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  1. #81  
    Hi All,

    I just installed linphone, and I have successfully connected to my ubity sip account. I was able to place an outgoing call to a conference system and could hear just fine (through the back speaker only). I'll test more over the next few days.

    I attempted to join a conference room, but dialing numbers while on did not seem to do anything. Will that be supported at some point, or is there another way of dialing a conference room?
  2. #82  
    Just made my first SIP call with LinPhone on my Pre 2. Had to set a hard IP address for the SIP server in stead of the name (sipNL.net). Windows mobile (5?) had the same problem. No issue if you know that.

    Since the call is routed through the back speaker, the receiver experienced a substantial echo, which seems logical and could be minimized by playing with the volume. Also I cannot get a bluetooth headset working, even though the BT headset connected icon is in top status bar.

    As reported by others,after terminating LinPhone, and restarting it, the App hangs on 'Waiting for the service to come up ...' (Pre 2, webOS 2.0.1). Once I could circumvent that by going in and out of airplane mode. Also restarting Luna seems to help sometimes.

    As reported by others, incoming call succeeded if the Pre/screen is not in sleep mode. If it is, there is no ringing but the call is there when pushing power and can be accepted. I have no response on quality yet.

    Seems like going to be a great addition to webOS. Certainly one I am waiting for. Until now I have a dedicated HP 514 GSM for making SIP over WiFi calls abroad. Thanks for the opportunity to help testing. Will try more calls and also calls through voipbuster.
    Last edited by Dick99999; 02/21/2011 at 02:10 AM. Reason: incoming eleborated and bluetooth headset
  3. #83  
    I'm running 0.1.6 on my Sprint FrankenPrePlus (formerly Verizon).

    I'm getting the REG_PENDING error when using both Gizmo and SipGate.

    What other info do you need?
  4. #84  
    You could inspect var/log/messages. My experience is that this type of error is caused by
    - invalid user/pw
    - firewall ports blocked
    - hard IP address needed for SIP provider setup in sip client (Pre)
    Most times I use the IP packet dump capability of my router (Frtizbox), which includes wireless packages. Very useful for these problems.
  5. #85  
    My username/pw are correct.

    I have used both nominal and numerical addresses for SIP provider setup.

    Being at school I have no control over which ports are open and closed on the wireless signal. I'll have to take it home and try it.

    If it doesn't work then obviously it is something else. We'll see.
  6. #86  
    I hooked myself up on my Callcentric account with no problems to report so far! Great experience thus far considering this is alpha. Looks like my Pre will be useful overseas after all!

    BLiip
  7. Cache's Avatar
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    #87  
    Hi All,

    Testing 1.6

    I have a DID through Callwithus.net (great service BTW), and I am testing 1.6 on Pre- 1.4.5. Outgoing calls work great. Incoming calls, I get no notification and cannot answer incoming calls. Did I mention Outgoing works great?? ;-)

    I run eyebeam on Windows 7 laptop, works great. I can see through the web portal to all of my account services at callwithus that linphone is registering properly with the SIP server.

    What additional info or logs can I provide to help troubleshoot?

    Lastly, GREAT JOB getting this port going. I can't wait for full featured client with contact management, front speaker/proximity, call logging, incoming calls, etc. Thx Thibaud and all WebOS Internalz.
  8. bitflung's Avatar
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    #88  
    Quote Originally Posted by Thibaud View Post
    Sounds quality: most probably depends on the CODEC used (I let the Linphone core decide for now, although there are internal options I'll make available later in the app).

