Today

• 2 hours ago

  issue 238 (No compression option) commented on by jahrome11   -   I have the same issue, editing org.sipdroid.sipua_preferences.xml on the device and changing compression from never to always made me able to activate compression but the option is still not changeable in the ui.

  I have the same issue, editing org.sipdroid.sipua_preferences.xml on the device and changing compression from never to always made me able to activate compression but the option is still not changeable in the ui.
• 2 hours ago

  issue 242 (Crash while loading pjlib) reported by jahrome11   -   I am testing sipdroid gsm codec support with an asterisk server (configured with 2 accounts restricted to GSM codec) with a G1 under cyanogen 4.9.1. While making a call, sipdroid rings but drops the call. Log shows : Cannot load library: reloc_library[1245]: 86 cannot locate '__android_log_print'... (full trace below) nm on libpjlib_linker_jni.so shows an undefined symbol U __android_log_print pjlib is distributed as binary and pjsip does not support building for android. I found a patch to build pjsip for android but havn't tested. I am interested in the project so would it be possible to share your sources of pjsip so I can help resolve the issue (I pointed other issues with strange behaviour while making calls and deregistering with a cirpack gateway) ? Jerome logcat: I/ActivityManager( 105): Displayed activity org.sipdroid.sipua/.ui.InCallScreen: 1492 ms (total 1749 ms) D/dalvikvm( 323): Trying to load lib /data/data/org.sipdroid.sipua/lib/libpjlib_linker_jni.so 0x43b5c9e0 I/ActivityManager( 105): Process com.google.android.partnersetup (pid 285) has died. D/VolumePanel( 105): onVolumeChanged(streamType: 3, flags: 0) I/dalvikvm( 323): Unable to dlopen(/data/data/org.sipdroid.sipua/lib/libpjlib_linker_jni.so): Cannot load library: reloc_library[1245]: 86 cannot locate '__android_log_print'... D/dalvikvm( 323): GC freed 2506 objects / 150344 bytes in 131ms D/AudioHardwareMSM72XX( 87): audpre_index = 0, tx_iir_index = 1 D/HTC Acoustic( 87): msm72xx_enable_audpre: 0x0000 D/dalvikvm( 323): +++ not scanning '/system/lib/libwebcore.so' for 'decode' (wrong CL) D/dalvikvm( 323): +++ not scanning '/system/lib/libexif.so' for 'decode' (wrong CL) D/dalvikvm( 323): +++ not scanning '/system/lib/libmedia_jni.so' for 'decode' (wrong CL) D/dalvikvm( 323): +++ not scanning '/system/lib/libsrec_jni.so' for 'decode' (wrong CL) D/dalvikvm( 323): +++ not scanning '/system/lib/libFFTEm.so' for 'decode' (wrong CL) W/dalvikvm( 323): No implementation found for native Lorg/sipdroid/pjlib/Codec;.decode ([B[SI)I W/dalvikvm( 323): threadid=27: thread exiting with uncaught exception (group=0x4001e170) E/AndroidRuntime( 323): Uncaught handler: thread Thread-29 exiting due to uncaught exception I/ActivityManager( 105): Process com.android.settings (pid 419) has died. E/AndroidRuntime( 323): java.lang.UnsatisfiedLinkError: decode E/AndroidRuntime( 323): at org.sipdroid.pjlib.Codec.decode(Native Method) E/AndroidRuntime( 323): at org.sipdroid.media.RtpStreamReceiver.run(RtpStreamReceiver.java:342) I/Process ( 105): Sending signal. PID: 323 SIG: 3 I/dalvikvm( 323): threadid=7: reacting to signal 3 What steps will reproduce the problem? 1. 2. 3. What is the expected output? What do you see instead? What version of the product are you using? On what operating system? Which SIP server are you using? What happens with PBXes? Which type of network are you using? Please provide any additional information below.

  I am testing sipdroid gsm codec support with an asterisk server (configured with 2 accounts restricted to GSM codec) with a G1 under cyanogen 4.9.1. While making a call, sipdroid rings but drops the call. Log shows : Cannot load library: reloc_library[1245]: 86 cannot locate '__android_log_print'... (full trace below) nm on libpjlib_linker_jni.so shows an undefined symbol U __android_log_print pjlib is distributed as binary and pjsip does not support building for android. I found a patch to build pjsip for android but havn't tested. I am interested in the project so would it be possible to share your sources of pjsip so I can help resolve the issue (I pointed other issues with strange behaviour while making calls and deregistering with a cirpack gateway) ? Jerome logcat: I/ActivityManager( 105): Displayed activity org.sipdroid.sipua/.ui.InCallScreen: 1492 ms (total 1749 ms) D/dalvikvm( 323): Trying to load lib /data/data/org.sipdroid.sipua/lib/libpjlib_linker_jni.so 0x43b5c9e0 I/ActivityManager( 105): Process com.google.android.partnersetup (pid 285) has died. D/VolumePanel( 105): onVolumeChanged(streamType: 3, flags: 0) I/dalvikvm( 323): Unable to dlopen(/data/data/org.sipdroid.sipua/lib/libpjlib_linker_jni.so): Cannot load library: reloc_library[1245]: 86 cannot locate '__android_log_print'... D/dalvikvm( 323): GC freed 2506 objects / 150344 bytes in 131ms D/AudioHardwareMSM72XX( 87): audpre_index = 0, tx_iir_index = 1 D/HTC Acoustic( 87): msm72xx_enable_audpre: 0x0000 D/dalvikvm( 323): +++ not scanning '/system/lib/libwebcore.so' for 'decode' (wrong CL) D/dalvikvm( 323): +++ not scanning '/system/lib/libexif.so' for 'decode' (wrong CL) D/dalvikvm( 323): +++ not scanning '/system/lib/libmedia_jni.so' for 'decode' (wrong CL) D/dalvikvm( 323): +++ not scanning '/system/lib/libsrec_jni.so' for 'decode' (wrong CL) D/dalvikvm( 323): +++ not scanning '/system/lib/libFFTEm.so' for 'decode' (wrong CL) W/dalvikvm( 323): No implementation found for native Lorg/sipdroid/pjlib/Codec;.decode ([B[SI)I W/dalvikvm( 323): threadid=27: thread exiting with uncaught exception (group=0x4001e170) E/AndroidRuntime( 323): Uncaught handler: thread Thread-29 exiting due to uncaught exception I/ActivityManager( 105): Process com.android.settings (pid 419) has died. E/AndroidRuntime( 323): java.lang.UnsatisfiedLinkError: decode E/AndroidRuntime( 323): at org.sipdroid.pjlib.Codec.decode(Native Method) E/AndroidRuntime( 323): at org.sipdroid.media.RtpStreamReceiver.run(RtpStreamReceiver.java:342) I/Process ( 105): Sending signal. PID: 323 SIG: 3 I/dalvikvm( 323): threadid=7: reacting to signal 3 What steps will reproduce the problem? 1. 2. 3. What is the expected output? What do you see instead? What version of the product are you using? On what operating system? Which SIP server are you using? What happens with PBXes? Which type of network are you using? Please provide any additional information below.
• 11 hours ago

