Understanding VOIP Peers

I have a number of questions on an h.323 gateway configuration.
1.
I have a 3660 connected to a PBX via T1 CAS. I have the necessary pots peers to send digits to the PBX. If I configured CM to send a string of digits to the Gateway for an outbound call (IP Phone to PBX) is it necessary to have (some kind) of VOIP dial-peer in the gateway pointing back to CM for the call to work properly.
2.
I have an h.323 gateway configured in CM that is pointing to an IP address of a gateway that CM cannot ping, yet I can place an outbound call on the gateway via an IP phone and ring a station off the PBX. How is this possible.
3.
What is required to provide ring pack to an IP phone when placing calls out an H.323 gateway to a PBX with T1 CAS. I generally place an ( progress_ind setup enable 3) statement on my Pots peers but I thought this was for ringback on calls placed from the PSTN in. I am placing calls from IP phone out the gateway an am getting no ring back.
4.
Lastly, My 3660 dos not allow me to us the (progress_ind setup enable 3) command on the pots peers. Only options are the two below.
progress_ind progress enable 8
progress_ind connect enable 8
Will these commands do the same thing. Version of code on the 3660 is 12.1(5)T10
Thanks in advance for any insights.

Hi!
One by one...
1.
You certainly need also a VOIP peer on 3660, but not in order to point back to CM. Every established call through GW has two call legs; VOIP and POTS. One of them is incoming and the other one is outgoing. So in case of IP phone calling to PBX, VOIP peer will be incoming and POTS outgoing.
It is possible of course to mix them and use them as both - incoming and outgoing, but I prefere not to.
So for example, if PBX phones have 4 digit numbers starting with 5, you should have something like this:
dial-peer voice 1 voip
incoming called-number 5...
dial-peer voice 2 pots
destination-pattern 5...
prefix 5
port 1/1:D
As you can see, VOIP peer does not point back to CM, it receives call from CM (or anyone else, actually).
Now, if you want to send calls from PBX to CM, you will need inversed situation - incoming POTS peer and outgoing VOIP peer. Now in this case, VOIP peer will point to CM with command
session-target ipv4:A.B.C.D
2.
Most probably, you can not ping the GW because some firewall along the way (or the GW itself, or the firewall on CM if you have one) blocks ICMP packets, so the GW is not pingable. None of my machines replys to ping, but they still do their job.
3.
Hard to tell without some more info. You should play some more with this commands and see what happens.
4.
Don't know. Try, and you will see?
And you could upgrade your IOS to something a little more up-to-date, couldn't you? Often helps!
Regards,
Nikola

Similar Messages

Voip in Nokia C7 00

Did Nokia C7-00 support outgoing internet calls?

I’ve just bought C7, but I think it’s the same matter of N8. and I have tried to use Voip with Skype, Fring, Mobilevoip.
All these 3 applications crash down when trying to make calls. Wifi network is correctly configured.
Skype logs in correctly, you can chat and send sms, but not make a phone or Skype call. Latest version was released last december.
Fring logs in correctly, you can chat but not make Fring calls. I have configured the Sip client but nothing to, finally crashes the same. I have recently updated Fring with latest version.
MobileVoip is a specific program for sip calls (eg voipstunt, smsdiscount, justvoip, smartvoip, voipalot, etc). The service is provided from Betamax. It is very simple and smart, there is no need of any configuration, only username and password are needed. It cannot log in, the phone freezes for about one minute, then application crashes.
I have read something about the problem on various forums, and as I can understand voip calls cannot be actually performed on Symbian ^3. I have installed a firmware update after purchase (from 012.004 to 014.002). So all is updated.
All such 3 applications run perfectly on Symbian 5th, I had a 5800XM before, and all is still all ok.
I have not understood if this is a Nokia Symbian ^3 problem, so we have to wait an update patch from Nokia, or if it is a problem concerning each voip application (Skype, Fring, etc.) so we have to wait updates from them.
Can anyone explain me more about this please?
To tell the truth if I knew this problem I would have not bought C7, or at least I would have waited to purchase. Sometimes I travel abroad and these voip services are very useful on Wifi. I have to bring my old 5800 if want to make voip calls, and left my new C7 in a drawer awaiting future updates.
I tried to install Nokia voip frame work (downloaded here: http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin... and configuring Sip Voip settings. I successfully used this feature in a N85 two years ago. I put the same parameters, but it seems not to run, there is an error in registration. Maybe parameters are different in Symbian ^3, can anyone help me please? Or is there a site where all parameters are indicated? The latest Pdf userguide is dated 2008, but parameters seem to be the same.
I disappoint all this, it would be a good device.
Besides, just to notice, the maps provided with Ovi Maps are not updated, I have some examples of new crossroads in my city that permit me do deduce that maps are 4 or 5 years old!!! It’s a pity, the platform would be very good.
Another thing: I tried to register a new account at https://email.nokia.com, but you can only log existing users. Really, I cannot found where to register new users. Can anyone help please?
Incredible, c’mon Nokia, what are you doing?? Before you offer some services and features, and then there is an impossibility to use them!!! We are talking about smartphone. What do we buy them for? Obviously to install and use applications. Now I cannot see why I have to renounce to use what I used before in the past. And now I’m loosing a lot of time trying to use these applications and services.
They say that a good service and consumer’s satisfaction is the best advertisement. I think that the opposite is true too.
What’s happening?
If this is not the correct place where to post all these things please tell me where to post.
Thanks a lot in advance

