May I know the Buildin VoIP of WM 6.0 support the service type of Nortel VoIP gateway? I using HKBN now, but it only show Searching...
VoIP is still very much developer stuff. We can't give you the "works with VoIP" list because we are still sorting why different users are experiencing different results with the same VoIP provider. Your best bet is to read and understand the "{Solved} How to configure VoIP/SIP client in WM6" thread. From that point, you will need to find the exact settings for your Nortel switch. Its service must be SIP based and have DNS referenced Public IP addressing.
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Now that there is a solution for VoIP available I can't help but get greedy So far I used a softphone with multi provider support for this task, but now I wonder: Is it possible to use two different providers at the same time? Right now I am helping myself by switching by hotkey assigned cab files (which works pretty well btw), but what I am really looking for is using one provider for inbound calls and the others for outbound telephony. There are of course providers which can act as a proxy for others, but that solution never really worked out for me as many voip vendors seem to block these attempts (especially pointing a finger at SipGate here ). So, is there any solution for handling multiple providers?
Theoreticall, they are "Backup SIP Settings" you can provision in Windows Mobile 6. Just look at http://msdn2.microsoft.com/en-us/library/aa926605.aspx.
Having only one working VOIP provider, I haven't tried this yet.
However, the format of the XML provisioning is very similar to the one I posed in http://cleanimport.xda/index.php?posts/299950/.
You just have to replace parm name="SIPSettings" by parm name="BackupSIPSettings".
If you try it out, please report here.
Thanks.
--eluth.
The CAB installs fine, but there is no difference between installing a new provider with just "SIPSettings" and "BackupSIPSettings" I tried this with Arcor and SIPGate to test ut whether both incoming numbers would be available at the same time.
Hello everyone!
Situation: I have several VoIP provider (sipgate, 1und1...) My problem is I can only setup a single provider in my phone with built-in WM6 VoIP client.
Solution attempt: I set up an account with pbxes.org. There I added my SIP providers and an extension with username and password. Now I tried to set this up using the Sip Config Tool V2.0.1. Here are the settings:
Sip Server: pbxes.org Port: 5060
(tried others like sip.pbxes.org(com) and the IP)
username: username-extension_number
password: password
That's it. It's not working. The Today Internet-Calling-plugin tells me "searching....", but never to "selected"
I also tried using x-lite on my desktop pc. With the same settings I successfully registered and I was able to use pbxes just as it is supposed to be.
When using sipgate directly with WM6 it works, but not with pbxes.
I am kind of running out of ideas. So any help is really appreciated.
Thanks in advance,
Stefan
Tried several times as well on my S620.
Now, two days ago got a S710 and was curious to test VOIP on it.
Very first time I tried I was thinking everything was fine, at least nearly.
The Today plugin showed "Searching" (like with providers it works with)
The icon for Internet Phone showed up.
Unfortunately it "semi worked" only that time, any other tests I tried the Today Plugin didn't even get into "Searching".
The bad thing is that it did work once, like meaning it could work. Now PBXES status shows as Registered but it isn't actually. The other bad thing is that I tried to post in their forum for help but didn't receive much info back.
Bad luck, but I'll try and try again. Maybe newer files one of these days and our probs would be solved.
Ka.
Subscribe message
I ever check the message from sip server, I found that windows mobile voip client not only send register message, but also try to send subscribe message, I guess that it is reason that some sip server only may work as registra, and not handle correctly subscribe. it seems the reason the client always display "searching", I will try to modify sip server to support subscribe message in the future.
try fring
try using fring...
it supports multiple sip accounts.. and actually register u not just subscribe you.
You are right, Fring or any other 3rd party app connecting to a SIP server can work maybe better than the "builtin" SIP functionality but personally would prefer being able to use a solution where I dial like usual and calls get automatically router to VOIP, not having to switch to a separate app.
This is why I'd rather like the builtin sip stack to work on its own with any SIP provider. I tested it with PBXES but doesn't work. I'm currently renting a VPS with Asterisk (Trixbox) installed and it does connect. Only prob is that even if it manages to register, seems MS VOIP implementation is still not really compatible or suitable for the use I wish to have.
Ka.
This is an old thread but this is the same problem I have on my HTC HD2 (Leo).
Installed sip drivers and voip config cabs and configured it with my account through pbxes and it is not working.
Can someone help?
Android (Moto droid, HTC Hero) I had no problem with pbxes and sipdroid...
Hi guys
I know many people have had one way voice issues using the various cabs for enabling the WM6 VoiP client but I don't think this issue is related to that.
I'm connected to a SIP PBX over an OpenVPN connection and everything connects but I the other party cannot hear me. I've done a packet capture and although the SIP INVITE is coming from the correct source address if you drill down into the SIP packet the owner creator etc is the original IP of the device (not the VPN one).
As a result the RTP stream is being directed towards the wrong IP and I'm getting one way voice. This issue doesn't happen with 3rd party SIP clients but I haven't found a good one yet (fring doesn't work on our PBX).
Has anyone came across this issue before?