  issue 237 (Support for proxy ( siproxd, etc )) commented on by costin   -   My network: 1. stupid ATT DSL router ( not modem ). Firewall set to forward most UDP/TCP to linux box 2. Linux box - running siproxd, mostly out-of-box config. For IP - it has a dyndns client, siproxd uses the dyndns hostname to rewrite. 3. a grandstream box - with outbound proxy set to the IP of the linux box 4. Android phone running sipdroid, with Server = IP of the linux box, domain=proxy01.sipphone.com, different username/password from the grandstream. Also seems to be important to login with the phone number as username, and dial the number - I couldn't get the non-phone username to work, which makes sense since siproxd only knows about one username. So far I can receive and make calls - I tried with the google voice number and a friend. I'll keep testing - there is some instability, maybe because I keep making changes. I'm trying to retire the grandstream box and only use the android(s). Let me know if you want additional details. Turning verbose logging ( including UDP dumps ) in siproxd helped a lot.

  My network: 1. stupid ATT DSL router ( not modem ). Firewall set to forward most UDP/TCP to linux box 2. Linux box - running siproxd, mostly out-of-box config. For IP - it has a dyndns client, siproxd uses the dyndns hostname to rewrite. 3. a grandstream box - with outbound proxy set to the IP of the linux box 4. Android phone running sipdroid, with Server = IP of the linux box, domain=proxy01.sipphone.com, different username/password from the grandstream. Also seems to be important to login with the phone number as username, and dial the number - I couldn't get the non-phone username to work, which makes sense since siproxd only knows about one username. So far I can receive and make calls - I tried with the google voice number and a friend. I'll keep testing - there is some instability, maybe because I keep making changes. I'm trying to retire the grandstream box and only use the android(s). Let me know if you want additional details. Turning verbose logging ( including UDP dumps ) in siproxd helped a lot.
• 18 hours ago

  issue 60 (Registration failed (timeout) - only on some wifi networks) commented on by sschuberth   -   Hmm, this issue disappeared for me with version 1.2.2, or maybe switching my AP from a hidden SSID to broadcasting the SSID also helped. Anyway, works now for me on my HTC Hero with a FritzBox.

  Hmm, this issue disappeared for me with version 1.2.2, or maybe switching my AP from a hidden SSID to broadcasting the SSID also helped. Anyway, works now for me on my HTC Hero with a FritzBox.
• 18 hours ago

  issue 60 (Registration failed (timeout) - only on some wifi networks) commented on by philipp.batroff   -   works seamless with openWrt! This thread helped a lot in finding the problem, although my "solution" isn't perfect, but works for me! Shiny :) Thanks a lot and keep up the good work!

  works seamless with openWrt! This thread helped a lot in finding the problem, although my "solution" isn't perfect, but works for me! Shiny :) Thanks a lot and keep up the good work!
• 19 hours ago

  issue 195 (Not able to hear anything from other party while on Full 3G) commented on by rolandomotaX3   -   I am in the same boat as you. Hardware: TMOBILE G1 ROM: CM 4.2.9.1 (also tested using many other CM roms and AOSP 1.6) SIP Provider: sipgate When using the app (v1.2.2) over wifi it works perfectly. On TMO's 3G/EDGE I am able to connect to the remote party but I have no sound from my device at all. Prior to v1.2.2 I was unable to connect to the remote party for more than a few seconds. The curious thing is that if I set up SIPDroid and GV Dialer by Evan Charlton with pbxes.org I can make/receive calls without any issues. What does this indicate?