Calls are not getting thru in Cisco voice GW for a particular Number

Cisco gateway is connecte to a PBX with an Qsig interface, for a particualr destination number the calls are not gettin estabilished.
the output of the Q931 debug :
Aug 16 16:17:46.145: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x7E05
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98396
                Exclusive, Channel 22
        Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
        Facility i = 0x9FAA068001008201008B0100A11D0202010002010080144E455453202
F204C4F4E472044495354414E4345
        Calling Party Number i = 0x2183, '8168911010'
                Plan:ISDN, Type:National
        Called Party Number i = 0x89, '18553808521'
                Plan:Private, Type:Unknown
        Sending Complete
Aug 16 16:17:46.149: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0xF
E05
        Channel ID i = 0xA98396
                Exclusive, Channel 22
Aug 16 16:17:55.709: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x
FE05
        Cause i = 0x80BF - Service/option not available, unspecified
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x7E0
5
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref =
0xFE05
The Qsig and dial-peer configration :
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vad

Hi Raj,
My name is Edson Pineiro, I understand that your problem description is in regards to failed incoming calls from a qsig trunk.
According to the received q931 setup message I can see the called party number is 18553808521 and as so the gateway should route the dnis based on the best match in destination-pattern. My first suggestion would be to ensure your outgoing dial-peers has a matching destination-pattern that matches the dialed number, for example:
dial-peer voice 1 voip
destination-pattern 1T
The T is a wild card for any digit any length
Or you can be very specific.
dial-peer voice 1 voip
destinaton-pattern 18553808521
The next suggestion would be to ensure that your incoming pots dial-peers contains 'direct-inward dial'. This is so that you don't receive secondary dial tone when dialing in, which I don't think is happening here.
Another suggestion would be to remove 'isdn overlap-receiving' from interface serial 0/0/0:23. Reason being is that the DNIS received is enbloc and not overlapping. You can clearly see that the complete e164 number is received within the setup and no further digits are needed.
But overall the disconnect cause code is 0x80BF the 80 portion is related to the source of the disconnect which is the router and BF "Service/option not available, unspecified" which is described as:
The network or remote equipment cannot provide the service option that the user requests, due to an unspecified reason. A subscription problem can cause this issue.
Any ways seems like the router does not support the protocol or type of message included in the Setup. After decoding one of the facility message:
        Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
decode -->
Facility IE first byte (protocol profile): 0x9f(Network Extentions), depends on Network Protocol Profile
**Note:
**0x91/0x9f both be used by older qsig spec, including:
**ISO 11582:1995, ETSI 300 239:1993/1995
**newer qsig spec use 0x9f only, including:
**ISO 11582:1995/Cor.1:1999, ECMA 165(4th), ETSI 300 239:2003
**see CSCeb58118 for CCM compatibility issue
NetworkFacilityExtension ::= {
sourceEntity: 0
destinationEntity: 0
NetworkProtocolProfile not present
APDU is a ROSE
0
DivertingLegInformation2Invoke ::= {
invokeID: 1793
operationValue: 21
argument: DivertingLegInformation2Arg ::= {
diversionCounter: 1
diversionReason: 1
originalDiversionReason: 1
divertingNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
originalCalledNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
redirectingName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
originalCalledName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
Looks like this is a redirected call (call forward or transfer), the redireted number is "50005998" and the other end of the PRI maybe attempting to do either a 2 B channel transfer or B channel optimization, which is not supported certain gateways or needs the use of a tcl scripts. Any ways is it possible to confirm if such features are enabled on the other end of the qsig trunk? and what the number 50005998 is assigned too. This may warrant a TAC case.
However please ensure your carry through the first three configuration changes before looking at the possible bad facility message.
Here are some good documents on ISDN, IOS dial-peers and call legs:
Understanding debug isdn q931 Disconnect Cause Codes
http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a008012e95f.shtml
Configuring Telephony Call-Redirect Features
Two B-Channel Transfer
http://www.cisco.com/en/US/docs/ios/voice/ivr/pre12.3_14_t/configuration/guide/ivrapp.pdf
Understanding Dial Peers and Call Legs on Cisco IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
Understanding Direct-Inward-Dial (DID) on IOS Voice Digital (T1/E1) Interfaces
http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_note09186a00801142f8.shtml
Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#prereq
Voice Translation Rules
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml
Let me know how you go.
Thanks again for asking the tuff questions.
Cheers
Edson

Implement Direct Inward System Access (DISA) in VoIP Environment

Hi,
May i know, is it possible to implement DISA Call in VoIP environment. If yes, how we can make it? Is it some configuration in CE Router at SRST Sites or CE Router at Main Sites? Also can you give me the information how to implement it?
As I understand DISA (Direct Inward System Access) allows someone calling in from outside the telephone switch (PBX) to obtain an "internal" system dialtone and dial calls as if from one of the extensions attached to the telephone switch. Frequently the user calls a number DISA number with invokes the DISA application. The DISA application in turn requires the user to enter his passcode, followed by the pound sign (#). If the passcode is correct, the user will hear dialtone on which a call may be placed.
Please advise me as soonest.
Thanks in advanced
Rgds,
Izazi Zainy

