apache244pkg/httpd-2.4.4"; then \
echo "" ; \
fi; \
list='srclib os server modules support'; for i in $list; do \
target="shared-build"; \
if test "$i" = "."; then \
made_local=yes; \
target="local-shared-build"; \
fi; \
if test "$i" != "srclib"; then \
(cd $i && make $target) || exit 1; \
fi; \
done; \
if test -f 'modules.mk'; then \
if test -n ''; then \
echo "Building shared: "; \
if test "$made_local" != "yes"; then \
make "local-shared-build" || exit 1; \
fi; \
fi; \
fi; \
if test `pwd` = "/local/apache244pkg/httpd-2.4.4"; then \
echo "" ; \
fi
make: Fatal error: Command failed for target `shared-build-recursive'
Current working directory /local/apache244pkg/httpd-2.4.4
*** Error code 1
The following command caused the error:
otarget=`echo all-recursive|sed s/-recursive//`; \
list=' srclib os server modules support'; \
for i in $list; do \
if test -d "$i"; then \
target="$otarget"; \
echo "Making $target in $i"; \
if test "$i" = "."; then \
made_local=yes; \
target="local-$target"; \
fi; \
(cd $i && make $target) || exit 1; \
fi; \
done; \
if test "$otarget" = "all" && test -z 'httpd shared-build '; then \
made_local=yes; \
fi; \
if test "$made_local" != "yes"; then \
make "local-$otarget" || exit 1; \
fi
make: Fatal error: Command failed for target `all-recursive'
your help will be highly appreciated.
thanks
Edited by: 990086 on May 28, 2013 2:01 AM

freaks wrote:if somebody can give me an example of ssl.conf
Hi freaks; you can find an example SSL configuration file here on your system:
/etc/httpd/conf/extra/httpd-ssl.conf
It’s full of comments describing what the different options are.
Apache’s documentation on SSL is full of good stuff, including a howto.
As far as selecting ciphers go, you could do worse than following Qualys’ advice on the subject.
Paul
Last edited by prelog (2014-03-19 04:05:40)

WRT54G rewriting sip headers

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