RE: [Asterisk-Users] RC2 - H323 channel broken
Hi, Can you tell me how you could get the RC1 to work because for me RC1 does not work as well as I got NO audio on both sides. I have use the stable version of asterisk and it works for me with H323 audio, but once I upgraded to RC1 , calls were not completing at all with H323 and I had to revert back. I am using the Native H323 as well. Please help, thanks. TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of administrator tootai Sent: Friday, August 13, 2004 4:17 AM To: Asterisk-Users Subject: [Asterisk-Users] RC2 - H323 channel broken Hi list, I updated few hours ago from RC1 to RC2 (tar.gz) and discover that native H323 (Nufone) is broken, having no audio on both sides. I came back to RC1 and it worked again. Someone else can confirm/infirm -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)
Dear All Now that RC1 is buggy, should we go back to the Dev cvs Head? OR how do we get the bug resolved in RC1 version and where could we get them? Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of hank Sent: Saturday, August 07, 2004 1:02 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio) 1 what ivr system did you use am curious, 2 what is there contact info for support? - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 06, 2004 10:34 PM Subject: Re: [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio) Jeremy McNamara and I spent some time tonight figuring this out. (aside: anyone claiming nufone doesn't have decent customer service is full of shit, who else is gonna spend over 3 hours helping a customer?) The gappy audio problem isn't quite what I first thought: It has nothing to do with IAX2, SIP does it too. It has nothing to do with a specific codec. It has *everything* to do with native bridging. home* -IAX2- colo* -IAX2- switch-1.nufone.net I am using xlite to home* but a zap interface does the same thing, the problem is with the native bridging between colo* and switch-1. switch-1 runs RC1 colo* runs CVS HEAD 20040806 but also ran CVS HEAD 20040604 with the same problem. The gappy audio problem started showing up around the time Nufone upgraded to RC1. home* - colo* GSM and colo* - switch-1 GSM: dead audio (from nufone) 1:02 into the conversation. It stays dead for a good long time but the remote end can hear me (I was calling an IVR and hitting # every 10 seconds or so, it could hear me). home* - colo* GSM and colo* - switch-1 iLBC or ULAW, no dead air. home* - colo* iLBC and colo* - switch-1 iLBC, dead air 1:02 in. home* - colo* iLBC and colo* - switch-1 ULAW, dead air 1:02 in. seeing a pattern? :-) I've privately sent steve my debug log and a pcap dump of a sample conversation. The conversation is about 6 minutes long and the bulk of it is dead air, starting at 1:02. There are a few (maybe 3 or 4) very short (2s) bursts of audio from the remote end but mostly dead air. So it's gappy alright... but very much so. Note that the gappy audio has changed a little as time went on. When we first started seeing it (around the 20th of June or so) it resembled dropped packets or a congested link. Short bursts of silence or garbled audio. The past few days it was getting to be two to three seconds of dead air and now tonight it dies 1:02 in and stays dead for a minute or two, then a burst of audio which quickly garbles (2s) and then silence for another minute or two. The IVR I called through Nufone could hear my # key presses. If I didn't hit anything the remote IVR would hang up after 20s or so. Note that there are no native bridging problems to VoipJet. They run 0.90 apparently. Again: nufone' s been great in helping diagnose this. Highly recommended. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 268 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)
Dear All, I have been using the stable version of the asterisk for these past couple of months, and it has been working okay. However whenever I try to work with CVS Head version, it will get NO audio at all at least from the caller side via H323 (using Jeremy H323 driver) and I have had no such problem with the stable version which does not exist anymore. I read that I should make sure that the code is configured no faststart and I did, but I am still not getting it to work at all. I am using g729 pass through. However with the stable version, it works even with the boolean NOT set to TRUE for faststart. I have this problem with alot of versions on CVS (versions at different dates) and the same problem exists on the RC1, and only the stable version works, what is the difference there? Can anyone help please? Once again, I am using it for passing through and g729 codec, I tried H323 in and H323 out and I tried sip in and H323 out but the same results I get. Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of T. Chan Sent: Saturday, August 07, 2004 4:27 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio) Dear All Now that RC1 is buggy, should we go back to the Dev cvs Head? OR how do we get the bug resolved in RC1 version and where could we get them? Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of hank Sent: Saturday, August 07, 2004 1:02 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio) 1 what ivr system did you use am curious, 2 what is there contact info for support? - Original Message - From: Andrew Kohlsmith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 06, 2004 10:34 PM Subject: Re: [Asterisk-Users] RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio) Jeremy McNamara and I spent some time tonight figuring this out. (aside: anyone claiming nufone doesn't have decent customer service is full of shit, who else is gonna spend over 3 hours helping a customer?) The gappy audio problem isn't quite what I first thought: It has nothing to do with IAX2, SIP does it too. It has nothing to do with a specific codec. It has *everything* to do with native bridging. home* -IAX2- colo* -IAX2- switch-1.nufone.net I am using xlite to home* but a zap interface does the same thing, the problem is with the native bridging between colo* and switch-1. switch-1 runs RC1 colo* runs CVS HEAD 20040806 but also ran CVS HEAD 20040604 with the same problem. The gappy audio problem started showing up around the time Nufone upgraded to RC1. home* - colo* GSM and colo* - switch-1 GSM: dead audio (from nufone) 1:02 into the conversation. It stays dead for a good long time but the remote end can hear me (I was calling an IVR and hitting # every 10 seconds or so, it could hear me). home* - colo* GSM and colo* - switch-1 iLBC or ULAW, no dead air. home* - colo* iLBC and colo* - switch-1 iLBC, dead air 1:02 in. home* - colo* iLBC and colo* - switch-1 ULAW, dead air 1:02 in. seeing a pattern? :-) I've privately sent steve my debug log and a pcap dump of a sample conversation. The conversation is about 6 minutes long and the bulk of it is dead air, starting at 1:02. There are a few (maybe 3 or 4) very short (2s) bursts of audio from the remote end but mostly dead air. So it's gappy alright... but very much so. Note that the gappy audio has changed a little as time went on. When we first started seeing it (around the 20th of June or so) it resembled dropped packets or a congested link. Short bursts of silence or garbled audio. The past few days it was getting to be two to three seconds of dead air and now tonight it dies 1:02 in and stays dead for a minute or two, then a burst of audio which quickly garbles (2s) and then silence for another minute or two. The IVR I called through Nufone could hear my # key presses. If I didn't hit anything the remote IVR would hang up after 20s or so. Note that there are no native bridging problems to VoipJet. They run 0.90 apparently. Again: nufone' s been great in helping diagnose this. Highly recommended. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter 268 spam mails have been blocked so far. Download free www.spamfighter.com today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http
RE: [Asterisk-Users] Asterisk scalability?
Hi, Scott Thanks for your information. I have worse luck in load testing with asterisk. I have tried both SIP and H323 inbound calls and terminating on PSTN PRIs. I am using a single Xeon 2.8G chip and 512M Ram and in both cases, once it gets more than a T1, call quality starts to degrade with choppiness, and Asterisk becomes very unstable and resets itself like every 5-15 minutes. Can you let me know more about your tests, like which version of Asterisk are you using for the test, and which version of H323 and your computer configuration please, thanks a million TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Saturday, July 31, 2004 2:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk scalability? Hi Roy- I've done a lot of load testing with asterisk and TE410P's. My guess, with no transcoding, is that you might be able to handle 8 E1's max on the PSTN side absolute max (ie: 2 TE410P's). This assumes you have a fast processor.If you're using T1's, scale these numbers up accordingly, as there are fewer channels per span. If this answer is lower than you might expect, consider that every byte of data has to pass through the processor. The 410's are capable of bus-mastering, and so are an improvement over the T400P's, but still I think you run into horsepower limitations. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Saturday, July 31, 2004 8:25 AM To: Asterisk Users Subject: [Asterisk-Users] Asterisk scalability? Hi I plan to setup an asterisk box to function as a SIP gateway forwarding lots of calls to/from a backend of several other asterisk boxes, each with a TE410 card for PSTN connectivity. It will only gateway the calls into the PSTN gateways. No transcoding is planned - only plain ALAW. How many concurrent calls would you think this can handle? I'm asked to plan a system that can handle 1000 concurrent calls... thanks for any input regards roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date: 7/29/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 channel