Giving users access to system dial tone via DISA is a security hole on PBX's and VOIP system so be careful how you use it. The following note describes how to use a TCL script and audio prompts to allow a user to call in and authenticate via an account number and PIN before they can dial an internal number. This will allow basic DISA type functions on a H323 gateway. Obviously you would also want to log the details of who made the call and when they made it, so syslog VOIP accounting is enabled to send a CDR to a syslog server.
We use an inbuilt TCL script that is inbuilt in IOS called 'clid_authen_collect'. This script authenticates the call with the ANI (Calling number) and DNIS (Called number) of the incoming call, or if this fails, it then prompts the user to enter an account number and then a PIN number. Since the call is coming in on an FXO (or FXS) port, there is no associated ANI and DNIS, so the script immediately prompts the user for the account number and PIN. We do the authentication by a local 'username XXX password YYY' command in the router config. The user keys in the account code and PIN (can use the # as a string terminator to speed the process up and if the values entered match a local username and password, it then prompts for the user to enter the actual destination telephone number.
I have also enabled syslog accounting for call detail records, so when the call completes you get a basic record of the called number and durations. If they wanted to use a full blown AAA server, they could run the authentication from this and this way keep a full log of all users calling in, and it would also log the CDR's for billing etc ...
The router needs the following audio .AU files on the flash memory :
Test#sh flash
System flash directory:
File Length Name/status
1 14097360 c2600-is-mz.122-11.T.bin
2 14150 enter_account.au
3 14869 auth_fail_retry.au
4 11510 enter_pin.au
5 52644 enter_destination.au
[14190860 bytes used, 2062068 available, 16252928 total]
16384K bytes of processor board System flash (Read/Write)
Test#
(obviously needs the IOS image but the important files are the audio prompts)
The .au files are the audio prompts that the IVR plays. These are in Sun/Next audio 64Kbps G711ulaw audio format. Use an audio editor to create the files and save them in this format.
When a call comes in on FXO port 1/0/0, you will hear a prompt to enter the account code. Key in the account number, followed by a #, then key in the PIN , followed by #. The caller will be prompted to enter the destination phone number, and this is matched on any subsequent voip or pots dial peers.
Configured user account numbers/passwords are 1000/1000 and 1001/1001
Refer to the attachment for the full router configs. Hope this helps.

VOIP over VPN need clarification

Hi,
Recently I have implemented Site-to-Site VPN between ASA and sonic wall firewall.
Problem: I can able to make call from ASA side(inside) Ip phone to sonic wall (inside) side Ip phone and vice versa and it’s ringing, But not able to hear voice. So I created VOIP over VPN configuration and applied appropriate service policy towards outside interface. But still I was not able to hear voice.
Tried below mentioned t’shot steps:
From ASA side we had two subnets (10.20.1.x/24 – Data and 10.20.2.x/24 – Voice ) and one subnet (192.168.x.x/24 ) from sonic wall side as interesting traffic ( lan to lan). When I configured site-to-site configuration on both ends my phase-1 and phase-2 came UP and can able to communicate between each other. (In interesting traffic I created two objects and bind those objects as one object-group for source i.e. ASA side lan subnet and one object for remote-Lan as destination)
My call manager is rest behind ASA and Ip phones needs to communicate from sonic wall side to inside ASA.
I can able to make call from ASA side(inside) Ip phone to sonic wall (inside) side Ip phone and vice versa and it’s ringing, But not able to hear voice. So I created VOIP over VPN configuration and applied appropriate service policy towards outside interface. But still I was not able to hear voice.
So, I done supernetting the data subnet and voice subnet into single network i.e. 10.20.x.x/16 at ASA side and applied the configuration changes (changed ACL, nonat rule, Voice QOS ACL accordingly), and I’m able to hear voice both end and I can communicate properly from ASA inside Ip phone to Sonic wall inside Ip phone and vice versa.
My question: I’m not understanding the logic how this supernetting resolved dead voice issue.
Pls clarify my question I’m bit confused on this.

It's not recommended. Although VPNs guarantee a secure pipe end-to-end, they don't guarantee latency and variations in latency (Jitter).

