Re: [asterisk-users] 1.8 and prematuremedia problem
hi: maybe you can try noload res_timing_timerfd in modules.conf and see what asterisk pick up for timing. in my system, if I disable res_timing_timerfd, then dahdi timing is selected and system become stable. Regards, tbskyd 2011/5/14 satish patel satish...@hotmail.com: You mean say i don't use res_timing_dahdi.so ? I guess this is just timing module nothing related to Card. _S From: tu...@canistec.com Date: Fri, 13 May 2011 18:30:52 +0200 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem sangoma cards do not use dahdi... 13.5.2011 v 17:16, satish patel satish...@hotmail.com: Thank you so much!! I found following (res_timing_timerfd.so in USE). But we have asterisk dahdi install and sangoma A102D pri card configured. Do you think i should use res_timing_dahdi.so ? campbx1*CLI module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 res_timing_dahdi.so DAHDI Timing Interface 0 3 modules loaded From: n...@njcolledge.net To: asterisk-users@lists.digium.com Date: Fri, 13 May 2011 15:11:19 + Subject: Re: [asterisk-users] 1.8 and prematuremedia problem At the asterisk CLI type “module show like timing” Whichever has a use-count 1 is the one you are using. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: 13 May 2011 16:03 To: tbs...@gmail.com; asterisk-users Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I
Re: [asterisk-users] 1.8 and prematuremedia problem
Thanks and I did that and my figure are cross now. Let see -- Sent from my iPhone On May 15, 2011, at 8:35 AM, d tbsky tbs...@gmail.com wrote: hi: maybe you can try noload res_timing_timerfd in modules.conf and see what asterisk pick up for timing. in my system, if I disable res_timing_timerfd, then dahdi timing is selected and system become stable. Regards, tbskyd 2011/5/14 satish patel satish...@hotmail.com: You mean say i don't use res_timing_dahdi.so ? I guess this is just timing module nothing related to Card. _S From: tu...@canistec.com Date: Fri, 13 May 2011 18:30:52 +0200 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem sangoma cards do not use dahdi... 13.5.2011 v 17:16, satish patel satish...@hotmail.com: Thank you so much!! I found following (res_timing_timerfd.so in USE). But we have asterisk dahdi install and sangoma A102D pri card configured. Do you think i should use res_timing_dahdi.so ? campbx1*CLI module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 res_timing_dahdi.soDAHDI Timing Interface 0 3 modules loaded From: n...@njcolledge.net To: asterisk-users@lists.digium.com Date: Fri, 13 May 2011 15:11:19 + Subject: Re: [asterisk-users] 1.8 and prematuremedia problem At the asterisk CLI type “module show like timing” Whichever has a use-count 1 is the one you are using. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: 13 May 2011 16:03 To: tbs...@gmail.com; asterisk-users Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch
Re: [asterisk-users] 1.8 and prematuremedia problem
Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php? id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
Re: [asterisk-users] 1.8 and prematuremedia problem
Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information
Re: [asterisk-users] 1.8 and prematuremedia problem
sangoma cards do not use dahdi... 13.5.2011 v 17:16, satish patel satish...@hotmail.com: Thank you so much!! I found following (res_timing_timerfd.so in USE). But we have asterisk dahdi install and sangoma A102D pri card configured. Do you think i should use res_timing_dahdi.so ? campbx1*CLI module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 res_timing_dahdi.soDAHDI Timing Interface 0 3 modules loaded From: n...@njcolledge.net To: asterisk-users@lists.digium.com Date: Fri, 13 May 2011 15:11:19 + Subject: Re: [asterisk-users] 1.8 and prematuremedia problem At the asterisk CLI type “module show like timing” Whichever has a use-count 1 is the one you are using. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: 13 May 2011 16:03 To: tbs...@gmail.com; asterisk-users Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can
Re: [asterisk-users] 1.8 and prematuremedia problem
You mean say i don't use res_timing_dahdi.so ? I guess this is just timing module nothing related to Card. _S From: tu...@canistec.com Date: Fri, 13 May 2011 18:30:52 +0200 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem sangoma cards do not use dahdi... 13.5.2011 v 17:16, satish patel satish...@hotmail.com: Thank you so much!! I found following (res_timing_timerfd.so in USE). But we have asterisk dahdi install and sangoma A102D pri card configured. Do you think i should use res_timing_dahdi.so ? campbx1*CLI module show like timing Module Description Use Count res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 res_timing_dahdi.soDAHDI Timing Interface 0 3 modules loaded From: n...@njcolledge.net To: asterisk-users@lists.digium.com Date: Fri, 13 May 2011 15:11:19 + Subject: Re: [asterisk-users] 1.8 and prematuremedia problem At the asterisk CLI type “module show like timing” Whichever has a use-count 1 is the one you are using. Nic. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: 13 May 2011 16:03 To: tbs...@gmail.com; asterisk-users Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Thanks for reply, How do i find asterisk using which timing res_timing_timerfd or res_timing_dahdi ? -S Date: Fri, 13 May 2011 22:13:47 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: satish...@hotmail.com; asterisk-users@lists.digium.com hi: I am using 64bit scientific linux 6 with default kernel. my loading is quite low, maybe 1~10 concurrent calls. I remember last time I have unstable problem about timer. my linux now use HPET clock. and asterisk use res_timing_dahdi instead of the default res_timing_timerfd. I don't know if these are related to you problem. hope you can find the key point to make a stable asterisk. Regards, tbskyd 2011/5/13 Satish Patel satish...@hotmail.com: Glad you solved it. Now I'm having high CPU load issue. I don't know why but sometime my asterisk process reached ~150% CPU load and just locked no calls nothing only solution is kill -9 I've 1000hz preemtive kerenel on ubuntu do you think it's the issue because of low through put ?? Which OS are you using? -- Sent from my iPhone On May 12, 2011, at 9:31 PM, d tbsky tbs...@gmail.com wrote: hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 and prematuremedia problem
I apply sig_pri.c patch for 1.8.3.2 manually. (the patch
can not apply to 1.8.3.2 or 1.8.4-rc3).
but the situation is the same. do I need to play with other
options with the patch? or I need newer asterisk versions to
solve the problem?
thanks a lot for information!!
What does your dialplan look like
Maybe a progress could help
exten = _9.,1,Progress()
exten = _9.,n,Dial(DAHDI/g0/${EXTEN}:1)
Alec
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Re: [asterisk-users] 1.8 and prematuremedia problem
I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 and prematuremedia problem
Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