  I am in the same boat as you. Hardware: TMOBILE G1 ROM: CM 4.2.9.1 (also tested using many other CM roms and AOSP 1.6) SIP Provider: sipgate When using the app (v1.2.2) over wifi it works perfectly. On TMO's 3G/EDGE I am able to connect to the remote party but I have no sound from my device at all. Prior to v1.2.2 I was unable to connect to the remote party for more than a few seconds. The curious thing is that if I set up SIPDroid and GV Dialer by Evan Charlton with pbxes.org I can make/receive calls without any issues. What does this indicate?
• 19 hours ago

  issue 237 (Support for proxy ( siproxd, etc )) commented on by osirisx11   -   coustin: Are you sure this is true support for a proxy? I am trying to get sipsorcery.com to work for outgoing calls and I am unable to. Please elaborate on your solution with more variables and less specific values. Cheers

  coustin: Are you sure this is true support for a proxy? I am trying to get sipsorcery.com to work for outgoing calls and I am unable to. Please elaborate on your solution with more variables and less specific values. Cheers
• 20 hours ago

  issue 236 (Mic volume too low when using headset with Sipdroid) changed by pmerl...@googlemail.com   -  

  Status: Duplicate

  Status: Duplicate
• 20 hours ago

  issue 241 (Sipdroid sound problems when using a headset (on Motorola Dr...) commented on by pmerl...@googlemail.com   -   Issue 236 has been merged into this issue.

  Issue 236 has been merged into this issue.
• 20 hours ago

  issue 60 (Registration failed (timeout) - only on some wifi networks) commented on by philipp.batroff   -   ups, I also made a typo on the sipdroid version there, obviosly I meant 1.2.2! I forget to mention that I can succesfully register (only after the update to 1.2.2), but I cannot receive calls and or phone out. then the yellow button appears, and afterwards I get the same "registration failure(timeout)" message as the other did ... I am sorry for the double post!

  ups, I also made a typo on the sipdroid version there, obviosly I meant 1.2.2! I forget to mention that I can succesfully register (only after the update to 1.2.2), but I cannot receive calls and or phone out. then the yellow button appears, and afterwards I get the same "registration failure(timeout)" message as the other did ... I am sorry for the double post!
• 20 hours ago

  issue 60 (Registration failed (timeout) - only on some wifi networks) commented on by philipp.batroff   -   Hello, I have the same problems with the Linksys54G with WPA2 private and TKIP/AES. I am running on Android 1.6 and Sipdroid 2.2beta. I can register fine, and it works on 3G networks. ALso I tried it on a different wireless, and that worked as well. That's why I found this posting...Just wanted you guys to know that the problem seems to still be there. I am gonna try OpenWRT or sth like that now

  Hello, I have the same problems with the Linksys54G with WPA2 private and TKIP/AES. I am running on Android 1.6 and Sipdroid 2.2beta. I can register fine, and it works on 3G networks. ALso I tried it on a different wireless, and that worked as well. That's why I found this posting...Just wanted you guys to know that the problem seems to still be there. I am gonna try OpenWRT or sth like that now
• 22 hours ago

  issue 241 (Sipdroid sound problems when using a headset (on Motorola Dr...) reported by d...@davidshields.us   -   Sipdroid works fine when listening to calls on the phone and speaking into the phone's mic. However, Sipdroid has sound problems when using a headset. There are two related problems: 1. mic volume is too low with headset under all conditions. 2. under special conditions, the caller cannot hear any sound -- not even the called phone ringing -- when using a headset. This happens when the VoIP call is initiated by Google Voice (using web page). I'll elaborate on this problem below. -- What steps will reproduce the problem? 1. plug any headset with mic into Droid. 2. go to Google Voice website (on phone or other computer) and initiate a call using Gizmo (which can be answered on the Droid). 3. answer the call. What is the expected output? Expect to hear called party's phone ringing and expect to hear the other person speaking after they answer. What do you see instead? Do not hear other phone ringing, although it is ringing. Other party answers, but I cannot hear them speaking. What version of the product are you using? Sipdroid 1.2.1 beta On what operating system? Android 2.01 on Droid Which SIP server are you using? Gizmo5 Which type of network are you using? WiFi Please provide any additional information below. Hopefully the formatting of the table below will be OK. Legend: After=Headset Plugged in Only After call was connected. Prior=Headset Plugged in Prior to dialing call (normal use case). * asterisk indicates lower microphone sound volume NoHeadset After Prior Call Initiated By: ---- ----- ----- Google Voice (Web) Good OK* no sound Sipdroid from phone Good OK* OK* Droid Phone App Good Good Good This table shows that with Sipdroid, there are always sound issues when using a headset, but no sound issues when not using a headset. The worst case is when using Sipdroid and the call is initiated by Google Voice and a headset is used. The caller hears no sound and sound transmitted by the mic is the lowest.

  Sipdroid works fine when listening to calls on the phone and speaking into the phone's mic. However, Sipdroid has sound problems when using a headset. There are two related problems: 1. mic volume is too low with headset under all conditions. 2. under special conditions, the caller cannot hear any sound -- not even the called phone ringing -- when using a headset. This happens when the VoIP call is initiated by Google Voice (using web page). I'll elaborate on this problem below. -- What steps will reproduce the problem? 1. plug any headset with mic into Droid. 2. go to Google Voice website (on phone or other computer) and initiate a call using Gizmo (which can be answered on the Droid). 3. answer the call. What is the expected output? Expect to hear called party's phone ringing and expect to hear the other person speaking after they answer. What do you see instead? Do not hear other phone ringing, although it is ringing. Other party answers, but I cannot hear them speaking. What version of the product are you using? Sipdroid 1.2.1 beta On what operating system? Android 2.01 on Droid Which SIP server are you using? Gizmo5 Which type of network are you using? WiFi Please provide any additional information below. Hopefully the formatting of the table below will be OK. Legend: After=Headset Plugged in Only After call was connected. Prior=Headset Plugged in Prior to dialing call (normal use case). * asterisk indicates lower microphone sound volume NoHeadset After Prior Call Initiated By: ---- ----- ----- Google Voice (Web) Good OK* no sound Sipdroid from phone Good OK* OK* Droid Phone App Good Good Good This table shows that with Sipdroid, there are always sound issues when using a headset, but no sound issues when not using a headset. The worst case is when using Sipdroid and the call is initiated by Google Voice and a headset is used. The caller hears no sound and sound transmitted by the mic is the lowest.