Wireless VoIP roaming - getting an 802.11g phone to roam with CCX / CCKM

Hello,
Our group is facing a pretty difficult issue at the moment, we are trying to deploy a wireless Avaya VoIP phone at one of our sites in Australia.
The site in question is located in Western Australia, which is on the other side of the continent to where our 5508 WLC is (in Sydney), so there is a latency of abouy 60-80ms between the sites. When testing the phone at the Sydney location, we're able to authenticate the phone via a number of methods (802.11x, 802.11x FT) and the phone can roam fine. When testing at the WA location, the 802.11x authentication method drops the call when roaming, and the 802.11x FT authentication method has about 2 seconds of dead air during roaming / reassociation.
The avaya phone supports a few different methods of "fast handoff" - CCKM and OKC - and we've only got the OKC (802.11 version of fast transition) method to work, not CCKM. The phone reports error messages when trying to connect to the network with CCKM enabled - specifically "No CCKM AP's" and "No CCX AP's".
So I have some questions...
1) Is a 60ms latency between sites a dealbreaker for wireless voice usually?
2) The voice engineer is convinced that CCX is disabled on the WLAN. My understanding of CCX (according to online Cisco docco) is that it can't be switched off. Is this the case? I know CCKM can be disabled (in the Layer 2 Authentication options) but can CCX or CCKM be enabled / disabled anywhere else?
3) The voice engineer has a point on one front - if we do a command "show ap ccx rm [APName] status", the command outputs "Beacon Request ..... Disabled." This seems to be related to CCX Location awareness - but even after switching this on in the WLC (both globally and for the WLAN), I cannot get this to enable.
3) We have had some minimal success getting the phone to connect with only CCKM Authentication enabled. The phone could not roam. When we did this, I looked at the Client information in the WLC of this device and the CCX Version read "Not Supported." Does this perhaps mean some kind of CCX compatibility problem?
4) Is there much of a difference between OKC (Fast Transition) and CCKM in terms of roaming time? I.e. are we better off trying to install a WLC at the location rather than getting this work remotely.
We're a bit lost on this. Anyone that's had experience in getting Wireless VoIP phones to roam on a wireless campus before, particularly where the WLC is remote, I'd love to hear from you.
Thanks,
Dave

Hi, thanks Steve. Aironet IE is enabled.
The encryption is news to me. We intially tried WPA2/AES & WPA/TKIP together but had no success at all. We then moved to WPA/TKIP, and we were able to get the phone to connect to the WLAN using the CCKM settings, but it doesn't roam at all & the WLC reported the Clients CCX version was "Not Supported".
I think the phone has CCX version 4. Are you certain that the phone will only support WPA & WPA2 with CCX v5? Just that this sounds a little strange, the phone's documentation claims to support CCKM, but if you're right then there's no way we can secure our phone conversations? In previous testing, we did actually temporarily drop all authentication / encryption and the roaming worked perfectly.
Thanks for your help...

How can I use Voip to communicat with my voice in Labview

i want to be able to call my computer using Voip. I know this service
cost money. I dont just want to call the computer, i want to call
labview and all labview to have access to my voice, so i can do speech
commands on it.

Hello,
Thank you for contacting National Instruments. I would like to apologize for the delay in responding to your support request. I understand that you would like to use Voip capabilities to access your computer remotely and then communicate with LabVIEW using speech recognition.
I know of a LabVIEW example file that describes the voice recognition capabilities of LabVIEW. Also, there is a link to example programs within this database entry. To access the entry, go to the National Instruments support portal: http://niweb2.natinst.com/ae/portal.htm and search for �voice recognition LabView.� Click the first entry in the list: �Voice Recognition in LabVIEW - Example - Development Library - National Instruments.� You will be able to find helpful information here.
I hope this helps! Please let me know if I can help you further. Have a great day!
Kind Regards,
Joe Des Rosier
National Instruments

Attention skype users! - MS patent to allow recording/spying of VoIP

hi everyone,
I thought Skype users using Archlinux might want to have a look at a technology Microsoft Corporation has patented on June 23 2011, that will possibly end up in skype and potentially would allow viewing of conversations using subversive techniques that abuse networking protocols to allow your conversations to be routed to a recording agent. (ie: Digital wire-tapping) Microsoft, Government and law enforcement can make use of this new technology to view/monitor/record your calls. this potentially will affect not only skype users, but any VoIP platform that Microsoft supports/develops. (ie: Xbox 360, Office 365 too)
here is a link to the article;
http://softwarefreedom.org/blog/2011/jun/29/Office-365/
for those of you who aren't familiar with www.softwarefreedom.com ... Software Freedom Law Center = Legal team for the Free Software Foundation, and they offer legal advice for free software developers - this source is legit.
here is a link to the Patent in question, itself;
http://appft1.uspto.gov/netacgi/nph-Par … 0110153809
the SFLC also has a link to this patent in their blog... If you have a look at the patent itself, it's pretty clear what the plan entails, and isn't overly difficult to read through.
Myself, i am not impressed with this news, and won't be using skype at all ever again, at home or at work, period. ~ Part of the reason for posting (aside from to share this news with those who are unaware) - i will be looking for a more secure free-alternative. If any Archers have advice on a good VoIP client for linux, by all means - feel free to post
also, i'd like to hear other Archer's opinions on the matter!
NOTE: ****Let's keep this conversation clean, if you have opinions on the matter as per the moderators request; No Microsoft bashing, attacking governments, etc. and/or drawing conclusions that may not be factual ie: stick to the forum guidelines - by opinions that isn't what i meant. i was more interested if any of you think that there are valid concerns to be discussed***
Last edited by triplesquarednine (2011-07-11 04:19:48)

