Re: VOIP Minutes [7:73210]
Chuck, I just returned from India doing some IP Telephony training. According to the students there your description is close, but not fully descriptive. What is being discussed is something called foreign end hop-off (in telephone jargon). This practice is loosely defined as calls originating in one location being transported across a private network to a distant location, and then being handed off to a local telco for connection to a PSTN phone in the distant city. (Thereby avoiding LD charges). The regulations that govern this vary depending on where you are in the world. According to a fellow VOX instructor that was familiar with the laws in the region, it was a beheading offense in Malaysia at one time. :-(. ;-( In the USA, this practice is legal as long as the calling party and called party are performing a function related to a mutual business arrangement. (A procurement person in LA is calling a vendor contact in Oklahoma City to check on shipping schedules). However, should an employee of the same company call an acquaintance in OKC over the same facilities, the organization responsible for the private transport network (No, not the WAN SP), is in violation of FCC regulations. The private transport network is now being used by an individual consumer to perform the job normally performed by an IXC (Inter-Exchange Carrier). The IXC industry is a tariffed business, the US government wants their tariff . (And we all thought that it was just the LD SPs that were concerned about Toll bypass). ;-) Now, back to India; VOIP systems can be connected to the local switch in India. The system is not supposed to allow a PSTN phone in India to call a PSTN phone elsewhere in the world, (or anywhere India Telephone considers long distance). Basically the system connected to the local telco is segregated from the Toll bypass VOIP system to prevent those connections. But, an employee in India can call a coworker in the USA, using a VOIP system (IP phone to IP phone) without fear of criminal prosecution. It is up to the business governing the VOIP system in India to prevent foreign end hop-off at the distant end by gentlemen's agreement. Discovery and Enforcement are the main issues here. It should be obvious that enforcing rules controlling foreign end hop-off through a gentlemen's agreement is not necessarily a realistic expectation on the part of anyone. India just makes it simple; the system connected to the local telco will not be part of a Toll bypass system, period. If you want to support Toll bypass for your company by incorporating IP Telephony or any VOX system, that is perfectly okay. Just don't connect that system to their local telco. (Unless they station someone to oversee each installation now and forever, how would they know you didn't allow communications between the two VOIP systems?) As you may have noticed in this whole scenario, the called party has very little to do with the discussion. That is because the destination Carrier of a LD phone call doesn't realize much, if any, income from terminating the LD phone call. They get their money from the subscriber for providing the phone connection in the first place. They only get additional income when that subscriber makes an outbound LD call. Most telcos get little or nothing for connecting an inbound LD call. As far as buying VOIP minutes into India. There are multitudes of LD calling card vendors that use VOIP networks for transport. (Last Mile, Nexxus Telecom. etc.) But, I am not aware of any SP that would have a gateway into India Telephone, that would allow a consumer to simply connect a VOIP gateway into their network. (They have far better control, and less compatibility hassles if you just dial-in from your PSTN telephone). But the whole VOIP and IP Telephony technologies have created some very surprising business opportunities, so keep looking there may be an SP out there interested in supporting your request. This is what happens when you get tied up with laws and lawyers. You get long winded answers to seemingly short questions!;-) Bruce Chuck Whose Road is Ever Shorter wrote: Curious wrote in message news:[EMAIL PROTECTED] ... Fellows Where is the best place to buy International VOIP minutes, e.g I have a voice gateway and i want to call India on a regular phone, i have to have a voip gateway in inda to make this call or if some one already has voip gateways in india and they are selling there minutes. does it make sence to any one. ? last I heard ( and my info could be obsolete ) is that India did not allow gateways between VoIP nets and their own telco network. You can have dedicated phone links using VoIP, but those phones on the Indian side are not allowed to connect into their telco net in any way shape or form. vestige of monopoly by a state run institution or some such. -- Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=73242t=73210
Re: does anyone know the pinout on a t1 cable bet/ a [7:72069]
