VOIP
Hi I am looking for Voice over Ip configuration for 3COM AS 400 Series of router.3 Com had already stopped this product , if any one has some comman set or some type of notes on that please let me know.. Himanshu _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP************
Hi All, I am working on VoIP solution, i am working out for E &M solution but i donot know how to calculate the memory required in router depending on the no of channels. also with E & M card how many simultaneous connections can happen. Can any one help me in giving url address where this info will be available. Thanks in Adv. __ Do You Yahoo!? Yahoo! Shopping - Thousands of Stores. Millions of Products. http://shopping.yahoo.com/ _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP
Does anyone have experience with VoIP on the 2600 series routers? I have a customer that wants to connect two building via GIG fiber and I have some questions on how to get the voice between the two buildings. If you can help let me know and I will send some diagrams and more specific questions. Thanks, Alex
VoIP
Hello everybody, I need to use a network for doing Voip and I have the RTT history for the network. I can't change the network (I do not administer it , and I can't change anything). The routers in the network don't do particular queuing(I think they all do FIFO, but it is possible some of them will do RED). Do you know of some table that gives me the correspondence MOS(subjective Quality of voice) and network condition (Medium RTT,percent of packet loss, maximum RTT, variance of end to end delay,.) or an indication of what range should the parameter of the network condition should have to do a decent or good VoIP conversation. -- Best regards, mamo mailto:[EMAIL PROTECTED] ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP
Hi everybody, I have 2 offices with a leased line between them in Frame-Relay. At one office I have got a Nortel PBX and at the other side an little Alcatel PBX. For the routeur I thought using a 3640 and a 1750. Here are my questions: How can i configure the routers(Bandwith, frame-relay trafic shapping, Be, Bc, CIR)? How can I avoid adding a card in the PBXs? Can I directly link my router with the telephone jack? What card should I use on the routers? Should i configure the PBXs? Thanks in advance. Arnaud __ Do You Yahoo!? Yahoo! Photos -- now, 100 FREE prints! http://photos.yahoo.com ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP
Hello , Has any one connected FXO to PABX and E&M on the other side . I cannot make a call from E&M to FXO but FXO to E&M works fine . Any ideas ? Thanks ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP problem
Hello, i'm planning one VoIP instalation and i don't want to use any PBX. I will use one 2640 with HDV but i came to this question and i don't know the answer: how can i connect to my network fax and modem's? What can i do? Antonio _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP Security
Hi Guys, in my company will make the VoIP. The solution that we want do is "full IP phones". In Other terms, we don't want have the PBX central!! In my organitazion, there is a firewall that we protect of the unauthorized access. The location of the firewall is in Germany (Emea). The location where we want make VoIP is in Rome. My question it is, how can i protect my Data network of the possibile access from Voice Network? Do i have to insert an other Firewall in Rome? It is enough that IP Phone and the PC for every user are in two different VLAN ? Regards, Carmelo P.S. Are there books that explain the security for VoIP? _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP Lab
Could someone tell me what is needed to use cisco's voice over ip . I would like to connect 5 phones in my office but i can not seem to find the necessary info to help me choose the components , I figure i should need 1 cisco ics7750 1 catalyst switch Where could i find more info. I have checked cisco.com but t become more confused..Does anyone have a case study or a design guide.. Please let me know jim k _ Get your FREE download of MSN Explorer at http://explorer.msn.com _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Voip Problem
Hai, I was in an interview when the manager asked we about this problem related to VOIP. (A) VOIP (B) Phone (+) PABX (+) Router <---> Router (+) PABX (+) Phone System System FR 128kb (+) PSTN (+) Phone System (Copper) (C) Situation : a) User from Phone System (A) connected to the Office PABX then connected to the router plus NT using Frame Relay connection to user Phone System (B). During the test call, the user A can hear echo from Phone B. where shall you start to check on the problem?. b) The second problem is from A --> B no echo but to C there is echo. When test from B--> C there is no echo, just from A--> C the echo come. What may be the problem?. For problem a) with the given assumption with out the PABX nor the router model but assuming it have a VOPI card install, I suggest the the problem may relay on the Frame Relay line and the router it self. I believed debug command will help here, but on the PABX site is there any one can give a suggestion on it?. I For problem b) I believed the problem lies between the B--> C since C is using Analogue going into the Pabx, converted to Digital and then been tunnel into the router(B) then decompress in router(A) thus the end user got the echo problem. Will the be the case when we used the PABX to call a local call from A through B pabx and then goes to the normal PSTN line. Or it may due to my country PSTN condition?. Is there any command that we can do on the either router to check why the echo is there. any help on implementing VOIP, connecting router to PABX would be much helpful. Thanks Have a nice day Nuurul Basar Bin Mohd Baki Customer Service Executive IT Help Desk Shell Services International Sdn Bhd (432283-T) MCP CCNA Tel : 603-20803214 Fax : 6032512957 Email : [EMAIL PROTECTED] _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP config
Hi All, Need some help in configuring VoIP I am testing the loopback connectivity between my router and EPABX by dialing a local extension number. As Soon as I dial the seizing code I get connected to the router. When I dial the destination pattern my call gets transferred to the router,s next port When I dial the local extension i do not get a response. I am using tone dialing,the Interface model is Type- 5 E& M Type of Signalling is Immediate Thanks for your clues in advance. Amit __ Do You Yahoo!? Thousands of Stores. Millions of Products. All in one Place. http://shopping.yahoo.com/ _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP Troubles
All right guys I need some help I have been working with Cisco for a while now on a VOIP issue. The problems lies in both call disconnects and voice distortion. We have followed all the steps for traffic shaping (QOS) and rtp header-compression but these do not seem to help. We have 150 remote sites all running 2600's with FXS modules that all come back into the host site where we have 2 7206's. Each of these links are 56k frame-relay links with 16k CIR running very few applications mostly small transactions and Citrix clients. The call must then traverse two internal Ethernet segments, routing through our 6509 backbone switch and then into a 3640 before hitting the PBX. Cisco seems to think that we need to increase our bandwidth to support the voice traffic, however, that is not something I have been able to sell to the "powers that be". We sold this idea on cutting cost and in our estimations for upping the CIR to even 32k will be significant cost increase. Right now I am shaping to 16k with an 8k committed burst so at any one time I should be able to burst to 24k. Assuming that I am able to burst to port speed (56k) why would I have call distortion unless there is some latency coming through the ISP's switch? We also have another company site that also comes back in this way and we have no problems with those calls.O.K. that being said (and hopefully not too confusing to follow) here come the questions: 1. Has anyone else implemented VOIP in slow links successfully? 2. Is anyone else having QOS problems with their VOIP implementations? 3. Do I need to prioritize the voice traffic through the local network? 4. Has anyone tried turning off traffic shaping and letting the voice and data compete for bandwidth? Thanks in advance for your feedback! Thanks, Chris Boyd, CCNANetwork SupportAlex Lee, Inc.120 4th Street SWHickory, NC 28601(828) 323-4103http://www.alexlee.com
Re: VoIP
Don't take this the wrong way, but I have a couple of questions. Since you're being compensated (paid) by your customer, how do you intend on compensating someone in this group for helping you? If you don't compensate someone for helping you, how do you know that your helper didn't just give you enough rope to go hang yourself in front of that customer? i.e. They gave you enough information to think you know what you're doing but not enough to be successful. And since money is involved, and hence the concept of harm (legal definition). Do you have adaquate liability insurance to protect not only yourself but also the person who helps you? That should just about cover it, thanks. Rodgers Moore ""Alex Madjeski"" <[EMAIL PROTECTED]> wrote in message 002301c04c37$648f5080$[EMAIL PROTECTED]">news:002301c04c37$648f5080$[EMAIL PROTECTED]... Does anyone have experience with VoIP on the 2600 series routers? I have a customer that wants to connect two building via GIG fiber and I have some questions on how to get the voice between the two buildings. If you can help let me know and I will send some diagrams and more specific questions. Thanks, Alex
Fw: VoIP
Oops was supposed to go to list Has been a really long day - Original Message - From: Laurel Redd To: Rodgers Moore Sent: Saturday, November 11, 2000 9:52 PM Subject: Re: VoIP Oh good grief. I think people who have the good sense to ask a question when they are over their heads should be applauded. I have been taught over the last 6 months that if I have no clue or can't figure it out in a certain amount of time that it is wise to ask for insite and help. *Rant Complete Thank You, Morgan - Original Message - From: Rodgers Moore Newsgroups: groupstudy.cisco To: [EMAIL PROTECTED] Sent: Saturday, November 11, 2000 5:41 PM Subject: Re: VoIP Don't take this the wrong way, but I have a couple of questions. Since you're being compensated (paid) by your customer, how do you intend on compensating someone in this group for helping you? If you don't compensate someone for helping you, how do you know that your helper didn't just give you enough rope to go hang yourself in front of that customer? i.e. They gave you enough information to think you know what you're doing but not enough to be successful. And since money is involved, and hence the concept of harm (legal definition). Do you have adaquate liability insurance to protect not only yourself but also the person who helps you? That should just about cover it, thanks. Rodgers Moore ""Alex Madjeski"" <[EMAIL PROTECTED]> wrote in message 002301c04c37$648f5080$[EMAIL PROTECTED]">news:002301c04c37$648f5080$[EMAIL PROTECTED]... Does anyone have experience with VoIP on the 2600 series routers? I have a customer that wants to connect two building via GIG fiber and I have some questions on how to get the voice between the two buildings. If you can help let me know and I will send some diagrams and more specific questions. Thanks, Alex
VoIP/voFR
Has anyone used the Num-Exp command? If so, how many Num-exp commands can a Cisco router handle? We are deploying a huge VoFR network. The customer has a private numbers scheme such as 745-2345 (2345 is the extension), Each location has 3 prefix numbers when they dial internally. But to dial outside, the extension is keeped and the front 3 digits are replaced with a real number such as 212-345-2345. So I want to use the command num-exp 745... 212345... The question is, how many time can put that global command in a Cisco? How will it affect the router?I have about 200 to 400 sites. I would rather have the PBX do what I am asking for, but customer does nto want the PBX translating the numbers. Thanks! Evan You - CCNA
voip details
can any one get information on voip ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP question...
