For non listed question, do not hesitate to join and ask on the users group :
http://groups.google.com/group/csipsimple-users (csipsimple-users@googlegroups.com).
If you don't know what is SIP and what this program does : WhatIsSIP
To solve this problem, you must activate STUN :
Go on setting Settings > Network - Tick "Use Stun" and fill a stun server on the field bellow.
CSipSimple has a default a stun server if you activate STUN option ! (but as soon as you do not activate the STUN setting it is not used), if you want to use another STUN server than the default one, you can see http://www.voip-info.org/wiki/view/STUN Public Stun server section to know what server you could use freely.
You can also try to use ICE in addition to STUN if STUN alone doesn't solve the problem : Settings > Network - Tick "Use ICE"
In this case, try to disable Bluetooth option of your phone (even if nothing is paired).
If you have problem with no audio on one of the side you can start to eliminate some use case before reporting a problem.
If it's the remote that doesn't ear you, focus on the micro activity :
If it's the app that doesn't produce sound, focus on the speaker activity :
This is probably a configuration problem. You should ask your sip provider for settings they expect (particularly whether they expect to be the sip proxy or have a different sip proxy address).
If the call cuts automatically after 30seconds it's because the app never received the confirmation of the fact the call is established. In such situation sip protocol indicates that the call is not valid anymore.
Just try to make a call using the standard android phone dialer, or clicking a contact in the official contact application ;).... And you'll see that a contact list is not needed at all in the SIP application !!! There is also a quick text search dialer if you switch to text dialing mode, that will search on android contacts.
There is a powerful rewriting/filter/auto-answer tool in CSipSimple (see the UsingFilters wiki page). To access this tool, go on the account on which you want to apply things. Press menu button and choose "Filters". You are now in the tool. Using this tool you can :
Yes, but you have to configure CSipSimple according to the fact you want to be able to close it !
Let me explain : CSipSimple can run under several setting configuration :
When you set "Always available", close option make no sense : indeed, the application will automatically restart when network will change of state... and you'll say me "Hey there's a bug... the app constantly restart itself". By using this profile you tell the software that you want to receive your calls, and it will do it's best to be running when you can receive calls.
However, if you choose "Only for outgoing"... Close option will appear in the menu of CSipSimple dialer! Since here, it make sense to use this option, since the app will never restart itself.
To do so, transform your account into an Expert account (Edit account, press menu, "Choose wizard" and choose Expert). Edit again the account, there is now more settings to set. Try this :
Switch to ExpertSettingMode and in User interface settings activate "Use partial wake lock".
You SIP server probably don't support fully the RFC and the registration method used by CSipSimple by default is not suitable for your server. Fortunately, there is a way to configure your account to avoid that :
To do so, transform your account into an Expert account (Edit account, press menu, "Choose wizard" and choose Expert). Edit again the account, there is now more settings to set. Try this :
If your server actually doesn't support the latest RFC you may have to wait for about 15 min for the current registration to timeout.
In order to improve CSipSimple, if you are experimenting this issue and that this workaround help, you can inform us about that and tell us what is your SIP provider so that we could add a wizard with this enabled by default.
You can also notify your SIP provider that they are not fully RFC compliant (their server doesn't respect the normalization : and precising them that they do not respect RFC 3261 about contact rewrite method)
It's possible that your SIP server doesn't support actually TCP (in terms of SIP RFC compliance). Fortunately, there is a way to configure your account to force the use of UDP only :
To do so, transform your account into an Expert account (Edit account, press menu, "Choose wizard" and choose Expert). Edit again the account, there is now more settings to set. Try this :
In order to improve CSipSimple, if you are experimenting this issue and that this workaround help, you can inform us about that and tell us what is your SIP provider so that we could add a wizard with this enabled by default.
If you are experimenting audio automatically routed through rear speaker instead of earpiece or other problem with audio driver on your phone, it's probably cause manufacturer implement a strange way its audio driver and android official API for routing audio.
Fortunately, there is some common behavior that can be managed by activating some "hacks" in CSipSimple.
To do so :
Try each of these settings independently and then in combination.
If one or a combination of that modes helps... well good news for you, you have CSipSimple running correctly on your phone ;). You can then share with me which setting helps so that I can automatically turn on one of the correct hack by default for your device.
To do so, I need the exact device info (the infos announced by android to the application about the device). An easy way to collect these info is to send me logs using my embedded tool : see HowToCollectLogs.
If it doesn't work you can install other apps on the market such as "Device Info" and tell me what are the "Device" and "Product" values.