I'm thinking that even mentioning a straight through cable when connecting two T1 interfaces together is a good way to confuse the person asking the question. A T1 crossover is always used to directly connect two interfaces. The pin-outs are correct for the Xover. Bob by The Bay wrote: I'm thinking for straight through you meant to say: Straight through T1 you will need 11, 22, 44 and 55 wrote in message news:[EMAIL PROTECTED] ... For a standard T1: Cross-over you will need 14 and 25 Straight through T1 you will need 11, 22, 33 and 44 Thanks, Mario Puras SoluNet Technical Support Mailto: [EMAIL PROTECTED]Direct: (321) 309-1410 888.449.5766 (USA) / 888.SOLUNET (Canada) -Original Message- From: [EMAIL PROTECTED] [ mailto:[EMAIL PROTECTED] ] Sent: Wednesday, July 09, 2003 3:16 PM To: [EMAIL PROTECTED] Subject: does anyone know the pinout on a t1 cable bet/ a [7:72069] 3660 an ls1010...the interfaces on both are t1 thx in advance Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] -- Bruce Enders Chesapeake Netcraftsmen, LLC Cell 443-994-0678 1290 Bay Dale Drive #312 HO 410-280-6927 Arnold, MD 21012 efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=72103t=72069 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Loosing router config [7:69850]
What kind of output do you get after the write mem or copy run start commands? Anything? Also, after you save the config, do a show start to see if the changes have in fact been written to NVRAM. (I suspect the problem is with NVRAM, although I personally have never encountered a write-protected NVRAM on a Cisco router before, but that doesn't mean it can't happen! And your symptoms certainly sound like that is the case)! Since the existing configuration is still there when you reboot, I doubt the problem is with the config-register. I will be interested in what you find, Bruce MADMAN wrote: That's a good one! After saving the config do you see the changes when you do a write term? What is the platform and the IOS? Dave Hitesh Arora wrote: Dear All, I need some expert comments from this group for my problem. The router is in working condition and 3 links are working fine on this router. Now I need to do some changes in the router configuration. After changing and saving the configuration, I gave a reboot to the router. But I find, that router is back to the previuos old configuration. Why so?? I have checked that the config-register setting is set to 0x2102. Sh Version command also shows me the config-register is set to 0x2102. I have applied the config-register 0x2102 command also to be doubly sure that the router is picking config from the same register. Pls. help Thanks Hitesh _ Got a wish? Make it come true. http://server1.msn.co.in/msnleads/citibankpersonalloan/index.asp Best personal loans! -- Bruce Enders Chesapeake Netcraftsmen, LLC Cell 443-994-0678 1290 Bay Dale Drive #312 HO 410-280-6927 Arnold, MD 21012 efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=69879t=69850 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Loosing router config (OT rants) [7:69850] OT funny rants [7:69900]
Priscilla, The only reason I remember the old syntax is because I am basically lazy when it comes to typing! (I am also not very good at it). :-( So, if I can type wr instead of copy run start I will! ;-) In truth, all of the old write x syntax was supposed to wind up in the undocumented commands bucket in the IOS way back when I was teaching IMCR! (We won't discuss how long ago that was)! :-[ Non-intuitive huh? That is being kind! I try not to lapse into old syntax in front of folks that are new to the IOS CLI. There are enough commands to try to remember without that! But just when you think you are safe, you run into a machine that won't take copy run start! :-D Sometimes being old enough to have forgotten the Hill you are supposed to be over is cool! Bruce Priscilla Oppenheimer wrote: Bruce Enders wrote: What kind of output do you get after the write mem or copy run start Wasn't Cisco supposed to depricate write mem? I never learned those forms of the commands because when I first started learning Cisco eight years ago, Cisco said not to bother learning them because they were going away! Then yesterday I discovered that my new PIX firewall wouldn't take copy run start? Or was I making a typo or something? I had to reach into the back of my mind and come up with write mem which I thought they were going to get rid of. And I approved of that plan since it's totally non-intuitive. :-) Speaking of non-intuitive, why DO we put up with the PIX? What a beast. It took me all day to get it to do some simple forwarding. The thing is expensive, slow, and almost impossible to configure. Why do we put up with it? :-) Not being able to do copy run start took the cake. Rantings from a frustrated Cisco fan. Priscilla commands? Anything? Also, after you save the config, do a show start to see if the changes have in fact been written to NVRAM. (I suspect the problem is with NVRAM, although I personally have never encountered a write-protected NVRAM on a Cisco router before, but that doesn't mean it can't happen! And your symptoms certainly sound like that is the case)! Since the existing configuration is still there when you reboot, I doubt the problem is with the config-register. I will be interested in what you find, Bruce MADMAN wrote: That's a good one! After saving the config do you see the changes when you do a write term? What is the platform and the IOS? Dave Hitesh Arora wrote: Dear All, I need some expert comments from this group for my problem. The router is in working condition and 3 links are working fine on this router. Now I need to do some changes in the router configuration. After changing and saving the configuration, I gave a reboot to the router. But I find, that router is back to the previuos old configuration. Why so?? I have checked that the config-register setting is set to 0x2102. Sh Version command also shows me the config-register is set to 0x2102. I have applied the config-register 0x2102 command also to be doubly sure that the router is picking config from the same register. Pls. help Thanks Hitesh _ Got a wish? Make it come true. http://server1.msn.co.in/msnleads/citibankpersonalloan/index.asp Best personal loans! -- Bruce Enders Chesapeake Netcraftsmen, LLC Cell 443-994-0678 1290 Bay Dale Drive #312 HO 410-280-6927 Arnold, MD 21012 efax 443-331-0651 -- Bruce Enders Chesapeake Netcraftsmen, LLC Cell 443-994-0678 1290 Bay Dale Drive #312 HO 410-280-6927 Arnold, MD 21012 efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=69900t=69900 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Voice Over Internet [7:61467]
Neil, In broad brushstrokes the answers are sort of: 1. Variable delay is the worst enemy of Voice QoS. Queuing delays are sometimes very common in ISP-to-ISP connections. Putting Voice traffic on the Internet is a risky proposition if you have significant concerns regarding Voice quality. Making sure that each remote has significant bandwidth for the VOIP traffic is the first step. ISPs may be capable of providing some levels of QoS, but may be reluctant to do so. Most ISPs have significantly less queuing delay within their network than they do across connections to other ISPs. (VOIP across the same ISP backbone usually results in better than acceptable voice quality). It is usually the links that connect different ISPs that create the most problems. I have seen large VOIP implementations that achieved very good voice quality over a very large geographic area that was all served by one ISP. (Choose your ISP wisely). 2. VPN could hurt voice quality as some concentrators inject delay into the audio streams. Check the delay specs on any VPN concentrator you are thinking about using to see how much delay you can expect to have to deal with. 3. Solution? Most new Cisco routers and switches support QoS configurations that enhance the probability of achieving good voice quality within a network. I do not know the specs on their VPN concentrators off the top of my head. HTH Bruce neil K. wrote: Hi Guys, I have a few questions regarding implementing VoIP. 1) Can I have different remote offices run VoIP if they have (DSL access of Cable modem access) to the Internet, I mean running VoiP over internet as there wouldn't be any QoS.I am not sure about the Quality of Voice in that case.Also can the service provider of DSL or Cable provide us with some kind of QoS so that the Voice quality can be improved. 2)Will implementing a VPN solution help in running VoIP and how and what are the different solutions and what vendors should I be looking at. 3) Does Cisco have a solution for this. Thanks in advance. Neil. K. -- Bruce Enders Email: [EMAIL PROTECTED] Chesapeake NetCraftsmeno:(410)-280-6927, c:(443)-994-0678 1290 Bay Dale Drive, Suite 312 WWW: http://www.netcraftsmen.net Arnold, MD 21012-2325 Cisco CCSI# 96047 Efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=61474t=61467 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP Question AGAIN... [7:57747]
Andrew, You are asking how to set something up without all the pieces necessary to make it work. And you are describing developing a dialplan that does not adhere to common conventions. That part is okay, you can define how you route calls in-to and out-of your own voice network pretty much any way you want. The 3640 with an analog FXO is okay to act as a gateway to the PSTN. The same router with FXS VIC ALSO will allow you to connect and originate calls internally. However, you state you want to place VOIP phones on the network behind the 3640. What VOIP phones would those be? If you mean 79XX series IP Telephones from Cisco, you also need a CallManager Server to control those phones. As far as the dialing rules in this network, it is up to you what you use to direct calls out to the world. Using 9 to classify all telephone numbers that follow it as being sent to the PSTN is a convention not a rule. The only real rule is that you have to give the system the means to route a call based on what the User dials. AND, if you are handing the call off to the PSTN or any other telephone switch, you have to give that switch the call routing information (telephone number) that it needs to route the call. In coming calls are a whole different topic. Try reading one of the many VoX books that are now available. Hope this helps, Bruce Andrew Dorsett wrote: Second call for this one. I never received any answers to my question. I want to know how to