Voip Question

At the office where I work we use voip phones. I'm in the process of getting setup with a phone at home to use and also ******* after an iphone I was hoping to possibly combine the two.
What I was wondering is if there was a generic voip client anywhere on the horizon for the iphone. We have everything setup through our office I just need to be able to connect to the PBX here. The only 'offical' voip app I saw in the app store seemed to be bundled with service, and quite expensive I might add at 6c/min.
I'd like to be able to configure the office calls to come in through the iphone via voip and then the regular calls come in normally. Does anyone know if this is, or will soon be, possible.
Thanks.

Calls are free between Truphone users like Skype.
Dave M.
MacOSG Founder/Ambassador  An Apple User Group  iTunes: MacOSG Podcast
Creator of 'Mac611 - Mobile Mac Support' (designed exclusively for an iPhone/iPod touch)

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ITouch Javascript and Voip Questions

I'd like to get an IPod Touch or an HP iPAQ 211.
I need to be able to transfer some small html files that utilize javascript and rely on a small directory structure. I'd be glad to post the files or send them to anyone who requests them.
Basically I have several construction calculators that are coded in html and utilize javascript to compute the answer. The directory structure is simple just a single folder that contains two folders. One of the folders has the javascript and the other folder has the html calculators.
Just open the html calculator and populate the fields and press calculate then the javascript populates the answer fields. I'd be glad to send a pair of files for testing as i'd like to take the device with me when I leave for Afghanistan.
That leads to my next question does the iTouch 3g support Voip and some sort of instant messaging platform if utilized over a wi-fi connection?
Message was edited by: Mr Bro

I've seen youtube vids where files are transfered to the touch. I've read it supports javascript. Apple has a no refund policy so I'd like to have two files both under 60k transfered and set up in folders in the root directory to test if the html file can be opened from within the device and if the calculator will function properly to populate the answer when the javascript is ran via the compute button on the html page.
Thx for the tips/info.

Avaya VOIP Question

Hi,
I am connecting avaya 9630 VOIP phone to our cisco network. what is the cos / dscp markings the avaya phone uses by default?

I have a very large multisite Avaya PBX system, working on Cisco router and switches. The first thing I do is to make the Avaya phones work on the network like Cisco phones. One the 'ip-network-region' page I set call control to 24 (CS3), Leave RTP set to 46, and set video to 34 (AF41). I set the setting in the next field or 3, 5, and 4. respectively. In the 46xxsettings.txt file, I also set the phones with the same settings. And then if you really want to make certain that the phones are behaving, you can take those same settings from the 46xxsettings.txt file and use them in DHCP, although you may not need to.
Sent from Cisco Technical Support iPad App

Questions about VoIP, Can anybody help me?

Hi All,
I am learning VoIP now and I have serveral questions:
1. when I use "destination-pattern" command in "dial-peer" command, how will the number forwarded? For example: If I want to forward the number 123456 to ip x.x.x.x. If I use "destination-pattern 12...." to identify the number, the router will forward 123456 to x.x.x.x or 3456 to x.x.x.x? I think the answer is 123456. Am I right? And If I am right, there is another issue. What should I do if I just want to forward 3456 when received 123456? I think this is helpful to implement the dialing plan to use some digits as a prefix and discard the prefix to forward to another destination when received the number.
2. Can cisco router only support e.164 numbering plan? Can I use another? How can I explicity config the numbering plan in the router?
3. If there is a case like this:
PhoneA--R1--R2--R3--R4--PhoneB
R2 and R3 is connected via frame-relay. So when I use the "dial-peer" command in R1 and R4, should I use "voip" or "vofr" parameter?
Thank You!
Best Regards
Teru Lei

1) Here is an example of a dial-peer
Dial-peer voice 123456 voip #this mean this is a voip type dial peer name #123456
destination-pattern 123456T # mean when you will call 123456 and any number after that it will go to this dial peer
session target ipv4:192.168.0.1 # this will fwd the packet to that IP when you #will call 123456......
If you like to call somthing and send somthing else you can do translation rule
translation-rule 1
Rule 1 123456 3456
and on the dial-peer you need to set
translate-outgoing calling 1
2) Yes it support E.164
3) I would reccomand to use VOIP

