Anyone know if using voipstunt or voipdiscount, can we send faxes?

lucky 206

The issue is that it works for about 30 minutes after resetting both the Aztech 600EW & Linksys PAP2 then its back to no ringing tone when I call the other party. I tried calling my mobile number and it rings BUT I can't hear any ringing tone.

I have already setup QOS, disabled NAT & firewall. I even tried to put it in the DMZ but the same thing still happens. I'm not sure whether it is a firmware issue. My Aztech firmware is 66.68.2 and the PAP2 is 1.12 (if I'm not mistaken).

It could also probably be due to latency issue as I was using VoipStunt.

I am using 1mbit package. I have tried all the available codecs and it is still the same.

Google for FoIP.

Okay laa, forgot today is Sunday so I'll have to ask my VoIP folks tomorrow. As for the Aztech thing "curmzmz" seems to have some strong feelings about Aztech not being very reliable/stable. Got me thinking of changing out to Cisco gear through and through... once again "dependant on budget"... btw, I'm using 66.68 as well (I believe, in the office just now so can't check) it's to bad Aztech isn't adjusting to market needs as I know they make some of the stuff here and wanted to support malaysian company subsiderary and all... oh well, money goes to "tried" and "true"...

I'll give more intricate details of my setup when I return home tonight... inshaAllah... using 512/256 StreamyHex connection; Aztech 600EW/Zyxel Prestige 2002; SIP server is in Australia; calls are crisp/sharp/no echo... when I call ummi in the USA she sounds like in next room. Aussie calls okay too... calling mobiles is a bit dicey as I think most mobile systems are already on IP based systems and there may be a little hardware interchange issue there (once again the VoIP guys will know; try to get more info tomorrow out of them as well). eh, what's the best setting to give priority to VoIP in the Aztech... I've looked at the QoS screen then got kind of scurred as everything "works" now and the Aztech seems kind of "sensitive" to changes; if you know what I mean... hehehehehe

Linksys PaP2 config (if asked you didn't get this from me):

(using Cisco Call Manger on backend; latest firmware on PaP2 etc. hope this helps may need to tweak stuff for you specific location; yadda.. yadda... yadda...)

Dial Tone: 400@-19,425@-19,450@-19;10(*/0/1+2­ +3)

Second Dial Tone: 420@-19,520@-19;10(*/0/1+2)

Outside Dial Tone: 420@-16;10(*/0/1)

Prompt Tone: 520@-19,620@-19;10(*/0/1+2)

Busy Tone: 425@-19;10(.375/.375/1)

Reorder Tone: 425@-19, 425@-29;60(.375/.375/1,.375/.375/2­ )

Off Hook Warning Tone: 480@-10,620@0;10(.125/.125/1+2)

Ring Back Tone: 400@-19,425@-19,450@-19;*(.4/.2/1+2+3,.4/.2/1+2+3,0/2/0)

Confirm Tone: 600@-16;1(.25/.25/1)

SIT1 Tone: 985@-16,1428@-16,1777@-16;20(.380/­ 0/1,.380/0/2,.380/0/3,0/4/0)

SIT2 Tone: 914@-16,1371@-16,1777@-16;20(.274/­ 0/1,.274/0/2,.380/0/3,0/4/0)

SIT3 Tone: 914@-16,1371@-16,1777@-16;20(.380/­ 0/1,.380/0/2,.380/0/3,0/4/0)

SIT4 Tone: 985@-16,1371@-16,1777@-16;20(.380/­ 0/1,.274/0/2,.380/0/3,0/4/0)

MWI Dial Tone: 400@-19,425@-19,450@-19;30(.1/.1/1+2);­10(*/0/1+2)

Cfwd Dial Tone: 350@-19,440@-19;2(.2/.2/1+2);10(*/­ 0/1+2)

Holding Tone: 600@-19;*(.1/.1/1,.1/.1/1,.1/9.5/1­ )

Conference Tone: 350@-19;20(.1/.1/1,.1/9.7/1)

Secure Call Indication Tone: 397@-19,507@-19;15(0/2/0,.2/.1/1,.­ 1/2.1/2)