Dear All, There is a question about the H323 channels (H323 driver, not OH323), it is not passing CallerID. If a call comes in on ZAP and out H323 to another gateway, the other gateway does not see the ANI, and if Asterisk is used as a passthrough, it receives callerID from the other gateway, but when sent out to the terminating endpoint, the terminating endpoint is not seeing it. Is there anywhere we should configure so that it is passing the callerID to a terminating endpoint on the outbound H323 channels. Please enlighten me, thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris A. Icide Sent: Tuesday, July 06, 2004 6:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] H323 channel On 03:23 AM 7/6/2004, administrator tootai wrote: Hello everybody, my * box is connected to gnugk with H323 channel. If I call from an H323 EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio start but noisy (scratch) , then became ok for callee (SIP EP) but still scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323 EP and it's ok. And from now, it's also ok when H323 EP call SIP one's! No need to say that H323-H323 is working, as well as SIP-SIP. Running CVS version from yesterday. Used codecs are G711U A, G723.1 and G729. If I just use G711 it's the same. SIP EP has to call first when * is started to make it work. Any hint? Also, H323 is still broken and working without FastStart. Is there a workaround existing? Just to help troubleshooting, which h323 implementation for asterisk did you use? -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.713 / Virus Database: 469 - Release Date: 6/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.713 / Virus Database: 469 - Release Date: 6/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] strange problem with oh323 loaded!
Hi, I am the unlucky one, I have similar problem, but I am mostly using safe_asterisk, and this stop now...restart now never works, with neither 0.6.3 nor 0.6.2 TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anthony Law Sent: Thursday, July 08, 2004 3:33 PM To: Mailing List Asterisk Subject: [Asterisk-Users] strange problem with oh323 loaded! Hi, As explained in my original post on June 30. When I used CVS 2004-06-16 with oh323-0.6.3a. I can compile and install without problem but when I am in the asterisk console whenever I issue stop now or restart now or extension reload I got stuck on the console and asterisk did not response to either shutting down or restarting. It stucked on Executing last minute cleanups == Cleaning up OpenH323 channel driver. == Unregistered channel type 'OH323' == Destroying any remaining musiconhold processes The same thing will not happen if I do not load the oh323-0.6.3.a module. Since I have this problem I have gone back to oh323-0.6.3 and it acts the same, finally yesterday I revert it back to oh323-0.6.2a and the above did not happen. Do you happen to know why? Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.713 / Virus Database: 469 - Release Date: 6/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.713 / Virus Database: 469 - Release Date: 6/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] strange problem with oh323 loaded!
Dear All, I don't know but I tried all 0.6.x version of OH323 and normally I use safe_asterisk to start asterisk, and everytime when I use 'stop now' to terminate asterisk, it does not do anything, and you are rite, I have to use kill -9 to kill the PIDs and threads. However, if I use asterisk -vvvgc to start, 'stop now' works. Thanks all TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Friday, July 02, 2004 4:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] strange problem with oh323 loaded! Same problem here - with latest 0.6.3a oh323. Locks up on exit. Had to kill -9 This didn't happen with 0.6.2a, but that's on a different machine. Maybe you could try this older version which worked fine (same PwLib and OpenH323) Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Law Sent: Friday, July 02, 2004 1:15 PM To: Mailing List Asterisk Subject: [Asterisk-Users] strange problem with oh323 loaded! Hi, Here it is when I start it with /etc/rc.d/init.d/asteriskd found in asterisk source contrib/init.d/rc.redhat.asterisk It started without problem and when i issue stop now It freezes, please see below, tai*CLI add debug dontdumpextensions helpiax2 include init loadlocal logger mgcpno oh323 reload remove save set showsip skinny softunload == Registered channel type 'OH323' (OpenH323 Channel Driver) == OpenH323 Channel Ready (v0.6.3) == Parsing '/etc/asterisk/enum.conf': == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/logger.conf': == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger restarted Asterisk Ready. tai*CLI stop now tai*CLI It freezes right here and does nothing else - If I do it with safe_asterisk , it died and loops [EMAIL PROTECTED] init.d]# /usr/sbin/safe_asterisk -vvc [EMAIL PROTECTED] init.d]# Asterisk ended with exit status 127 Asterisk died with code 127. Automatically restarting Asterisk. Asterisk ended with exit status 127 Asterisk died with code 127. Automatically restarting Asterisk. Asterisk ended with exit status 127 As I have mentioned, if I noload oh323 this won't happen *CLI stop now Beginning asterisk shutdown Executing last minute cleanups == Destroying any remaining musiconhold processes Asterisk cleanly ending (0). Any ideas? Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.713 / Virus Database: 469 - Release Date: 6/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.713 / Virus Database: 469 - Release Date: 6/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] strange problem with oh323 loaded!