I am currently reading throught the CVoice notes and have just passed the section on RSVP. You can reserve bandwidth with this protocol to give to active voice channels for the duration of the call - this much I understand, no problem. What isn't clear in the notes is whether or not the bandwidth is available to other traffic if the voice traffic is not using it all - ie: during silence suppressed moments. Anybody know..? Thanks. keith. ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP books
Can anyone recommend a good VoIP book which covers content of the CVOICE exam? Thanks! Russ.. ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
AS5300 voip
Hi All I have a E1 line with a telephone number 4561743 in Singapore site. I got a AS5300 in Singapore and Hong Kong. I would like all call from the E1 to terminate it to a specific telephone in Hong Kong (55666777). I would like to configurate it as when anyone call in Singapore As5300 with the number 4561743, it will auto forward the call to Hong Kong number(55666). That mean after dailing 4561743, i do not need to dial the Hong Kong number. That mean once all call will auto forward to 55666777 when 4561743 is dial in singapore. Pls direct to any sources that can help me to solve this. Thank , with regards chan ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP(urgent)
Dear all, Anyone having running config of VoIP using Cisco AS5300 router Pls. mail it to me as I am in a great need. P.S: Right now I do not have E1/T1 line. I wanted the AS5300 to be used as access server. I mean I want AS5300 to switch the calls from one remote to other remote. The remote sites are already configured for VoIP. Thanks in advance Hitesh ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP
Wow, this sounds familar!!! Yeah, check your wiring. We had the same issue at one of our client sites. Make sure the PBX is connected to the E&M correctly (wiring). -Brad ""pinoal"" <[EMAIL PROTECTED]> wrote in message 8jp8at$qku$[EMAIL PROTECTED]">news:8jp8at$qku$[EMAIL PROTECTED]... > Hello , > > > Has any one connected FXO to PABX and E&M on the other side . > > I cannot make a call from E&M to FXO but FXO to E&M works fine . > > > Any ideas ? > > > Thanks > > > > > ___ > UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html > FAQ, list archives, and subscription info: http://www.groupstudy.com > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > --- ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP problem
You will need to install the FXS voice cards into your router for as many as devices you need to connect. But then this will be the limitation on your router. Rajeev ALMEIDA Antonio Jose wrote: > Hello, > i'm planning one VoIP instalation and i don't want to use any PBX. I will > use one 2640 with HDV but i came to this question and i don't know the > answer: how can i connect to my network fax and modem's? What can i do? > > Antonio > > _ > FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VOIP Troubles
Wait for 12.1.5(T) Cisco IOS. [skipped] follow) here come the questions: 1.Has anyone else implemented VOIP in slow links successfully? 2.Is anyone else having QOS problems with their VOIP implementations? 3.Do I need to prioritize the voice traffic through the local network? 4.Has anyone tried turning off traffic shaping and letting the voice and data compete for bandwidth? Thanks in advance for your feedback! Thanks, Chris Boyd, CCNA Network Support Alex Lee, Inc. 120 4th Street SW Hickory, NC 28601 (828) 323-4103 http://www.alexlee.com -- --- WBW, xander CCNP+Voice, CCIE very soon :-) _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP: Need help
Hello People! Need your valued advice. VoIP, though frequently heard, I have never had the opportunity to work on it. Books are rare here in Bangladesh and only source of information is the Internet. So could anyone please let me know the useful resources on the web(just want to be familiar with the technology). My requirement is simple: 1. What is the underlying technology of VoIP? 2. What sort of Cisco Hardware do I need? Thank you in anticipating. Merry Christmas and happy New Year to you all. Regards Ishtiaque _ Get Your Private, Free E-mail from MSN Hotmail at http://www.hotmail.com. _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: VoIP Lab
It kinda depends what you want to do. Are you doing straight VOIP; i.e. connecting standard phone to travel over an IP backbone, or do you want to do IP telephony, i.e. The phones actually have an IP address. Are you connecting to a pbx? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of jim klane Sent: Thursday, December 28, 2000 7:30 AM To: [EMAIL PROTECTED] Subject: VoIP Lab Could someone tell me what is needed to use cisco's voice over ip . I would like to connect 5 phones in my office but i can not seem to find the necessary info to help me choose the components , I figure i should need 1 cisco ics7750 1 catalyst switch Where could i find more info. I have checked cisco.com but t become more confused..Does anyone have a case study or a design guide.. Please let me know jim k _ Get your FREE download of MSN Explorer at http://explorer.msn.com _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP Lab
While at AT&T I took in some training on Voice Over IP and according to that, there needed to be a PBX and a Voice Over IP module from Cisco. In the Cisco 2620 or 2621 (do not remember which one was used) the module was a card. On the Test and Turn-up things went smooth. jim klane <[EMAIL PROTECTED]> wrote: Could someone tell me what is needed to use cisco's voice over ip . I would like to connect 5 phones in my office but i can not seem to find the necessary info to help me choose the components , I figure i should need 1 cisco ics7750 1 catalyst switch Where could i find more info. I have checked cisco.com but t become more confused..Does anyone have a case study or a design guide.. Please let me know jim k _ Get your FREE download of MSN Explorer at http://explorer.msn.com _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] = = = = = = = = = = = = = = = = = = [EMAIL PROTECTED] = = = = = = = = = = = = = = = = = = - Do You Yahoo!? Yahoo! Photos - Share your holiday photos online! _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP Lab
It really depends on what you want to do. Using an ICS7750 sounds like you want to do something more like IP Telephony. So in addition to the server and a switch, you'll have to use get a handful of Cisco IP phones, not your typical office phone. If you're just interested in voice over IP, you can use a couple of routers with FXS cards (1700s work well for this). These will accept your standard analog phones and provide dialtone. Set the routers up back to back and make calls over your little mock-up wan. Tommy > jim klane <[EMAIL PROTECTED]> wrote: > Could someone tell me what is needed to use cisco's voice over ip . > I would like to connect 5 phones in my office but i can not seem to find the > necessary info to help me choose the components , > I figure i should need > 1 cisco ics7750 > 1 catalyst switch _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Help for VOIP!
Hi, members, I have one question about VOIP. Actually, I have two 3600s with three exact modules- Voice card(NM-1V), ATM (1A-OC3MM), and FastEthernet, but no serial ports. The NM-1V has two FXS. Based on these, how do I set up for VOIP lab? I have several 2503, 2513, and 2514. Because I don't have serial ports on 3600s, do I need to use FastEthernet ports or use ATM interface? Please let me have cabling info too. Thanks for your help. = Gene Park [EMAIL PROTECTED] __ Do You Yahoo!? Yahoo! Photos - Share your holiday photos online! http://photos.yahoo.com/ _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
help for VOIP!
Hi, members, I have one question about VOIP. Actually, I have two 3600s with three exact modules- Voice card(NM-1V), ATM (1A-OC3MM), and FastEthernet, but no serial ports. The NM-1V has two FXS. Based on these, how do I set up for VOIP lab? I have several 2503, 2513, and 2514. Because I don't have serial ports on 3600s, do I need to use FastEthernet ports or use ATM interface? Please let me have cabling info too. Thanks for your help. = Gene Park [EMAIL PROTECTED] __ Do You Yahoo!? Yahoo! Photos - Share your holiday photos online! http://photos.yahoo.com/ _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Help regarding VOIP
Hai, I will be changing job in a near time and now looking for info regarding VOIP and ICT( Information Communication Technology). It have been 4 month since I left Cisco stuff and fell a bit left behind. Can some give me some url regarding VOIP implemntation on Cisco and other routers. Thanks have a nice day Nuurul Basar CCNA, MCP __ Do You Yahoo!? Yahoo! Messenger - Talk while you surf! It's FREE. http://im.yahoo.com/ _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP config
Reply in-line. Rodgers Moore "Amit Gupta" <[EMAIL PROTECTED]> wrote in message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > Hi All, > > Need some help in configuring VoIP > I am testing the loopback connectivity between my > router and EPABX by dialing a local extension number. > > As Soon as I dial the seizing code I get connected to > the router. > When I dial the destination pattern my call gets > transferred to the router,s next port Right here. Do you hear PBX dial-tone? When you dial the first digit does dial-tone go away? Also at this point you should do a "show voice calls", "show voice dps". What is the state of all of the ports & dsp's? Does everything look good? 90% of the time I see this problem it is incomplete or incorrect PBX programming. 9% its that the PBX set for 2 wire and router 4 wire, or the reverse, or incorrect wiring in a 4 wire config. (Cisco was putting out incorrect wiring diagrams for E&M 4 wire a year ago. I assume that it's been fixed, I reported it to TAC) Low volume level, the PBX can't hear the DTMF digits. PBX is made by NEC or Lucent. Both are rather picky about DTMF frequency accuracy and volume. To test, change the codec to G.711 on the ports so that no compression is being used. Or turn on local call compression bypass. This way the PBX's DTMF just passes through unmolested back to itself. > When I dial the local extension i do not get a > response. > I am using tone dialing,the Interface model is Type- 5 > E& M > Type of Signalling is Immediate > > Thanks for your clues in advance. > > Amit > > __ > Do You Yahoo!? > Thousands of Stores. Millions of Products. All in one Place. > http://shopping.yahoo.com/ > > _ > FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VOIP Troubles