VoIP PIN Tone: 600@-10;*(0/1/1,.1/.1/1,.1/.1/1,.1­ /.5/1)

PSTN PIN Tone: 600@-10;*(0/.7/1,.2/.1/1,.2/.1/1,.­ 2/.5/1)

PSTN Warning Tone: 600@-10;5(0/.5/1,.05/.05/1,.05/.7/­ 1)

Feature Invocation Tone: 350@-16;*(.1/.1/1)

Other Bits
Control Timer Values (sec)

Ring 1 Cadence: 60(.4/.2,.4/2)
Ring 2 Cadence: 60(.3/.2,1/.2,.3/4)

Hook Flash Timer Min: .07
Hook Flash Timer Max: .13
Time Zone: GMT + 8:00
FXS Port Impedance: 220+820||120nF

well i never tested dlink since i dont have any source frm Dlink
only can get Aztech both V300/310 ATAs also combo 600EVW
also Grandstream ATA (which is the best i ever tested but quite expensive)
SMC wireless ATA is good but damn expensive
and some china brand ATA like Welltek which give me headache juz to configure it to work

well one thing is true that aztech brand u have to update its firmware if anything is wrong wz it

as for my fav ehmm still yet to find ... i think for now ill juz stick to softphone Eyebeam rulezzz

Hi, Long time no see the VoIP sifu's (cruzzmz,Xybirium and lucky206). I happen to have experience with Dlink-DVG2001s. Well its ok, but the only thing that sucks is the configuration which is abit or I shall say very tedious.

And the other cons is that it only have 1 port. Happen to drop by in lowyatt latest market price for the Linksys PAP2 is RM 330+- (just an update for the folks.) Anyone need a Dlink DCG2001s (open box) or PAP2 brand new do contact me. I can give cheaper.

Anyway lucky 206 regarding your issue, it could be a few issues :
1) PAP2 Firmware
2)Codecs

I suggest try playing around with the codecs. Might help

I will also do a search on this see if I can find any information on this issue.

BTW off topic,

Just got my new WRT54GL and a spanking new PAP2. Sifu's in here, is there any setting I could use to optimise my VoIP experience since both are from the same manufacturers (Linksys). Do suggest me.

Just a gentle warning to fellow mates, do turn off your PAP2 and also you wireless router/modem as mu DLink DI-524 and PAP2 got struck lightning and it blew causing me a total lost clost to RM450 in splits seconds right infront of my eyes. (Even heard the pap2 popping) hehe....

CHeers

This post has been edited by rattan: Oct 17 2006, 08:45 PM

uh ur pap2 blew up infront of u eh that is why we ought to shutdown our modem and unplug the tel line frm it since lightning can stuck ur tel post .... neway about ur Q, is there any qos mechanism inside ur wireless router that u can tweak or trottle ur packet as in #1 VoIP packet #2 http #3 ftp etc if there is any do try the QoS thingy ... not sure ur router got it or not ... coz this features only include in VoIP ready modem

cruzzmz : there is a function on QoS. I have set port 5061 to highest. Hope it will help as the VoIP provider im using is using port 5061

uhmmm i dont think that is setting up QoS on ur router .... cos that is like port fwding ... u actually want to class ur netwrk packet right ??? as in while u r ftp-ing or torrent-ing u wanna make a quick VoIP call ... so u dont want ur ftp or torrent kacau ur VoIP quality right ???
For that u need a router that can classify ur packet ... as in if there is any VoIP packet in ur q it will jump the q n will go 1st right eventhough there r other tarffics .. well 4 that u need an adv router that can classified ur traffic

Well... even if your router classify your packet, does Streamyx classify your packet or is it the same as any other packet? If Streamyx does classify packets, then maybe we would all get better VoIP service.

I would think that they classify VoIP packet only if you are subscribed to their BB Phone service.

Hello rattan,

It was actually "Xybirium" whom was having a couple of issues with his PaP2. I do look forward to getting one of these though; unless I can get a Supria as I've heard good things about this model as well (still kind of wavering). First thinks first I have to do something about this Aztech600EVW and with the current TMnut throttling issue everything is on hold for the moment thanks for that though...