Questions about VOIP and recording in Connect

Hi,
We have been using Connect for a long time, but have been doing the audio using AT&T teleconferencing. We were thinking about using VOIP for the webinars we do. I just have a few questions.
1. Is there a limit of the number of connections to VOIP like there is with the teleconferencing? Is there a point where the number on there could decrease the quality of the VOIP?
2. When we record the webinar, where can we save it to? Can we save it to our server?
Any help with these questions would be great. Anything else that would be good to know for doing VOIP and recording?
Thanks.
Lisa

We don't natively integrate with AT&T by the way - so you have to use Universal Voice in this case and set it up yourself. Not hard to do.
1. No limit on VOIP but we recommend using the optional SPEEX audio codec built into the Host Add-In if using VOIP a lot. It requires everyone to be using the Host Add-In though. Over 150 though and it might get unruly!
2. Recordings are saved to the Adobe Connect Central portal that every customer has for their rooms. You can create a local copy from there and then copy the local copy to another server. It's a Flash file so you may need to convert it if you don't have a player.

VoIP chalenging questions

Dear All,
I've got some chalenging questions about implementing Cisco VoIP and I looking for some fair answers:
1. What visible indications are invoked if IP phones default to survivable telephony mode?
2.How long will active calls last if primary call control is lost?
3.How does VoIP solution provide effortless IP phone addressing without having to change the addressing scheme of the existing IP data network?
4.Under what circumstances would a telephone outage affect the LAN/PC communications to the network?
5.Does the system support clustering? If so, across how many systems?
6.What is the highest number of IP devices currently supported under a single instance of the software? How many IP devices are supported by a cluster?
7.Does the system have a web-based or GUI-based administration tool? Is this included in the base price?
8.Can a single administration portal/tool be used to manage all systems within an enterprise simultaneously? If not, explain the administrative process?
9.How does the system provide for fault tolerance? f.i. Is it ?hot standby redundancy?
10.If/when a switchover event occurs, describe the events and detail what a subscriber and caller would experience.
11.How are databases synchronized?
12.How is time clocking designed?
13.How is clocking synchronized?
14.What virus and intrusion protection is included ?
15.What is the response process and what activities are conducted in the immediate identification of new viruses, denial of service and similar attacks?
16.What activities or applications do you recommended / conduct to detect and block VoIP spam?
17.What toll fraud protection is available? Identify its process. How is it managed & monitored?
18.How many pairs of wires are needed to support non- IP telephones: digital telephones, analog stations, terminals and consoles?
19.Identify and describe the distance limitations and wire gauge limitations to your station equipment, consoles, administrative terminals, etc.
20.Do digital or IP telephone ?sidecar expansion? units require local power? If so, identify & explain the requirement.
21.Describe the power requirements for all system components.
22.Wwhat are the two-hour UPS or ?battery? backup capabilities ?
23.What system management diagnostic tools can be use?
a.What types of event logs does the system support?
b.What diagnostic reports are available to aid in isolating faults? Can diagnostics be remotely accessed?
c.Are the system?s diagnostic tools SNMP compliant?
24.What specific VoIP system diagnostic & management tools can be use?
a.What types of event logs does the system support?
b.What VoIP reports are available to aid in isolating faults? Can diagnostics be remotely accessed?
c.Are the system?s diagnostic tools SNMP compliant?
25.What remote diagnostics are available? Can administrators see and access any alarms or alerts on the system from remote terminals?
26.What system monitoring report(s) are available ? How about any third party monitoring products for the followings:
-Status trunking
-Real-time traffic
-Status of call center
-Status of call center components
-Status of all routing components
-Status of all remote components
-Status of individual stations
-Status of all gateway ports
-Call trace capability
I would like to thanks in advance to each of you for your time and consideration.
Regards,
Mike

Someone would spend a day typing everything up for you. If I were you, I would pick up a CallManager book. "Cisco IP Telphony: Planing, Design, Implentation, Operation and Optimization" There is also "Cisco CallManager Best Practice."
Or call your local Cisco Sales guy, they have local SE's that can give a briefing of the product in more details.
Most of know all the answers to those questions, but it would take any of us 2-3 hours to pull up. The books Cisco have out are great for reference and quick lookups for this stuff.
good luck!