Yesterday

• 32 hours ago

  issue 240 (Search and replace feature only seems to support one rule) Labels changed by pmerl...@googlemail.com   -  

  Labels: Type-Enhancement −Type-Defect

  Labels: Type-Enhancement −Type-Defect
• 32 hours ago

  issue 237 (Support for proxy ( siproxd, etc )) Status changed by pmerl...@googlemail.com   -  

  Status: Done

  Status: Done
• 35 hours ago

  issue 240 (Search and replace feature only seems to support one rule) reported by david.sainty   -   It would be nice if multiple regular expression search+replace operations could be specified, as one is not always enough.

  It would be nice if multiple regular expression search+replace operations could be specified, as one is not always enough.
• 35 hours ago

  issue 237 (Support for proxy ( siproxd, etc )) commented on by costin   -   Never mind - found the solution: set Server to the IP address of the siproxd-running, and domain to proxy01.sipphone.com. Seems to work. Changing the bug to 'better documentation for the "domain" option and how to get it to work with a proxy'

  Never mind - found the solution: set Server to the IP address of the siproxd-running, and domain to proxy01.sipphone.com. Seems to work. Changing the bug to 'better documentation for the "domain" option and how to get it to work with a proxy'

Last 7 days

• Dec 17, 2009

  issue 239 (Sipdroid on WiFi causes me to miss calls all the time) commented on by d...@davidshields.us   -   Changing my WiFi sleep policy to "never" seems to have resolved this. I'm using UDP. I will report back if this is not totally resolved.

  Changing my WiFi sleep policy to "never" seems to have resolved this. I'm using UDP. I will report back if this is not totally resolved.
• Dec 17, 2009

  issue 239 (Sipdroid on WiFi causes me to miss calls all the time) Status changed by pmerl...@googlemail.com   -   Issue solved by chaning sleep policy.

  Status: Done

  Issue solved by chaning sleep policy.

  Status: Done
• Dec 17, 2009

  issue 239 (Sipdroid on WiFi causes me to miss calls all the time) commented on by pmerl...@googlemail.com   -   Do you observe this with UDP or TCP protocol, or both?

  Do you observe this with UDP or TCP protocol, or both?
• Dec 17, 2009

  issue 147 (Incoming Calls still not working right ) changed by pmerl...@googlemail.com   -  

  Status: Fixed

  Status: Fixed
• Dec 17, 2009

  issue 239 (Sipdroid on WiFi causes me to miss calls all the time) commented on by d...@davidshields.us   -   I just changed my WiFi sleep policy to "never". Maybe this will resolve the issue... ? In my tests so far, it seems to help, but the phone doesn't make a ringing sound. I have to watch the screen to see the incoming call. So this may be one small step in the right direction, but there are still issues.

  I just changed my WiFi sleep policy to "never". Maybe this will resolve the issue... ? In my tests so far, it seems to help, but the phone doesn't make a ringing sound. I have to watch the screen to see the incoming call. So this may be one small step in the right direction, but there are still issues.
• Dec 17, 2009

  issue 239 (Sipdroid on WiFi causes me to miss calls all the time) reported by d...@davidshields.us   -   I use my Droid in WiFi mode all the time. (Airplane mode is on, WiFI is on.) I always miss incoming calls. -- What steps will reproduce the problem? 1. Set up Sipdroid. 2. put Droid in airplane mode, then turn ON WiFi. 3. Open Sipdroid and let it register (shows green light in status bar). 4. wait for incoming phone calls What is the expected output? Expect to see phone ring when calls come in. What do you see instead? Sipdroid becomes unregistered after a while (not sure how long) and I miss incoming calls. I have to open Sipdroid again and let it register again. But it is too late -- I have already missed incoming calls. What version of the product are you using? 1.2.1 beta On what operating system? Android 2.01 Which SIP server are you using? Gizmo5 Which type of network are you using? WiFi

  I use my Droid in WiFi mode all the time. (Airplane mode is on, WiFI is on.) I always miss incoming calls. -- What steps will reproduce the problem? 1. Set up Sipdroid. 2. put Droid in airplane mode, then turn ON WiFi. 3. Open Sipdroid and let it register (shows green light in status bar). 4. wait for incoming phone calls What is the expected output? Expect to see phone ring when calls come in. What do you see instead? Sipdroid becomes unregistered after a while (not sure how long) and I miss incoming calls. I have to open Sipdroid again and let it register again. But it is too late -- I have already missed incoming calls. What version of the product are you using? 1.2.1 beta On what operating system? Android 2.01 Which SIP server are you using? Gizmo5 Which type of network are you using? WiFi
• Dec 17, 2009

  issue 9 (Sent-by Address: 127.0.0.1) commented on by bennett16   -   Thank You Devs!!! :-) This fixed my problem and made my phone usable with our asterisk server. Now all I have to do is convince the boss to pay for the phone!

  Thank You Devs!!! :-) This fixed my problem and made my phone usable with our asterisk server. Now all I have to do is convince the boss to pay for the phone!
• Dec 17, 2009

  issue 231 ( public AttributeField getAttribute(String attribute_name) ) commented on by pmerl...@googlemail.com   -   How did you find that? Did you get force closes, if yes with which provider?