Blµb wrote:
Personally I wasn't a fan of skype anyway. I understand that what really makes it that successful is the phone-calling.
From my point of view: I don't care about calling phones, since I never ever ever EVER use up the free minutes I have on my phone anyway. And I really prefer teamspeak 3 / gtalk with voip support, etc.
I currently refuse to add anybody in skype, and tell them to get jabber instead. If they have an android phone they *most likely* have a gtalk account anyway. (almost certainly actually...) (- and that works well together with "regular" jabber)
Also I'd really like to point out that "M$ probably doesn't care about us {playing <insert game>,talking about <insert topic here>}" is NOT a valid argument FOR allowing them to be ABLE to spy on you!,
which btw. is the only thing people I talk to seem to respond with...
I'm also not really a fan of skype. At work we use it, and because of that at home i have had to have it as well. Not only that, but i also do a bit of contracted work (here and there) for another company ~ whom relies on skype. everyone works from home, and we essentially have a virtual-office space over skype. My 'main' employer is in Germany right now, and once the IT department can get him to sign off on it, we will be replacing Skype with something else (we already know he is not going to be impressed with this skype news. He already pushed a corporate mandate down that we will never be migrating to Office365 either, being as we see absolutely no logic in having microsoft host all of our internal data on their servers/ the cloud. <- it's bad idea in general.
I whole heartedly agree with your last statement about the gaming perspective. whether they care about who's playing X game, talking about Y topic, is completely besides the point. for me, the concerns tend to be centered around a few issues i see:
1. invasion of privacy - which applies to both individuals, groups, and businesses.
2. creating a backdoor in their implementations of VoIP, creates more possibilities for exploitation by 3rd parties... which could have number of ill-effects.
3. who is to say that this technology won't be abused? (and not necessarily by microsoft, but in general). it's not a stretch of the imagination, to think of ways this might apply. ~ a couple quick examples, which have nothing to do with MS, nor this technology or any particular group specifically, but instead looking at how people have behaved throughout history in in general, would be ... look how airports were doing racial profiling post 9/11. or how way back when in the 50's the US GOV't targeted anyone who they suspected was a communist ~ from the average Joe, to folk singers, politicians, to movie stars. Most of whom, actually had nothing to do with communism, but there political views didn't gel with the government at the time. they were demonized.. i just feel this technology isn't good for society in general and 'could' create a conflict of interest... I'm not saying something of this nature 'WILL' happen, but the possibly for abuse is there, ya know?
and besides all of that, this whole situation just creates a lot of distrust, it's very invasive, and should be considered malware (because it is).
..maybe i'll check out teamspeak / gtalk - thanks for the tip. I've heard of both, but yours is the first review i have heard (which is helpful, so thanks!)
take care and thanks for your own insights on the matter, it is appreciated!

Small office VoIP system

Good morning everyone,
I am currently defining the architecture for a very simple VoIP network that is to be installed in a small office (about 70 VoIP extensions). Initially, we don't want to include any special feature, just the internal voice IP service and the ability to make up to 4 simultaneous external phone calls through PSTN.
We are going to acquire a Cisco 2921 Router, the SL-29-UC-K9 Unified Communications licence and a VIC2-4FXO.
I have been reading about ISR G2 licensing, but I am still not sure of fully understanding what we need for this project. My question is, does the UC licence for the Cisco 2921 include the FL-CME licence to use the Call Manager Express functionality? do we need something else? any special license for the FXOs functionality?
Thank you for your help
Pablo

Thank you guys for your answers,
Actually, it's my first contact as a developer with VoIP and UC Cisco technology... so I would like to keep it simple at the beginning: just internal voip phone calls and the possibility to ring pstn extensions. Our budget is very limited too... so I need to stick to the hardware that I told you in the previous message: 2921 + fxo card.
As far as I have read, SRST functionality is very useful when you have several nodes, to provide survival to each local node that loses connectivity with a centralized call manager. But in our scenario, with a single node... I think we can give up this functionality.
So, we only have budget to purchase any other license that could be necessary to accomplish this small project. so the key question is: does SL-29-UC-K9 Unified Communications License include Call Manager Express functionality and the ability to use de FXO card to make pstn phone calls as well? because that will be more than enough for us...
thank you very much
Pablo

Would like to understand WHY changing the IP address seems to make a difference.

I have had significant difficulty getting my WRT54G v5 to pass to the internet multiple pcs connected by cable and/or wireless. Default settings seemed to work fine to log into the LAN, but only 1 connection would go to the internet. I called Linksys tech support and they successfully got me over at least the initial hurdle so that my 2 main pcs can now function, and a 3rd seems to also get through. I've seen in this forum a BUNCH of similar posts, including some which say that success only lasts for brief periods of time.
What changed to get mine to work???.......192.168.1.1 was changed to 192.168.2.1. What I'd like to understand is.......why does that make a difference?
I am not a networking technical guru by any means, but am I a novice to technical discussion. Simply changing the address to go from not working to working does not make sense to me, and I'd like it to make sense so I can support this thing without spending hours every weekend.
Linksys, can you please explain?