VOIP Design Question in reference to DHCP

Background:
I have setup VOIP at 3 different remote sites connected via Layer 3 routing and single mode fiber. Each site has it's own voice gateway. I have it setup that the DHCP scopes for data and voice are received from our central data center offsite.
Question:
From a design standpoint, what is best practices in reference where the DHCP server for the voice vlan should be located? i.e. When voice phones request an IP address they are having to go out via Layer 3 and get the address via the DHCP server located off site.
Issue:
If the remote site were to lose power and network connectivity thus losing a way to the DHCP server that is off site. If power were restored without network connectivty the phones will not be able to get an address to be able to reach the local voice gateway.
Thank you for any and all help!
Travis

You can configure your voice gateway with the local DHCP scope, that way DHCP is significant to the local site. This would provide you with DHCP functionality should you loose your WAN connections.
--Matt

CISCO VOIP GW AS5400 Questions

Dear all,
I have some questions with Cisco VoIP GW AS 5400:
1) Our customer request that “MUST NOT put any silence suppression attributes in the SIP offer( specifically Invite ),
since A fmtp SDP attribute for silence suppression should be defined if silence suppression is on."
Can CISCO GW guarantee that not put any silence suppression attributes in the SIP offer and how to fulfill this?
2) Our customer request that “ptimes MUST NOT be a odd numbered one”
Can CISCO GW guarantee it ?
3) Support Rport extension RFC3581
Check it and find seems not on GW compliance RFC list, please confirm
4) Support NATed SIP and RTP Traffic?
5) For DTMF-relay, can Only found the "SIP-notify" item , is GW support DTMF-relay by SIP Info/ Subscribe?
Is there anyone who can help me about these questions?
Thank you very very much!
Jennifer

Hi
It is a switch port on the phone as you can read in the following link:
http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html
But it for sure has limitations and is not comparable to normal switch.
So you can for sure discover your PC behind the ip phone via SNMP. All you need to make SNMP Queries is an IP connection which is given.
As i posted bevor i don't think you can query the phone via SNMP because
i do not know about options to configure SNMP on the phone. I also used a mib browser to query an ip phone...but as expected the phone did not answer.
Hope that helps
Roger

VOIP in US; several questions

I had a first gen iphone and had to return it because zero bars in my area.
I live in Salt lake city, utah.
Dead spot where I live for all Cell phone carriers.
I currently us T-mobiles hot spot phone; which allows me to us my Apple airport to make WIFi calls.
Question 1. Is it possible to make VOIP calls using the 3G iphone.?
Question 2. If possible what Software; VOIP carrier etc. is needed to make this work?
Question 3. Any and all advice is appreciated.
I am a newbe in this area and am looking for all the help I can get.
I don't want to purchase another iphone and suffer the $100 + cost to return it when it doesn't work.
ATT says there is excellent coverage in my area. WRONG.
thanks in advance for any and all help.

Fring will also allow FREE calls for Fring-to-Fring users and Fring-to-Skype users. Its only when you call a land line you have to buy skype minutes.
See also this review of various software for the iphone:
http://www.voip-sol.com/iphone-voip-apps-the-top-softphones-for-iphone/

Questions about Gateways and VOIP

First~~~ please do not be mad with me.since my questions sound some how stupid.
1. We use gateway between PSTN and IP network. But in PSTN, the voice is analogue. But in IP network, the data is digital. Which role does gateway act? The D/A converter, beside some necessary protocol translation?
2. What is the difference of PSTN and ISDN? I think there is no requirement of gateway between the connection of ISDN and IP network.
3. In PSTN, the signaling protocol is SS7. In IP network, the signaling protocol is TCP for data transmission, and SIP/H.323 for voice transmission such as VOIP. What is the signaling protocol for ISDN?
4. When we talk each other in MSN or ICQ,by microphone and speaker, which protocol in use in this case? still H.323? I do not think so. I guess still TCP?!
thanks for your answer~~~~

