Re: [Asterisk-Users] No audio? Update your Asterisk
Roger Hill wrote: I'm picking up the tail end of a thread, so apologies if this is offtrack... Have you perhaps got an old set of EXECUTABLES in your path, that are being picked up before your newly compiled ones? If you are under linux rm /usr/lib/asterisk/modules/* rm /usr/include/asterisk/* cd asterisk-1.2.4 make clean make upgrade asterisk -r stop now safe_asterisk that's all Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
FIXED!!! Finally got some free time to really look into it while not rusing around with work. The problem boiled down to this: IN sip.conf: externip = X.X.X.X ;Outside address localnet = 10.73.73.133 ;Inside address localmask = 255.255.255.0 ;Inside subnet This worked for months months months (since mid 2005) with no problems for me. I do remember seeing the warnings about changing it to the new format quite awhile back but forgot about it. What I had to do was simple: Changed it to the new format -- externip = X.X.X.X ;Outside address localnet = 10.73.73.0/255.255.255.0 ;Inside address Now everything works! and I have working sip to sip audio once again! The world smiles on my RTP packets once again. Was strange how it all failed on Jan, 25 along with the 'time bomb' failure/problem that everyone else had. Seems that this had something to do with it! Simple fix but a major hair pulling experience to track it down in spare time. :-) Thanks for the help and I hope I didn't waste your time too badly here. Steve Good question, But the answer is no. I have went through the trouble to make sure that all traces of other asterisk libraries/modules, config files excutables are removed from the system before compiling running testing anything. I am also being sure to unload ztdummy zaptel modules before removing the files and recompiling. For good/bad measure I am also completely powering off the system between attemps to ensure the USB hardware being used for timing with our 2.4 kernel is getting reset properly before trying again. Probably not really needed but I'm at the point of desperation and am trying not to leave anything out. Thanks for your time! Steve Steve: I'm picking up the tail end of a thread, so apologies if this is offtrack... Have you perhaps got an old set of EXECUTABLES in your path, that are being picked up before your newly compiled ones? Roger Steve Gladden wrote: Yes I have. I have been battling this issue since wednesday 1-25 And so far have tried many things. Have also tried RTP debug and do not see ANY RTP when the call is made. I will keep working at this until I figure it out but right now am very stumped and frusterated. The software update SHOULD have fixed it as it has for many others. Steve Have you tried increasing the debug level and watching the cli? No Firewalls involved, the test has been simplified down to two sip phones on a LAN and still no audio. For waht it's worth IAX2 still works fine. Steve - Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users
Re: [Asterisk-Users] No audio? Update your Asterisk
Sure enough we lost ALL sip-sip audio on 1-25 Pulled my hair out for hours before looking here or at the website to find this problem reported... Very greatful to find this I have upgraded to 1.2.3 but still have no sip-sip audio! what?! Now I'm back to contnued hair pulling what culd I possible be missing? Have started over re-compiled and reinstalled rebooted, tried it all over again -- yet cannot get any audio between sip phones on the LAN! I must be doing something stupid but what???! Any pointers appreciated! Thanks! Steve -- Vontage*CLI show version Asterisk 1.2.3 built by root @ Vontage on a i686 running Linux on 2006-01-29 10:38:54 UTC Vontage*CLI show version files File Revision func_callerid.c Revision: 7221 cdr_custom.c Revision: 7221 cdr_manager.c Revision: 7221 cdr_csv.c Revision: 7221 format_g723.c Revision: 7221 format_jpeg.c Revision: 7221 format_au.c Revision: 7221 format_sln.c Revision: 7221 format_ilbc.c Revision: 7221 format_g726.c Revision: 7221 format_h263.c Revision: 7221 format_pcm_alaw.c Revision: 7819 format_g729.c Revision: 7221 format_pcm.c Revision: 7819 format_vox.c Revision: 7221 format_wav_gsm.c Revision: 7221 format_wav.c Revision: 7221 format_gsm.c Revision: 