  How did you find that? Did you get force closes, if yes with which provider?
• Dec 16, 2009

  issue 103 (Connection to 3CX Phone systems) commented on by abedfarah   -   Any Updates on this issue, I am having the same issue with 3cx server

  Any Updates on this issue, I am having the same issue with 3cx server
• Dec 16, 2009

  issue 238 (No compression option) reported by adam.rybicki   -   NOTE: This form is only for reporting bugs. For problems, questions, or comments, please visit: http://groups.google.com/group/sipdroid-users ************************************************* * Read the FAQ * * Check if the bug has already been reported * * No doubles please * ************************************************* Did you carefully read above and decide this means of communication is the right for your notice? Don't just list an incompatible SIP server or device, please. Try to look behind the scenes if you like to contribute here. Thanks! -- What steps will reproduce the problem? 1.Registered with pbxes.org and entered the info into Sipdroid (got green light) 2.Turned on Use WLAN, USE 3G, and Use EDGE 3.Voice Compression set to Never and disabled What is the expected output? What do you see instead? Expected to enable compression for EDGE, but cannot What version of the product are you using? On what operating system? 1.2.2 on G1 with CyanogenMod-4.2.8.1, but 4.2.7.1 was the same Which SIP server are you using? What happens with PBXes? pbxes.org Which type of network are you using? WLAN and 3G Please provide any additional information below. Looking at the source code, it seems that a failure to load the native library could cause this. No idea how to test that. Uninstalling 1.2.2, which has been upgraded from numerous older versions over time, and reinstalling did not help.

  NOTE: This form is only for reporting bugs. For problems, questions, or comments, please visit: http://groups.google.com/group/sipdroid-users ************************************************* * Read the FAQ * * Check if the bug has already been reported * * No doubles please * ************************************************* Did you carefully read above and decide this means of communication is the right for your notice? Don't just list an incompatible SIP server or device, please. Try to look behind the scenes if you like to contribute here. Thanks! -- What steps will reproduce the problem? 1.Registered with pbxes.org and entered the info into Sipdroid (got green light) 2.Turned on Use WLAN, USE 3G, and Use EDGE 3.Voice Compression set to Never and disabled What is the expected output? What do you see instead? Expected to enable compression for EDGE, but cannot What version of the product are you using? On what operating system? 1.2.2 on G1 with CyanogenMod-4.2.8.1, but 4.2.7.1 was the same Which SIP server are you using? What happens with PBXes? pbxes.org Which type of network are you using? WLAN and 3G Please provide any additional information below. Looking at the source code, it seems that a failure to load the native library could cause this. No idea how to test that. Uninstalling 1.2.2, which has been upgraded from numerous older versions over time, and reinstalling did not help.
• Dec 16, 2009

  r391 (string completion for french translations) committed by vherilier   -   string completion for french translations

  string completion for french translations
• Dec 16, 2009

  r390 (fix errors by escaping the "'" character) committed by vherilier   -   fix errors by escaping the "'" character

  fix errors by escaping the "'" character
• Dec 16, 2009

  issue 237 (Support for proxy ( siproxd, etc )) Labels changed by pmerl...@googlemail.com   -  

  Labels: Type-Enhancement −Type-Defect

  Labels: Type-Enhancement −Type-Defect
• Dec 16, 2009

  issue 237 (Support for proxy ( siproxd, etc )) reported by costin   -   -- What steps will reproduce the problem? Wifi with a NAT. I have a router that forwards all SIP/RTP ports to a siproxd, and few SIP phones (grandstream, etc) that connect using the proxy. Sipdroid doesn't seem to have any option for that. Looking at the code, it looks like there is a method SipdroidEngine.setOutboundProxy() which seem to do what is needed, using the "dns" setting. There is also code to add the dns to the settings - the only thing missing is an entry in preferences.xml. Any reason for not having it exposed ? I can test it and submit a patch if it works. What version of the product are you using? On what operating system? Market version - now testing with SVN head. Which SIP server are you using? What happens with PBXes? gizmo, SER, asterisk. Which type of network are you using? Wifi. Please provide any additional information below.

  -- What steps will reproduce the problem? Wifi with a NAT. I have a router that forwards all SIP/RTP ports to a siproxd, and few SIP phones (grandstream, etc) that connect using the proxy. Sipdroid doesn't seem to have any option for that. Looking at the code, it looks like there is a method SipdroidEngine.setOutboundProxy() which seem to do what is needed, using the "dns" setting. There is also code to add the dns to the settings - the only thing missing is an entry in preferences.xml. Any reason for not having it exposed ? I can test it and submit a patch if it works. What version of the product are you using? On what operating system? Market version - now testing with SVN head. Which SIP server are you using? What happens with PBXes? gizmo, SER, asterisk. Which type of network are you using? Wifi. Please provide any additional information below.
• Dec 16, 2009