It's possible that none of these settings helps (for example it's currently a known issue on samsung moment). In this case we have to cross finger for your manufacturer or a custom ROM maker to fix that in the ROM directly. Maybe another hack could be found for your device too, but it requires devs skills and a phone to test on. If you are in this case, you can contact us so that we can give you what to test on source code to troubleshoot the problem.
That's cause of the fact your device has special policy when screen goes off. There are a few well known reasons for it:
Fortunately, there is an existing workaround in CSipSimple to prevent screen going off and so keeping up the good call quality. To activate this workaround if not automatically detected by CSipSimple (there is an autodetection done for the PSP behavior).
Actually you can, you just don't know how :).
The first prompt you get is there cause both CSipSimple and the stock SIP application can intercept the outgoing call to treat it.
If you choose "Dialer", you actually choose the stock SIP application !! If you choose CSipSimple, you choose the standard process of telephony intent.
So just choose CSipSimple ! You'll see, next you'll have the CSipSimple chooser which allow to choose Mobile call.
If you don't want to be bothered anymore with the stock chooser that propose you between the stock 2.3 SIP application and CSipSimple, in the first popup (which CSipSimple don't and CAN'T manage), there is checkbox to remember your choice.
Activate it and choose CSipSimple. Then you'll not be bothered anymore with this chooser and benefit all powerful features of CSipSimple in terms of Filters and Rewriting rules.
In future release this will not be relevant anymore. You can try nightly builds if you are hurry to get something working well on gingerbread (there is also cool features for gingerbread included in latests nightlies)
That's a known limitation. Due to the fact ZRTP is considered as a plugin for pjsip that already manages SRTP it's own way, you should never try to enable SRTP and ZRTP at the same time.
In fact if you do so, pjsip will ignore ZRTP and use SRTP as media adapter.
It's recommanded to use ZRTP that includes features of SRTP if the remote side is known to also support ZRTP. If your remote side is a sip server that only use SRTP and announce it with SIP mechanism, you'll hove to choose SRTP however.
This distribution includes cryptographic software. The country in which you currently reside may have restrictions on the import, possession, use, and/or re-export to another country, of encryption software. BEFORE using any encryption software, please check your country's laws, regulations and policies concerning the import, possession, or use, and re-export of encryption software, to see if this is permitted. See <http://www.wassenaar.org/> for more information.
The U.S. Government Department of Commerce, Bureau of Industry and Security (BIS), has classified this software as Export Commodity Control Number (ECCN) 5D002.C.1, which includes information security software using or performing cryptographic functions with asymmetric algorithms. The form and manner of this CSipSimple distribution makes it eligible for export under the License Exception ENC Technology Software Unrestricted (TSU) exception (see the BIS Export Administration Regulations, Section 740.13) for both object code and source code.
Wiki: ExpertSettingMode
Wiki: HowToCollectLogs
Wiki: HowToInstallDevVersion
Wiki: MainSideBar
Wiki: UsingFilters
Wiki: WhatIsSIP
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: j.soo...@gmail.com
Incoming voice is played via speaker instead of earpiece causing the other party to hear themselves speak. This forces me to use headphones. How can I make csipsimple use the earpiece for playback like standard calls?
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: r3gis...@gmail.com
@soong1 : you should try one last dev version (see HowToInstallDevVersion). I've recently this kind of problem on some devices. If it doesn't not help, you can use the issue section to report a detailed issue (specially what device you are using).
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: j.soo...@gmail.com
I'm already using the latest dev version and it doesn't help. I'm running on a Samsung Intercept with OS v2.1. I'll try to write an issue report.
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: j.soo...@gmail.com
After filling out the 'new issue' form the submit issue button is greyed out and not clickable for me.
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: PureLone...@googlemail.com
Hi there
I have the app set to be selectable each time I choose to dial - But I would rather that it didn't launch the application unless I choose it...
Is this possible? I have a very specific SIP setup and only want it to launch when I specifically tell it to.
Thanks
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: radu_ram...@yahoo.com
How can I choose what codec is used ? For instance for me it only works with 8khz pcm .( pressing info during call I can see this) When I enter the menu there are 4 codecs available and some unavailable. But is there any way I can convince the softphone to use GSM instead o 8khz pcm as I guess GSM is more bandwidth efficient over EDGE network.
On wifi it works awesome regardles the settings or codec.
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: r3gis...@gmail.com
@LoneWolf? : In this case you should probably choose to disable android integration and use CSipSimple dialer when you want specifically to dial out with SIP. You can also play with rewriting/filtering rules (see the corresponding section above).
