fredtheman

I testing axon 1.09 on win2003. Im new here. My question is we are planning as a voip provider.

You want to start a business as a VoIP provider... with Axon? You gotta be kidding... Get yourself a bunch of Linux servers, and try some serious software like SER, sipXPBX, or Asterisk.

Im asking an advice, do we need to be in public ip or private. Can anyone recommend me about my setup.

For obvious security reasons, SIP servers should be in the public part of the network, or in a DMZ. Besides, it's one less headache if the server isn't itself NATed.

Has anyone installed it on windows server system and what has been the results?

I have 1.09 running on W2003 Server, and sometimes, clients can no longer register (Error 408). No idea why.

Let's suppose I have assigned the number xxx....it would be nice that when calling xxx101 only the extension 101 would ring, like in a traditional pbx.

Most likely, Asterisk can be configured to do this (extracting information from the called number and ringing a given extension), but Axon doesn't do this, and since it's closed-source...

Depending on the codec you are using typically you need about 64k per call for good voice quality

Unless I'm mistaken, bandwidth can be lower than this with more efficient codecs (G729a?), but those are typically commercial and not available by default.

The question is, those axon work with winmodem (I have read that the quality is too poor)?

PBX's can't work with voice modems unless they have a driver for them that can convert analog voice into SIP messages.

Unless I'm mistaken, there's no PCI FXO card available for PBX's other than Asterisk, so you're left with getting a box device. I can recommend the Linksys 3102, but there are a few brands out there.

http://www.voip-info.org/wiki/

Hi

Do you also experience this with Axon 1.09? After a couple of days, clients that connect through the Net can no longer register with Axon. It could be the NAT firewall in front of the LAN where Axon sits, but since I mapped UDP 5060 on the firewall and registration resumes OK after I restart Axon, I suspect it's a bug in Axon.

For those with remote users, have you seen this too? If yes, do you know of a good, free watchdog program that could restart Axon every night?

Thanks.

Any takers?

If they're behind NAT firewalls, make sure the remote clients use a STUN server so that their public address is used in SIP messages. If that doesn't solve the issue, you might have to force their use of specific ports for RTP, and configure your router/firewall to map those ports to the clients.

Fred.

Im convinced the issue is NAT related, or something else related to SIP packet loss, but it must have something to do with Axon in particular, or my network with these users. Does Axon send some kind of Keep alive packets, which are easily dropped by some routers/networks, which would cause lost calls? Why is Axon sensitive like this?

You are using hardphones, and you tried moving them to a DMZ so as to rule out the NAT firewall from the picture (ie. phones and the Axon server are both directly accessible from the Net with no NAT in between), but calls still fail at some point. So I would guess that Axon is the culprit.

FWIW, here's what I observe using 1.09:

- sometimes, clients that are located behind a NAT can no longer register to Axon accross the Net

- sometimes, Axon no longer reports incoming calls that are routed by a Linksys SIP gateway.

In both cases, restarting Axon solves the issue. That doesn't definitely confirm that Axon has a bug there, but as a test, you could restart Axon every hour or so, and see if calls no longer go quiet.

They are two extension in Axon, they register perfectly and I can call from the remote one to the local and I can see the video in the two ways, but I only can hear the audio in the remote, in the local one I can't hear anything.

If the two sites use private addresses and you go through the Net without a VPN tunnel, make sure the two applications use STUN to translate their private address into the public one, and make sure your routers/firewalls either can open RTP ports dynamically or they are set up to map the right ports to the apps.

Hello

I use those two pieces of technology, and I need to rewrite caller ID information on the fly: Some customers don't have names that are descriptive enough in their Name section of CID (eg. "Mc Donald's"), so I would need to...

1. look up the CID number in a database

2. fetch the city this customer is located

3. rewrite the CID Name from "McDonald's" to "McDonald's 78th St"

4. Have this SIP packet be sent to either the Axon PBX software, or to all the clients that are part of the ring group.

Does anyone have an idea how I could do that, either on the Linksys or the Axon side? Unfortunately, Axon is a closed-source software, so I can't really do much on that end.

I was thinking maybe of having some kind of proxy that would parse and rewrite data between the Linksys and the Axon. What do you think?

Thank you.

Which one is the best ?

No idea, but I was looking for another free PBX for Windows and never heard of 3CX. Looks very good. I'll give it a shot.

Found a work-around: Just install a copy of a softphone like X-Lite, make that a member of the 701 ring group, and configure the softphone to forward all incoming calls to a remote VoIP account. Not a good solution if you have several users who all want to be reachable on their cellphone, but it solves the issue for one-two users... until Axon allows adding foreign VoIP accounts to a ring group.

Hello

I noticed that NCH put out a new software called Carousel, but I don't understand what it does. It seems to stand between an SIP gateway to handle incoming PSTN calls, and then route them to Axon, but I don't understand what it does that Axon doesn't already do. Anybody knows?

Cheers.

Hello, we installed FX-300 GSM Gateway that works seamlessly with VoIP Handytones or Linksys SPA-2000, and of course with AXON. Check out www.citivoip.com

Thanks for the link. It's also available at http://www.sipcpe.com/fx300GSM.html for $350 retail.

Hi

I'd like to add a VoIP account to an Axon group so that users who register with other SIP servers can still be be part of an Axon ring group.