Some things that might help: 1. Are you using CODEC like G.729 that works at an 8Kbps bit rate? If not, you'll be hating life. 2. Bandwidth at full rate & Multilink PPP (MLPPP) or FRF.12 will still be 17.2kbps. 3. You need QoS, even if you can burst to port speed at 56kbps. 4. Consider LLQ for Frame Relay for your QoS, which is now supported in IOS as of 12.1(2)T. We had to use this to get good voice quality on a moderately saturated FR link with a 1.544mbps CIR. 5. You may need QoS on your LAN if the links are congested near the 3640. Look at QoS as an end-to-end solution. 6. Perform ping tests from the 3640 to your remote sites to get a better idea of how many ms the round trip is. Try pings from both sides. Things to keep in mind: - VoIP overhead will kill you - you need to do testing to figure out exactly how much bandwidth you are using. - Good luck doing this over 56kbps, even if you use the full line. - Original Message - From: Chris Boyd To: [EMAIL PROTECTED] Sent: Friday, November 10, 2000 1:08 PM Subject: VOIP Troubles All right guys I need some help I have been working with Cisco for a while now on a VOIP issue. The problems lies in both call disconnects and voice distortion. We have followed all the steps for traffic shaping (QOS) and rtp header-compression but these do not seem to help. We have 150 remote sites all running 2600's with FXS modules that all come back into the host site where we have 2 7206's. Each of these links are 56k frame-relay links with 16k CIR running very few applications mostly small transactions and Citrix clients. The call must then traverse two internal Ethernet segments, routing through our 6509 backbone switch and then into a 3640 before hitting the PBX. Cisco seems to think that we need to increase our bandwidth to support the voice traffic, however, that is not something I have been able to sell to the "powers that be". We sold this idea on cutting cost and in our estimations for upping the CIR to even 32k will be significant cost increase. Right now I am shaping to 16k with an 8k committed burst so at any one time I should be able to burst to 24k. Assuming that I am able to burst to port speed (56k) why would I have call distortion unless there is some latency coming through the ISP's switch? We also have another company site that also comes back in this way and we have no problems with those calls.O.K. that being said (and hopefully not too confusing to follow) here come the questions: 1. Has anyone else implemented VOIP in slow links successfully? 2. Is anyone else having QOS problems with their VOIP implementations? 3. Do I need to prioritize the voice traffic through the local network? 4. Has anyone tried turning off traffic shaping and letting the voice and data compete for bandwidth? Thanks in advance for your feedback! Thanks, Chris Boyd, CCNANetwork SupportAlex Lee, Inc.120 4th Street SWHickory, NC 28601(828) 323-4103http://www.alexlee.com
Re: VOIP Troubles
1) Yes, and we had some of the same issues. The biggest was that we had to hard code the codec on all interfaces. For some reason the default codec G729ar8 wasn't always being selected or detected and since it's the default you can't hard code it. We had to opt for G729r8. We had point to point 56K circuits so it was much easier to deal with. Oh yea, we also tried to do a voice class to change the default codec for a whole router, but we had a couple of routers that ignored the voice class config, which is why we ended up hard coding every interface. 2) I think this is more an issue that IOS is a work in progress. We've always had to use an Early Deployment release to fix an issue. ( and in the process broke something else...) 3) Do it anyway. 4) Yep, and you'll be sorry you did. Even on full point to point T1's, I've seen issues. Just a suggestion, but have you considered putting voice cards in the 7206's and going VOFR? Rodgers Moore ""Chris Boyd"" <[EMAIL PROTECTED]> wrote in message 001d01c04b5a$6df29c50$[EMAIL PROTECTED]">news:001d01c04b5a$6df29c50$[EMAIL PROTECTED]... All right guys I need some help I have been working with Cisco for a while now on a VOIP issue. The problems lies in both call disconnects and voice distortion. We have followed all the steps for traffic shaping (QOS) and rtp header-compression but these do not seem to help. We have 150 remote sites all running 2600's with FXS modules that all come back into the host site where we have 2 7206's. Each of these links are 56k frame-relay links with 16k CIR running very few applications mostly small transactions and Citrix clients. The call must then traverse two internal Ethernet segments, routing through our 6509 backbone switch and then into a 3640 before hitting the PBX. Cisco seems to think that we need to increase our bandwidth to support the voice traffic, however, that is not something I have been able to sell to the "powers that be". We sold this idea on cutting cost and in our estimations for upping the CIR to even 32k will be significant cost increase. Right now I am shaping to 16k with an 8k committed burst so at any one time I should be able to burst to 24k. Assuming that I am able to burst to port speed (56k) why would I have call distortion unless there is some latency coming through the ISP's switch? We also have another company site that also comes back in this way and we have no problems with those calls.O.K. that being said (and hopefully not too confusing to follow) here come the questions: 1. Has anyone else implemented VOIP in slow links successfully? 2. Is anyone else having QOS problems with their VOIP implementations? 3. Do I need to prioritize the voice traffic through the local network? 4. Has anyone tried turning off traffic shaping and letting the voice and data compete for bandwidth? Thanks in advance for your feedback! Thanks, Chris Boyd, CCNANetwork SupportAlex Lee, Inc.120 4th Street SWHickory, NC 28601(828) 323-4103http://www.alexlee.com
Re: VoIP config
Rodgers , Do you have the wiring diagrams for E&M 4 wire. I have done a few installations and got the wiring right by trial and error. thanks ""Rodgers Moore"" <[EMAIL PROTECTED]> wrote in message 8uhh3t$76f$[EMAIL PROTECTED]">news:8uhh3t$76f$[EMAIL PROTECTED]... > Reply in-line. > > Rodgers Moore > > "Amit Gupta" <[EMAIL PROTECTED]> wrote in message > [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > > Hi All, > > > > Need some help in configuring VoIP > > I am testing the loopback connectivity between my > > router and EPABX by dialing a local extension number. > > > > As Soon as I dial the seizing code I get connected to > > the router. > > When I dial the destination pattern my call gets > > transferred to the router,s next port > > Right here. Do you hear PBX dial-tone? When you dial the first digit does > dial-tone go away? > > Also at this point you should do a "show voice calls", "show voice dps". > What is the state of all of the ports & dsp's? Does everything look good? > > 90% of the time I see this problem it is incomplete or incorrect PBX > programming. > 9% its that the PBX set for 2 wire and router 4 wire, or the reverse, or > incorrect wiring in a 4 wire config. (Cisco was putting out incorrect > wiring diagrams for E&M 4 wire a year ago. I assume that it's been fixed, I > reported it to TAC) > Low volume level, the PBX can't hear the DTMF digits. > PBX is made by NEC or Lucent. Both are rather picky about DTMF frequency > accuracy and volume. To test, change the codec to G.711 on the ports so > that no compression is being used. Or turn on local call compression > bypass. This way the PBX's DTMF just passes through unmolested back to > itself. > > > When I dial the local extension i do not get a > > response. > > I am using tone dialing,the Interface model is Type- 5 > > E& M > > Type of Signalling is Immediate > > > > Thanks for your clues in advance. > > > > Amit > > > > __ > > Do You Yahoo!? > > Thousands of Stores. Millions of Products. All in one Place. > > http://shopping.yahoo.com/ > > > > _ > > FAQ, list archives, and subscription info: > http://www.groupstudy.com/list/cisco.html > > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > > > > > _ > FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP config
Believe it or not, yes I do, and it's only hard copy. Cisco TAC has this document, again only in hard copy form. That's where I got my copy from. If you don't already know this, the wiring is different for each E&M type. If and when I get to it, I'll create an electronic version. I might be persuaded to share it too. ;) Rodgers Moore ""pinoal"" <[EMAIL PROTECTED]> wrote in message 8unip9$j3t$[EMAIL PROTECTED]">news:8unip9$j3t$[EMAIL PROTECTED]... > Rodgers , > > > Do you have the wiring diagrams for E&M 4 wire. I have done a few > installations and got the > wiring right by trial and error. > > > thanks > > > > > ""Rodgers Moore"" <[EMAIL PROTECTED]> wrote in message > 8uhh3t$76f$[EMAIL PROTECTED]">news:8uhh3t$76f$[EMAIL PROTECTED]... > > Reply in-line. > > > > Rodgers Moore > > > > "Amit Gupta" <[EMAIL PROTECTED]> wrote in message > > [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > > > Hi All, > > > > > > Need some help in configuring VoIP > > > I am testing the loopback connectivity between my > > > router and EPABX by dialing a local extension number. > > > > > > As Soon as I dial the seizing code I get connected to > > > the router. > > > When I dial the destination pattern my call gets > > > transferred to the router,s next port > > > > Right here. Do you hear PBX dial-tone? When you dial the first digit > does > > dial-tone go away? > > > > Also at this point you should do a "show voice calls", "show voice dps". > > What is the state of all of the ports & dsp's? Does everything look good? > > > > 90% of the time I see this problem it is incomplete or incorrect PBX > > programming. > > 9% its that the PBX set for 2 wire and router 4 wire, or the reverse, or > > incorrect wiring in a 4 wire config. (Cisco was putting out incorrect > > wiring diagrams for E&M 4 wire a year ago. I assume that it's been fixed, > I > > reported it to TAC) > > Low volume level, the PBX can't hear the DTMF digits. > > PBX is made by NEC or Lucent. Both are rather picky about DTMF frequency > > accuracy and volume. To test, change the codec to G.711 on the ports so > > that no compression is being used. Or turn on local call compression > > bypass. This way the PBX's DTMF just passes through unmolested back to > > itself. > > > > > When I dial the local extension i do not get a > > > response. > > > I am using tone dialing,the Interface model is Type- 5 > > > E& M > > > Type of Signalling is Immediate > > > > > > Thanks for your clues in advance. > > > > > > Amit > > > > > > __ > > > Do You Yahoo!? > > > Thousands of Stores. Millions of Products. All in one Place. > > > http://shopping.yahoo.com/ > > > > > > _ > > > FAQ, list archives, and subscription info: > > http://www.groupstudy.com/list/cisco.html > > > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > > > > > > > > > _ > > FAQ, list archives, and subscription info: > http://www.groupstudy.com/list/cisco.html > > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > > > > > _ > FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: VoIP config