Hi, Anthony, can you try issuing stop now on safe_asterisk and see if it works please? I am used to using safe_asterisk and with this new version and when I tried issuing stop now, it did not do it. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anthony Law Sent: Wednesday, June 30, 2004 4:17 PM To: Mailing List Asterisk Subject: [Asterisk-Users] strange problem with oh323 loaded! Hi, I am using asterisk CVS 2004-06-16 with oh323-0.6.3a I have a strange problem if I start asterisk with oh323 loaded /usr/sbin/asterisk -vc once I am in the console and issue restart now or reload asterisk hangs and it not stoping or restarting at all, below is the console logging when it happens, as you can see it stucks on Destroying any remaining musiconhold processes [chan_oh323.so] = (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.13.5, PWlib v1.6.6 == Registered channel type 'OH323' (OpenH323 Channel Driver) == OpenH323 Channel Ready (v0.6.3) == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger restarted Asterisk Ready. *CLI restart now Beginning asterisk restart Executing last minute cleanups == Cleaning up OpenH323 channel driver. == Unregistered channel type 'OH323' == Destroying any remaining musiconhold processes If I do not load oh323 the above will not happen. Does anyone knows how to why or how to fix? Even if I use safe_asterisk it acts the same. Is this a problem with oh323 or asterisk itself? Regards, Anthony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk addon mysql
cvs checkout -D mm/dd/yy asterisk-addons -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: Monday, June 28, 2004 1:03 AM To: [EMAIL PROTECTED]; T. Chan Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] asterisk addon mysql Tommy, Thanks, how do i get the older version of asterisk-addons? -- Harold Workman Quoting T. Chan [EMAIL PROTECTED]: Hi, I got the same thing, so what I did was for the asterisk-addons, I used CVS April instead of the most current CVS and it worked. Of course, I would have liked to use the most current CVS of asterisk-addons as well, but since the old version works with the most current version of asterisk anyways, I left it like that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: Sunday, June 27, 2004 3:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk addon mysql hi, ive read through the last few posts with people having problems compiling the asterisk-addons for mysql support, and none of them have helped me resolve my compile problem. I currently have -- CVS-06/24/04-22:20:31 and downloaded asterisk-addons. I compiled * first then asterisk-addons, have added CFLAGS+=-I../asterisk/include When I try to make install for asterisk-addons i get [EMAIL PROTECTED] asterisk-addons]# make clean ; make install rm -f *.so *.o .depend cc -fPIC -I../asterisk -D_GNU_SOURCE -I../asterisk/include -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function decla ration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 I have MySQL-server and devel upgraded at version 4.0.20 on a Fedora Core 1. I would really love to have mysql support Harold Workman This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk addon mysql
Hi, I got the same thing, so what I did was for the asterisk-addons, I used CVS April instead of the most current CVS and it worked. Of course, I would have liked to use the most current CVS of asterisk-addons as well, but since the old version works with the most current version of asterisk anyways, I left it like that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Harold Workman Sent: Sunday, June 27, 2004 3:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk addon mysql hi, ive read through the last few posts with people having problems compiling the asterisk-addons for mysql support, and none of them have helped me resolve my compile problem. I currently have -- CVS-06/24/04-22:20:31 and downloaded asterisk-addons. I compiled * first then asterisk-addons, have added CFLAGS+=-I../asterisk/include When I try to make install for asterisk-addons i get [EMAIL PROTECTED] asterisk-addons]# make clean ; make install rm -f *.so *.o .depend cc -fPIC -I../asterisk -D_GNU_SOURCE -I../asterisk/include -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function decla ration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 I have MySQL-server and devel upgraded at version 4.0.20 on a Fedora Core 1. I would really love to have mysql support Harold Workman This message was sent using IMP, the Internet Messaging Program. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 Audio problem UPDATE