setup the link between the VoIP phones and the FXO's. Basically a dialplan, but how do I route inbound calls from the PSTN to the VoIP phones? And how do I route outbound calls from the VoIP phones over the FXO to the PSTN? I would like to avoid a system that uses 9 to dial an outside line. I want to do direct dialing to the PSTN without any special steps. Thanks, Andrew On Thu, 14 Nov 2002, Andrew Dorsett wrote: Hey everyone, I'm playing with an idea. I want to get ahold of a 3640 with FXO's and interface it to the PSTN and connect to some VOIP phones on a network behind it. I have done all of my research on the CCO and have found how to configure everything for phone connection and FXO configuration. However I haven't found out how to configure dialplans to dial the outside world. I basically need one that would say all 4 digit dialed calls are VoIP phones and all other numbers are outside PSTN phone numbers. And another question that I haven't found is how to link inbound calls from the PSTN to my VoIP phones. Say I have 555-1221 for one line and I want it as line 1 on my phones, and 555-1234 as the other line on my phones. I haven't found how to map the inbound calls to a VoIP extension. 555-1221 -- | || | || | 3640 ||SWITCH|-|IP Phone| 555-1234 -- | || | || My primary info source has been: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_example09186a00800ffdcc.shtml#ITS3660 Thanks, Andrew --- http://www.andrewsworld.net/ICQ: 2895251 Cisco Certified Network Associate Learn from the mistakes of others. You won't live long enough to make all of them yourself. http://www.andrewsworld.net/ ICQ: 2895251 Cisco Certified Network Associate Learn from the mistakes of others. You won't live long enough to make all of them yourself. -- Bruce Enders Email: [EMAIL PROTECTED] Chesapeake NetCraftsmeno:(410)-280-6927, c:(443)-994-0678 1290 Bay Dale Drive, Suite 312 WWW: http://www.netcraftsmen.net Arnold, MD 21012-2325 Cisco CCSI# 96047 Efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=57765t=57747 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: FXS to key-system connection [7:57522]
Neil, I assume you are considering connecting an FXS VIC port to a key system. In that case you would connect to an FXO port on the key system. This would qualify as a trunk circuit connecting the two telephone switching devices. You are correct that this is not the best of possible connections. Dialtone is intended to be used as a signal between a machine, (a telephone switch) and a human listening to the telephone handset. It does not do well from machine to machine. In the analog world the best scenario for telephone trunk circuits is the EM interface. But, your key system nay not support that. Good luck with the FXS/FXO trunking. (Pay close attention to any calls that get refused, as it could indicate that digits are getting lost between the two systems). Bruce neil K. wrote: Hi All, How can you interface the FXS to the key system.I mean FXS provides dial-tone so I was thinking only a phone or fax can be interfaced to an FXS port. Can anyone explain. Thanks in advance for your help. neil -- Bruce Enders Email: [EMAIL PROTECTED] Chesapeake NetCraftsmeno:(410)-280-6927, c:(443)-994-0678 1290 Bay Dale Drive, Suite 312 WWW: http://www.netcraftsmen.net Arnold, MD 21012-2325 Cisco CCSI# 96047 Efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=57524t=57522 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP+QoS+xDSL+H.323Gatekeeper [7:57104]
Mark, I saw something similar to this on a customer's pilot of AVVID. The symptoms were such that if a call between IP phones was established prior to the traffic flood, everything worked just fine. If the traffic flood came first, the destination IP phone rang, but no voice packets were received by either phone, period. And, this was in a pure LAN environment! Looking at the display on the 7960s, we discovered that not one UDP packet was arriving at either phone! (The fact that the destination phone rang would seem to indicate that TCP traffic was arriving OK). Unfortunately there was no sniffer available to capture the traffic to dissect the problem. The fix was to change the parameters on the traffic generator. The customer was using Network Observer. It was a new tool for them. The traffic being generated was designated as raw ethernet frames. As soon as the traffic type was changed to TCP or any other selection, the problem disappeared. What are you using to saturate the WAN link? What I saw might trigger some observation in your network. Bruce Mark S wrote: Well, this should give you enough to chew on since voice is becoming a hot topic. I am trying to configure VoIP with QoS. Why over IP and not over ATM, you say? I have to controll the call with a H.323 Gatekeeper, and that is IP. My problem appears to be that the call setup (or maybe signalling?) appears to be delayed. The test results are as follows: If the WAN link is saturated with data packets PRIOR to establishing the voice call, the first 10 to 15 (approximately) seconds of the call are lost. After the call is established, voice is rock solid and no voice packets are delayed or lost. If the voice call is established PRIOR to saturating the WAN link with data packets, the voice call is rock solid and no voice packets are delayed or lost. Thoughts or configs would be appreciated. --Mark version 12.