And this is an excellent point!!!... VoIP class 101 has just conclueded. You can do all sorts of things locally to improve quality; QoSing packets; having a top notch ATA; etc... at the end of the day when it's out on the wire in the big INET cloud all that goes up in smoke laaa.... sad but true... hehehehehe and with the recent TMnut clamp down on bandwidth it's really got me curious as the phones in the office use IP phones (they connect to the CallManager) and the ATA at home connects to the SIP server all travelling on port 80 (I believe). A forumer on another thread said they were monitoring usage by protocol... hhhmmm???? so if they see a whole lot of 80 or 8080 associated with my usage will I get hit over the head?... ayouyouyouyouy Thanks for that cruzzmz about the QoS that is in relation to ftp; torrenting etc... I thought as much but still don't trust this blasted Aztech

Checked with VoIP guru's today. Confirmed that the "backend" controls the usable "feature sets" given to an end users VoIP package. So essentially if you look at a VoIP package (Jaring for example) and they claim to allow you to do everything per normal; fax, voice mail, call forwarding, yadda, yadda, yadda, and you find that a "feature" doesn't work. This is do to mis-configuration on the "backend". Further IP Phones generally (if I understood the guru's breakdown) use a Call Manager or similar device on the backend (Cisco 7905 IP Phone, for example uses Cisco Call Manager on the "backend"). An ATA such as the PaP2 or Zyxel Prestige 2002 would use the SIP server on the "backend" (as would a softphone as well, I think). Within my company the VoIP guru's created a special "pipe" between the CM (CallManager) and SIP implementation to allow for certain services to be handed off and vice versa. As before, I'm not the/an expert. Just a monkey in the IT machine trying to get it to Go!... Mush?!... Move!... heheheheh Selamat Hari Raya... and Happy Depavalli... (saying it early as I'm no good with dates)

This post has been edited by lucky206: Oct 19 2006, 08:32 PM

Not a guru/sifu??? You seem to be a very expert in this VoIP. Me... I just learn by reading here and there on the Internet.

Really good of you to join us in this thread. Since you have better experience, we would all be really grateful if you share your experience here. By the way, as per my previous question, is there any VoIP proxy server or does it exist?

I would guess that even if my connection is in the 512kbps range BUT if my latency is excellent; there wouldn't be much of an issue in using VoIP. As it is now; even my ping to TMNet is around the 60ms range.

Please correct me if I'm wrong.

PS - rattan, you seem to have a lot of things blowing up in your face. Hehe... but I think you have a lot of stuff anyway, so that was probably a very "interesting" experience. Hopefully your eyebrows and eyelashes didn't get burnt.

as i have said earlier ... aztech is not that so good brand but who am i to comment on a brand

lucky206 u use port 80 to do ur rtp steaming ehmm ... so how is the quality ?? it is not that advisable to do that i thing if TMNet is using a level 7 switch then although u r using any common port also they can see ur packet header and will know which packet is which ... but i thing now they only give a damn only for torrents ... usually VoIP will use 5060 for signaling and 10000 above for streaming RTP ...

main point is yes we can do all sort of thing to our netwrk but when going to the big cloud if no class of services is offer at the edge router ehmm no point ....