After further investigation the WM6 client works perfectly when used over a wireless access point. The problem only occurs when using OpenVPN as the original IP address of the wireless adaptor is sent in the invite packet (rather than the VPN IP address), this causes the audio to be streamed to the wrong IP and as such one way voice is experienced. If I find a workaround I'll post it up.
shippyt said:
Hi guys
I know many people have had one way voice issues using the various cabs for enabling the WM6 VoiP client but I don't think this issue is related to that.
I'm connected to a SIP PBX over an OpenVPN connection and everything connects but I the other party cannot hear me. I've done a packet capture and although the SIP INVITE is coming from the correct source address if you drill down into the SIP packet the owner creator etc is the original IP of the device (not the VPN one).
As a result the RTP stream is being directed towards the wrong IP and I'm getting one way voice. This issue doesn't happen with 3rd party SIP clients but I haven't found a good one yet (fring doesn't work on our PBX).
Has anyone came across this issue before?
Click to expand...
Click to collapse
This is a known SIP issue. As SIP had been designed to work over end-to-end connections. As SIP is a self routing protocol, the SIP server and user agents use the source IP stated in the SIP header instead of the source IP stated in the IP header for their routing
Most 3rd party SIP clients use a technique called STUN to discover their global IP (behind the NAT, or as in your case the VPN local IP) and they put that IP in the source IP in SIP header.
Tokko said:
This is a known SIP issue. As SIP had been designed to work over end-to-end connections. As SIP is a self routing protocol, the SIP server and user agents use the source IP stated in the SIP header instead of the source IP stated in the IP header for their routing
Most 3rd party SIP clients use a technique called STUN to discover their global IP (behind the NAT, or as in your case the VPN local IP) and they put that IP in the source IP in SIP header.
Click to expand...
Click to collapse
is there any workaround? I do not want to install STUN server on a local pc. I connect to my router openvpn (dd-wrt firmware). mega version has also a milkfish sip server. I have been thinking maybe I could use it as proxy to resolve this one-way voice problem?
Hey, I had the same issue recently and I have found a solution: do not enable VoIP calls over 3G/GSM when using SIP Config Tool. I was actually able to make and receive VoIP calls over an OpenVPN connection - the sound was a little choppy at the callee's side, but in general it was quite good. The only problem I have so far is that the phone unregisters itself from server when going to standby mode and registers back when turning on, so I can receive calls only in the latter case.
shurik_1 said:
is there any workaround? I do not want to install STUN server on a local pc. I connect to my router openvpn (dd-wrt firmware). mega version has also a milkfish sip server. I have been thinking maybe I could use it as proxy to resolve this one-way voice problem?
Click to expand...
Click to collapse
Not to mention native SIP client you are using probably doesn't support STUN anyway.
Proxy idea sounds doable, but then WM device is still going to use the ip of its wireless interface in the INVITE message. Unless this is fixed by the router, milkfish will still be unable to encapsulate incoming RTP traffic into the VPN.
Perhaps you can mangle SIP INVITEs with sip_conntrack available at http://www.iptel.org/sipalg/?
I'd like to save everyone the trouble of figuring out how to get this done by sharing my setup with you. The VOIP itself works great over 3G/4G and Wifi, but to use the VPN you'll need to be on Wifi, 3g VPN doesn't seem to work.
Requirements:
SipDroid app in the market
A free account at pbxes.org
A paid account at callcentric.com (~$.019 a minute to call anywhere in the US)
A paid account at SuperVPN.net ($4 /mo if you pay for the whole year)
First set up a pbxes.org account, and connect to it with the SipDroid app, I recommend using this guide to walk you through the process.
http://guardianproject.info/2010/05...e-mobile-phone-system-for-android-and-beyond/
Once you have that working there is one crucial adjustment to be made within SipDroid. For some reason it comes default with all sorts of audio codecs, but only ONE of them seemed to work on the EVO, the Speex codec. So go into audio codecs and switch everything but speex to "never".
After that you should have a working VOIP system but you'll still need some kind of trunk if you want to make outgoing calls to land lines or cell phones. There are many solutions for this but I recommend callcentric.com, they seem to be the most recommended for this type of setup, and they worked great for me. You can pay $20 a month for unlimited US calling, this means you can be anywhere in the world and call the US for just $20 a month. Or you can prepay (this is what I did), then you pay a flat rate of about $.019 a minute to call the US from anywhere, and if you reach you pre paid limit, it just cuts off until you recharge it.
Once you have your callcentric account purchased, just go into your pbxes.org admin area and under trunks add one for call centric, use your callcentric # as the username, and callcentric.com as the sip server. Then go under Outbound routing, add a new one, name it whatever and choose your callcentric trunk from the pulldown menu, submit the changes and you're done.
Now you should be able to successfully make outbound calls to anywhere using SipDroid.
