Hello everybody! First of all I would like to thank you all for the already given support...
I've installed the O2 rom today on a M2000 and everything's fine except the only thing which was motivating me to install it, BBconnect!
I've changed the gprs settings to fit to my provider (swisscom, switzerland), subscribed to the bb service, when connecting getting the message "successfuly registered", Getting the PIN and an IP address, asked the Administrator to set up my account on our bb server but nothing's happening... no email received!
I must mention that I don't get any Service in the Services Lists, No IT policy, in the Option tab the connection is set to "BlackBerry" (and unable to change it, details: APN blackberry.net and the IP adresses plus ports).
The version of BBconnect is 2.0.1.39)
Can someone help me on this? That would be great! As I'm a newbie with BB a help from you would be much appreciated...
Hi guys.
I have Intsalled WM6 on my universal do you know how to user Voip Functions?
I wonder about that, too. I have read into the matter a bit and it seems that the test ROM might still be missing an actual sip core? As of now the today plugin just looks like a mask with nothing behind it (unless I am missing some hidden options). It would be great if I could ditch the 3rd party voip solutions as they are totally uncomfortable to use when compared to a native integration.
Me too! I'm dead keen to hook up my Universal with my asterisk PBX. Anyone know any more about the client? It appears to either be absent or just non-configurable atm?
I think it is simply a VoIP through MSN messenger? Hardly a SIP client I know...
If you have MSN Live set up the today plugin allows you to make calls through MSN Live if it is available.
So its still SJPhone for me to hook up to our asterisk server... (works with Midgets ROM BTW)
All the Reviews said that has sip function.
The MSN option would at least be something, but I also think that what was meant is real SIP support as searching through the registry reveals at least a few corresponding entries (sip port for example). My guess is that as the ROM was not finished when it got leaked that the VoIP functions were still not implemented... the reviews didn't show detailed configuration pages for this either if I recall it correctly.
if I remember correctly the msn voip client is a SIP client anyway? it just points at a microsoft sip proxy and only handles uris of other msn voip users (ie, the sip proxy doesn't route outside the msn voip client uri namespace?). I have to admit I have not used it myself tho. I may sign up for msn voip to see if the windows mobile client starts automagically working (its only a tenner or something)
Would be great if someone could test it out to confirm which version is actually true. If the MSN phone function is SIP indeed there should be ways to hack it and use our own services; but that's going into speculations a bit too much for now. I hope to hear your report
MSN SIP for VOIP
I can confirm that Microsoft MSN Messenger uses standard SIP for voice. It's one thing the M$ got right, deciding to use an open, properly implemented standard protocol.
Hi ... I have a problem with the configuration of my email account.
I need to change the default SMTP port 25 with another because my ISP use another ... How can I change It?
The version of my ROM is 3.30.
THNX.
ekkelon said:
Hi ... I have a problem with the configuration of my email account.
I need to change the default SMTP port 25 with another because my ISP use another ... How can I change It?
The version of my ROM is 3.30.
THNX.
Click to expand...
Click to collapse
Apologies if I am wrong on this one but don't you just add it to the end of the mail server address in options?
ie. if mail server is : mail.server.com and you want to use port 2525, then you put in:
mail.server.com:2525
I cant get this to work - anyone know if it would be possible to create a cooked rom with a given port hardcoded into the pocket outlook app?
Dave.
port 2525 sounds like a socketmail email service
nah, 2525 is a common mail services alternate port to get around the ISP's blocking of port 25.
My mail service will also use port 587.
Anyway adding :2525 after the mail.server.com works fine here.
serverort does work
I am using servernameortnum currently in WM6 on my hermes, so it does work.
I am using port 465, the secure SMTP port. that, or 587 (the mail submission agent port) is likely unblocked by ISPs, but you would have to make sure that the server is listening there, and you will almost certainly need to authenticate to the smtp server. Many servers will only allow authentication over a secure link, so you might need SSL enabled too.
I had trouble however with SSL and SMTP AUTH in pocket outlook. If anyone else does, this might be useful: http://lists.exim.org/lurker/message/20040609.135310.ba09a6e4.html
Based on that link I ended up needing to make sure my mail server started conversations on port 465 in SSL/TLS mode (rather than waiting for a STARTTLS). For anyone using Exim, that means setting "tls_on_connect_ports = 465" in your config. For people using a commercial mail server, you might just have to try lots of ports and setting permutations.