take a look at http://www.cisco.com/warp/public/788/signalling/21.html http://www.cisco.com/univercd/cc/td/doc/product/access/acs_mod/cis3600/voice /4712voic.htm#xtocid221719 Stefano > -Original Message- > From: Rodgers Moore [mailto:[EMAIL PROTECTED]] > Sent: martedì 14 novembre 2000 04.59 > To: [EMAIL PROTECTED] > Subject: Re: VoIP config > > > Believe it or not, yes I do, and it's only hard copy. Cisco > TAC has this > document, again only in hard copy form. That's where I got > my copy from. > If you don't already know this, the wiring is different for > each E&M type. > If and when I get to it, I'll create an electronic version. > I might be > persuaded to share it too. ;) > > Rodgers Moore > > ""pinoal"" <[EMAIL PROTECTED]> wrote in message > 8unip9$j3t$[EMAIL PROTECTED]">news:8unip9$j3t$[EMAIL PROTECTED]... > > Rodgers , > > > > > > Do you have the wiring diagrams for E&M 4 wire. I have done a few > > installations and got the > > wiring right by trial and error. > > > > > > thanks > > > > > > > > > > ""Rodgers Moore"" <[EMAIL PROTECTED]> wrote in message > > 8uhh3t$76f$[EMAIL PROTECTED]">news:8uhh3t$76f$[EMAIL PROTECTED]... > > > Reply in-line. > > > > > > Rodgers Moore > > > > > > "Amit Gupta" <[EMAIL PROTECTED]> wrote in message > > > [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > > > > Hi All, > > > > > > > > Need some help in configuring VoIP > > > > I am testing the loopback connectivity between my > > > > router and EPABX by dialing a local extension number. > > > > > > > > As Soon as I dial the seizing code I get connected to > > > > the router. > > > > When I dial the destination pattern my call gets > > > > transferred to the router,s next port > > > > > > Right here. Do you hear PBX dial-tone? When you dial > the first digit > > does > > > dial-tone go away? > > > > > > Also at this point you should do a "show voice calls", > "show voice dps". > > > What is the state of all of the ports & dsp's? Does > everything look > good? > > > > > > 90% of the time I see this problem it is incomplete or > incorrect PBX > > > programming. > > > 9% its that the PBX set for 2 wire and router 4 wire, or > the reverse, or > > > incorrect wiring in a 4 wire config. (Cisco was putting > out incorrect > > > wiring diagrams for E&M 4 wire a year ago. I assume that > it's been > fixed, > > I > > > reported it to TAC) > > > Low volume level, the PBX can't hear the DTMF digits. > > > PBX is made by NEC or Lucent. Both are rather picky about DTMF > frequency > > > accuracy and volume. To test, change the codec to G.711 > on the ports so > > > that no compression is being used. Or turn on local call > compression > > > bypass. This way the PBX's DTMF just passes through > unmolested back to > > > itself. > > > > > > > When I dial the local extension i do not get a > > > > response. > > > > I am using tone dialing,the Interface model is Type- 5 > > > > E& M > > > > Type of Signalling is Immediate > > > > > > > > Thanks for your clues in advance. > > > > > > > > Amit > > > > > > > > __ > > > > Do You Yahoo!? > > > > Thousands of Stores. Millions of Products. All in one Place. > > > > http://shopping.yahoo.com/ > > > > > > > > _ > > > > FAQ, list archives, and subscription info: > > > http://www.groupstudy.com/list/cisco.html > > > > Report misconduct and Nondisclosure violations to > [EMAIL PROTECTED] > > > > > > > > > > > > > _ > > > FAQ, list archives, and subscription info: > > http://www.groupstudy.com/list/cisco.html > > > Report misconduct and Nondisclosure violations to > [EMAIL PROTECTED] > > > > > > > > > _ > > FAQ, list archives, and subscription info: > http://www.groupstudy.com/list/cisco.html > > Report misconduct and Nondisclosure violations to > [EMAIL PROTECTED] > > > > > _ > FAQ, list archives, and subscription info: > http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: VoIP/voFR
Hello Evan, I would use num-exp for other reasons, not to accomplish what you want to do. You can do what you want on the dial-peer configuration. Here is an example: dial-peer voice 1 pots destination-pattern 745 port 1/0:0 prefix 212345 That's what happends: when the route receives the calls from another router or from any other source (e.g. a phone connected to it or pbx), it will look for the dial-peer statement that has the better match. When the dial-peer 1 pots is executed, the router sends only the , NOT the 745. But before sending, it will add the prefix you want, on the this case, it will send the following string: 212345 One interesting thing is that on the dial-peer pots statement, the router removes the 745, if you had the same config on the dial-peer voip, the router would NOT remove the 745. Hope it helps, let me know if you have any questions. Guilherme -Original Message-From: Evan You [mailto:[EMAIL PROTECTED]]Sent: Thursday, November 16, 2000 9:29 AMTo: [EMAIL PROTECTED]Subject: VoIP/voFR Has anyone used the Num-Exp command? If so, how many Num-exp commands can a Cisco router handle? We are deploying a huge VoFR network. The customer has a private numbers scheme such as 745-2345 (2345 is the extension), Each location has 3 prefix numbers when they dial internally. But to dial outside, the extension is keeped and the front 3 digits are replaced with a real number such as 212-345-2345. So I want to use the command num-exp 745... 212345... The question is, how many time can put that global command in a Cisco? How will it affect the router?I have about 200 to 400 sites. I would rather have the PBX do what I am asking for, but customer does nto want the PBX translating the numbers. Thanks! Evan You - CCNA
Re: VOIP Troubles
You can also try calculating bandwidth at www.erlang.com John Deatherage wrote: > Some things that might help: 1. Are you using CODEC like G.729 that > works at an 8Kbps bit rate? If not, you'll be hating life.2. > Bandwidth at full rate & Multilink PPP (MLPPP) or FRF.12 will still be > 17.2kbps.3. You need QoS, even if you can burst to port speed at > 56kbps.4. Consider LLQ for Frame Relay for your QoS, which is now > supported in IOS as of 12.1(2)T. We had to use this to get good voice > quality on a moderately saturated FR link with a 1.544mbps CIR.5. You > may need QoS on your LAN if the links are congested near the 3640. > Look at QoS as an end-to-end solution.6. Perform ping tests from the > 3640 to your remote sites to get a better idea of how many ms the > round trip is. Try pings from both sides. Things to keep in mind:- > VoIP overhead will kill you - you need to do testing to figure out > exactly how much bandwidth you are using.- Good luck doing this over > 56kbps, even if you use the full line. > > - Original Message - > From: Chris Boyd > To: [EMAIL PROTECTED] > Sent: Friday, November 10, 2000 1:08 PM > Subject: VOIP Troubles > All right guys I need some help I have been working > with Cisco for a while now on a VOIP issue. The problems > lies in both call disconnects and voice distortion. We have > followed all the steps for traffic shaping (QOS) and rtp > header-compression but these do not seem to help. We have > 150 remote sites all running 2600's with FXS modules that > all come back into the host site where we have 2 7206's. > Each of these links are 56k frame-relay links with 16k CIR > running very few applications mostly small transactions and > Citrix clients. The call must then traverse two internal > Ethernet segments, routing through our 6509 backbone switch > and then into a 3640 before hitting the PBX. Cisco seems to > think that we need to increase our bandwidth to support the > voice traffic, however, that is not something I have been > able to sell to the "powers that be". We sold this idea on > cutting cost and in our estimations for upping the CIR to > even 32k will be significant cost increase. Right now I am > shaping to 16k with an 8k committed burst so at any one time > I should be able to burst to 24k. Assuming that I am able > to burst to port speed (56k) why would I have call > distortion unless there is some latency coming through the > ISP's switch? We also have another company site that also > comes back in this way and we have no problems with those > calls.O.K. that being said (and hopefully not too confusing > to follow) here come the questions: 1.Has anyone else > implemented VOIP in slow links successfully?2.Is anyone > else having QOS problems with their VOIP > implementations?3.Do I need to prioritize the voice > traffic through the local network?4.Has anyone tried > turning off traffic shaping and letting the voice and data > compete for bandwidth? Thanks in advance for your > feedback! > Thanks, Chris Boyd, CCNA > Network Support > Alex Lee, Inc. > 120 4th Street SW > Hickory, NC 28601 > (828) 323-4103 > http://www.alexlee.com > _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
URGENT: regarding VoIP
Hello guys Greetings.. I have my CCIE Exam on April 10 and 11. I need your urgent help. I am trying to setup a Voice over IP. Well everything works fine. When I put access list on one of the router to act as a IOS firewall, I can't make calls. Here is a sample access list access-list 140 permit tcp any any range 11000 11999 access-list 150 permit ospf any any access-list 150 permit icmp any any echo-reply access-list 123 permit ip host 138.1.249.6 host 138.1.252.4 access-list 150 permit udp any any range 16384 2000 access-list 150 permit tcp any any eq 1720 access-list 150 permit tcp any eq 1720 any access-list 150 permit tcp any any range 11000 11999 access-list 150 deny ip any any with this configuration, I can ring both phone from either location but I can't hear anything. Also does any one has info on IP OSFP Demand Circuit over ISDN. My ISDN line keep flapping even after putting the demand circuit. I can see that my routes in OSPF Database has DNA mark beside it but the line keep coming up. When do a show dialer, I see the d=224.0.0.5 Any idea what could cause this? I do have access list that permit only ISDN Network. Anyway help would be highly appreciate. Thank you faisal = __ Do You Yahoo!? Get email at your own domain with Yahoo! Mail. http://personal.mail.yahoo.com/?.refer=text _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP....... [7:39741]
Hi, Any one can guide me for the VOIP Basic Technology. So that it can help me for understanding the technology. Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=39741&t=39741 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP [7:31615]
Hi all, I have to configure VOIP at my end, i am working on CISCO equippment.. Router: Cisco 3660 RAS: AS 5300 can anyone tell me how to go about configuring it at my end.. Regards, Amit Bhasin _ Do You Yahoo!? Get your free @yahoo.com address at http://mail.yahoo.com Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=31615&t=31615 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: voip details
Here's a place to start: http://www.itprc.com/voice.htm Annlee ""v srinivasarao"" <[EMAIL PROTECTED]> wrote in message 8lbumr$o06$[EMAIL PROTECTED]">news:8lbumr$o06$[EMAIL PROTECTED]... > can any one get information on voip > > > ___ > UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html > FAQ, list archives, and subscription info: http://www.groupstudy.com > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > --- ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: VoIP books
One of the best VoIP books is Voice over IP by Uyless Black ISBN 0-13-022463-4 I found some of the best prices at booksamillion.com, pay the extra 5.00 for the membership and the prices come down even more. Its pretty great! -Ryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Russell Lusignan Sent: Tuesday, August 01, 2000 6:29 AM To: [EMAIL PROTECTED] Subject: VoIP books Can anyone recommend a good VoIP book which covers content of the CVOICE exam? Thanks! Russ.. ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP/VPN products
Dear All, Which are cost-effective and reliable VOIP and VPN products that can be deployed over a Wireless Ethernet system (Cisco/Aironet 342 series) ? Chow, James Smith James Smith System Administrator Excel Internet Service __ FREE Personalized Email at Mail.com Sign up at http://www.mail.com/?sr=signup ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
CIPT (VoIP) Class
I'm getting ready to sign up for a CIPT (Cisco IP Telephony) course from Telcordia Training and Education Ctr in Chicago. Before I fly all the way from Los Angeles to take it I was hoping to get some feedback from you fine folks. I'm one test short of my CCNP and have a technical background and I hope to see the system in action and experience the implementation, not just get an advertising dog-and-pony. Here are my questions: * Is the class worthwhile? * Has anybody had experience with this trainer? * Is there anything I simply must see in Chicago while I'm there? :-) - Dennis ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP and VOFR
Hi, How different is VOIP and VOFR? I thought IP networks are possible over Frame Relay (WANs). In this case, does VOIP is to be interpreted as a LAN implementation of Voice network? Can some please explain the differences between VOIP and VOFR? Thank you in advance. -- Oscar Rau [EMAIL PROTECTED] ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP Error Desc.