7221 codec_g726.c Revision: 7221 codec_a_mu.c Revision: 7221 codec_alaw.c Revision: 7221 codec_ulaw.c Revision: 7221 codec_adpcm.c Revision: 7221 codec_lpc10.c Revision: 7221 codec_gsm.c Revision: 7221 codec_ilbc.c Revision: 7221 app_sms.c Revision: 7634 app_page.cRevision: 7274 app_zapscan.c Revision: 7221 app_zapbarge.cRevision: 7221 app_flash.c Revision: 7221 app_meetme.c Revision: 8194 app_zapras.c Revision: 7221 app_mixmonitor.c Revision: 7740 app_directed_pickup.c Revision: 7550 app_externalivr.c Revision: 7634 app_dictate.c Revision: 7221 app_settransfercapability Revision: 7221 app_chanspy.c Revision: 7740 app_readfile.cRevision: 7221 app_md5.c Revision: 7221 app_setrdnis.cRevision: 7221 app_while.c Revision: 7221 app_waitforsilence.c Revision: 7605 app_dumpchan.cRevision: 7221 app_realtime.cRevision: 7221 app_math.cRevision: 7221 app_forkcdr.c Revision: 7221 app_test.cRevision: 7221 app_verbose.c Revision: 7221 app_userevent.c Revision: 7221 app_alarmreceiver.c Revision: 7221 app_talkdetect.c Revision: 7221 app_controlplayback.c Revision: 7221 app_txtcidname.c Revision: 7221 app_groupcount.c Revision: 7221 app_exec.cRevision: 7221 app_sendtext.cRevision: 7221 app_nbscat.c Revision: 7221 pp_eval.cRevision: 7221 app_ices.cRevision: 7221 app_random.c Revision: 7221 app_setcdruserfield.c Revision: 7221 app_read.cRevision: 7221 app_cut.c Revision: 7497 app_sayunixtime.c Revision: 7221 app_hasnewvoicemail.c Revision: 7608 app_cdr.c Revision: 7221 app_setcidnum.c Revision: 7221 app_transfer.cRevision: 7221 app_enumlookup.c Revision: 7221 app_chanisavail.c Revision: 7221 app_db.c Revision: 7221 app_privacy.c Revision: 7771 app_waitforring.c Revision: 7221 app_lookupblacklist.c Revision: 7221 app_softhangup.c Revision: 7221 app_authenticate.cRevision: 7221 app_macro.c Revision: 7221 app_lookupcidname.c Revision: 7221 app_setcidname.c Revision: 7221 app_parkandannounce.c Revision: 7221 app_senddtmf.cRevision: 7221 app_queue.c Revision: 8445 app_festival.cRevision: 8140 app_setcallerid.c Revision: 7221 app_zapateller.c Revision: 7221 app_milliwatt.c Revision: 8232 app_getcpeid.cRevision: 7221 app_adsiprog.cRevision: 7221 app_disa.cRevision: 7221 app_url.c Revision: 7221 app_image.c Revision: 7221 app_record.c Revision: 7221 app_echo.cRevision: 7221 app_system.c Revision: 7221 app_mp3.c Revision: 7221 app_directory.c Revision: 7221 app_voicemail.c Revision: 7999 app_playback.cRevision: 7221 app_dial.cRevision: 8608 chan_zap.cRevision: 8573 chan_phone.c
RE: [Asterisk-Users] No audio? Update your Asterisk
To check if it's the same problem, set your system clock back 2 weeks. If it gets better, then the upgrade didn't take. If it doesn't get better, it's something else. --Rob -Original Message- Very greatful to find this I have upgraded to 1.2.3 but still have no sip-sip audio! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Thanks! Steve To check if it's the same problem, set your system clock back 2 weeks. If it gets better, then the upgrade didn't take. If it doesn't get better, it's something else. --Rob -Original Message- Very greatful to find this I have upgraded to 1.2.3 but still have no sip-sip audio! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
No Firewalls involved, the test has been simplified down to two sip phones on a LAN and still no audio. For waht it's worth IAX2 still works fine. Steve - Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
Have you tried increasing the debug level and watching the cli? No Firewalls involved, the test has been simplified down to two sip phones on a LAN and still no audio. For waht it's worth IAX2 still works fine. Steve - Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
Yes I have. I have been battling this issue since wednesday 1-25 And so far have tried many things. Have also tried RTP debug and do not see ANY RTP when the call is made. I will keep working at this until I figure it out but right now am very stumped and frusterated. The software update SHOULD have fixed it as it has for many others. Steve Have you tried increasing the debug level and watching the cli? No Firewalls involved, the test has been simplified down to two sip phones on a LAN and still no audio. For waht it's worth IAX2 still works fine. Steve - Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