Lastly, this was the most challenging for me, the VPN. Apparently android, including 2.2, has some major issues with maintaining vpn connections, especially when you try and use them for VOIP. There is a huge issue queue in the android google groups forum where the problem is openly discussed without a real solution. BUT, while it appears the majority of VPN connections will fail, they CAN work if you get it set up just right. Setting up VPN's, specifically VPN's tailored for mobile devices, is not something I know how to do. In the android group thread someone mentioned SuperVPN.net as a working solution, I checked it out and sure enough they work great, I had zero problems with them the whole time I was out of the country.
So create a supervpn.net account, and then on your phone go into Menu -> Wireless & Networks -> VPN -> Add VPN -> Add PPTP VPN, create the connection and you are good to go.
*I didn't set up an inbound call # with callcentric as I didn't need one, I assume after you upgrade your callcentric account, adding the inbound trunk is similar to the outbound. Be sure and look into getting a free inbound number from sipgate.com before you go and pay for one, you'll be locked to a California area code, but free is free.
An alternative I use is having an Asterisk server at home and use IAXAgent from the market. IAX does not have the problems that SIP does when going over NAT. I can make calls over 3G or wifi. A lot of SIP providers also provide IAX accounts. IAX is just a better way to go for making calls over the Internet. SIP is excellent for the LAN.
ChrisDos said:
An alternative I use is having an Asterisk server at home and use IAXAgent from the market. IAX does not have the problems that SIP does when going over NAT. I can make calls over 3G or wifi. A lot of SIP providers also provide IAX accounts. IAX is just a better way to go for making calls over the Internet. SIP is excellent for the LAN.
Click to expand...
Click to collapse
Oops.. the method I posted actually works on 3g and 4g also, it's only the VPN that requires Wifi, I had worded it incorrectly, now it's fixed. (thank you)
I looked into setting up an asterisk server, but I didn't want to have to depend on my own server or home connection being available whenever I needed it, especially when I was traveling for more than a week.
Is IAX the same as a trunk, does it cost anything to connect to land lines or cellphones?
True, you method does make SIP work because you are using a VPN. IAX is an alternative to SIP. It is NAT friendly, and as long as the port is not blocked, it just works. Though, there are a fewer choices for clients compared to SIP. IAX was created by the Asterisk team. I do not know of any VOIP systems that support IAX, bug that does not mean they dont exist. I am a heavy Astersk guy, so IAX was my cup of tea.
What advantages does this have over google voice?
I'm curious cause i'll be going to england soon and would be nice to make calls over wifi.
ShoxV said:
What advantages does this have over google voice?
I'm curious cause i'll be going to england soon and would be nice to make calls over wifi.
Click to expand...
Click to collapse
None, in fact, it his disadvantages (See below). Also, most businesses, schools, etc. will block just about every VPN method. OpenVPN is the most flexible one I have found, which might be able to sneak around by using alternate sub-1000 ports (which most places won't block, since they require root access on whatever server they're running from).
OP: Might wanna take a look at this...
http://www.mywot.com/en/scorecard/supervpn.net
drmacinyasha said:
None, in fact, it his disadvantages (See below). Also, most businesses, schools, etc. will block just about every VPN method. OpenVPN is the most flexible one I have found, which might be able to sneak around by using alternate sub-1000 ports (which most places won't block, since they require root access on whatever server they're running from).
OP: Might wanna take a look at this...
http://www.mywot.com/en/scorecard/supervpn.net
Click to expand...
Click to collapse
Not sure what you're talking about, it saved me hundreds of dollars in roaming minutes while I was in Belize. Some places do block vpns I'm sure but I never had any issues, but you don't need the vpn itself unless you're in a country that blocks voip altogether, at which point occasional vpn is greater than no vpn.
Also supervpn was the only method I found that actually works on android, I think the risks referenced in that link you posted really only apply to desktop vpn use, not phones. Openvpn is great for somewhat advanced users and if you have a computer you can depend on as a server while you're out of the country for days or weeks, this guide isn't meant for someone capable of managing that.
As for Google voice, it just initiates an inbound call to your actual cell number, which does zero good when you're trying to avoid roaming. Now the new gmail implementation of voice shows promise as an actual voip solution, but currently that version is desktop only from what I can tell.
Hi,
From Internate search I found this forum which is having very useful research and Discussion. I have one question and need solution and reply setp by step for the following issue:-
That I am trying to send voip call by using IP 192.168.0.20 using Port:6800 as SIP port to the Gateway from VPS (softswitch). At the destination we are using a Router (Dlink) for connectivity of internate and Gateway.
When I configure GW with IP 192.168.0.20 SIP port 6800 with username and password, and tried to connect Gateway as SIP client from Softswitch, the messages comes "The GW/Registrar Client does not register as SIP client". Thats means switch is not reaching to Gateway.
As I understand from some study that SIP-based communications cannot reach LAN users behind firewalls and NATs automatically because firewalls are designed to prevent inbound unknown communications.
I would much appreciate if any colleague from this forum may guide me step by step and let me know how this scenario can works.
Thanks
Sameel
I've Googled it and there are two ranges of ports you need open for it to work. Usually though ICE/Stun can get past that. Have you tried that?