Dear All, Can Anybody can inform me where can I find Error details of Voice Calls debugging. I have encountered with different Voice errors but could not find details of disconnections: (cause disconnection 0*10) (cause disconnection 0*22) (cause disconnection 0*15) (cause disconnection 0*11) obtained through command: debug voip ccapi inout Rgds, Hasan __ Do You Yahoo!? Get Yahoo! Mail Free email you can access from anywhere! http://mail.yahoo.com/ ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP [7:64080]
Everyone, I am seeking a recommendation on a voip book, preferably something that explains the different types of technologies and how they can be applied 'in the real world'. Right now, we are doing some very rudimentary voip stuff with a variety of routers, 1760, 2600, and a 3600 seriers which is connected to a pri. We have no call manager (yet), so we have a bunch of dial-peer groups set up on all our routers to interconnect the remote offices. All offices have their own pbx's of different types, and most of the routers at the remote locations have the vic fxs cards. I feel like this is a good opportunity for me to learn a great deal, but I want to make sure that I REALLY understand what I am learning. And right now, for example, I have no idea what the difference is between an FXS card and an E&M card. I'm starting to feel a bit like the village voip idiot, and the tons of docs I've read on the cisco web site do not seem to be helping. Any recommendations will be highly appreciated. Thanks, Kris ** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the sender by email, delete and destroy this message and its attachments. ** Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=64080&t=64080 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP [7:41934]
Hi I have a question regarding the 827-4V router with VoIP. Can you define a VoIP gateway on the 827-4V router? Thanks in advance Geoff Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=41934&t=41934 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP [7:45991]
1 - What routers support VoIP? (Looking for a Cisco Link, can't find one) I have a lot of old 2501's and have been told they don't support VoIP but have found nothing in writing. 2 - Has anyone used the ATA 186? Steve Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=45991&t=45991 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP [7:18426]
Does anyone know around how many hours it takes to tie in Voice over IP (VoIP) on a 2600 series router over Frame Relay? Less than 50 users and 10 simultaneous lines connecting with a PBX? I bidded this out and received one bid and do not want to pay for excess hours. I appreciate your constructive responses and knowledge. Julius __ Do You Yahoo!? Get email alerts & NEW webcam video instant messaging with Yahoo! Messenger http://im.yahoo.com Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=18426&t=18426 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP [7:56129]
Dear fiends, Who can send me some information about VOIP. In my Network, One of the Routers is Cisco 3661. I like to offer VOIP service by this Router(Cico 3661) but I don't have any information about VOIP service. Please let me to know about the hardwares and softwares that I need them to offering this service to my users.. Regards, Hamed Sedghi Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=56129&t=56129 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP [7:48091]
Team, I am currently trying to implement VOIP for the first time using Cisco routers. Can you please give me a URL or any resource where i can read about this step by step or go through a simulation. THIS IS URGENT, I APPRECIATE YOUR RESPONSE. Cisco Die-Hard Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=48091&t=48091 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP ????? [7:51621]
Hi All, Over the weekend, i paid $22AUD to get a haircut, which has effected my thought processing skills come monday. Thus, I need help. Its not even a good haircut. My boss, without me, has purchased a Cisco 2950 switch and 10 IP Telephones for a branch office. IP telephones, is something I have never worked with b4. I hope this is not stupid, however I am pretty sure that the IP telephones will not work by simply plugging into a 2950 switch. I am assuming, the branch office will need some sort of router/specialised software to intercept and manage the routing and IP assignment of each phone. Is it possible, if some one can lend any advice on the very basic requirments that they have used ? Any help/advice is much appreciated. John Sydney, Australia ** visit http://www.solution6.com visit http://www.eccountancy.com - everything for accountants. UK Customers - http://www.solution6.co.uk * This email message (and attachments) may contain information that is confidential to Solution 6. If you are not the intended recipient you cannot use, distribute or copy the message or attachments. In such a case, please notify the sender by return email immediately and erase all copies of the message and attachments. Opinions, conclusions and other information in this message and attachments that do not relate to the official business of Solution 6 are neither given nor endorsed by it. * Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=51621&t=51621 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
voip [7:51729]
Is there a way to tell a router (3810) , which is running voip, to reroute a voip call if the destination router is down. This is how I see it. The call is made from a typical digital phone. The pbx sends the digits to the router. The router processes the digits and sends them to the destination router. What happens if the destination router is down. The PBX does not know if the destination router is down , so it will send the digits to the local router. But, how do I tell the local router to reroute the phone call?? If you need a more info please specify.. Thanks Jake Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=51729&t=51729 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
force a AS5300 involment in a VOIP-VOIP call [7:37385]
Dear Group, In h323 enviroment I have to " bill" all the calls, VOIP calls too. The billing system is "synchronized" with AS5300 and not with the Gatekeeper. The question is "is there a way to force the intra voip calls (H323) through the AS5300"? Any idea Thanks in advance, Teresa Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=37385&t=37385 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
force a AS5300 involment in a VOIP-VOIP call [7:37386]
Dear Group, In h323 enviroment I have to " bill" all the calls, VOIP calls too. The billing system is "synchronized" with AS5300 and not with the Gatekeeper. The question is "is there a way to force the intra voip calls (H323) through the AS5300"? Any idea Thanks in advance, Teresa Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=37386&t=37386 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP isdn numbering types
Hi All I'm trying to send the ISDN numbering types and translate numbers from the ingress to the egress gateway. I have a gateway which accepts an incoming call from the PSTN then routes it across the IP network to the egress gateway and to an IVR system, this part works fine. However when I get an international call the call is still routed across the network but the translation rule removes the international prefix and puts the local country code, hence the IVR system gets an incorrect number. Has anyone any experience of this or know any good references? Thanks Patrick Donlon [EMAIL PROTECTED] _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: what is "VOIP" ?
A few lines. Voice-over-IP (VoIP) enables a router to carry voice traffic (for example, telephone calls and faxes) over an IP network. Cisco's voice support is implemented using voice packet technology. In VoIP, the digital signal processor (DSP) segments the voice signal into frames and stores them in voice packets. These voice packets are transported using IP in compliance with the International Telecommunications Union-Telecommunications (ITU-T) specification H.323, the specification for transmitting multimedia (voice, video, and data) across a network. Because it is a delay-sensitive application, you need to have a well-engineered, end-to-end network to successfully use VoIP. Fine-tuning your network to adequately support VoIP involves a series of protocols and features to improve quality of service (QoS). Traffic shaping considerations must also be taken into account to ensure the reliability of the voice connection. regds Hp > -Original Message- > From: Robert Nickson [SMTP:[EMAIL PROTECTED]] > Sent: Monday, February 19, 2001 2:47 PM > To: [EMAIL PROTECTED] > Subject: RE: what is "content distribution" ? > > Can someone give me a few lines on VOIp don't know much about it ? > > _ > FAQ, list archives, and subscription info: > http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: what is "VOIP" ?
Please does anyone have any link or information on configuring VOIP for the Solaris environment. Thank you "Hitesh Pathak (CSD-BBYRO-RTSG)" <[EMAIL PROTECTED]> on 02/19/2001 11:12:05 AM Please respond to "Hitesh Pathak (CSD-BBYRO-RTSG)" <[EMAIL PROTECTED]> To: "'Robert Nickson'" <[EMAIL PROTECTED]>, [EMAIL PROTECTED] cc:(bcc: Babashola Madariola/C/Africa/Mobil-Notes) Subject: RE: what is "VOIP" ? A few lines. Voice-over-IP (VoIP) enables a router to carry voice traffic (for example, telephone calls and faxes) over an IP network. Cisco's voice support is implemented using voice packet technology. In VoIP, the digital signal processor (DSP) segments the voice signal into frames and stores them in voice packets. These voice packets are transported using IP in compliance with the International Telecommunications Union-Telecommunications (ITU-T) specification H.323, the specification for transmitting multimedia (voice, video, and data) across a network. Because it is a delay-sensitive application, you need to have a well-engineered, end-to-end network to successfully use VoIP. Fine-tuning your network to adequately support VoIP involves a series of protocols and features to improve quality of service (QoS). Traffic shaping considerations must also be taken into account to ensure the reliability of the voice connection. regds Hp > -Original Message- > From: Robert Nickson [SMTP:[EMAIL PROTECTED]] > Sent: Monday, February 19, 2001 2:47 PM > To: [EMAIL PROTECTED] > Subject: RE: what is "content distribution" ? > > Can someone give me a few lines on VOIp don't know much about it ? > > _ > FAQ, list archives, and subscription info: > http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP over Satellite link
Hi All, Help needed on the prerequisites in the form of IOS for configuring VoIP over an International Leased Private Circuit. Do the Cisco Routers at both the sides have to have a minimum IOS version. We are using the 3640 Router at both ends. Thanks & Regards Amit __ Do You Yahoo!? Get email at your own domain with Yahoo! Mail. http://personal.mail.yahoo.com/ _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
QOS for VoIP Traffic
Hi All, I am using VoIP over a satellite link. I am planning to configure some queuing mechanism on the router which can provide dedicated bw to voice traffic and at the same time use the entire bw for data traffic when no traffic from higher priority flows is available( voice ) Please send me your suggestions and comments on any other feature that I can enable to improve quality. Thanks & regards Amit __ Do You Yahoo!? Get email at your own domain with Yahoo! Mail. http://personal.mail.yahoo.com/ _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP and Security Exams
I have only days to go before my Support exam which will compplete my CCNP. I have been wondering the 'what next' and thought that a couple of specialist exams like VoIP and Security would be fun. If anyone can give recommendations on good books for Cisco VoIP and Security, it would be much appreciated. Thanks in Adv. Rod _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Question About VOIP Problem
Hai, Today, I have observed a network eng from a vendor install a E&M Type II card on a Cisco 3600 and connected to my E&M II Toshiba card. We did a loop test from port 1 and port 2 and it works. The Eng configured the first port dial peer as 300..., second port dial peer as 301.., 302.. and 303.. . So when we did a call from our Key phone unit to another remote site connecetd to a FSX int card we were able to get connected using both FXS numbers was 311 and 312. Funny thing is when the other side dial the first port i.e 300242--> 242 is the ext. We were able to see the call get into the router and pass through to the KST. >From the ext 242 we can see the light showing a call but the call was never establish. The other side advice getting busy tone. But when we test the other dial peer number i.e 301242, 302242 it works. And when we need do a local loop from one key phone to another one we must add up another number i.e 3011242, insted off 301242. As I observed the sh run for both router, the config for the dial peer was the same and the port was config correctly (just observed since it by another vendor). But each time the net eng configured or change any setting he never do any shutdown and no shut. Thus my question is : a) Is there any barr from using 300 flow by the wild card ...?. b) When I asked the eng why he keep congif a new dial peers insted off duing any debug he reply that for VOIP there is no debug commands. I may not be VOIP experts but as a CCNA i did observed a few debug commands for VOIP. Any info on this. c) Would the command no shut and shutdown help in this situation. d) What would happend to a 64Kbps line with 32 CIR with 4 voice channels compress up to 11 KBPS?. We only test the voice quality with extended ping as dumy data. If the first, second and third channels is busy the fourth persons who is using the voice channel will have a good or bad quality off voice. Heck after this any problem the first people to be blame will be the PBX guy. Remarks: We have test our E&M card before and we do not ned to set any config on the card just change some jumper. Soon I will be installing some cisco router for my company for VOIP and need some help on the problem. Thanks for the help Nuurul Basar Network Engineer CCNA and MCP Kuala Lumpur Malaysia P/S I am reading the Integrating Voice and Data Network by Scot Keagy, it's a good books pity it arrived here a bit late ( arround Nov ). Is there any more book's like this arround. __ Do You Yahoo!? Yahoo! Shopping - Thousands of Stores. Millions of Products. http://shopping.yahoo.com/ _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP: Need help
Try the following link http://www.cisco.com/warp/public/793/voip/ Larry Osei-Kwaku --- Ishtiaque Mahbub <[EMAIL PROTECTED]> wrote: > Hello People! > > Need your valued advice. > > VoIP, though frequently heard, I have never had the > opportunity to work on > it. Books are rare here in Bangladesh and only > source of information is the > Internet. > So could anyone please let me know the useful > resources on the web(just want > to be familiar with the technology). My requirement > is simple: > 1.What is the underlying technology of VoIP? > 2.What sort of Cisco Hardware do I need? > Thank you in anticipating. > Merry Christmas and happy New Year to you all. > > Regards > > Ishtiaque > > _ > Get Your Private, Free E-mail from MSN Hotmail at > http://www.hotmail.com. > > _ > FAQ, list archives, and subscription info: > http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to > [EMAIL PROTECTED] > > > = "Wear a smile and have friends; wear a scowl and have wrinkles." - George Eliot "the greatest glory is not in never falling, but rising up each time we fall." "The greatest man is not he who does not fall but he who falls and rises again to win" __ Do You Yahoo!? Yahoo! Shopping - Thousands of Stores. Millions of Products. http://shopping.yahoo.com/ _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP Installation and Configuration
Hi Guys I intend installing a voice over IP gateway and I have with me here the = following software: 1. Gateway 2. Gate Admin 3. Gatekeeper 4. SQL Server All from Innomedia. Question. 1. Do I need any other vital software to complete the installation? 2. Can I install all the Software Packages in one Computer System or it = will be better otherwise? If otherwise, then which one(s) will go into one = Computer=20 system and which will go into the others? 3. Other useful information / Documentation will be well appreciated. I lost the manual that came with the software and that is why I am doing = this. Thanks in advance. OLETU Hosea Godswill, CCNA _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: Help for VOIP!