  issue 236 (Mic volume too low when using headset with Sipdroid) reported by d...@davidshields.us   -   I'm using a Motorola Droid with Sipdroid, Gizmo5 and Google Voice. I have a v-moda Vibe II headset, an Etymotic Ety-Com headset and a couple other headsets for testing. What steps will reproduce the problem? 1. Plug in headset to Droid. 2. Place an outbound call using Sipdroid. (I am calling into my own voicemail and leaving a message that I can play back later.) 3. Speak at a consistent volume level in a quiet office environment. 4. Repeat steps without a headset. Speak into phone's mic. Use same call settings (with Sipdroid). 5. Compare volume levels with and without a headset. 6. Repeat the above steps but do not use Sipdroid. Make the call using cellular service. 7. Compare volume levels again (with and without headset). What is the expected output? What do you see instead? With any headset, the mic volume is too low when using Sipdroid. Without a headset the mic volume is fine. When not using Sipdroid, mic volume is similar without and with a headset. The low volume only happens when using Sipdroid. I expect that my voice volume when using a headset will be equally as loud as when I speak into the phone's built-in mic. (By voice volume I mean the sound level as heard by the party I called, or as recorded on the voice mail message.) So far I have recorded over 100 messages into my voice mail testing every possible combination of headset, call settings, etc. When I do not use Sipdroid (e.g., make a regular cellular call), the headset mic volume and the phone's internal mic volume are very similar. With Sipdroid and any headset, mic volume is too low. What version of the product are you using? On what operating system? Sipdroid 1.2.1 beta (I didn't update to 1.2.2 because so many people are reporting problems) Android 2.01 Which SIP server are you using? Gizmo5 What happens with PBXes? not able to use it yet - haven't figured out how to set it up and get it to work with Google Voice, etc. Which type of network are you using? WiFi mostly. 3G sometimes. Please provide any additional information below. Sipdroid is my most essential app! Keep up the great work! Thanks.

  I'm using a Motorola Droid with Sipdroid, Gizmo5 and Google Voice. I have a v-moda Vibe II headset, an Etymotic Ety-Com headset and a couple other headsets for testing. What steps will reproduce the problem? 1. Plug in headset to Droid. 2. Place an outbound call using Sipdroid. (I am calling into my own voicemail and leaving a message that I can play back later.) 3. Speak at a consistent volume level in a quiet office environment. 4. Repeat steps without a headset. Speak into phone's mic. Use same call settings (with Sipdroid). 5. Compare volume levels with and without a headset. 6. Repeat the above steps but do not use Sipdroid. Make the call using cellular service. 7. Compare volume levels again (with and without headset). What is the expected output? What do you see instead? With any headset, the mic volume is too low when using Sipdroid. Without a headset the mic volume is fine. When not using Sipdroid, mic volume is similar without and with a headset. The low volume only happens when using Sipdroid. I expect that my voice volume when using a headset will be equally as loud as when I speak into the phone's built-in mic. (By voice volume I mean the sound level as heard by the party I called, or as recorded on the voice mail message.) So far I have recorded over 100 messages into my voice mail testing every possible combination of headset, call settings, etc. When I do not use Sipdroid (e.g., make a regular cellular call), the headset mic volume and the phone's internal mic volume are very similar. With Sipdroid and any headset, mic volume is too low. What version of the product are you using? On what operating system? Sipdroid 1.2.1 beta (I didn't update to 1.2.2 because so many people are reporting problems) Android 2.01 Which SIP server are you using? Gizmo5 What happens with PBXes? not able to use it yet - haven't figured out how to set it up and get it to work with Google Voice, etc. Which type of network are you using? WiFi mostly. 3G sometimes. Please provide any additional information below. Sipdroid is my most essential app! Keep up the great work! Thanks.
• Dec 16, 2009

  issue 212 (HTC Hero problem: dialing using preffered Call Type set to S...) commented on by krabica   -   I have the same issue on my HTC Hero 1.5.

  I have the same issue on my HTC Hero 1.5.
• Dec 15, 2009

  issue 189 (Auto-answer calls from specific number(s) [enhancement]) commented on by brilthor   -   This is INCREDIBLY hackish and a terrible way to do it, but if you don't want to wait as I didn't you can include this line as part of the if statement at the beggining of "public void onResume()" in org.sipdroid.sipua.ui.inCallScreen.java else if (Receiver.ccConn.getAddress().equals("<10digitGVnum>") && !mKeyguardManager.inKeyguardRestrictedInputMode()) mHandler.sendEmptyMessageDelayed(MSG_ANSWER, 1000); I repeat; this is a terrible way to do it but it works ps. if you want to disable the ringer on the gv calls too you can do a similar edit in Receiver.java inside the "onState" section

  This is INCREDIBLY hackish and a terrible way to do it, but if you don't want to wait as I didn't you can include this line as part of the if statement at the beggining of "public void onResume()" in org.sipdroid.sipua.ui.inCallScreen.java else if (Receiver.ccConn.getAddress().equals("<10digitGVnum>") && !mKeyguardManager.inKeyguardRestrictedInputMode()) mHandler.sendEmptyMessageDelayed(MSG_ANSWER, 1000); I repeat; this is a terrible way to do it but it works ps. if you want to disable the ringer on the gv calls too you can do a similar edit in Receiver.java inside the "onState" section
• Dec 15, 2009

  issue 181 (Port 5060/udp not open/listening) commented on by brilthor   -   posted an enhancement before locating this one, same issue but from a different direction, please merge issue 235 to this one

  posted an enhancement before locating this one, same issue but from a different direction, please merge issue 235 to this one
• Dec 15, 2009

  issue 234 (Strange problem with Sipdroid call connections) Status changed by pmerl...@googlemail.com   -   Too strange to open an issue.

  Status: Invalid

  Too strange to open an issue.

  Status: Invalid
• Dec 15, 2009

  issue 235 (Option to run without registration (listen on 5060 for call)) commented on by brilthor   -   found dupe in defects, or at least similar; please merge with issue 181

  found dupe in defects, or at least similar; please merge with issue 181
• Dec 15, 2009

  issue 235 (Option to run without registration (listen on 5060 for call)) Status changed by pmerl...@googlemail.com   -   This is a duplicate of two other issues (listen to port 5060 and run without registration).

  Status: Invalid

  This is a duplicate of two other issues (listen to port 5060 and run without registration).