Since Axon won't let me add that kind of extension directly in the "Ring On Extensions" section, I added this type of extension under the... Extension section:

SIP User ID or Number: sip:myremotevoipaccount@myvoipprovider.com

... and added it to the 701 ring group, but it won't ring when a PSTN call comes in.

Does it mean Axon won't dial out remote SIP accounts?

Thank you.

Hello

Among the users who must be part of the 701 default group when a call comes in from the PSTN is one who moves around a lot and, hence, wants calls to ring his cellphone instead of his IP phone at home/work.

Instead of getting my very own, personal IP-to-GSM gateway, I'd like to check out...

1. What VoIP providers have in store for this

2. How to set up Axon so that it sends notifications to that VoIP account for that precise users, while ringing regular extensions for the other users part of the 701 group.

Thank you!

i can get this to transfer call from outside and not call out, i get a dial sound two times then fast busy sound on disconnect. what the issue here.

I don't use the Linksys to dial out, so I don't know, but it's possible that your dialplan is wrong. You'll get more luck getting help on the Linksys by asking at Voxilla:

http://forum.voxilla.com/linksys-sipura-spa-users-group/

However with Sipgate, I get no incoming Audio!!!

Easy on the exclamation marks ;-)

As an SIP newbie, I would first investigate NAT not allowing RTP data packets in as it should. Make sure you use STUN and have a UPnP-capable router.

If still no luck, go ask in a VoIP-related forum, as you won't get much help over here, judging by the number of threads that didn't get any answer.

I am in country A and wish to call multiple numbers in country B.

You don't even need a PBX to do this, provided the numbers are phone numbers and not extensions in location B.

4: One SPA-(1/2/3)000 in loc.A & one SPA-3000 in loc.B, how to call to SPA-3000's PSTN?

http://www.sipura.com/support/spa3000faq/Section_2.html#4

5: Two SPA-3000 in two locations, how to call from PSTN in loc.A to PSTN in loc.B ?

http://www.sipura.com/support/spa3000faq/Section_2.html#5

I can call from outside line to PSTN and the call is transfered to Axon extensions (in group 701 for example), the extension rings but I don't see the Caller ID: however the softphone displays the name I gave to the gateway and the User ID for external line in Axon. I set "PSTN CID for VOIP CID" to yes, but the same.

It might be due to wrong regional settings. You'll have to find someone in the same country as yours and copy/paste his settings. This is especially true for caller ID encoding.

The other problem is how to dial from my Axon extension to an outside line via Linksys gateway.

No idea. I never used the Linksys for outgoing calls.

You'll get more help there:

http://forums.whirlpool.net.au/forum-threa...f=107&g=100

http://forum.voxilla.com/linksys-sipura-spa-users-group/

I'd made more tests, in the remote site I've got a C&S vizufon 5300 (a videophone) and I can call to a local extension and I have video but not audio (the problem with the video was I was using too much bandwidth). but if I try to call from the local videophone to the remote one I still have the error message: "temporarily unavariable" I've noticed that it happens everytime I try to call to something outside the site. Any idea?

Remember, I'm just as much a newbie as you are, but if you can make phones ring each other (either a phone on the Axon side to the remote side, or the remote side to the Axon side), it means that you have SIP working ok. However, if you get no sound once the calls are established, I would first guess it's an RTP issue. You have to check which ports each SIP device is supposed to use (ideally, you should force which ports it uses: For instance, the GrandStream IP phone uses 5004 for RTP - and probably 5005 for RTCP), so that you can map just that port on the NAT firewall. Do this on both sides, obviously.

If you still get no sound, maybe it's a codec issue. Make sure they both use G711u or G711a by default, as this is the most standard codec.

If you still can't get them working, go ask in VoIP-related forums like those:

news://comp.dcom.voice-over-ip

http://voxilla.com/PNphpBB2.html

HTH

Fred.

Did you also solve the problem that in the log-window of AXON every 11 minutes an error line is coming by to tell you that the SPA can't register? If not, how did you solve this.

Nope. I guess the Linksys doesn't allow SIP devices to register with it?

Which version are you using?

Upgraded from 3.2.6 to 3.2.10.

At this point, I'd like to get rid of the Axon (:-))... and have the Linksys call the IP phone directly through the Net. It works when they're both on the same LAN, but it doesn't work over the Net. I'm looking into this, but if someone has already done it...

Among Call Express users. Having two active calls with two different express calls when I generate a Transfer, to connects User 1 with User 2 the call between then gets disconnected.

Does Axon even support this feature, namely getting two calls through call waiting, and then connect the two and get out of the loop?

I've got a problem. I've Axon in a location with fixed IP and other location with Express talk, Xlite and a videophone.

Off the top of my head:

1. If the remote clients are using dynamic IP's, maybe their IP changes between the time they register with Axon and the time a call comes in for them, making the hosts unreachable?

2. Correct me if I'm wrong, but SIP uses UDP 5060 to handle call progress (calling out, etc.), and RTP for actual voice data. If a call can't even be established, make sure UDP 5060 is routed through the firewall to the right device inside. If you have more than one SIP device in one location, make sure they are configured to each listen on a different port (eg. device #1 -> UDP 5060, device #2 -> UDP 5061, etc.) and that the firewall maps ports accordingly. Until this works, you won't get anywhere

3. RTP is similar: Each device and the firewall must be configured to allow one UDP port and the next one (eg. 5004 and 5005 on GrandStream IP phone; one port for RTP, the next for RTCP). But if you can't even get phones to ring, don't worry about RTP yet.