    Call (not) received: can you elaborate on your configuration and how you manage to initiate the call on the other side? Please PM if you want to keep your personal data ... private!
    i'm new to webos (only have the emulator to play with for now; my pre plus will arrive monday) and so i haven't had a chance to try linphone on webos quite yet, but:

    on pretty much every SIP device i've used, issues with receiving calls has stemmed form NAT traversal issues. there are a bunch of ways to get around those issues if that's what's going on, but a simple fail-safe way to test if that's it is to get rid of NAT. doing that on webos might be tricky, so someone correct me if i'm wrong, but i would think placing the pre's IP address in the router's DMZ would essentially mimic connecting the pre outside of NAT.

    if i'm right, just login to your router's admin page and look for DMZ settings. find that and you'll see that you can put an IP address as the DMZ host. now you need your pre's ip address (i'll let you figure out how to get it - having never held a pre i'm really the wrong person to aid in that department) and enter the ip address here.

    don't reboot anything. your pre will only have this ip address as long as the connection is active (next time you leave the wifi range and return you could get a different ip address). try receiving a call now. if it works, NAT traversal is almost certainly the primary issue.

    i'll have my pre on monday and plan to get involved with linphone immediately after that. if you have NAT traversal issues, i'll be able help then. until then, maybe someone else around here knows a thing or two about STUN or ICE (or you can google these terms along with linphone and your SIP provider).

    on a side note: anyone try using linphone on the pre with gizmo5? i joined them long before google bought them out, so that's the service i'll be testing on.
  9. pman_lt's Avatar
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    #89  
    After multiple tries on installing Linphone 1.0.6 (service seems to be not starting), and then installing 1.0.5 and 1.0.4, restarting phone and Luna only somehow it magicaly fixed itself, and Linphone service is now running fine..

    Was able to login to two services that Im currently running/have access to (succesfull registration on 3G/EDGE/Wi-Fi).

    Both inbound and outbound calls working, but not on all tries, if it did, it did only work on speaker, didn't try to call to phone when on screen off (Sleep).

    Great start, but it has some way to go

    Some things that would be GREAT:

    - multiple SIP account support (thats a must for me, since its a complete waste of time reentering the account info everytime when i wont to call from other service)
    - option to enable/disable registration to the server (to recive inbound calls)
    - dialing plan support (for sufixes and etc)

    Keep up the good work.
  10. #90  
    I have determined that my REG_PENDING problem was a NAT issue. However, now I have several other problems.

    Phone: Verizon Pre+ but with a Sprint comms board, webos 1.4.5

    SIP Provider: Gizmo5, SIPGate

    1) I am successfully receiving calls through both Gizmo5 and SIPGate, but although I have CALL_IN_CONNECTED, when I answer there is no sound at all.

    2) When I try to hang up (tapping the end call button), the button responds but Linphone never changes from CALL_IN_CONNECTED and the green button doesn't come back (both Gizmo and SIPGate).

    3) When I close and then restart Linphone after the above, it gets stuck at "Waiting for the service to come up..." I have to restart the phone to get it to REG_OK again. (both Gizmo5 and SIPGate).

    I have not tested outgoing calls yet.
  11. #91  
    I just installed 0.1.6 on my Pre- using my Asterisk 1.8.x server at home. It is working quite well but I have found a couple of issues. One, I believe is related to Linphone and the other is the Uberkernel.

    Normally, I run my Pre- at 800 MHz with the Uberkernel but I found lots of latency and lag as the duration of the call increased. I moved the speed down to 500 MHz and everything worked great.

    The Linphone problem, which I need to test more, is showing the device losing registration every once in a while. I had to go back into the preferences and then go back out for the device to re-register. I need to pay closer attention and write down exactly what I see. I will update with my findings.

    I have only been calling internal to the Asterisk box but by and large it is working great.
  12. #92  
    Sometimes LinPhone hangs while the status shows Call-out-Ringing. I can press end call once, the message stays on. And then no response anymore.
  13. ensign's Avatar
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    #93  
    Just installed linphone 0.1.6. I am able to register correctly to the draytel.org domain. But because i am behind a nat router a need to use nat.draytel.org:5065 as a proxy.

    When a type nat.draytel.org:5065 in the proxy field and enable proxy, i can't get past reg_pending. Disabling proxy allows a successful registration. Despite this i need to enable the proxy to get linphone to work correctly with my server.