So the short version is that serverort does work, but there are other concerns as well.
note that is "colon" "p". or a dumb looking little face.
I have TYTN with Schaps WM6 ver 3.57a
Has anyone know how to configure the VOIP to support Gizmo project?
i second that
When I had Schapps I never set up VOIP, but with Black Dymond way back when I downloaded and installed the SipConfigTool and set username at my Gizmo number 1747XXXXXXX, password as your own password, and the gateways as proxy01.sipphone.com (same for sip server, domain and realm if asked). If you use another tool and can set a stun server, set it to stun01.sipphone.com. Schapps probably has his own voip settings tool, but if you run a search on the SipConfigTool I'm sure it will still do the trick.
I was never very happy with the WM6 voip on any provider, and have since relied on fring, which I felt worked better especially for receiving calls reliably. See www.fring.com for more info. I've got multiple call in numbers in several countries through both Gizmo and Skype all of which I route through fring which I use with a 3G connection. Depending on the connection there's some mild latency, but I never miss a call and the Gizmo connections are far superior to anything I was able to get with just the WM6 client or Skype.
Quist you right also with Schap's Gizmoproject SIP account is working well for me with Shap's 3.54c
Shap's have the built-in SIP confiuration tools under "system tools"
please note that you can replace your number with your account name.
Was glad to find this thread, but a bit disappointed on how old it is and without resolution. I have an 8525, with Schap's on it. I would like to set up the Internet Calling built in to this ROM with my Gizmo Project information as if it were an ATA.
I also have an ATA at home, a Innomedia left over from Sunrocket days, configured and working with Gizmo.
I am particularly interested in using this with WiFi, and not over 3G.
Using the information on GP website:
http://support.gizmoproject.com/index.php?_a=knowledgebase&_j=questiondetails&_i=88
I have gone into Schap's Settings/Connections/VoIP to get the config screen.
Provider name is irrelevant, and I put in "gizmo"
for user account, I put in my area 747 number, both with a leading 1 and without. I have put in my correct PW.
For registrar address, I was unsure what to put in, and I ended up filling in the STUN information from the GP website: stun01.sipphone.com port 3478 / UDP.
Proxy is set up as proxy01.sipphone.com:5060 / UDP. <- sure that one is right.
The URI automatically fills in with the information from above.
However, I still can't make things work. I'm not sure if it is the registrar/STUN stuff that is messing me up, or if my userID should be some other variation?
Has anyone made this work, and if so, how? I have a feeling it is some silly setting I've just messed up here.
Thanks.
I'm pretty sure your username needs to include the domain. These settings are for a different phone, but should apply the same.
Profile Name: sipphone
Service Profile: IETF
Default Access point: [your WLAN AP name]
Public user name: sip:[email protected]
Use Compression: No
Registration: "Always on"
Use Security: No
--
Proxy Server: siproxy01.sipphone.com
Realm: proxy01.sipphone.com
User Name: 1747xxxxxxx
Password: [You SIPPhone password]
Allow loose routing: Yes
Transport Type: Auto
Port: 5060
--
Registrar Server: siproxy01.sipphone.com
Realm: proxy01.sipphone.com
User Name: 1747xxxxxxx
Password: [Your SIPPhone password]
Transport Type: Auto
Port: 5060
My q
vdub144 said:
Was glad to find this thread, but a bit disappointed on how old it is and without resolution. I have an 8525, with Schap's on it. I would like to set up the Internet Calling built in to this ROM with my Gizmo Project information as if it were an ATA.
I also have an ATA at home, a Innomedia left over from Sunrocket days, configured and working with Gizmo.
I am particularly interested in using this with WiFi, and not over 3G.
Using the information on GP website:
http://support.gizmoproject.com/index.php?_a=knowledgebase&_j=questiondetails&_i=88
I have gone into Schap's Settings/Connections/VoIP to get the config screen.
Provider name is irrelevant, and I put in "gizmo"
for user account, I put in my area 747 number, both with a leading 1 and without. I have put in my correct PW.
For registrar address, I was unsure what to put in, and I ended up filling in the STUN information from the GP website: stun01.sipphone.com port 3478 / UDP.
Proxy is set up as proxy01.sipphone.com:5060 / UDP. <- sure that one is right.