Steve: I'm picking up the tail end of a thread, so apologies if this is offtrack... Have you perhaps got an old set of EXECUTABLES in your path, that are being picked up before your newly compiled ones? Roger Steve Gladden wrote: Yes I have. I have been battling this issue since wednesday 1-25 And so far have tried many things. Have also tried RTP debug and do not see ANY RTP when the call is made. I will keep working at this until I figure it out but right now am very stumped and frusterated. The software update SHOULD have fixed it as it has for many others. Steve Have you tried increasing the debug level and watching the cli? No Firewalls involved, the test has been simplified down to two sip phones on a LAN and still no audio. For waht it's worth IAX2 still works fine. Steve - Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
Good question, But the answer is no. I have went through the trouble to make sure that all traces of other asterisk libraries/modules, config files excutables are removed from the system before compiling running testing anything. I am also being sure to unload ztdummy zaptel modules before removing the files and recompiling. For good/bad measure I am also completely powering off the system between attemps to ensure the USB hardware being used for timing with our 2.4 kernel is getting reset properly before trying again. Probably not really needed but I'm at the point of desperation and am trying not to leave anything out. Thanks for your time! Steve Steve: I'm picking up the tail end of a thread, so apologies if this is offtrack... Have you perhaps got an old set of EXECUTABLES in your path, that are being picked up before your newly compiled ones? Roger Steve Gladden wrote: Yes I have. I have been battling this issue since wednesday 1-25 And so far have tried many things. Have also tried RTP debug and do not see ANY RTP when the call is made. I will keep working at this until I figure it out but right now am very stumped and frusterated. The software update SHOULD have fixed it as it has for many others. Steve Have you tried increasing the debug level and watching the cli? No Firewalls involved, the test has been simplified down to two sip phones on a LAN and still no audio. For waht it's worth IAX2 still works fine. Steve - Yep, tried that. blew away all my source code, re-downloaded re compiled and re installed. it's behaving exactly the same, calls go through but no audio in either direction for sip-sip calls on the LAN or to-from the Internet SIP providers tested. I'm at a loss I feel like I have tried everything. even stripped down my configs and tried to make them as simple as possible with nothing more than two SIP phones and a default context. I'm running a 2.4 kernel with USB timimg for ztdummy Another interesting note is that I am getting no DTMF decode with PAP2 devices set to AVT. It was working before Jan 25th along with audio before all suddenly quite working. I set my system and hardware clock back to 00:00 Jan, 01 2006 and rebooted the system Anything else I should be checking for? Sounds like maybe a firewall is involved somewhere. Are you sure there are none in the path (including on your asterisk box)? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
Same situation. Asterisk 1.2.1 ([EMAIL PROTECTED] 2.2) apparently doesn't have this problem. Thanks Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Tanner Sent: Wednesday, January 25, 2006 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No audio? Update your Asterisk For what it's worth, I've been messing around with my install all night and haven't had a single issue. [EMAIL PROTECTED] 2.2, Asterisk version 1.2.1. Even set the date ahead, still no problems. Could be a fluke, I'm interested if anyone else is using 1.2.1 and has these issues, but for now I'm sticking with what I have. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