The address pool of the DHCP server are just the IP addresses which the DHCP server uses to assign them to devices which ask for an IP address via DHCP (e.g. computers which are set to obtain their IP address automatically). The DHCP server automatically supplies those devices with an IP address and assigns the correct subnet mask, gateway address, and dns servers.
This has nothing to do which IP address actually have internet access or not. After the device got its IP configuration from the DHCP server this configuration works just like any other configuration you may statically configure on some other devices.
The router connects all IP addresses inside your LAN to the internet as long as the devices use the router IP address as default gateway.
If you router has an LAN IP address 192.168.1.1 and a subnet mask 255.255.255.0 these two numbers together define that all addresses 192.168.1.* are part of the LAN. 192.168.1.0 is a special purpose address. 192.168.1.255 is the broadcast address of this LAN. Everything from 192.168.1.1-254 are normal IP addresses inside your LAN (with .1 obviously allocated by your router). Operating as gateway between the LAN and the internet the router accepts traffic from any of the LAN addresses and forwards it into the internet and receives the responses and sends them back into your LAN.
For the routing/gateway part of the router it is irrelevant where the device got its IP address from. This part does not know about the DHCP server. All that matters here is that the IP address is valid and inside the LAN subnet. If the device has a static IP address or DHCP assigned IP address does not matter.
It does not matter which IP address range the DHCP server assigns. Linksys uses 100-149 as default. Some other routers assign 2-100. Some others assign 2-254. Assign whatever you like. Orderly people split their subnet into various parts, e.g. assigning 10-19 to NAS devices with static IP, 20-29 for printers with static IP, 30-39 for VOIP phones with static IP, 50-99 for DHCP, 240-249 for other networking gear. Or in any other way.
The only important thing is not to assign a static IP address inside the DHCP server address pool. The DHCP server checks whether an IP address is used before it assigns it. If, however, the device with the static IP address inside the DHCP address pool is temporarily down, the DHCP server does not know that the IP address is used otherwise and thus may assign this address to another device. Now, once the device with the static IP address comes up again you have an IP address conflict. Therefore, never assign a static IP address inside the DHCP server address pool.

Lync in combination with voip

We are using Lync but just installed a new VOIP system called SWYX
Everything is working fine except if a user picks up his phone (usb Phone) Lync is also popping up behind the screen of the SWYX software client.
Problem is that when Lync is also popping up this makes a loud sound through the usb phone, if you just minimize Lync the sound is gone.
Work around for this problem is go to Lync options - Ringtones and sounds and disable the option play sound in Lync, if you do so and you pick up your phone and Lync is popping up, no annoying sound anymore but when a user calls someone through Lync they
have no ringing sound on Lync.
So my question is can you disable a reg key that Lync is not popping up when picking up your usb Phone
tried already to set audio settings from Lync to a headset an from SWYX to handset but if I pick up handset,Lync is popping up.
I know you can also use Lync as a voip system but we use SWYX now in our European offices and in the US we have only Lync now for the moment, story is to long to explain why using SWYX in Europe, main question is how can I prevent Lync from popping up when
I pickup the usb phone.
THX all for your support

Hi,
You may custom your own lync client to archive this feature. I suggest you post the issue on
Lync MSDN forum and more developing expert will help to verify if this can be achieved. Thank you for your understanding.
http://social.msdn.microsoft.com/Forums/en-US/communicatorsdk/threads
Kent Huang
TechNet Community Support

Cluster nodes discover peers outside cluster domain

All, cross-posted from the ColdFusion Server Administration
forum:
I've run into an issue with CFMX7 clustering on a subnet with
multicast disabled. In our configuration, we have two physical
Windows Server 2003 Enterprise Edition servers hosting nine
ColdFusion MX 7 Enterprise clusters. Each server hosts one of two
instances in a cluster. i.e.:
server1 [1.2.3.4] - instance1-1 <- cluster1 -> server2
[1.2.3.5] - instance1-2
server1 [1.2.3.4] - instance2-1 <- cluster2 -> server2
[1.2.3.5] - instance2-2
server1 [1.2.3.4] - instance3-1 <- cluster3 -> server2
[1.2.3.5] - instance3-2
server1 [1.2.3.4] - instance4-1 <- cluster4 -> server2
[1.2.3.5] - instance4-2
server1 [1.2.3.4] - instance5-1 <- cluster5 -> server2
[1.2.3.5] - instance5-2
server1 [1.2.3.4] - instance6-1 <- cluster6 -> server2
[1.2.3.5] - instance6-2
server1 [1.2.3.4] - instance7-1 <- cluster7 -> server2
[1.2.3.5] - instance7-2
server1 [1.2.3.4] - instance8-1 <- cluster8 -> server2
[1.2.3.5] - instance8-2
server1 [1.2.3.4] - instance9-1 <- cluster9 -> server2
[1.2.3.5] - instance9-2
My first step in enabling peer discovery was to add the
unicastPeer attribute to the ClusterManager service under each
instance.
e.g. jrun.xml on instance1-1:
<service class="jrunx.cluster.ClusterManager"
name="ClusterManager">
<attribute name="bindToJNDI">true</attribute>
<attribute name="enabled">true</attribute>
<attribute
name="clusterDomain">cluster1</attribute>




<attribute name="unicastPeer">1.2.3.5</attribute>
<service class="jrunx.cluster.ClusterDeployerService"
name="ClusterDeployerService">
<attribute
name="deployDirectory">{jrun.server.rootdir}/SERVER-INF/cluster</attribute>
<attribute name="deactivated">false</attribute>
</service>
</service>
e.g. jrun.xml on instance1-2:
<service class="jrunx.cluster.ClusterManager"
name="ClusterManager">
<attribute name="bindToJNDI">true</attribute>
<attribute name="enabled">true</attribute>
<attribute
name="clusterDomain">cluster1</attribute>