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname Router ! logging buffered 4096 debugging ! memory-size iomem 25 ip subnet-zero ! no ip domain lookup ! ip cef ! voice call carrier capacity active voice rtp send-recv ! no voice hpi capture buffer no voice hpi capture destination ! vc-class atm vip vbr-rt 256 256 10 precedence 5 no bump traffic no protect vc no protect group ! vc-class atm normal vbr-nrt 192 192 precedence other no protect vc no protect group ! interface ATM0/0 ip address 1.1.1.254 255.255.255.0 ip nat outside no atm ilmi-keepalive bundle-enable bundle qosmap protocol ip 1.1.1.1 encapsulation aal5snap pvc-bundle data 0/37 class-vc normal pvc-bundle voice 0/36 class-vc vip ! dsl equipment-type CPE dsl operating-mode GSHDSL symmetric annex A dsl linerate AUTO h323-gateway voip interface h323-gateway voip id Gatekeeper ipaddr x.x.x.x 1718 h323-gateway voip h323-id Gateway ip rsvp bandwidth 64 64 ip rsvp resource-provider wfq pvc ! interface FastEthernet0/0 ip address 10.200.100.1 255.255.255.0 ip nat inside speed auto ! ip nat inside source list 1 interface ATM0/0 overload ip classless ip route 0.0.0.0 0.0.0.0 1.1.1.1 no ip http server ip pim bidir-enable ! access-list 1 permit 10.200.100.0 0.0.0.255 ! call rsvp-sync ! voice-port 2/0 station-id name StaID station-id number 111222 caller-id enable ! voice-port 2/1 station-id name StaID station-id number 111222 caller-id enable ! dial-peer cor custom ! dial-peer voice 1 voip destination-pattern T session target ras ! gateway ! line con 0 line aux 0 line vty 0 4 login ! no scheduler allocate end -- Bruce Enders Email: [EMAIL PROTECTED] Chesapeake NetCraftsmeno:(410)-280-6927, c:(443)-994-0678 1290 Bay Dale Drive, Suite 312 WWW: http://www.netcraftsmen.net Arnold, MD 21012-2325 Cisco CCSI# 96047 Efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=57184t=57104 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: can 7507 support voice directly? [7:57001]
Charles, The 7507 does not accept the Network Modules for Voice (NM-1V, NM-2V) in order to give you someplace to install the FXS, FXO or EM VIC. It also does not currently support the HDV modules that would allow it to operate in the Digital Voice world. But, then on the other hand, those routers that can handle voice directly can't push packets in comparison to the 75xxs. HTH Bruce cer wrote: Hi, all, Pardon if this has been asked but archive searching did not turn up anything. Kind of a dumb question, but here goes. Can a 7507 support VOIP directly? I know that you can configure it to support voice traffic passing through it (QoS, RSVP, etc.) but can you configure it for voice as you would a 36xx?In other words, are there VIC cards (FXS, FXO, and EM) for it as there are cards for the 26xx, 36xx, etc. I searched the cisco site, but that was fruitless. I envision a scenario where I install a FXS card into a 7507, configure VOIP, connect my phones, and talk... nothing fancy. Is that possible? TIA, Charles -- Bruce Enders Email: [EMAIL PROTECTED] Chesapeake NetCraftsmeno:(410)-280-6927, c:(443)-994-0678 1290 Bay Dale Drive, Suite 312 WWW: http://www.netcraftsmen.net Arnold, MD 21012-2325 Cisco CCSI# 96047 Efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=57004t=57001 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: FXO vs other Analog Voice Card [7:56536]
I knew this question was going to lead to trouble! Okay, here goes. First thing, if the incoming call digit string will be matched to the destination-pattern of a VoX dial-peer, there is no prefix or forward command available. In a VoX dial-peer you are mapping a telephone address (the destination-pattern) to a Network address (session target with Layer 2 or Layer 3 address). This is your voice routing table! This is the logical routing plan you created when you designed this voice network. This is a static routing table you created. Why would you map an address one way, then try to modify it before you sent it to the ultimate destination? Dial plans are simply static routing tables. There is no EIGRP for voice networks, yet. Digit stripping happens ONLY on POTS dial-peers. Digits described explicitly (333 in 333) are consumed during the digit analysis. The prefix command can be used to replace the consumed digits, or the forward digits command can direct that X number of digits are to be preserved and sent across the POTS line once it is active. The EM should do what you want if you want to only send 4 digits to the other end. The VoIP dial-peer will send all digits that match the destination-pattern statement. This all depends on what you are trying accomplish. This isn't clear because you are mixing FXOs and EMs which you can't do. Maybe this will get you aimed in the right direction. Feel free to email me with what you are trying to do. the configuration may be very simple. Bruce Paul Oh wrote: Hello All, When FXO receives a phone call, it strips out corresponding called-number that matches destination pattern settings.. For instance, If call string that matches 333 , it will strip 333 and pass on last four digit. IF there is next hop voip router only sees last four digit. (Isn't that correct?. Now, how can we make that happen for EM card? (VIC-2EM)? digit-strip is enabled by default, but next router only sees 333- instead of . Help me out. Thank you. -Paul -- Bruce Enders Email: [EMAIL PROTECTED] Chesapeake NetCraftsmeno:(410)-280-6927, c:(443)-994-0678 1290 Bay Dale Drive, Suite 312 WWW: http://www.netcraftsmen.net Arnold, MD 21012-2325 Cisco CCSI# 96047 Efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=56603t=56536 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP Clarification. [7:55682]
