I flashed my HD with L26_THDV3 WWE, I inputed SIP server information inside Sip Conf Tool, and added HD inside CallManager as a Basic Sip Device, but HD still show no service? May I know How can I use it with Cisco CallManager?
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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...
Hello world.
Since I can't hear the other party while placing VoIP calls through the integrated SIP client of WM6 via FON hotspots (probably due to the double NAT-routing on most hotspots), I'd like to know if there's any possibility to implement STUN support.
regards,
Inquisitor
Hi there, I think you should post your question in this blog:
http://blogs.msdn.com/cenet/Default.aspx?p=3
I'm bypassing the issue by using OpenVPN on my S710 which connects to my home gateway.
STUN Support in 6.1?
I have been searching around quite a bit.
Is there any support for a STUN server with the VOIP "Internet Calling" Feature on some of the newer ROMs. I'm on a Kaiser and it seems to register. I just can't seem to make a call to an asterisk system.
I believe this is because I am behind a NAT, so I think STUN is required for it to work properly.
doing same but no voice
jockyw2001 said:
Hi there, I think you should post your question in this blog:
http://blogs.msdn.com/cenet/Default.aspx?p=3
I'm bypassing the issue by using OpenVPN on my S710 which connects to my home gateway.
Click to expand...
Click to collapse
i m also having the same issue of no voice, could you please guide on how you accomplish this, my current setup is like this,
i have my server configure as openvpn server that is behind a router, ssh, ovpn ports are forwarded to server, ovpn client is xp and which is behind another router, xlite as softphone and working perfectly, ovpn installed on pocket, vpn tunnled is connect can connect through to server using putty, pocket pc is showing signal means registered with asterisk server (openvpn server is same) can dial also and it shows line connected as well, but no dial tone and no voice.
linux server with asterisk and openvpn
client htc universal (jasjar)
i hope i've given a clear picture of my setup, thanx in advance for any help...
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/?
What application do you people use here for making true voip calls (not skype)?
On my old XDA 2S I used to have something that was nicely intergrated in the system with the phone interface, and that I used with my Asterisk setup.
Unfortunately I can not remember the name of the application.
How about the Voipbuster Mobile???? http://www.voipbuster.com/en/mobile.html
Nop.. I want to use it with my own server.
The Voipbuster client will only work with VB.
I have developed a simple client sip application on andorid by using SipDemo example which is provided by Google.
My application(everything such as Authentication, Streaming, ...) works perfect over my WiFi network, but when I switch it in 3G network, just authentication of SIP session works fine and the audio streaming does not work!!??
By the way I have a desktop client SIP application(written in C#) that works perfect with my android client app over 3G network, it means that streaming does not work when two client is android in 3G network, but when one of clients changes to my desktop SIP client application(written by C#) streaming works perfectly.
Does any body run SIP stack which provided in android 2.3 in 3G network? in the following links it's been told that Sip Stack does not work in 3G network, if so why I can use SIP in 3G network when one client is SipDemo and another is c# client in windows desktop application?
http://stackoverflow.com/questions/5139718/android2-3-sip-implementation
http://stackoverflow.com/questions/...included-in-android-2-3-does-not-work-over-3g
No body knows
Please someone gimme a hint (((
Why no one helps?
As I know, Google disable the option for the native SIP to work over 3G, in some deviced, the all SIP was disable.
Check that: http://www.inspiredgeek.com/2011/07...laxy-s2-sii-for-making-calls-on-wi-fi-and-3g/
Maybe it will solve your problem also.
alto said:
As I know, Google disable the option for the native SIP to work over 3G, in some deviced, the all SIP was disable.
Check that: http://www.inspiredgeek.com/2011/07...laxy-s2-sii-for-making-calls-on-wi-fi-and-3g/
Maybe it will solve your problem also.
Click to expand...
Click to collapse
Thanks for your reply but as I have mentioned SIP auth works fine over 3G and laso I can trace stream packets which is sent by client to another client, so it means it is not related to the "config_sip_wifi_only"