  Status: Invalid
• Dec 15, 2009

  issue 235 (Option to run without registration (listen on 5060 for call)) commented on by brilthor   -   I can't seem to find how to list it as an enhancement, does this need to be done by a mod of some kind?

  I can't seem to find how to list it as an enhancement, does this need to be done by a mod of some kind?
• Dec 15, 2009

  issue 235 (Option to run without registration (listen on 5060 for call)) reported by brilthor   -   a possible enhancement: I would love to see the possibility of setting up without registration (just listening on 5060 for incoming calls / sending outgoing through configured proxy), two distinct benefits being: 1) battery power, no registration means no need to transmit sure tcp is better than udp, but if you need neither it's even better 2) no more issue with the registration timing out or jumping between ports at a ridiculous rate this clearly would only work in an environment where the phone a static ip (or ddns) and no nat, but given those two criteria are met it is a superior method of operating if this is not of sufficiently widespread use to be included into the mainline program if someone could point me to where in the source I can hack something like this up for personal use that would be beneficial

  a possible enhancement: I would love to see the possibility of setting up without registration (just listening on 5060 for incoming calls / sending outgoing through configured proxy), two distinct benefits being: 1) battery power, no registration means no need to transmit sure tcp is better than udp, but if you need neither it's even better 2) no more issue with the registration timing out or jumping between ports at a ridiculous rate this clearly would only work in an environment where the phone a static ip (or ddns) and no nat, but given those two criteria are met it is a superior method of operating if this is not of sufficiently widespread use to be included into the mainline program if someone could point me to where in the source I can hack something like this up for personal use that would be beneficial
• Dec 15, 2009

  issue 234 (Strange problem with Sipdroid call connections) commented on by rojajimmy   -   Also, I would like toa dd that I had no problems using Pbxes before but now when I use Pbxes (Which is configured correctly with my Gizmo credentials) it registers but I cannot make any calls using the call back method or if I call my own GV number it does not ring which is absolutely silly because it should if its registered and the green dot is on. No. 2 If I use Auto answer option to avoid the call getting disconnected as soon as it gets connected, I get the same problem... "No data" and then the phone gets disconnected. I am using Cyanogen's latest modded firmware on HTC magic in Australia on Hutchison 3. Thanks

  Also, I would like toa dd that I had no problems using Pbxes before but now when I use Pbxes (Which is configured correctly with my Gizmo credentials) it registers but I cannot make any calls using the call back method or if I call my own GV number it does not ring which is absolutely silly because it should if its registered and the green dot is on. No. 2 If I use Auto answer option to avoid the call getting disconnected as soon as it gets connected, I get the same problem... "No data" and then the phone gets disconnected. I am using Cyanogen's latest modded firmware on HTC magic in Australia on Hutchison 3. Thanks
• Dec 15, 2009

  issue 234 (Strange problem with Sipdroid call connections) reported by rojajimmy   -   NOTE: This form is only for reporting bugs. For problems, questions, or comments, please visit: http://groups.google.com/group/sipdroid-users ************************************************* * Read the FAQ * * Check if the bug has already been reported * * No doubles please * ************************************************* Did you carefully read above and decide this means of communication is the right for your notice? Don't just list an incompatible SIP server or device, please. Try to look behind the scenes if you like to contribute here. Thanks! -- What steps will reproduce the problem? 1.Strange problem with calls. It gets connected but after a while the data loss is around 50% or more and in less than a minute it shows no data. 2. I cant hear anything from the other end when the "no data" thing comes up 3. The call sometimes just gets disconnected as soon as it gets connected. What is the expected output? What do you see instead? What version of the product are you using? On what operating system? Sipdroid. 1.2.2 Which SIP server are you using? What happens with PBXes? Gizmo5 and Pbxes. With Pbxes, I never get a call to ring on Sipdroid Which type of network are you using? 3G, HSDPA and WIFi. Tried everything so far. Please provide any additional information below. I m on Hutchison 3 in Sydney, Australia and it provides VOIP

  NOTE: This form is only for reporting bugs. For problems, questions, or comments, please visit: http://groups.google.com/group/sipdroid-users ************************************************* * Read the FAQ * * Check if the bug has already been reported * * No doubles please * ************************************************* Did you carefully read above and decide this means of communication is the right for your notice? Don't just list an incompatible SIP server or device, please. Try to look behind the scenes if you like to contribute here. Thanks! -- What steps will reproduce the problem? 1.Strange problem with calls. It gets connected but after a while the data loss is around 50% or more and in less than a minute it shows no data. 2. I cant hear anything from the other end when the "no data" thing comes up 3. The call sometimes just gets disconnected as soon as it gets connected. What is the expected output? What do you see instead? What version of the product are you using? On what operating system? Sipdroid. 1.2.2 Which SIP server are you using? What happens with PBXes? Gizmo5 and Pbxes. With Pbxes, I never get a call to ring on Sipdroid Which type of network are you using? 3G, HSDPA and WIFi. Tried everything so far. Please provide any additional information below. I m on Hutchison 3 in Sydney, Australia and it provides VOIP
• Dec 15, 2009

  issue 9 (Sent-by Address: 127.0.0.1) commented on by Pimmetje   -   [username] callerid= canreinvite=no context=internal dtmfmode=auto host=dynamic mailbox=2099 nat=yes port=5060 qualify=no record_in=Never record_out=Never secret=password type=friend username=username I used those settings with my asterisk server with is not in the same subnet (asterisk 192.168.1.asterisk) My own network 102.168.2.client the router of my 2 network has a 192.168.1.router adres the registration works now but calling still won't work. The server see the 192.168.2.sipdroid. My guess is that asterisk does not know this adress so wont know where to send the response. Dont know if i should make a other issu for this cause it is very related to this one. Sorry for my not so well English.