    What is the correct syntax for entry of a server with a non-standard port? Does linphone currently support a proxy server?
  14. #94  
    Quote Originally Posted by bitflung View Post
    i'm new to webos (only have the emulator to play with for now; my pre plus will arrive monday) and so i haven't had a chance to try linphone on webos quite yet, but:

    on pretty much every SIP device i've used, issues with receiving calls has stemmed form NAT traversal issues. there are a bunch of ways to get around those issues if that's what's going on, but a simple fail-safe way to test if that's it is to get rid of NAT. doing that on webos might be tricky, so someone correct me if i'm wrong, but i would think placing the pre's IP address in the router's DMZ would essentially mimic connecting the pre outside of NAT.

    if i'm right, just login to your router's admin page and look for DMZ settings. find that and you'll see that you can put an IP address as the DMZ host. now you need your pre's ip address (i'll let you figure out how to get it - having never held a pre i'm really the wrong person to aid in that department) and enter the ip address here.

    don't reboot anything. your pre will only have this ip address as long as the connection is active (next time you leave the wifi range and return you could get a different ip address). try receiving a call now. if it works, NAT traversal is almost certainly the primary issue.

    i'll have my pre on monday and plan to get involved with linphone immediately after that. if you have NAT traversal issues, i'll be able help then. until then, maybe someone else around here knows a thing or two about STUN or ICE (or you can google these terms along with linphone and your SIP provider).

    on a side note: anyone try using linphone on the pre with gizmo5? i joined them long before google bought them out, so that's the service i'll be testing on.
    Yep, sounds like there are NAT traversal issues, or the like. A STUN server might be helpful there, need to give it a try. I'll be working on that too since a showstopper for many of you.

    BTW, Linphone is running on the emulator too (service startup issue on 0.1.6, though but fixed in later revisions). My configuration is Ubuntu 9.10 with headset + mic plugged to the soundcard. Your mileage may vary...
    Last edited by Thibaud; 03/02/2011 at 04:53 PM.
    Palm Pilot Pro -> M505 -> T|X -> (franken) Pre & Pixi -> Pre2 2.2.4 & TP
    Linphone port to webOS (1.4.x -> ...), lately working on NAVIT
  15. #95  
    Quote Originally Posted by Ricyteach View Post
    I have determined that my REG_PENDING problem was a NAT issue. However, now I have several other problems.

    Phone: Verizon Pre+ but with a Sprint comms board, webos 1.4.5

    SIP Provider: Gizmo5, SIPGate

    1) I am successfully receiving calls through both Gizmo5 and SIPGate, but although I have CALL_IN_CONNECTED, when I answer there is no sound at all.

    2) When I try to hang up (tapping the end call button), the button responds but Linphone never changes from CALL_IN_CONNECTED and the green button doesn't come back (both Gizmo and SIPGate).

    3) When I close and then restart Linphone after the above, it gets stuck at "Waiting for the service to come up..." I have to restart the phone to get it to REG_OK again. (both Gizmo5 and SIPGate).

    I have not tested outgoing calls yet.
    1. try to increase the volume when call is connected, sound is handled as a media sound (the note icon, not the bell) and might be too low. I need to improve that too...
    2. a click on the dial/disconnect button will change it to an empty button, then an event from the linphone core will make it change back to an action button (so you can do only one action at a time and not overshoot the app with actions). But there's obviously an issue that locks up the service (see below)
    3. I have a found a bug in the service that may explain your issue (asynchronous event within a synchronous request to the service not properly handled, sorry, my bad). This locks up the service and only a service restart (kill/killall or reboot as you do) can fix it for now.


    Please try an outgoing call too if you can (digit-based only for now, I'll add fully qualified URI asap so you can test SIP to SIP calls at no-charge).
    Palm Pilot Pro -> M505 -> T|X -> (franken) Pre & Pixi -> Pre2 2.2.4 & TP
    Linphone port to webOS (1.4.x -> ...), lately working on NAVIT
  16. #96  
    Quote Originally Posted by onelander View Post
    I just installed 0.1.6 on my Pre- using my Asterisk 1.8.x server at home. It is working quite well but I have found a couple of issues. One, I believe is related to Linphone and the other is the Uberkernel.

    Normally, I run my Pre- at 800 MHz with the Uberkernel but I found lots of latency and lag as the duration of the call increased. I moved the speed down to 500 MHz and everything worked great.