Priscilla, Thanks. High praise considering the source. ;-) The reason the telephone number is mapped to the MAC is so that the number follows the DEVICE on the network. Please note that I stated that the IP phone identified itself to the CM by announcing its MAC address. If you leave Extension Mobility out of the equation, it is a pretty ironclad way to guarantee that the phone number assigned to the phone will be consistent even if the phone moves to a new network segment, ANY network segment that can route IP traffic to the CallManager. Bruce Priscilla Oppenheimer wrote: Great answer. Finally an explanation that makes sense for the marketing babble about IP Telephony making Moves, Adds, and Changes easier. ;-) One quesiton though, does CallManager really care about MAC addresses? Unless the receiving phone is on the same network segment as the calling phone, the MAC address won't help matters. ARP would take care of getting the MAC when it's needed. Priscilla Bruce Enders wrote: B. J. The only trick here is to remember that the User phone number is mapped to the MAC address and IP address of the ethernet interface associated with the hard phone, or the laptop in the case of Softphone. (Both are PCs running specific applications software). Whenever either is disconnected from the network long enough for link to drop, they have to check in with DHCP when they are re-connected to the network. Both also have to check in with their CallManager. During that process, they identify themselves using their MAC address, and announce their current IP address. After that, the CM can simply forward based on the IP address. This capability is one of the primary reasons that Moves, Adds, and Changes in an IP Telephony system are far more simple than in a legacy PBX environment. (The logic behind your response sounds like it comes from the legacy telephone world, which is very used to working in a very static addressing environment). Bruce B.J. Wilson wrote: Hi Vance - I too am studying All Things VoIP, and I'm curious how this would work. Say you have User A trying to call User B. User B is currently in the office. So User A dials '' which is User B's phone number (or route pattern if you want to be specific). CallManager picks up the route pattern, looks up User B's location, and forwards the call on. All is good. Now, say User B is telecommuting. How does CallManager know this? How does your RAS (remote access) server notify CM that User B's geographical location has moved? Is there something in User B's RAS (Registration, Admission and Status) setup that alerts CM to the fact that they're dialing in from home? Thanks, BJ - Original Message - From: Vance Krier To: Sent: Wednesday, October 16, 2002 4:08 AM Subject: Re: VoIP Clarification. [7:55682] Hey Stu, In simple terms, yes you are correct. However, as I'm sure you know, you need to take this type of setup with a grain of salt. If you have a decent bandwidth, low latency, consistent connection between the phone and CM, it works fine. There's absolutely no guarantees for QoS on the Internet. Now, FWIW, I use softphone on my laptop when I travel and I've gotten satisfactory results (IMO) better than 75% of the time. I always pitch this as being a *kewl* feature, but never as a selling point. I'm very, very cautious with customers over this. As long as the user using it is understanding and realizes there will be times when it doesn't work or the quality is really crappy, then typically they stay happy. Not something I'd give to Internet/computer/technology illiterate executive. I love it, by the way. Good luck, Vance Stuart Pittwood wrote in message news:200210160746.HAA10542;groupstudy.com... Good Morning all, I am just starting to look into VoIP as I have been asked by my manager to do some research and find out if there are any benifits from VoIP for our firm. Am I right in saying that if we had a solution based on Cat 6000 (or similar) switches, with a cisco VPN solution for the home workers, that users who use their laptop at home with cisco softphone or hardware phone could have their telephone extenstion follow them? Please forgive
Re: VoIP Clarification. [7:55682]