<attribute name="unicastPeer">1.2.3.4</attribute>
<service class="jrunx.cluster.ClusterDeployerService"
name="ClusterDeployerService">
<attribute
name="deployDirectory">{jrun.server.rootdir}/SERVER-INF/cluster</attribute>
<attribute name="deactivated">false</attribute>
</service>
</service>
. . . and so on for each instance and cluster. This is where
the problem begins. When I start the instances, every instance
discovers every other instance as a cluster peer, regardless of
cluster domain.
Another forum user suggested using host:port, where port is
the JNDI listening port. That doesn't work. Using the Jini
listening port, however, does work, e.g.:
<attribute
name="unicastPeer">1.2.3.4:4160</attribute>
That presents another problem. The Jini listening port
defaults to 4160. If 4160 is taken, a port is chosen at random.
I can't find documentation on setting a static Jini listening
port, if that's even the correct action to take.
Thoughts?
From what I can tell, the version of Reggie (the Jini lookup
service) shipped with JRun only supports setting the unicast
listening port programmatically. Reggie is started by
jrunx.cluster.ClusterManager.init--actually, the private method
startLookupService--and JRun doesn't appear to ever call Reggie's
setUnicastPeer method.
Assuming we can't tweak Reggie, I guess a more appropriate
question is how do we get JRun's RMI service (?) to honor
groups/domains in a call to getPeers? I'll cross-post to the JRun
forums and investigate JRun Updater 6.
Trev

. . . and it appears I'm exposing my ignorance of Jini in
general. :-)
If I now understand the Jini discovery process correctly, a
multicast request includes one or more service IDs and one or more
groups. The registrar will respond if and only if its service ID is
not in the request and its group memberships exactly match one or
more of the groups in the request.
A unicast request includes nothing more than the protocol
version, and the registrar will respond as if a valid multicast
request had been received.
In both cases, the response packet includes a marshalled copy
of the ServiceRegistrar object and the names of all groups of which
the registrar is a member.
Without looking at more of JRun, I'm guessing that in some,
if not all cases, either JRun's discovery implementation assumes
that any response from a unicast query is valid, regardless of the
server IDs or group names received, or the logic that sorts out the
response isn't 100% correct.

Can I add VOIP Gateway SPA2100-SU to the DMZ of Router BEFSR41 Ver 4 ?

I am having multiple problems with dropped calls; need to reboot my LinkSys VOIP Gateway and so on, and my VOIP provider has suggested the following: "Add your Gateway (the physical device I guess) to the DMZ of your router." We are trying to tell the Router to give this VOIP Gateway, which gets its IP address via DHCP from the router, to essentially give this Gateway carte blanche to any port or destination it wants. I have NO idea how to do this on the BEFSR41, which is a CA model with Version 4 appended to its model number ? Can anyone send me how to advice ? - Mike BRYAN Ottawa Canada PM me for my email.
(Edited post for guideline compliance. Thanks!)
Message Edited by JOHNDOE_06 on 01-22-2008 02:54 PM

First thing is to assign a static IP on the WAN / Internet side of your SPA-2100 so that you can set that IP to the DMZ of the BEFSR41. If you are not sure how to assign a static IP address on the spa2100 or how to open the web interface of the spa2100 and befsr41, then you better refer to the KB articles from www.linksys.com/kb. Just type in something like spa2100 or befsr41.
By the way, you may also enable NAT mapping and NAT Keep alive in your spa2100 aside from opening / forwarding ports in the befsr41 (check out Answer ID 5242). Lastly, enabling Send Resp To Src Port in the spa2100 is another option to try (this is under Voice > SIP > NAT Support Parameters).
It really requires much reading for us beginners to fully understand these devices. I am happy that I learned how to use the KBase site of Linksys since it gives me useful info about their products.