  [username] callerid= canreinvite=no context=internal dtmfmode=auto host=dynamic mailbox=2099 nat=yes port=5060 qualify=no record_in=Never record_out=Never secret=password type=friend username=username I used those settings with my asterisk server with is not in the same subnet (asterisk 192.168.1.asterisk) My own network 102.168.2.client the router of my 2 network has a 192.168.1.router adres the registration works now but calling still won't work. The server see the 192.168.2.sipdroid. My guess is that asterisk does not know this adress so wont know where to send the response. Dont know if i should make a other issu for this cause it is very related to this one. Sorry for my not so well English.
• Dec 15, 2009

  issue 64 (Sipdroid on Android 1.0 release [Project 'Sipdroid' is missi...) commented on by brilthor   -   gen is just that; it's a folder with files generated by the compiler / eclipse, do a build and it will appear

  gen is just that; it's a folder with files generated by the compiler / eclipse, do a build and it will appear
• Dec 15, 2009

  issue 204 (Volume classification) commented on by brilthor   -   is there a technical limitation which makes it impossible to have the audio classified as a call? else why is it not

  is there a technical limitation which makes it impossible to have the audio classified as a call? else why is it not
• Dec 15, 2009

  issue 189 (Auto-answer calls from specific number(s) [enhancement]) commented on by brilthor   -   as far as additions go this one should be pretty quick; just a change to the ui and a if statement in the auto- answer code, I'll download the source and hardcode for now but I would really like to see this included

  as far as additions go this one should be pretty quick; just a change to the ui and a if statement in the auto- answer code, I'll download the source and hardcode for now but I would really like to see this included
• Dec 15, 2009

  issue 195 (Not able to hear anything from other party while on Full 3G) commented on by Surie.Lee   -   I also suffer this problem, - I cannot hear the connection alarm when I call to others(btw, call is connected and talking each other is possible, sound quality is good) - When others call me, I cannot hear their voices, they cannot hear me as well - I didn't test on WiFi, I usually use on 3G connection I'm using Samsung Galaxy(i7500), Firmware ver. is 1.5(I7500XXII5), and 3G by Cellcom, Israel. SIP server is 123.142.129.22:5060 (it's a Korean public VoIP provider, mylg070). When I use previous ver. of sipdroid, it didn't work at all(no voices each other), I appreciate better performance(now I can make a call, even though I cannot get!), but please make much better. Thanks in advance! :)

  I also suffer this problem, - I cannot hear the connection alarm when I call to others(btw, call is connected and talking each other is possible, sound quality is good) - When others call me, I cannot hear their voices, they cannot hear me as well - I didn't test on WiFi, I usually use on 3G connection I'm using Samsung Galaxy(i7500), Firmware ver. is 1.5(I7500XXII5), and 3G by Cellcom, Israel. SIP server is 123.142.129.22:5060 (it's a Korean public VoIP provider, mylg070). When I use previous ver. of sipdroid, it didn't work at all(no voices each other), I appreciate better performance(now I can make a call, even though I cannot get!), but please make much better. Thanks in advance! :)
• Dec 15, 2009

  issue 233 (code style for android) reported by yuxiao100   -   just some suggustions for coding sipdroid. hopes 1. add mXXXX prefix to standard for class member. 2. if possible, use Log.v,Log.e those native handy android logging functions 3. add more comments to code,

  just some suggustions for coding sipdroid. hopes 1. add mXXXX prefix to standard for class member. 2. if possible, use Log.v,Log.e those native handy android logging functions 3. add more comments to code,
• Dec 15, 2009

  issue 232 (createOffer,createAnswer audio media format agreement) reported by yuxiao100   -   version 1.22 in UserAgent.java createOffer,createAnswer function when you get a media description like m=audio 11576 RTP/AVP 98 3 0 8 101 a=rtpmap:98 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 you need to parse rtpmap description rtpmap:8 PCMA/8000 to know 8 is for pcma instead of just parse the "98 3 0 8 101" to say 3 is for GSM/8000. it could change the mapping by the voip server.

  version 1.22 in UserAgent.java createOffer,createAnswer function when you get a media description like m=audio 11576 RTP/AVP 98 3 0 8 101 a=rtpmap:98 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 you need to parse rtpmap description rtpmap:8 PCMA/8000 to know 8 is for pcma instead of just parse the "98 3 0 8 101" to say 3 is for GSM/8000. it could change the mapping by the voip server.
• Dec 15, 2009

  r389 (Update Russian translation) committed by and...@sitnik.ru   -   Update Russian translation

  Update Russian translation
• Dec 15, 2009

  issue 231 ( public AttributeField getAttribute(String attribute_name) ) reported by yuxiao100   -   line 747 of SessionDescriptor.java: public AttributeField getAttribute(String attribute_name) { for (int i = 0; i < media.size(); i++) { AttributeField af = (AttributeField) av.elementAt (i); if (af.getAttributeName().equals(attribute_name)) return af; } return null; } should it be for (int i = 0; i < av.size(); i++) { or not ???

  line 747 of SessionDescriptor.java: public AttributeField getAttribute(String attribute_name) { for (int i = 0; i < media.size(); i++) { AttributeField af = (AttributeField) av.elementAt (i); if (af.getAttributeName().equals(attribute_name)) return af; } return null; } should it be for (int i = 0; i < av.size(); i++) { or not ???