On Wed, 2006-01-25 at 14:10 -0600, Kevin P. Fleming wrote: Aaron Daniel wrote: We had the bug on 1.2.2, but when I rolled back to 1.2.1 to fix the problem, everything started working. Doesn't seem like it's a bug in 1.2.1 :) It is not. The bug was introduced during the 1.2.1-1.2.2 transition. Observation ... Had a problem with asterisk 1.2 trunk from approx 10 Dec, (from memory), when the problem occurred, a restart of asterisk did not fix it, and a reboot of the machine still did not fix it. A svn update and restart of asterisk still did not fix it. I wonder, was it really only 2^22 seconds, or how exactly did that work?? I really was quite stumped when it happened, since nobody had changed anything for ages... Anyway, thankfully it broke at 4:30pm, and everyone went home at 5pm (the after hours announce only and hangup was working), so I left it until the next day. A few debug attempts, and a login to IRC, and suddenly the solution was shouting at me (in the subject/title of the IRC channel to svn update). Mainly was confused as to why a asterisk + zaptel restart and a reboot didn't fix the problem (even just temporarily)? Regards, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
hello, Could you give us the path to the patch quickly ? Harry Gaillac --- Olle E Johansson [EMAIL PROTECTED] a écrit : This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
Olle E Johansson wrote: This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. What versions of Asterisk does this affect? Darren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
hello, Could provide us how to fix this serious bug my server is out of order please to post how to solve quickly this problem . Harry --- Olle E Johansson [EMAIL PROTECTED] a écrit : This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No audio? Update your Asterisk
Olle E Johansson wrote: This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. What versions of Asterisk does this affect? Darren ___ Hopefully not stable ;-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
All versions released in the last 2 weeks i think. take the newest versions from svn or the ftp. (1.2.3 is released). Cheers, Zoa Darren Ellis wrote: Olle E Johansson wrote: This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. What versions of Asterisk does this affect? Darren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
On 1/25/06, Darren Ellis [EMAIL PROTECTED] wrote: Olle E Johansson wrote: This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. What versions of Asterisk does this affect? It affects 1.2 and /trunk versions. You can do an svn update if you've downloaded your src tree from SVN. A tarball of 1.2.3 I'm guessing will be made available later on this morning. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
On 1/25/06, Darren Ellis [EMAIL PROTECTED] wrote: Guys, I'm not familiar enough with mantis to tell what version of asterisk are affected by this bug? I have 1.09, 1.10, 1.2.1 and 1.2.2 (as a test) deployed. Can someone tell me what the real impact is going to be? Thanks 1.2.1 and 1.2.2 are likely affected. I didn't see this code in a pre-1.2 system I reviewed earlier this morning for a client. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
Darren Ellis wrote: Olle E Johansson wrote: What versions of Asterisk does this affect? I've looked at the mydiff.txt on the bug report, it refers to channel.c, I don't find this code that it's trying to update in SVN-trunk-r7498. Hopefully I'm safe, but will need to wait for confirmation from those that know better. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
Zoa wrote: All versions released in the last 2 weeks i think. take the newest versions from svn or the ftp. (1.2.3 is released). Thanks Zoa, /emergency mode off ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
Hi all asterisk users On 1/25/06, Darren Ellis [EMAIL PROTECTED] wrote: Olle E Johansson wrote: This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
FYI, just upgraded from 1.2.2 to 1.2.3 and audio problems in sip channels gone way. Thanks a lot, -- Antonio José dos Santos Brandão Virgos Tecnologia da Informação www.virgos.com.br - São Carlos,SP On 1/25/06, Olle E Johansson [EMAIL PROTECTED] wrote: This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