Hi, You may use 3600 and 2500. Let voice backhaul throu FE of 3600s for both end. On 2500 you may want to config for QoS. On 2500 you would use standard cables, if connecting back to back then DTE and DCE cables. For VIC-2FXS it is usual RJ-11. Bye -Mak Gene Park wrote: > Hi, members, > > I have one question about VOIP. > Actually, I have two 3600s with three exact modules- > Voice card(NM-1V), ATM (1A-OC3MM), and FastEthernet, > but no serial ports. The NM-1V has two FXS. > > Based on these, how do I set up for VOIP lab? > I have several 2503, 2513, and 2514. > Because I don't have serial ports on 3600s, do I > need to use FastEthernet ports or use ATM interface? > Please let me have cabling info too. > > Thanks for your help. > > = > Gene Park > [EMAIL PROTECTED] > > __ > Do You Yahoo!? > Yahoo! Photos - Share your holiday photos online! > http://photos.yahoo.com/ > > _ > FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Bandwidth constraints for VoIP
Hello Group! A very happy new year to you all! I was wondering if any one could advise what is the minimum Bandwidth required for Voice Over IP installation on Cisco Routers (Router Series will be 2600). Is 64kbps is too steep for 4 simultaneous voice operations? Suggestion, advices welcome. Regards Ishtiaque _ Get Your Private, Free E-mail from MSN Hotmail at http://www.hotmail.com. _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
about voip dial-peer
On an origination only AS5300 gateway, i just need to configure VOIP peers ,right? On a distribution only AS5300 gateway, i just need to configure pots peers ,right? _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Slightly OT: VoIP Quality
We have implemented VoIP at two of our branches as a test. We are using Nortel ITG cards in the branch PBX to convert the calls to IP and then we connect the card to a Cisco 2924XL switch with all voice traffic in its own VLAN. Then the traffic hits a 2620 router with LLQ configured. The voice calls then go through another branch with custom queueing configured, then to the destination branch with the same setup as the first branch. This is now up and running without any serious glitches, but the users at the branches complain that all incoming calls sound like cell phone calls. Is this the type of quality we can expect from this technology? Is it a natural result of packetizing real-time voice traffic? Or, can we expect better? Any thoughts or tips would be appreciated. Thanks, John ___ Send a cool gift with your E-Card http://www.bluemountain.com/giftcenter/ _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Cisco VoIP mailing list
Are there any mailing lists that focus primarily on Cisco VoIP, not VoIP in general? Thanks, John **NOTE: New CCNA/CCDA List has been formed. For more information go to http://www.groupstudy.com/list/Associates.html _ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: Help regarding VOIP
Check the following URLs: http://www.cisco.com/pcgi-bin/Support/PSP/psp_view.pl?p=Internetworking:VoX: VoIP http://www.cisco.com/pcgi-bin/Support/PSP/psp_view.pl?p=Internetworking:VoX: VoIP http://www.cisco.com/pcgi-bin/Support/PSP/psp_view.pl?p=Internetworking:VoX: VoATM http://www.cisco.com/warp/public/cc/pd/iosw/ioft/mmcm/tech/h323_wp.htm http://www.cisco.com/public/products_tech.shtml Hope this helps... Chris Lemagie Systems Engineer Cisco Systems Seattle Commercial Region (425) 468-0959 [EMAIL PROTECTED] http://www.cisco.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Nuurul Basar Sent: Thursday, October 26, 2000 11:04 PM To: [EMAIL PROTECTED] Subject:Help regarding VOIP Hai, I will be changing job in a near time and now looking for info regarding VOIP and ICT( Information Communication Technology). It have been 4 month since I left Cisco stuff and fell a bit left behind. Can some give me some url regarding VOIP implemntation on Cisco and other routers. Thanks have a nice day Nuurul Basar CCNA, MCP __ Do You Yahoo!? Yahoo! Messenger - Talk while you surf! It's FREE. http://im.yahoo.com/ _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP PC cards... Help
Dear All, We are planning to implement VoIP solution with low budget. Could any one give me the details about the VoIP PC Cards like Micom Voice Cards. I was told that "Micom Voice Cards" are not available in the market is that's true..? Thanks, Chitla Sudhir CCNA, NCIP, CNA Network Analyst, India. Do You Yahoo!? Get your free @yahoo.co.in address at http://mail.yahoo.co.in _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: URGENT: regarding VoIP
i'm not sure PIX can support H323 or not, check it out it does or not. good luck! "Faisal Khan" <[EMAIL PROTECTED]> ¼¶¼g©ó¶l¥ó [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > Hello guys > > Greetings.. > I have my CCIE Exam on April 10 and 11. I need your urgent help. I am trying to setup a Voice > over IP. Well everything works fine. When I put access list on one of the router to act as a IOS > firewall, I can't make calls. Here is a sample access list > > access-list 140 permit tcp any any range 11000 11999 > access-list 150 permit ospf any any > access-list 150 permit icmp any any echo-reply > access-list 123 permit ip host 138.1.249.6 host 138.1.252.4 > access-list 150 permit udp any any range 16384 2000 > access-list 150 permit tcp any any eq 1720 > access-list 150 permit tcp any eq 1720 any > access-list 150 permit tcp any any range 11000 11999 > access-list 150 deny ip any any > > with this configuration, I can ring both phone from either location but I can't hear anything. > Also does any one has info on IP OSFP Demand Circuit over ISDN. My ISDN line keep flapping even > after putting the demand circuit. I can see that my routes in OSPF Database has DNA mark beside > it but the line keep coming up. When do a show dialer, I see the d=224.0.0.5 Any idea what could > cause this? > > I do have access list that permit only ISDN Network. > > Anyway help would be highly appreciate. > Thank you > faisal > > > = > > > __ > Do You Yahoo!? > Get email at your own domain with Yahoo! Mail. > http://personal.mail.yahoo.com/?.refer=text > > _ > FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: URGENT: regarding VoIP
you can check the "cause code" from both end voice switch, or u can try to use a PA to see are there any problem of "call processing" ""May"" <[EMAIL PROTECTED]> ¼¶¼g©ó¶l¥ó 9a19s8$ts5$[EMAIL PROTECTED]">news:9a19s8$ts5$[EMAIL PROTECTED]... > i'm not sure PIX can support H323 or not, check it out it does or not. > good luck! > > > "Faisal Khan" <[EMAIL PROTECTED]> ¼¶¼g©ó¶l¥ó > [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > > Hello guys > > > > Greetings.. > > I have my CCIE Exam on April 10 and 11. I need your urgent help. I am > trying to setup a Voice > > over IP. Well everything works fine. When I put access list on one of the > router to act as a IOS > > firewall, I can't make calls. Here is a sample access list > > > > access-list 140 permit tcp any any range 11000 11999 > > access-list 150 permit ospf any any > > access-list 150 permit icmp any any echo-reply > > access-list 123 permit ip host 138.1.249.6 host 138.1.252.4 > > access-list 150 permit udp any any range 16384 2000 > > access-list 150 permit tcp any any eq 1720 > > access-list 150 permit tcp any eq 1720 any > > access-list 150 permit tcp any any range 11000 11999 > > access-list 150 deny ip any any > > > > with this configuration, I can ring both phone from either location but I > can't hear anything. > > Also does any one has info on IP OSFP Demand Circuit over ISDN. My ISDN > line keep flapping even > > after putting the demand circuit. I can see that my routes in OSPF > Database has DNA mark beside > > it but the line keep coming up. When do a show dialer, I see the > d=224.0.0.5 Any idea what could > > cause this? > > > > I do have access list that permit only ISDN Network. > > > > Anyway help would be highly appreciate. > > Thank you > > faisal > > > > > > = > > > > > > __ > > Do You Yahoo!? > > Get email at your own domain with Yahoo! Mail. > > http://personal.mail.yahoo.com/?.refer=text > > > > _ > > FAQ, list archives, and subscription info: > http://www.groupstudy.com/list/cisco.html > > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > > > > > _ > FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: URGENT: regarding VoIP
um . or u can try to by pass the PIX firewall, to . isolate the problem, this the problem of ur PIX or Configure in ur Gateway, it seem something wrong in call processing, or u hvnt enable the "Answer Supervision" ""May"" <[EMAIL PROTECTED]> ¼¶¼g©ó¶l¥ó 9a1a16$ud1$[EMAIL PROTECTED]">news:9a1a16$ud1$[EMAIL PROTECTED]... > you can check the "cause code" from both end voice switch, or u can try to > use a > PA to see are there any problem of "call processing" > > > ""May"" <[EMAIL PROTECTED]> ¼¶¼g©ó¶l¥ó 9a19s8$ts5$[EMAIL PROTECTED]">news:9a19s8$ts5$[EMAIL PROTECTED]... > > i'm not sure PIX can support H323 or not, check it out it does or not. > > good luck! > > > > > > "Faisal Khan" <[EMAIL PROTECTED]> ¼¶¼g©ó¶l¥ó > > [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > > > Hello guys > > > > > > Greetings.. > > > I have my CCIE Exam on April 10 and 11. I need your urgent help. I am > > trying to setup a Voice > > > over IP. Well everything works fine. When I put access list on one of > the > > router to act as a IOS > > > firewall, I can't make calls. Here is a sample access list > > > > > > access-list 140 permit tcp any any range 11000 11999 > > > access-list 150 permit ospf any any > > > access-list 150 permit icmp any any echo-reply > > > access-list 123 permit ip host 138.1.249.6 host 138.1.252.4 > > > access-list 150 permit udp any any range 16384 2000 > > > access-list 150 permit tcp any any eq 1720 > > > access-list 150 permit tcp any eq 1720 any > > > access-list 150 permit tcp any any range 11000 11999 > > > access-list 150 deny ip any any > > > > > > with this configuration, I can ring both phone from either location but > I > > can't hear anything. > > > Also does any one has info on IP OSFP Demand Circuit over ISDN. My ISDN > > line keep flapping even > > > after putting the demand circuit. I can see that my routes in OSPF > > Database has DNA mark beside > > > it but the line keep coming up. When do a show dialer, I see the > > d=224.0.0.5 Any idea what could > > > cause this? > > > > > > I do have access list that permit only ISDN Network. > > > > > > Anyway help would be highly appreciate. > > > Thank you > > > faisal > > > > > > > > > = > > > > > > > > > __ > > > Do You Yahoo!? > > > Get email at your own domain with Yahoo! Mail. > > > http://personal.mail.yahoo.com/?.refer=text > > > > > > _ > > > FAQ, list archives, and subscription info: > > http://www.groupstudy.com/list/cisco.html > > > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > > > > > > > > > _ > > FAQ, list archives, and subscription info: > http://www.groupstudy.com/list/cisco.html > > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > > > > > _ > FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: URGENT: regarding VoIP
Hi Faisal, It seems that their is something wrong with your pix. You can rule it out by bypassing the pix or the access-lists. You can do some of the debugging on your voice routers acting as h.323 gateway. debug voip spi debug voip ccapi inout With these commands you will be able to know whether the call is getting bridged or not.I hope this will help you out. Regards Rajeev BHaradwaj Faisal Khan wrote: > Hello guys > > Greetings.. > I have my CCIE Exam on April 10 and 11. I need your urgent help. I am trying to >setup a Voice > over IP. Well everything works fine. When I put access list on one of the router to >act as a IOS > firewall, I can't make calls. Here is a sample access list > > access-list 140 permit tcp any any range 11000 11999 > access-list 150 permit ospf any any > access-list 150 permit icmp any any echo-reply > access-list 123 permit ip host 138.1.249.6 host 138.1.252.4 > access-list 150 permit udp any any range 16384 2000 > access-list 150 permit tcp any any eq 1720 > access-list 150 permit tcp any eq 1720 any > access-list 150 permit tcp any any range 11000 11999 > access-list 150 deny ip any any > > with this configuration, I can ring both phone from either location but I can't hear >anything. > Also does any one has info on IP OSFP Demand Circuit over ISDN. My ISDN line keep >flapping even > after putting the demand circuit. I can see that my routes in OSPF Database has DNA >mark beside > it but the line keep coming up. When do a show dialer, I see the d=224.0.0.5 Any >idea what could > cause this? > > I do have access list that permit only ISDN Network. > > Anyway help would be highly appreciate. > Thank you > faisal > > = > > __ > Do You Yahoo!? > Get email at your own domain with Yahoo! Mail. > http://personal.mail.yahoo.com/?.refer=text > > _ > FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP Configuration - Line Busy ...