Hi lily dont worry we r here to learn!
1. PSTN and IP
A gateway is the point at which a circuit-switched call is encoded and repackaged into IP packets.
For that we use DSP (Digital Signal Processor)
The Gateway converts media provided in one type of network to the format required for another type of network. For example, a Gateway could terminate bearer channels from a switched circuit network (i.e., DS0s) and media streams from a packet network (e.g., RTP streams in an IP network). This gateway may be capable of processing audio, video and T.120 alone or in any combination, and is capable of full duplex media translations. The Gateway may also play audio/video messages and performs other IVR functions, or may perform media conferencing.
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 one of the specifications for transmitting multimedia (voice, video, fax and data) across a network: H.323 (ITU), MGCP (level 3,Bellcore, Cisco, Nortel), MEGACO/H.GCP (IETF), SIP (IETF), T.38 (ITU), SIGTRAN (IETF), Skinny (Cisco) etc.
I always think in Analog (FXO, FXS) and Digital T1, E1, E&M, etc) and IP packets.
2. PSTN = Public Switch Telephony Network
The worldwide voice telephone network. Once only an analog system, the heart of most telephone networks today is all digital.
ISDN (Integrated Services Digital Network) is an all digital communications line that allows for the transmission of voice, data, video and graphics, at very high speeds, over standard communication lines. ISDN provides a single, common interface with which to access digital communications services that are required by varying devices, while remaining transparent to the user
http://www.ralphb.net/ISDN/defs.html
Yes, you need a GW for the conversion.
3. ISDN for voice transmission Q921/Q931 (Layer 2 and Layer 3)
4. I believe now MSN uses SIP and SIMPLE.
http://www.protocols.com/pbook/VoIPFamily.htm
http://www.packetizer.com/voip/
HTH
//G

Auto QoS Voip Trust question

Hello,
I am a little confused on the auto qos voip trust command. I am reading that it should be used on interior switchports like trunk ports connected to another switch. I guess my question is this: If the auto qos voip command tells the switchport to trust the incoming CoS values, then isnt it redundant to tell the port to "trust" the incoming CoS value when the auto qos voip command is already accomplishing this? Or does this have something to do with trusting DSCP values instead of CoS values? If so, why are we trusting DSCP values at the uplinks and not CoS values coming from the IP Phone? Im so confused. Thanks for any light you can shed on this topic.
Chris.

Chris
Yes, the trust value used for "auto qos voip trust" depends on the operation of the port as you say.
And yes when you use the "cisco-phone" option it will trust the values but as you say only if it detects a Cisco IP phone and it does this by using CDP.
What is not entirely clear, at least to me, is exactly which markings the "cisco-phone" option trusts. There seems to be conflicting information but the configuration guide talk about DSCP markings so it may be those but i can't say for sure as i have limited experience in that area.
Jon

C3-01 WiFi / WLAN & VoIP / SIP Question

Hello,
I am very interested in purchasing a C3-01, as I don't want a smart phone, but I like the advanced features this phone offers. Specifically, I want to make use of the VoIP SIP client that comes built in.
My question is this: I will be using VoIP / SIP on the WLAN (Wi-Fi) connection. In another post on this forum, I saw a user saying that Wi-Fi gets turned off automatically when the phone is locked.
So, does this mean that if I've got a VoIP account set up, and the C3-01 locks (either manually or automatically), that the WLAN will drop out and I won't be able to get incoming VoIP calls?
If this is how it works, this is a pretty major oversight, and I will look for something else.
Thanks
Solved!
Go to Solution.

Yeah, Sadly Some Country still have the v05.68 as the latest firware, But the reason might be that the Firmware released in your Country may contain the fixes included in the Latest Firware for Our Country.
So thats ok, If Nokia Finds any fixes to be done then, they will release a new Firware Update for your Country. Be Rest Assured.
And Buy C3-01.
If I've helped in any way, a click upon the White star to the left would always be appreciated.
If however my answer also solves your problem clicking below " ACCEPT AS SOLUTION " it will benefit other users!

VOIP on iPhone question

Hi all,
Using the latest v3.1.2 OS on my iPhone 3GS and have a question.
Does anyone know of any good SIP client or other VOIP client that wil work over 3G or are all VOIP clients limited to Wifi?
Many thanks!

Ok... This thing is a bit more complicated than just that.
It's true AT&T has changed its stance regarding VOIP over 3G. It is however also true that a lot of other carriers with whom Apple has distribution contracts are not that happy about VOIP over 3G. Up until april or may this year there were a few applications that allowed VOIP over 3G (Truephone, for example) but than Apple, following demands from carriers, has asked these applications to be modified.
I suppose right now a lot of diplomacy is in action to make it possible to "roll-back" the veto. It's going to take time, however. One solution Truephone has found to the problem is called "Truephone anywhere": http://www.truphone.com/applications/pricing/truphone_anywhere.html
I would also suggest you'd think a bit about using VOIP over 3G and possible downsides of such a use. Firstly 3G data network is not really as fast as your home broadband connection both in terms of bandwidth and in terms of round-trip times. Now – a decent quality VOIP call requires not as much bandwidth (64KBit/sec is enough) as it requires *short roundtrip* to prevent something that is called latency. And that's something 3G network can hardly offer. A usual roundtrip on a 3G network is of 500-600ms. Add to this switching latency and we're somewhere around 1 second of latency in best case. If the conditions are not perfect, we can go towards 2-3 seconds of latency and that's no good as two problems would present:
1) Echo
2) Latency glitches
The first problem is that annoying hearing your own voice from the other side a few seconds late. When the latency is too much for echo-cancellation to cut it out you start getting plenty of that stuff.
The second problem is really annoying as it prevents you from conversing normally. The pauses in the conversation become longer and it gets really hard to carry on a normal discourse.
So basically I'd say that to see real good VOIP quality on a cellular network we'd have to wait for 4G network to kick in. Given it is going to be a fully featured IP network, the phones are going to be fully featured VOIP terminals with the possibility to keep several lines (including lines from different operators and countries) on a single mobile phone and all the rest. But that's something we're going to see in a couple of years from now...