For what it's worth, I've been messing around with my install all night and haven't had a single issue. [EMAIL PROTECTED] 2.2, Asterisk version 1.2.1. Even set the date ahead, still no problems. Could be a fluke, I'm interested if anyone else is using 1.2.1 and has these issues, but for now I'm sticking with what I have. BTW, all my testing tonight involved SIP (Sipura SPA-2000) and either PSTN using an X100P card, IAX account with voxee, or a SIP account with vbuzzer. Had audio both ways all the time. Joseph Tanner On 1/25/06, BJ Weschke [EMAIL PROTECTED] wrote: On 1/25/06, Darren Ellis [EMAIL PROTECTED] wrote: Guys, I'm not familiar enough with mantis to tell what version of asterisk are affected by this bug? I have 1.09, 1.10, 1.2.1 and 1.2.2 (as a test) deployed. Can someone tell me what the real impact is going to be? Thanks 1.2.1 and 1.2.2 are likely affected. I didn't see this code in a pre-1.2 system I reviewed earlier this morning for a client. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
The patch for channel.c is right here: http://bugs.digium.com/file_download.php?file_id=9099type=bug Michel Belleau SERVICES INFORMATIQUES MALAIWAH.COM (418) 261-6412 -- http://www.malaiwah.com [EMAIL PROTECTED] a écrit : hello, Could you give us the path to the patch quickly ? Harry Gaillac --- Olle E Johansson [EMAIL PROTECTED] a écrit : This morning we discovered a serious bug that stopped all bridged audio in our Asterisk servers. Mark found the problem and soon fixed it. If you get this problem today, please update your Asterisk server. A fix has been commited to the subversion repository for 1.2 as well as trunk. A fixed 1.2.3 release will be published on ftp.digium.com as soon as we can find a release engineer (consider the time zone problem). A big thank you to everyone in the IRC channel that helped us locate this issue and to Mark that fixed it so quickly. /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
For what it's worth, I've been messing around with my install all night and haven't had a single issue. [EMAIL PROTECTED] 2.2, Asterisk version 1.2.1. Even set the date ahead, still no problems. Could be a fluke, I'm interested if anyone else is using 1.2.1 and has these issues, but for now I'm sticking with what I have. Just to expand on this a little, I installed 1.2.2 as soon as it was released, and ran it on a production server quite happily until this morning, when I suddenly lost audio on certain calls. Restarts did not fix the problem, which really confused me. I saw the upgrade to 1.2.3 and installed it right away. Everything was fixed with 1.2.3. On another production server, I've been running 1.2.1 for a few weeks with no problems. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
We had the bug on 1.2.2, but when I rolled back to 1.2.1 to fix the problem, everything started working. Doesn't seem like it's a bug in 1.2.1 :) Aaron Joseph Tanner wrote: For what it's worth, I've been messing around with my install all night and haven't had a single issue. [EMAIL PROTECTED] 2.2, Asterisk version 1.2.1. Even set the date ahead, still no problems. Could be a fluke, I'm interested if anyone else is using 1.2.1 and has these issues, but for now I'm sticking with what I have. BTW, all my testing tonight involved SIP (Sipura SPA-2000) and either PSTN using an X100P card, IAX account with voxee, or a SIP account with vbuzzer. Had audio both ways all the time. Joseph Tanner On 1/25/06, BJ Weschke [EMAIL PROTECTED] wrote: On 1/25/06, Darren Ellis [EMAIL PROTECTED] wrote: Guys, I'm not familiar enough with mantis to tell what version of asterisk are affected by this bug? I have 1.09, 1.10, 1.2.1 and 1.2.2 (as a test) deployed. Can someone tell me what the real impact is going to be? Thanks 1.2.1 and 1.2.2 are likely affected. I didn't see this code in a pre-1.2 system I reviewed earlier this morning for a client. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
Hi! On 1/25/06, Aaron Daniel [EMAIL PROTECTED] wrote: We had the bug on 1.2.2, but when I rolled back to 1.2.1 to fix theproblem, everything started working.Doesn't seem like it's a bug in 1.2.1 :) Our 1,2,1 doesn't seem to be affetcted either... -- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio? Update your Asterisk
Aaron Daniel wrote: We had the bug on 1.2.2, but when I rolled back to 1.2.1 to fix the problem, everything started working. Doesn't seem like it's a bug in 1.2.1 :) It is not. The bug was introduced during the 1.2.1-1.2.2 transition. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