This is a lab for VoIP. 1) I have 3640 with one VIC card which has 2 FXS ports.I connect one of those to telephone instrument and I get a dial tone once port is up. 2) I have one 3810 with 6 AVM ports out of which 2 are FXS. At one of the FXS I connect telephone instrument I get dial tone. My question is, is there any way to configure VoIP in the lab using these two routers? I have tried configuring both the fxs port in 3640 and 3810.I am pasting the config also.But some how I am not able to dial from one place to another. I am getting busy line. I have configured frame-relay cloud bet. 3640 and 3810 also and tried. First I tried with just serial IP cloud in between and then F/R I have done. I am doubtful at 3810 side. This is runn conf from 3640. version 12.1 service timestamps debug uptime service timestamps log uptime no service password-encryption ! hostname 3640 ! ! ! ! ! ! ip subnet-zero ! ! ! interface FastEthernet0/0 no ip address shutdown duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! interface ATM2/0 no ip address shutdown no atm ilmi-keepalive no scrambling-payload ! interface ATM2/1 no ip address shutdown no atm ilmi-keepalive no scrambling-payload ! interface ATM2/2 no ip address shutdown no atm ilmi-keepalive no scrambling-payload ! interface ATM2/3 no ip address shutdown no atm ilmi-keepalive no scrambling-payload ! interface Serial3/0 no ip address shutdown ! interface Serial3/1 encap frame-relay bandwidth 64 ip address 192.168.11.2 255.255.255.0 clockrate 64000 ip rtp header-compression ip rtp compression-connections 25 ! interface Serial3/2 no ip address shutdown ! interface Serial3/3 no ip address shutdown ! ip classless no ip http server ! ! voice-port 1/0/0 ! voice-port 1/0/1 ! dial-peer voice 402 voip destination-pattern 140855 session target ipv4:192.168.11.3 dial-peer voice 401 pots destination-pattern 1408553737 port 1/0/0 ! router rip netwrok 192.168.11.0 line con 0 transport input none line aux 0 line vty 0 4 ! end This is from 3810 sho ru Building configuration... Current configuration: ! version 12.0 service timestamps debug uptime service timestamps log uptime no service password-encryption ! hostname 3810 ! ! network-clock base-rate 56k ip subnet-zero ! ! ! ! ! interface Ethernet0 no ip address no ip directed-broadcast shutdown ! interface Serial0 ip address 192.168.11.3 255.255.255.0 no ip directed-broadcast encapsulation frame-relay no ip mroute-cache no fair-queue frame-relay traffic-shaping frame-relay ip rtp header-compression ! interface Serial1 no ip address no ip directed-broadcast shutdown ! interface Switch0 no ip address no ip directed-broadcast encapsulation frame-relay no fair-queue ! interface FR-ATM20 no ip address no ip directed-broadcast shutdown ! ip classless ip route 0.0.0.0 0.0.0.0 Serial0 no ip http server ! ! ! gatekeeper shutdown ! ! line con 0 transport input none line aux 0 line 2 3 line vty 0 4 ! ! voice-port 1/1 timeouts call-disconnect 0 ! voice-port 1/2 timeouts call-disconnect 0 ! ! dial-peer voice 501 pots destination-pattern 140855 port 1/1 ! dial-peer voice 502 vofr (i have my doubts here as I can not define voip pots here) destination-pattern 1408553737 session target Serial 0 201 (this is DLCI value) ! end I can dial but I get all the time busy tone from either side. As i am new, i might have made some silly mistake here so if u can help me then it would be really appreciated. Thanx a lot in advance ... KEYUR. _ FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP books [7:4260]
hi can any body suggest me one good book for VOIP? tanx ravee Get free email and a permanent address at http://www.netaddress.com/?N=1 Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=4260&t=4260 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP redudancy [7:5621]
VoIP requires a lot of redundancy, from my experience so far. Granted, each site has all it's voice and data trunks going to the same CO. For instance in a current install: 12 Small offices (10-20 phones): 4 voice trunks 2 backup BRIs (for extra voice trunks and/or backup data) Full/Fractional T1 to frame relay cloud Main site (80 phones): T1 PRI for voice w/DID T1 PRI for data (analog dialup users mainly, backup lines for remote BRIs) Multiple T1s to frame clouds If all local trunks at a branch are in use, the last line forwards to a DID on the PRI and forwards back internally to that site. If the frame goes down, we've got enough BRIs to still function (in a much more limited capacity, but the VoIP network has priority and won't go down, data will suffer but still work and is primarilly terminal/text based or Citrix clients). We purposely have a small number of trunks at each site (most sites originally had 6-10 trunks), but part of the goal here is to save money with less trunks and also do toll-bypass (this customer has suppliers/customers/sites all over California, Neveda and Oregon). Also there is the advantage of faster data paths when the dynamically allocated VoIP isn't in use. So far, it seems to work good (only 2 remote sites up so far, working out kinks with other items like Unity voicemail/faxing and just waiting on frame circuits to be upgraded). 3 remote sites will probably be brought up in the next month, so we'll see how well it all scales (shouldn't matter, the hardest was just getting it all working right with the first 2 remote sites). Even though it's not dynamic, if for instance one sites local telco was having analog trunk problems, we could block all calls from being routed out those trunks (which I guess you can do in a traditional PBX system, but you probably wouldn't be doing with key systems which would be going in these small offices). We're actually doing that for the second of two sites to keep the local trunks open for incoming calls as the local telco goofed on a date to have the trunks forward long distance, so if all 4 local trunks are in use the 5th+ caller get forwarded to the other 4 trunks no longer in use and get a "disconnect" message. -- Jason Roysdon, CCNP+Security/CCDP, MCSE, CNA, Network+, A+ List email: [EMAIL PROTECTED] Homepage: http://jason.artoo.net/ ""John Neiberger"" wrote in message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > >Aside from Priscilla (not Geoff Huston): What if the phone system had > >evolved this way? How many companies have redundant trunk lines? Don't > we > >just assume that the "phone company" will always provide service? We > don't > >multihome to the phone system, (do we?) > > [Warning: Slightly OT] > > Actually, here at our corporate headquarters we have redundant fiber > connections to separate telco central offices. The fiber links exit on > opposite sides of the building to frustrate Backhoe Bob in case he tries > to cut through them. These links carry two separate channelized DS-3s > that carry our voice and data circuits. Theoretically, you could bomb > one of the CO and we'd be just fine; both DS-3s would stay up and > running without a hiccup. I hope we never have occasion to test this. > > > John > FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=5621&t=5621 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP QoS [7:6586]
Hi Everybody, I have configured the following parameters on the serial interface for VoIP.The quality of the calls is not very good during working hours you can feel some delay/small interruptions while using it. interface serial 0 ip tcp header-compression iphc-format no ip mroute-cache no fair-queue ip rtp header-compression iphc-format ip rtp priority 16384 16383 64 Could anybody suggest any other alternative to improve the quality. Will removing the compression help ? Do I need to have something like Link Fragmentation and Interleaving configured. Thanks Amit __ Do You Yahoo!? Get personalized email addresses from Yahoo! Mail - only $35 a year! http://personal.mail.yahoo.com/ Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=6586&t=6586 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
cisco VOIP [7:10059]
Hi, Do you have any configuration sample for VOIP . Thanks. [GroupStudy.com removed an attachment of type text/x-vcard which had a name of khairuy.vcf] Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=10059&t=10059 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP certification?? [7:10113]
Is there an industry certification for VOIP? If you never worked with this technology, how do people show they have adequate knowledge if they don't get practical experience?? Mark B Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=10113&t=10113 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP certification? [7:10115]
Is there an industry certification for VOIP? If you never worked with this technology, how do people show they have adequate knowledge if they don't get practical experience?? Mark B Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=10115&t=10115 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOip&VoFR [7:34947]
Hi, Gang, I have a small thing I am still clear about, even though I thought it over. what's the difference between VOip over frame Relay and VoFR? Does it mean that for Voip the voice is taken as the data regarded as layer 4 data, then encapsulated in layer 3 ip,then further be encapsulated as layer 2 frame relay frames. but for VoFR, the voice don't need to be encapsulated by layer 3 protocol, and directly passed to layer 2 frame Relay network? could you shed me light over it? I could not turn it over on my brain. Many thanks, __ === PB@KCb7Q5gWSSJOd (http://mail.sina.com.cn) PB@K7V@`PEO"#:GaKI6)TD#,A?Im6(VF,:CPEO"@4URDc! (http://classad.sina.com.cn/) Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=34947&t=34947 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP Certification [7:35879]
Has anyone out there attempted the CIPT, CVOICE, and QOS tests yet for the CCNP / Voice Specialization certification? If so, what training did you use and what was the tone of the tests? Thanks, Kelley. Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=35879&t=35879 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP problem [7:36396]
Hi all I've a problem with a voice router I'm getting DSP timeout errors on the far end (egress) router and I was wondering if anyone has any ideas. See the text below for the error, it appears after the call is disconnected with "normal call clearing", we use E1s. A reboot will make the problem go away for a short while and we using 12.2(4)T on a 3640. The call routing is fine and I can make csim calls from the far end router to my local router and to my phone no problem, in the other direction I get DSP timeouts. Cheers Pat 10w5d: %VTSP-3-DSP_TIMEOUT: DSP timeout on event 0x6: DSP ID=0x1: DSP Disc (call mode=0) 10w5d: %VTSP-3-DSP_TIMEOUT: DSP timeout on event 0x6: DSP ID=0x1: DSP error stats (call mode=1658181684), chnl info(1, 0, 0) Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36396&t=36396 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP monitoring [7:36625]