B. J. The only trick here is to remember that the User phone number is mapped to the MAC address and IP address of the ethernet interface associated with the hard phone, or the laptop in the case of Softphone. (Both are PCs running specific applications software). Whenever either is disconnected from the network long enough for link to drop, they have to check in with DHCP when they are re-connected to the network. Both also have to check in with their CallManager. During that process, they identify themselves using their MAC address, and announce their current IP address. After that, the CM can simply forward based on the IP address. This capability is one of the primary reasons that Moves, Adds, and Changes in an IP Telephony system are far more simple than in a legacy PBX environment. (The logic behind your response sounds like it comes from the legacy telephone world, which is very used to working in a very static addressing environment). Bruce B.J. Wilson wrote: Hi Vance - I too am studying All Things VoIP, and I'm curious how this would work. Say you have User A trying to call User B. User B is currently in the office. So User A dials '' which is User B's phone number (or route pattern if you want to be specific). CallManager picks up the route pattern, looks up User B's location, and forwards the call on. All is good. Now, say User B is telecommuting. How does CallManager know this? How does your RAS (remote access) server notify CM that User B's geographical location has moved? Is there something in User B's RAS (Registration, Admission and Status) setup that alerts CM to the fact that they're dialing in from home? Thanks, BJ - Original Message - From: Vance Krier To: Sent: Wednesday, October 16, 2002 4:08 AM Subject: Re: VoIP Clarification. [7:55682] Hey Stu, In simple terms, yes you are correct. However, as I'm sure you know, you need to take this type of setup with a grain of salt. If you have a decent bandwidth, low latency, consistent connection between the phone and CM, it works fine. There's absolutely no guarantees for QoS on the Internet. Now, FWIW, I use softphone on my laptop when I travel and I've gotten satisfactory results (IMO) better than 75% of the time. I always pitch this as being a *kewl* feature, but never as a selling point. I'm very, very cautious with customers over this. As long as the user using it is understanding and realizes there will be times when it doesn't work or the quality is really crappy, then typically they stay happy. Not something I'd give to Internet/computer/technology illiterate executive. I love it, by the way. Good luck, Vance Stuart Pittwood wrote in message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... Good Morning all, I am just starting to look into VoIP as I have been asked by my manager to do some research and find out if there are any benifits from VoIP for our firm. Am I right in saying that if we had a solution based on Cat 6000 (or similar) switches, with a cisco VPN solution for the home workers, that users who use their laptop at home with cisco softphone or hardware phone could have their telephone extenstion follow them? Please forgive the simplicity of my question, just making sure I am thinking along the right lines. Thanks Stu -- Bruce Enders Email: [EMAIL PROTECTED] Chesapeake NetCraftsmeno:(410)-280-6927, c:(443)-994-0678 1290 Bay Dale Drive, Suite 312 WWW: http://www.netcraftsmen.net Arnold, MD 21012-2325 Cisco CCSI# 96047 Efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=55720t=55682 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP, Call Manager etc [7:55539]
Stuart, This should get you started in your quest. http://cisco.com/en/US/customer/products/sw/voicesw/ps556/prod_instructions_guides.html Stuart Pittwood wrote: Hi all, I have been asked to look into VoIP with Cisco hardware/software, can anyone recomend any good books/resources relating to the cisco Voice Hardware and software (Call Manager etc). I am looking through the Cisco site and the information there on these products seems to be quite well hidden Thanks in advance Stu -- Bruce Enders Email: [EMAIL PROTECTED] Chesapeake NetCraftsmeno:(410)-757-3050, c:(443)-994-0678 1290 Bay Dale Drive, Suite 312 WWW: http://www.netcraftsmen.net Arnold, MD 21012-2325 Cisco CCSI# 96047 Efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=55545t=55539 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Config-register???? [7:54632]
Shawn, There one other gotcha where the break enabled bit 8 of the config-register can bite you. You are correct that having break enabled and having a console device connected could be dangerous because someone could inadvertently send a break sequence and drop the router into Rommon. Be aware that NOT having a console device connected could be even more dangerous. In a significant number of instances where bit 8 was left off in a router, (0x2002) I have seen the router drop into Rommon when a console device was connected to the console interface! It turns out that a break sequence can be very closely approximated by the noise generated in plugging or unplugging the console cable! There was even one report of an extended console cable in a large NOC that was plugged into a console and laying on the floor that was ultimately linked to a Core router intermittently dropping into Rommon. All I can report for certain is that once this cable was removed from the console connection on