Newb VoIP End to End Qos Questions

I'm at a site that has Layer 3 Cisco switches and routers end to end. Cisco IP phones plug directly into Layer 3 switches with various routers in the core.
1) If all connections are Layer 3 do I need to concern myself with trusing COS? I would trust DSCP in this case yes?
2) Cisco phones by default send voice rtp audio with DSCP marked as EF, is that correct? If not, I know how to create policy-maps and reclassify it. My thought was it is already classified as high as possible.
3) I've read that QoS must be end to end or you don't really have a full implementation, as in the one device your traffic passes through that is not performing QoS will likely be running FIFO or WFQ, which do not give precedence to class based traffic. So in theory I need to implement CBWFQ or LLQ on every Layer 3 device end to end that RTP audio will touch. Do I have this correct?
4) If the voice traffic has the potential to travel to most Layer 3 devices in the network as it travels between the Call Manager and phones, I need to apply QoS outbound on each Layer 3 interface that is likely to be in a voice path yes?
Sorry, I'm a newb, I understand it in theory but I still hit the wall on occassion with my light saber while blindfolded.
Thanks, Mike

Hi
1) Yep - no trunk (dot1q/ISL) = no CoS as it's in the trunk header, so use DSCP.
2) Yep, traffic will be marked. You can trust it, or trust it conditionally in a service policy and police it.
3) Yep, more or less. It's very important that on any switches that have QoS enabled, you have proper trust states set on all the ports, otherwise the DSCP will be remarked to 0. This is worse that not particularly prioritizing voice on a particular device, as no other devices further down the chain will see any DSCP markings to act on.
4) Erm.. kind of. If it's a layer 3 link on a switch, then you don't configure it as you would on a router. The QoS is done in hardware, so your common/garden autoqos type config should see you right if you really don't know what you are doing. If you use auto qos voip trust, make sure you set it to mls qos trust dscp after as it will probably default to CoS. Best to keep it consistent...
Regards
Aaron
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P35 Neo2-FR and VOIP card questions

Hi,
I'm evaluating buying this motherboard and would like to know a couple of things:
- Do all P35 Neo2-FR include ALC-888T codec instead of ALC-888? If not, how do I find out which ones do and which ones don't? I'm asking this because it clearly says "Optional" in the product web page, but it doesn't specify which ones have the ALC-888T codec
- If I understand correctly the only difference between ALC-888 and ALC-888T is VOIP card support, is this so?
- Has anyone tried using this MB with a VOIP card? How did it go? (Ease of installation, ease of use, sound quality, etc.)
- Are VOIP cards common? I just did a quick search at Newegg and I couldn't find any. Not a single one ???
- I think the P35 Diamond MB comes bundled with a VOIP card. I guess the P 35 Neo2-FR DOESN'T come bundled with one, am I correct?
- If anyone has bought a VOIP card to use with this board, where did you buy it and approximately how much did you pay for it?
Any help GREATLY appreciated

Thanks everyone for your replies.
Quote from: Maesus on 30-November-07, 14:51:39
that's only P35 Platinum, not P35 Neo2-FR :(
city_zen, is VOIP a must-have feature for you?
No, not really a must-have feature, but it would be a very nice thing to have on my board. I'd definitely use it if it was in fact available (i.e. ALC-888T) and I can get a hold of a VOIP card. However, it's looking like that's not very likely to happen ... Oh, well ...
Anyway, this is a very good board even w/o the VOIP feature, so I'm still considering buying it.

Voip Question

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