Hi I'm after some tips for monitoring a couple of VoIP routers, as there are only two routers buying tools isn't going to be very cost effective. I've used the early versions of CVM (which was very funny), we use Cisco Works 2000, but don't have the add on CVM product, and Openview. I'm planning on automatically re-route calls on failure, but I'd like to know about the failure so we can react, any ideas or pointers? Cheers Pat -- email me on : [EMAIL PROTECTED] Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36625&t=36625 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP help... [7:36997]
Hi Experts, My organisation is going for VoIP implementation, Can any one send me links & Docs for implementation of VoIP. TIA. It __ Do You Yahoo!? Yahoo! Greetings - Send FREE e-cards for every occasion! http://greetings.yahoo.com Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=36997&t=36997 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VOIP billing [7:38756]
Hai, Is there any billing solution available for VOIP in cisco products.?? Thanks, Kiran Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=38756&t=38756 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: VOIP....... [7:39741]
Depends on how deep you want to go... Syngress' Configuring Cisco Voice over IP is a good start, if a little old (2000). If you are coming from the data side, The Essential Guide to Telecommunications by Dodd is a good primer on how the PSTN works. For the hard-core, Cisco Press has Cisco Voice over Frame Relay, ATM, and IP. This is a must for the CVoice exam. Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=39804&t=39741 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VOIP....... [7:39741]
You may want to check out the Partner E-Learning section of Cisco's website. If you have a CCO id you can access some of their VoD and web based FREE training such as IP Telephony Fundamentals. Cheers ""Mahesh"" wrote in message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > Hi, > > Any one can guide me for the VOIP Basic Technology. So that it can help me > for understanding the technology. Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=39909&t=39741 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Voip troubleshooting [7:11429]
Hi Everyone, I used the following cmd to get the following output. Can somebody elaborate on what the parameters lost/delay shown below indicate. show call history voice brief 57EB : 288131106hs.11447 +845 +4266 pid:22 Originate +78877462513 dur 00:00:34 tx:592/11840 rx:749/14980 10 (normal call clearing.) IP 136.225.219.169:18240 rtt:636ms pl:13880/830ms lost:23/30/40 delay:99/89/170 ms g729r8 Thanks & Regards Amit __ Do You Yahoo!? Get personalized email addresses from Yahoo! Mail http://personal.mail.yahoo.com/ Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=11429&t=11429 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
cisco voip [7:31100]
Hello, I'd like to study Cisco VOIP. But there are too many papers on CCO. Anyone can recommend a good URL or book? Thanks in advance. Jim __ Do You Yahoo!? Send FREE video emails in Yahoo! Mail! http://promo.yahoo.com/videomail/ Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=31100&t=31100 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VOIP [7:31615]
Under global configuration mode, type in "enable voip" and you are all set... Seriously, though I'm not VOIP expert, I think you need to provide a whole lot more information before anyone can help you... Jim --- Amit Bhasin wrote: > Hi all, > I have to configure VOIP at my end, i am working on > CISCO equippment.. > Router: Cisco 3660 > RAS: AS 5300 > can anyone tell me how to go about configuring it at > my end.. > > Regards, > Amit Bhasin > > > _ > Do You Yahoo!? > Get your free @yahoo.com address at > http://mail.yahoo.com [EMAIL PROTECTED] __ Do You Yahoo!? Send FREE video emails in Yahoo! Mail! http://promo.yahoo.com/videomail/ Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=31620&t=31615 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VOIP [7:31615]
Jim's correct. If you can't come up with some more information regarding your scenario, its difficult for members of the group to point you in the right direction. Check out this link for some help: http://www.cisco.com/univercd/cc/td/doc/product/voice/ip_tele/ Regards. ""Jim Bond"" wrote in message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > Under global configuration mode, type in "enable voip" > and you are all set... > > Seriously, though I'm not VOIP expert, I think you > need to provide a whole lot more information before > anyone can help you... > > Jim > > --- Amit Bhasin wrote: > > Hi all, > > I have to configure VOIP at my end, i am working on > > CISCO equippment.. > > Router: Cisco 3660 > > RAS: AS 5300 > > can anyone tell me how to go about configuring it at > > my end.. > > > > Regards, > > Amit Bhasin > > > > > > > _ > > Do You Yahoo!? > > Get your free @yahoo.com address at > > http://mail.yahoo.com > [EMAIL PROTECTED] > > > __ > Do You Yahoo!? > Send FREE video emails in Yahoo! Mail! > http://promo.yahoo.com/videomail/ Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=31699&t=31615 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP & Discontinous IP address
Hi, We setup small VoIP gateway using Cisco 1750 & FXO Card. Here is the diagram : Cisco 1750 A (WAN Interface using Public IP) -- LAN (Public IP) | | Internet | | Cisco 1750 B (WAN Interface using Private IP) -- LAN (Public IP) From Cisco 1750 A, we can make connection to Cisco 1750 B ... but after the phone ring, we can't hear any sound (I suspect the RTP can't be sent to cisco 1750 A). And from Cisco 1750 B can't make connection to Cisco 1750 A. The VoIP configuration is working well on the normal LAN (not using private IP). Is it because on the Cisco 1750 B WAN site using PRIVATE IP (discontinous IP) address ? regards, Ibam
RE: CIPT (VoIP) Class
I've changed my plans. Since I sent this I found out the class from Telcordia uses an older version of the course curriculum and software. I have since changed my plans. I'm now getting ready to sign up for a CIPT (Cisco IP Telephony, version 2) course from Information Innovation, Inc (III) in St. Louis. Before I fly all the way from Los Angeles to take it I was hoping to get another bit of feedback from you fine folks. Here are my questions: * Is the class worthwhile? * Has anybody had experience with this trainer? * Is there anything I simply must see in St. Louis while I'm there? :-) - Dennis -Original Message- From: Dennis Laganiere Sent: Thursday, August 24, 2000 11:54 AM To: Group Study Subject: CIPT (VoIP) Class I'm getting ready to sign up for a CIPT (Cisco IP Telephony) course from Telcordia Training and Education Ctr in Chicago. Before I fly all the way from Los Angeles to take it I was hoping to get some feedback from you fine folks. I'm one test short of my CCNP and have a technical background and I hope to see the system in action and experience the implementation, not just get an advertising dog-and-pony. Here are my questions: * Is the class worthwhile? * Has anybody had experience with this trainer? * Is there anything I simply must see in Chicago while I'm there? :-) - Dennis ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: CIPT (VoIP) Class
Information Innovation is terrific. Both Howard B. and I have done work with them and we know their staff, which consists of people who have been in the industry for years and know data and telephony really well. They have many CCIEs and Cisco Press authors on their staff. They helped develop the CIPT course. See http://www.infoii.com/ for more info. Priscilla At 06:49 PM 8/25/00, Dennis Laganiere wrote: >I've changed my plans. Since I sent this I found out the class from >Telcordia uses an older version of the course curriculum and software. > >I have since changed my plans. I'm now getting ready to sign up for a CIPT >(Cisco IP Telephony, version 2) course from Information Innovation, Inc >(III) in St. Louis. Before I fly all the way from Los Angeles to take it I >was hoping to get another bit of feedback from you fine folks. > >Here are my questions: >* Is the class worthwhile? >* Has anybody had experience with this trainer? >* Is there anything I simply must see in St. Louis while I'm there? :-) > - Dennis > > > >-Original Message- >From: Dennis Laganiere >Sent: Thursday, August 24, 2000 11:54 AM >To: Group Study >Subject: CIPT (VoIP) Class > > >I'm getting ready to sign up for a CIPT (Cisco IP Telephony) course from >Telcordia Training and Education Ctr in Chicago. Before I fly all the way >from Los Angeles to take it I was hoping to get some feedback from you fine >folks. I'm one test short of my CCNP and have a technical background and I >hope to see the system in action and experience the implementation, not just >get an advertising dog-and-pony. > >Here are my questions: >* Is the class worthwhile? >* Has anybody had experience with this trainer? >* Is there anything I simply must see in Chicago while I'm there? :-) > - Dennis > >___ >UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html >FAQ, list archives, and subscription info: http://www.groupstudy.com >Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] Priscilla Oppenheimer http://www.priscilla.com ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re:AS 5300 Voip config
hi all Actually what i wanted is to do a one stage dialing from Singapore to Hong > Kong with the As5300. > I would also attach a diagram for your info. > > > AS i got a AS 5300 in Singapore and Hong Kong. > In Singapore i got a E1 connected to the AS 5300. Normaly service provider > will configure the AS5300 for two stage dialing. First Dial into AS5300 > by the E1 DID then receive IVR , then they will be prompt to enter pin code > follow by destination phone number. Then the AS 5300 will establish a > connection to Hong Kong As 5300. Hong Kong AS 5300 will dial the destination > throught the T1 that is connected to As 5300 to the PSTN to reach the > destination. > > Now i would like to do a one stage dialing to my hong kong destination > number. > That mean when anyone dial my E1 number to access the AS5300, the As 5300 > will > able to do a direct connection to Hong Kong As 5300. Hong Kong As 5300 will > dial throught the T1 via PSTN to the destination . Like say my destination > number is 852 30027171. This destination number will be fixed. This number > will be my customer service number in Hong Kong. So i would like the public > in Singapore would able to just dial my E1 via PSTN have a direct connection > to mt cHong Kong customer service number (852 30027171) > > I would like to know how can i config AS 5300 to do this. If AS 5300 can't > do it , i hope that there are other solution for it. > > Thank You > with regards > chiaoliang > "king Net" <[EMAIL PROTECTED]> wrote in message [EMAIL PROTECTED]">news:[EMAIL PROTECTED]... > Any group outside the list who want to study together in the Tristate area > let me know. > > Get Your Private, Free E-mail from MSN Hotmail at http://www.hotmail.com > > ___ > UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html > FAQ, list archives, and subscription info: http://www.groupstudy.com > Report misconduct and Nondisclosure violations to [EMAIL PROTECTED] > ___ UPDATED Posting Guidelines: http://www.groupstudy.com/list/guide.html FAQ, list archives, and subscription info: http://www.groupstudy.com Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