The URI automatically fills in with the information from above.
However, I still can't make things work. I'm not sure if it is the registrar/STUN stuff that is messing me up, or if my userID should be some other variation?
Has anyone made this work, and if so, how? I have a feeling it is some silly setting I've just messed up here.
Thanks.
Click to expand...
Click to collapse
put the registrar and the proxy the value - proxy01.sipphone.com:5060 / UDP
edward
edward: Have you actually gotten this working? Can you show us exactly what settings you are using?
In my setup VoIP program I have these options, and this is how they are set:
user account: [gizmo username]
password: [passwod]
registrar address: proxy01.sipphone.com
Port: 5060 UDP
proxy address: proxy01.sipphone.com
Port 5060 UDP
URI address: 1747[my sip number]@proxy01.sipphone.com
I am getting "no service" on the home screen. What have I missed?
gthing said:
edward: Have you actually gotten this working? Can you show us exactly what settings you are using?
In my setup VoIP program I have these options, and this is how they are set:
user account: [gizmo username]
password: [passwod]
registrar address: proxy01.sipphone.com
Port: 5060 UDP
proxy address: proxy01.sipphone.com
Port 5060 UDP
URI address: 1747[my sip number]@proxy01.sipphone.com
I am getting "no service" on the home screen. What have I missed?
Click to expand...
Click to collapse
user account: 1747[my sip number]
password: [passwod]
registrar address: proxy01.sipphone.com
Port: 5060 UDP
proxy address: proxy01.sipphone.com
Port 5060 UDP
URI address: 1747[my sip number]@proxy01.sipphone.com
also enable the option in Schaps VoIP Settings
in options Enable VoIP over 3G
or if you use only wifi connect first to the network
i am using schaps 4 RC with this configuration and it work ok
edward
Nice, I reset and tried again - seems to work now.
A few problems though:
1. The settings seem to be reset when you reset your phone. How do you solve this?
2. After completing two test calls successfully, I let my phone "go to sleep" and when I turned it back on I had lost connectivity for VoIP (no service). Do I have to keep it awake to be able to receive calls using this method?
gthing said:
Nice, I reset and tried again - seems to work now.
A few problems though:
1. The settings seem to be reset when you reset your phone. How do you solve this?
2. After completing two test calls successfully, I let my phone "go to sleep" and when I turned it back on I had lost connectivity for VoIP (no service). Do I have to keep it awake to be able to receive calls using this method?
Click to expand...
Click to collapse
1.The settings are not reseting when i reset my phone
2.dont have this problem
Gizmo5
Gizmo just launched a midlet for it's service.... I have it but haven't check to see if it works yet... this could be a good solution for some.
http://www.gizmo5.com
I think it what it does is call your phone first and when you pick up it calls the other phone... not 100% on that though.
Mark
Where do I put stun settings?
okay got it to work, however
, I need to use g.729 codec, is there anyway to do so?
how to open .jad
i got Schap's 4.0 there is no midlet service. so could i install the gizmo execution program such as .jad. by the way, anyone can tell me how to open the .jad on tytn.
thank you very much.
I looked and saw there were a number of posts on the subject of sending POP3/IMAP email via Cingular/ATT and no actual working answers. When trying to send email via the carrier's wireless connection, not WiFi or Activesync, the status line would say "looking for changes" for a long time and then an error message reported that it could not connect to a server with the current settings or something to that effect. I too had this problem and found the solution on their support boards. The outgoing server for mail must be set to "cwmx.com" in order for POP3/IMAP mail to be sent correctly. The incoming server is your normal POP3 server like you'd set it up on your desktop for Outlook etc., such as "mail.bellsouth.net" or whatever you're using. Only the outgoing server is changed. I found no need for SSL or outgoing server authentication. This solution should work on WM5 and 6, although the setup screens are different. Make sure to set this up accordingly on all your POP3/IMAP accounts.
Hope this helps and sorry if it's a repeat.
Great tip, thanks
Excellent tip. Probably belongs in the Hermes WIKI. Shortly after I got my 8525, a friendly Cingular tech support person told me about this. I use it for my personal and office e-mail accounts, and it works great.
Setup is much easier on WM6. In WM5, I had to go into the advanced settings for the outgoing server (cwmx.com) and give it the name of the real server, as well as the user name and password. You don't have to jump through that particular hoop with WM6.