Understanding 5505 firewall-site to site and internet traffic

Hi,
My question is mutli-faceted. I apologize for the lengthy intro here but i think the info is necessary to understand where I am headed in this.
I am new to the cisco 5505. I have had very limited exposure to a 5510 that was preset. I have managed to make modifications to it here and there, but dont completely understand how it was put together. I learn by watching, listening, and gleaning what I can from others. I have had no formal training in CLI, but I have learned some of the commands. I know enough to be dangerous, but I respect my limitations.
That being said, I have been charged with setting up a 5505 at a remote site. I need to accomplish several things. Our ultimate goal is to use this device as a site to site with the 5510 at the corporate office. However, I need to accomplish this in baby steps, test, test real users and then maybe convert in full. Where I could outsource this in its entirety, that would preclude me from learning so I can address this in the future on my own.
We need to have this in place by the end of February 2013.
Currently the remote site is connected via a very slow (by todays standards) T1 line on a MPLS. Stable. Works, but slow. All internet traffic as well as work traffic is routed through that connection. We have added a 50mb cable connection (with static ips) to the office. First we want to set up the 5505 so that it can be used as follows:
1, Internet traffic can be routed out through this device and all other "work" traffic routed through the MPLS.
2, Test using this connection as a route out to the internet AND use it as a site to site VPN connection to the home office. (or anyconnect vpn)
          I need to be able to have users in both environments. IE, some still using step 1 and some starting to use and test step 2.
3, long term, use this as the main connection per number 2, but add the IP address so that if the cable connection drops, the office can access internet via the VoIP T1 line as a life line.
In all cases, I dont want internet going through the home office as it currently is traveling.
I have done a lot of searching but so far have come up empty with answers.
Question 1:     (This one probalby shows my ignorance the worst) - in using the 5505 firewall, will it segregate normal internet traffic from the VPN traffic when used by the workstation? Using the Gui, I didnt see where this was necessarily happening. Do I need to use CLI language (and what) to make this happen? Or is that a basic function that happens during the setup of the firewall using the GUI. Do I need to do some sort of "split tunneling"?
Question 2:     Do I use this device as the Default gateway for both step 1 and 2/3) for normal use and then change the gateway on the Pcs to the VoIP network during emergency use,(that would bypass the firewall though or is there a way to have it route to that router if there is no connection through the Outside port? Or as long as I have some access to the device, can I make a change remotely to help accomplish this failsafe?
Question 3:     We have 25 Anyconnect VPN licenses. Should we use these and not the Static site to site, if so, why or why not? They dont need to be used at all.
Question 4:     In setting up the VoIP line for backup, would using that on the "DMZ" connection help in making this viable so that the device could still ultimately control the internet traffic?
Question 5:     In setting up the VPN connections, unless i am getting the two methods confused, I will need the 5505 to hand out IP addresses for the vpn connection. I see in using a class c schema that i can use 92.168.0.0 to 192.168.255.0. So for instance, I could use 101.1.20.0 for the inside network Vpn addresses?? I need to stay away from 192.168.0.0 networks as we use that in our normal structure.
Reasons for setting this up:
Slow speeds over the T1.
increasing demand for Skype, Video conferencing etc that the T1 pipe couldnt adequately handle
Lack of backup pathways for downed connections - ie, backhoe chopping through wire at a construction site).
I read through the Getting started guides on both the 5510 and the 5505 and feel I can likely get the site to site setup (I have a list of all the Ip addresses i need for inside networks and outside networks etc.
additional notes:
I have to email ATT anytime I want a change made on the MPLS router, so doing as little to that as possible would be good.
I will be onsite for testing at the end of February and will have direct access to the home office via other methods to work on the asa5510 if any additional work needs to be done on it once i am onsite.
Thanks for taking the time to read through all of this. please forgive my lack of knowledge...
Dave

Thanks for getting back to me and so quickly!
1) I am not sure if I understand the “ACL” portion of your question, but this is how I want to access info via the VPN tunnel:
192.168.D.0 inside(NJ) to outside 5505 - 12.175.X.X to outside 5510 - 12.200.X.X to inside network (HQ)192.168.X.0. Routes are needed to find subnets 192.168.A.0, 192.168.B.0 and 192.168.C.0. The default gateway to those subnets right now is: 192.168.X.XX4 inside of HQ. This would be so that the NJ office could find resources of the other offices if needed. This will change as we wean off the MPLS. Inside the ASA 5505, the IP addresses are 192.168.D.0 for data, 10.X.X.0 for the Phone system. All other traffic would be sent out through the internet. Phone system uses the XOcomm conection to route phone traffic.
2) I did some reading on SLA. Thanks for pointing that out. For purposes of learning here, I am showing this as 12.175.XXX.XXX for Comcast and 12.200.XXX.XXX for XO comm.
4) I guess I would use an Outside 2 as that makes sense, in description, I would label them “ComCast” for outside 1 and “XOcomm” for outside 2.
5) I am still not sure I understand this part. Are additional IP addresses needed for the Site to site VPN to talk to the local hosts, or will it use the IP addresses assigned by the local server?
Next Steps
1-         Configure the ASA5510 for the 5505 connection
2-         Configure the ASA5505 for the 5510 connection
3-         Configure SLA for Comcast and XOcomm outside connections
4-         For this I need help….I think this is from step 1, but I need help to configure the internet to be segregated via my question from #1. Have I given enough information to do so? Please advise on ACL entries, and route statements needed so that NJ can talk to all the offices when using this connection, not just the Headquarters.
Thanks
dave

Help for basic VOIP function on 2821

Hi,
I'm new to VOIP. Sorry for this simple question.
We have two Cisco 2821 routers, two 7960G IP phones.
Each 2821 router has PVDM2-8 DSP module, HWIC-4ESW 4 port swtiching card, IOS 12.4(22)T3.
No any Call Manager software is loaded.
Can anyone point me to a sample configuration/how-to so I can make phone call from Router A to Router B?
Something as below:
Phone A <----> Router A <---- internal IP network ----> Router B <------> Phone B
Please note we don't have Call Manager software and we don't need fancy phone call features.
Thank you very much!
-Andrew

You need two analog phones. The VOIP phones you have 7960G will not work. They are dumm terminals. They need something to tell them what to do. A call agent (CUCM, CCM,) or any other call agent that can control cisco VOIP phones eg asterics.
When you do get 2 analog phones configure voip and pots dial peers
example
Configure this on both routers with the IP pointing to each other
dial-peer voice 10 voip
destination-pattern 5678 --No the analog will send to the other sides fxs port to ring the analog phone
session target ipv4:10.10.10.1 ------ ip address of the other router.
dial-peer voice 11 pots
destination-pattern 234 --- the number that will ring the phone plugged here.
port 1/0 -------fxs port where your analog phone is plugged.
Dial away.
You might find ths url helpful.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800e00d0.shtml

Understanding VOIP Peers

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