the router, the problem disappeared. There are a very limited set of circumstances where having the break enabled is a viable configuration on a Cisco router used for production. :-( I actually had one person tell me that he left the break enabled so that he could regain control in the eventuality that his routers went berserk. I tried to point out that if he was close enough to be connected to the console in that scenario, he could probably manage to flip the power switch instead. :-P I'm not certain the message got through,,, :-D Bruce Shawn Heisey wrote: Mark, Actually, the 'break disabled' is the default setting. It means that after rommon passes control to the IOS, you can't issue a break to get back to rommon. You can always issue the break before control is passed to IOS, regardless of this setting. If you turn this setting off, you can send a break at any time to get to rommon -- even after the router is up and running. This can be a Very Bad Thing (tm), especially if you leave something connected to the console port all the time. Thanks, Shawn Mark W. Odette II wrote: Set your terminal app's baud rate to 19200 and see if that doesn't fix ya. Also, according to the nifty Config-Register calculator (from Boson's website), the Break Key is disabled. So, you'll need to let the router boot normally, and then, via the console, go into config mode and change the config register to your desired setting. HTH's Mark -Original Message- From: Frank Lodato [mailto:[EMAIL PROTECTED]] Sent: Tuesday, October 01, 2002 10:10 AM To: [EMAIL PROTECTED]Subject: Config-register [7:54632] I broke in to a Cisco 2600 router today, but I didn't have access to my handy sheet that tells me exactly what config-register setting to type in. Instead of 0x2142 I put 0x2124. Now when I hard bott the router it gives me'JJJ^^' . Now, I've never seen this before so I'm very confused as to what to do next. I can't really type anything either so it wont take commands that I know. What did I do? How can I fix it? Help! -- Bruce Enders Email: [EMAIL PROTECTED] Chesapeake NetCraftsmeno:(410)-757-3050, c:(443)-994-0678 1290 Bay Dale Drive, Suite 312 WWW: http://www.netcraftsmen.net Arnold, MD 21012-2325 Cisco CCSI# 96047 Efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=54749t=54632 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: EM [7:54475]
Analog = 1 Digital = 24 Ismail M Saeed wrote: All, Does anyone know how many voice channels the EM interface carry ? Thanks and best regards -- Bruce Enders Email: [EMAIL PROTECTED] Chesapeake NetCraftsmeno:(410)-757-3050, c:(443)-994-0678 1290 Bay Dale Drive, Suite 312 WWW: http://www.netcraftsmen.net Arnold, MD 21012-2325 Cisco CCSI# 96047 Efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=54553t=54475 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: OT: FXO FXS terminology - comments? [7:54331]
The simplest way I know of to explain these is to take the last letter (O or S) and associate that to where it will connect TO. So, an FXO connects to an Office (PBX or CO) and an FXS connects to a Station device (Telephone, Fax, or answering machine). As Chuck suggests, if you are connecting from an O it will connect to an S, and vice versa, just like DTE and DCE. (Remembering it this way comes in handy when you are connecting two PBXs, or PBX to CO, or voice gateway to PBX or CO). OBTW, that voice gateway is a microscopic size PBX. Bruce Chuck's Long Road wrote: I did some quick looks into a couple of books I have to see what they say. Scott Keagy's book Integrating Voice and Data Networks has nothing to say about FXO and FXS in particular. The Cisco Call Manager Fundamentals book makes the rather brief assertion that FXS ports provide connection to loop-start or ground-start telephone lines, ... ( PBX ) ports, and other analogue telephone devices. FXO ports provide connection to central office ports or PBX extensions Interesting wording, and seems to apply to what I was told. Learn something new, some better way to think about things, every day. Chuck Jennifer Mellone wrote in message [EMAIL PROTECTED]">news:[EMAIL PROTECTED] ... That sounds great and makes more sense now! I always like reading your posts :-) I always confuse which device plugs into which port. I remember it like this: Plug phone or Station into FXS (where Station=S) Plug PBX/CO into FXO (where Office=0) - Jennifer -- Bruce Enders Email: [EMAIL PROTECTED] Chesapeake NetCraftsmeno:(410)-757-3050, c:(443)-994-0678 1290 Bay Dale Drive, Suite 312 WWW: http://www.netcraftsmen.net Arnold, MD 21012-2325 Cisco CCSI# 96047 Efax 443-331-0651 Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=54352t=54331 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: CCIE lab w/Voice [7:52696]
An analog MC3810 that could provide both analog and digital voice interfaces for testing. Since the MC3810 is EOS it is also CHEAP. Bruce Ben W wrote: If I were to put together a lab for CCIE RS, what would you recommend to get to cover the voice portion? Any specific voice cards? VWIC-xMFT-E1? Message Posted at: http://www.groupstudy.com/form/read.php?f=7i=52715t=52696 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