Nope... beginner in VoIP (but have been trying to get into this for a while actually). I would say my expertise lies more in networking and internetworking. Get the packet from A to Z in the shortest time etc. Yes, "cruzzmz" you were right on point with your port info on the VoIP traffic (i.e. 5060 or something like that). My point was more in the method TMnut was using to shape/control/throttle traffic. It seems more like a "blanket" approach; and if monitoring the usage of a oft repeated protocol/port/site (as in website) is being used then that of course would have a negative effect on any/all traffic which is repetative in nature.
Having said that cruzzmz and I agree on the "cloud" issue as well. If the ISP's or in this case the great monopoly of TMnut doesn't have classifull packet switching going on (to include it's border routers and proxy servers) then a VoIP packet gets tossed into the spaghetti just like normal http traffic. Thus far I've found all of TMnut's DNS servers; and Proxy servers. Now I'm working on Border routers ... Essentially trying to create a map for myself just cause I like to know what's what and who is where... hehehehehe
I recently learned that Malaysia has got the fiber run all over the country already. Cruzzmz commented on another thread (or maybe earlier in this thread) that there were a few "known" problems with the current layout. The most obvious is the over-shadowing TMnut and their control of the "last mile"; two would be the copper lines (old, out-dated, possibly mis-matched [meaning some may have different core diameter opposed to others]; mis-configured routers; dslams; etc.).
As you can probably tell, I'm not from here. However, my wife is a bumi; and we've chosen Malaysia to be the place for kids and kin. Thusly, I feel as though I've got to do my part to try to implement change when/where cause is due. Their are plenty of highly intelligent Malaysians here (so the excuses for experts coming in and setting up TMnut then leaving it in the hands of incompetents is old hat). What we as bill paying Malaysians need to do is demand of this company (or maybe the govn' since it's got a controlling interest as well), better QoS.
Started going Political there... eerrgghhh excuse me... lastly on the VoIP thing which I can say definately makes a difference is the "codec". We use G729 only (although the ATA product we push has an option; G729=>G711). Now, G711 is lighter in weight (takes less of your broadband resources) however we found the G729 gave an overall "better" experience to folks. Thusly, the suggestion for a minimal 512 connection and G729 "only" on any/all ATA solutions pushed from us. I'm running on low bandwidth now so will sign off... Good holidays everyone... drive safe and slow... Peace!!!...

This post has been edited by lucky206: Oct 20 2006, 09:17 PM

Almost forgot these questions: VoIP Proxy Server=>I would presume that this would either be another way of naming a SIP Server; or someone has placed a true Proxy Server in front of the SIP server as an added layer of protection. I haven't heard of a specific VoIP Proxy Server implementation as of yet. Maybe others in the thread know more about this one.
The only thing I found in relationship to the package size (like 256, opposed to 512, opposed to a 1Mb package) is in direct correlation too the codec, number of nodes and activity on the network. For example, at the house I've got two desktops and a notebook all sharing one 512/256 StreamyHex package, right. When on my main box, having my email client open; one instance of a browser; IM I can do VoIP calls okay laaa. If the wifey jumps on her computer and starts up Limewire then the call quality gets shot. If I'm running my Torrent and wifey has LW running then don't even try.
However, if all three clients are say just browsing the web via Firefox, IE etc then it's okay laaa. This is due to QoS which I have yet to figure out on this damn Aztech as well and the fact that although it's advertised as a 512 package it has and will probably be for some time a 384 package (and I'm not upgrading to a 1Mb package to get the speed I should rightfully be getting now, you know). My hope is that getting QoS setup correctly on the router will fix the issue of what "others" are doing while I'm making calls etc; will see laaa....
Last note. Blowing shyte up. Or break/fixing is what I've done throughout my short but experienced little carreer. Of course I can't speak for others but for me; this is one of the best ways to really learn a product (especially hardware; I love breaking some hardware.). Like the ZyXel ATA I have at home right now. The boss gave it to me; I got home and had it busted in like 20 minutes (I broke it good too; he had to take it back to Aussie to let our "real" experts fix it... hehehehehe). What did I do?... I simply tried to change the web interface IP. That should be a simple task, right?... at any rate, breaking stuff can be a great way of learning stuff. The key is not to break it past the point of "YOU" being able to fix it... I'm sure we all hate forking over those hard earned RM's to the computer shop... alrighty... now I'm outta here...

Good evening everyone...

I just bought a Nokia 6280 and also signed up for the unlimited Celcom 3G at RM88/month. Luckily it is not tied to a contract so I can just leave if I feel disappointed.

Has anybody tried to install any softphone on the Nokia 6280 (something like Skype or SJPhone)? I was browsing around and that was why I was a bit late going through this forum.

If anyone has successfully tried it, please share your experience with us.

lucky206, cruzzmz & rattan - Maybe something new you would like to get your hands on. rattan could help by "donating" the equipment.... Hehe...

This post has been edited by Xybirium: Oct 20 2006, 10:34 PM