@Radu : To change codec orders : in Settings > Media > codec drag and drop using the little handle at the left of the screen. To disable a codec : long press and deactivate/activate codec. Codec order here are just what CSipSimple advise. According to the settings (and supported codecs) of your remote party negociation can finally always lead to the same choose. To force the use of a codec you can try to disable all others. However be careful if you do that : if other party doesn't support the only codec you have left activated, call will directly hang up.
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: j.soo...@gmail.com
I would just like to update that I've been trying each new dev release and using the various combination of commands like "Use routing API" and "Use Mode audio API" but the problem of incoming audio going to speakers instead of earpiece persists. Thanks for you consideration.
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: ladams0...@gmail.com
@j.soong1 and @r3gis.3R - Same exact issue with Samsung Moment (older brother to Samsung Intercept). Use Routing API and Mode audio API combinations ineffective. Using build 0.00-16 r410
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: dr.k...@gmail.com
ditto @j.soong1 @ladams0000: audio only via external speaker, silent handset when speaker is disabled.
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: cristina...@gmail.com
Hello,
Is there any way of looking at the logs? I can send message to another devices but not received messages... instead of receiving the message the app is closed.
I've try to debug the code using the Android eclipse emulator but the welcome page doesn't allow me to use the app because is allways trying to download the info.
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: dengx2....@gmail.com
Hello. I pretty much thanks for your contribution.
I experienced so long delay about 500 ms. It means one way delay.
I think it is not a network problem. Is there any way to solve this problem?
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: ken.laberteaux
What is Tone hack?
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: r3gis...@gmail.com
See issue 371 For expert settings most of the time it's a good idea to do a quick search over all issues in issue list (not only opened issues) ;)
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: iiordanov@gmail.com
Hi guys,
I've written a guide on how to combine a SIP client like CSipSimple with Google Voice and the Gizmo5 SIP service to make free calls to USA and Canada over WiFi? or data without using your voice carrier at all here:
http://iiordanov.blogspot.com/2009/07/sipdroid-gv-guava.html
This works from anywhere in the world as long as you have or can create a Google Voice and Gizmo5 account.
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: pindi...@gmail.com
hi, just started using it first time with a betamax voip provider named powervoip, it seems to be working but the call quality was horrible, both of the sides were not able to listen to each other.....(the same voip account works ok on my pc though). Further when i tried to put headphones during a call....it crashed i think and restarted my wildfire...(froyo). Thanks..waiting for the improvements....
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: pindi...@gmail.com
in addition to the last comment, I just tried Stun server settings and that worked great for me and now the voice quality issue is gone.....:)
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: nilcasd...@gmail.com
I have in my Sony Ericsson Xperia X10 mini Pro the same earpiece/speaker problem the Samsungs have. Can you help me ?
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: r3gis...@gmail.com
@nilcasdias : is your X10 mini pro running android 1.6 or 2.1? (On my X10 mini it works correctly but android version is up to date : 2.1, previously sony had bugs with their audio driver on X10, X10 mini, mini pro and X8 that they have fixed AFAIK in android 2.1 update).
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: gregmaw...@gmail.com
@nilcasdias - I have an X10 mini and upgraded to 2.1 a couple of days ago. I have loaded CSip today and so far only made 1 call. The sound quality was generally very good. Very clear and understandable and was using the inbuilt speaker/mic with no problems after I enabled STUN. There was a little choppiness in the other persons voice but otherwise I'm very happy so far.
Thanks very much to the devs of this app!!
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: gauvi...@gmail.com
Bad gateway on 3g. Sipdroid connects without problem. Bug or settings?
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: r3gis...@gmail.com
@gauvin : it's a setting issue. Can you open an issue on issue list on issue tab of this website ? And precise your sip provider? I'll explain you how to configure and we will create a new wizard for your provider so that other users will not experiment the same problem.
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: ongr...@gmail.com
I get an voice mail icon and i dont have any .. anyway to disable it? its happening only with the rc versions scened Q is it possible to disable the "slide" graphics ?
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: r3gis...@gmail.com
@ongrass : for voice mail for now no option to disable voice mail integration. This new feature is based on what is sent by your sip provider : if csipsimple show a voice mail notification it mean that the sip provider announce you have an unread voice mail. But I'd be interested by logs if you can collect and send me some logs I'd be interested. For slide graphics : there is an option : see ExpertSettingMode wiki page to switch to expert mode and then in User interface choose "don't use slider for call answering" option.
View and moderate all "wiki Discussion" comments posted by this user
Mark all as spam, and block user from posting to "Wiki"
Originally posted by: azeem.za...@gmail.com
Cannot connect with voipbuster at my HTC Desire HD. www.voipbuster.com This one of the service provider from Betamax Germany. works OK with voipalot.com and siptraffic.com