    The Linphone problem, which I need to test more, is showing the device losing registration every once in a while. I had to go back into the preferences and then go back out for the device to re-register. I need to pay closer attention and write down exactly what I see. I will update with my findings.

    I have only been calling internal to the Asterisk box but by and large it is working great.
    I'll focus first on non O/C kernels until stable if you don't mind, then we'll have a look into the O/C issue you report. Looks similar to another issue reported some time ago with O/C kernels, but I can't seem to remember what it was.

    Yes, please report the de-registration issue with more details (/var/log/messages if you know how to do it, or better use Lumberjack for that -- thank you oil, this simply is great stuff!).
    Palm Pilot Pro -> M505 -> T|X -> (franken) Pre & Pixi -> Pre2 2.2.4 & TP
    Linphone port to webOS (1.4.x -> ...), lately working on NAVIT
  17. #97  
    Quote Originally Posted by Dick99999 View Post
    Sometimes LinPhone hangs while the status shows Call-out-Ringing. I can press end call once, the message stays on. And then no response anymore.
    Yes, most probably an issue in the service. Working on it. Please try again with 0.1.7 when it is announced.
    Palm Pilot Pro -> M505 -> T|X -> (franken) Pre & Pixi -> Pre2 2.2.4 & TP
    Linphone port to webOS (1.4.x -> ...), lately working on NAVIT
  18. scoinva's Avatar
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    #98  
    This reminds me of a link I recently found about LinkSys ATA VOIP router:
    Configure your Linksys VoIP ATA the right way!

    May not apply but interesting.
  19. #99  
    Quote Originally Posted by ensign View Post
    Just installed linphone 0.1.6. I am able to register correctly to the draytel.org domain. But because i am behind a nat router a need to use nat.draytel.org:5065 as a proxy.

    When a type nat.draytel.org:5065 in the proxy field and enable proxy, i can't get past reg_pending. Disabling proxy allows a successful registration. Despite this i need to enable the proxy to get linphone to work correctly with my server.

    What is the correct syntax for entry of a server with a non-standard port? Does linphone currently support a proxy server?
    Are you sure you need to address the 5065 port? It does not seem that obvious looking at Draytel's FAQ (this item and the following one).

    Furthermore, I signed-up at Draytel.org and got an account so to perform a quick test. I only had to provide username, password and mention "draytel.org" as the domain (no explicit proxy) to get a successful call-in (with working audio) on my Pre from my Ubuntu machine. I'm behind a router from my DSL provider (external IP address, internal private network with DHCP).

    Please can you retry and report with information about your configuration and possibly excerpts from /var/log/messages (Lumberjack's your friend) if appropriate? PM if privacy needed.

    PS: BTW, you used the appropriate syntax to change the default 5060 port to 5065. I'll update the Linphone page accordingly.
    Last edited by Thibaud; 02/27/2011 at 05:12 PM.
    Palm Pilot Pro -> M505 -> T|X -> (franken) Pre & Pixi -> Pre2 2.2.4 & TP
    Linphone port to webOS (1.4.x -> ...), lately working on NAVIT
  20. #100  
    Quote Originally Posted by Thibaud View Post
    try to increase the volume when call is connected, sound is handled as a media sound (the note icon, not the bell) and might be too low. I need to improve that too...
    I'll try that tomorrow and report back but I'm pretty sure it was already all the way up. However I am running modeswitcher and it may have had it low for some reason.

    Quote Originally Posted by Thibaud View Post
    I have a found a bug in the service that may explain your issue (asynchronous event within a synchronous request to the service not properly handled, sorry, my bad). This locks up the service and only a service restart (kill/killall or reboot as you do) can fix it for now.
    Yup that sounds to me like it lines up with the problem I"m experiencing.

    Quote Originally Posted by Thibaud View Post
    [LIST=1]Please try an outgoing call too if you can (digit-based only for now, I'll add fully qualified URI asap so you can test SIP to SIP calls at no-charge).
    I don't have any purchased time on either service so I'll have to wait until the free SIP to SIP is available (I use Gizmo and SIPGate for free Google Voice service on my iPhone currently- I don't have any cell service- and so I never use outgoing calls).
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