Hi, Scott. Are you telling me that this native h.323 has been hardcoded with fast start? Can you tell me where in ast_h323.cpp it is that you disabled this faststart? Have you tried using the Stable cvs of the Asterisk. Can you let me know which version of the OH323 are you using ? Is it the 0.6.2A? Which version of the Pwlib and OpenH323 you used, is it the newest version as stated? Did you apply the patch? I tried using this driver, but I have problem with cdr_mysql, it is not recording cdr. Please share your information, thanks alot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Sunday, June 27, 2004 6:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H.323 Audio problem UPDATE Update on this problem: I gave up on the native h.323 because, like others, I couldn't get audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no change) So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say that everything seems to work so far. Not only does audio work, but even the handshaking is now working in both OpenPhone and even NetMeeting (for the first time). Notes to others who want to try OH323: * The installation is a bit more complicated than h323. Follow the instructions in the ReadMe file exactly. * You must choose and install the proper versions of PWLib and OpenH323, as stated. * Don't forget to edit the Makefile as stated. Some load testing to following this week, but I'm encouraged! Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 Audio problem UPDATE
Hi, Scott, I am very interested in knowing the result of your loading test, please share after you have done it. Are you using Asterisk as a pass-through (kinda softswitch) or do have have digium hardware and use it as an endpoint, because I believe the maxiumum number of channels you can run stably could be different, please share, thanks. TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Sunday, June 27, 2004 9:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] H.323 Audio problem UPDATE Scott Stingel wrote: Some load testing to following this week, but I'm encouraged! This is where you are going to be discouraged with that other H.323 driver. I guarantee it. Disabling fast-start has solved the problems for quite a few other ppl using 5300s, so you must be doing something really nasty with them to still not work. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS
Hi, Jeremy, thanks for your help and dedication in resolving the problem. There must be something that could have caused the problem. Why don't I provide detailed information on what hardware I use and how I installed the Asterisk and I would suggest that other colleagues who had or are having this problem might want to do the same in order for Jeremy to help us. I have tried two different hardware configuration with the same result. The first Asterisk server I use a Pentium Xeon 2.4G with 512M Ram without any digium card, I use Redhat 7.3 with Kernel upgraded to 2.4.20-28.7smp, ie. enabling Hyperthreading. The second Asterisk server, I use a Pentium4 3.0G with 512M Ram with same OS version and Kernel version. I read somewhere that the system should be more stable without hyperthreading, I have tried using 2.4.20-28.7 Kernel but do not find any difference in terms of stability nor voice quality at all. I have tried many many times the following steps on both servers. 1. Get pwlib 1.5.2 and openh323 1.12.2 (ones as suggested by Jeremy) and under pwlib, do ./configure, make clean, and then make both (I even tried doing just a make opt here), and then openh323, do ./configure, make clean, and then make opt. 1. Obtain asterisk, libpri, zaptel (although I don't need without digium card) from cvs development head by doing CVS checkout asterisk libpri zaptel. Everytime when I do this step, I will erase old directories to make sure I have everything cleaned. 2. Do, make clean and make install on all directories, except that for asterisk directory, I will go in ../asterisk/channels/h323 and do a make clean and then make (without the install) before doing a make install under the asterisk directory. 3. Asterisk ready. I tried calling from another Asterisk running a January cvs into one of these servers and out to cisco (or quintum or yet another Asterisk with digium), but I got no audio on both servers. I tried calling from SJPhone into one of these servers and out to cisco or quintum or another Asterisk with digium and same thing with no audio both ends on both servers. I tried calling from cisco, passing through one of these servers and out to another cisco or endpoints above, same thing. No matter what I did, there was just no audio on both ends at all. Now, I have kept everything the same except that I changed to the stable CVS and did the same thing as described above, and I could get audio now. Jeremy, I hope that would give you some idea if I did anything wrong, and probably most colleagues out there were doing similar to what I did and have unknowingly made some mistakes somewhere. By the way, I was using a h323.conf that is practically the same as your sample. Meantime, can you tell us if you will be incorporating options like fast start, h245 tunneling, early alerting...into the driver? Thanks, and I hope you can resolve this as soon as possible, thanks again for your support. TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Saturday, June 26, 2004 4:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS T. Chan wrote: Jeremy, any way to fix that? Thanks again. I've spent many many days trying to duplicate any of these problems and absolutely cannot. I have tried everything from my mini-itx to my celeron based laptop to my dual xeon dell 1750s and every single one of them work 100% successfully in both directions with the cvs -head and chan_h323. I've also very successfully tested interop with 5300s, Quintium A800, some multi-tech box someone in IRC let me push a few calls thru (sorry forgot your nick) and even my 7960 here now runs the H.323 load and then OpenPhone works perfectly... I simply cannot duplicate any such problems. I even manage a few different production systems with 5300s and they are running absolutely perfectly on asterisk cvs -head with chan_h323. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS
Dear All, I second that, I have spent months on this. Since a few months ago when the recommended H323 libraries were changed to pwlib 1.5.2 and OpenH323 12.2.2, I have not been able to get audio. I tried different configurations and no luck. I have spent months and lots of hours on it, but I was not able to get it to work properly and was forced to stay with January CVS. I have followed this suggestion and tried the stable cvs head instead of the dev cvs, and bingo, I got audio. I have written in this forum at least 3 or 4 times asking for advices, but there has been no response, Jeremy has responded once but his advice was to look at the h323.conf.sample which was not too helpful. I thank Hekuran for this. You made my day. If Jeremy reads this, please when convenient, let us know why there would be difference when using the two different branches and let us know if it will be changed on the dev cvs, thanks Jeremy. Meantime, I would also like to take this opportunity to try raising this issue again which I have raised in this forum at least 4 times without any response. This version of H323 driver does not seem to pass CallerID, meaning that if an asterisk server receives the call (either via zap or VOIP), it will get the CallerID (depends on the whether the originating interface is sending of course), but when it sends the call out via H323 with this driver to other servers, whether it be cisco, asterisk, quintumor other softswitches, it will NOT pass the callerid, the other H323 driver by Michael is passing Callerid very reliably. Jeremy, any way to fix that? Thanks again. TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hekuran Doli Sent: Friday, June 25, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS After about 8 hours trying to fix this issue it succeded after compiling the stable version of asterisk. Its working perfect now. I would sujest all of you having this problem to compile the sable version of asterisk. I think there is a problem with the latest cvs version of asterisk. Best regards Hekuran Doli - I'm here with you on this one. I've not been able to figure this out - I triple quadruple checked that I have the right versions of pwlib openh323, I've followed all recommendations in the README, yet I still do not have audio in both directions. I'm also using a cisco 5300, there is no firewall. Tcpdump has revealed the following: when calls are made from the 5300 to asterisk, the 5300 sends continual udp packets, but asterisk doesn't seem to be responding. when calls are made from asterisk to the 5300, no udp packets are sent. It should be noted that when the calls are made using sip, everything works just fine. -g On Fri, 2004-06-25 at 10:54, Sebastian Nocetti wrote: hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS
HI, Scott, I am sorry to hear that you are not able to use the stable version. I really hope that Jeremy could solve the problem with the development cvs. Meantime, I would like to point out that faststart or not, this problem exists. I have used fast start, slow start , early h245 alerting, disabled early h245 alerting, h245 tunneling, disabled h245 tunneling, calling from cisco to asterisk and it just does not work. Now, it is working on this cvs. I am certain that Jeremy should be able to resolve this which will make everyone happy. Meantime, I wonder if this H323 driver has options with fast start, slow start, h245 tunneling, early alerting, and if Jeremy can incorporate this as soon as possible, and more importantly, please investigate into the passing of callerid which is important. Meantime, Scott, please try out the other version of Oh323 driver too, I tried that, and it is not bad, it is passing callerid and all, and have faststartoptions. More importantly, Michael is a really nice guy and he gives excellent support. Thanks, cheers TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Friday, June 25, 2004 11:48 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS I just posted a note to bug 1334, telling Jeremy that lots of people were having problems, and he responded that using quickstart might be the problem (he bases this on one earlier post). I can't dictate to my customer how to initiate calls from the Cisco, so I'm stuck too. Using stable CVS head is not an option for me - too many other bugs that effect me. Thanks - we'll see what Jeremy comes up with. I think I'll try the other version of H.323, perhaps I'll have better luck. Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Chan Sent: Friday, June 25, 2004 8:35 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS Dear All, I second that, I have spent months on this. Since a few months ago when the recommended H323 libraries were changed to pwlib 1.5.2 and OpenH323 12.2.2, I have not been able to get audio. I tried different configurations and no luck. I have spent months and lots of hours on it, but I was not able to get it to work properly and was forced to stay with January CVS. I have followed this suggestion and tried the stable cvs head instead of the dev cvs, and bingo, I got audio. I have written in this forum at least 3 or 4 times asking for advices, but there has been no response, Jeremy has responded once but his advice was to look at the h323.conf.sample which was not too helpful. I thank Hekuran for this. You made my day. If Jeremy reads this, please when convenient, let us know why there would be difference when using the two different branches and let us know if it will be changed on the dev cvs, thanks Jeremy. Meantime, I would also like to take this opportunity to try raising this issue again which I have raised in this forum at least 4 times without any response. This version of H323 driver does not seem to pass CallerID, meaning that if an asterisk server receives the call (either via zap or VOIP), it will get the CallerID (depends on the whether the originating interface is sending of course), but when it sends the call out via H323 with this driver to other servers, whether it be cisco, asterisk, quintumor other softswitches, it will NOT pass the callerid, the other H323 driver by Michael is passing Callerid very reliably. Jeremy, any way to fix that? Thanks again. TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hekuran Doli Sent: Friday, June 25, 2004 5:33 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS After about 8 hours trying to fix this issue it succeded after compiling the stable version of asterisk. Its working perfect now. I would sujest all of you having this problem to compile the sable version of asterisk. I think there is a problem with the latest cvs version of asterisk. Best regards Hekuran Doli - I'm here with you on this one. I've not been able to figure this out - I triple quadruple checked that I have the right versions of pwlib openh323, I've followed all recommendations in the README, yet I still do not have audio in both directions. I'm also using a cisco 5300, there is no firewall. Tcpdump has revealed the following: when calls are made from the 5300 to asterisk, the 5300 sends continual udp packets, but asterisk doesn't seem to be responding. when calls are made from asterisk to the 5300, no udp packets are sent. It should be noted that when the calls are made using sip, everything works just fine. -g On Fri, 2004-06-25 at 10:54, Sebastian Nocetti wrote: hello all, I am
RE: [Asterisk-Users] oh323
Jeremy I speak for myself, I have been testing with oh323 driver as well, because in my case, your h323 driver is not working, it was working before, but then when I started to upgrade to 0.7.0 version of asterisk and from that point onwards (beginning of January), calls have had no audio. I tried making calls and I was getting no audio at all when the call was connected. Since then, I have not been able to upgrade the asterisk version, because if so, I would not be able to run h323. That is why in my case, I have been trying to explore the other alternative. If you have some idea to it, please let me know, thanks alot TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Thursday, June 17, 2004 10:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 Michael M. Saunders wrote: Can I just pay you to fix it for me. I cant see anywhere where I use the debug Why do you see a need to run a 3rd party channel driver? Asterisk has native H.323 support. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] oh323
Jeremy, Yes, I felt that it was important to report my trouble and I did it three times, reporting to the asterisk community, but for some reasons, I was not being responded to at all. I thought my messages were embedded among the hundreds of them and were missed out or everyone was having the same problem and was not able to help. Jeremy, I have followed all instructions of yours by compiling the correct verson of pwlib and openh323 (by doing make clean opt under each directory), I have then gone into H323 and done a 'make' before going back to /usr/src/asterisk to do a 'make install'. I tried using sjphone, I tried using another asterisk, I tried using cisco to call into it, but I just was not able to get any audio at all, when using the old version, I was able to do so no problem with all the equipment above. Jeremy, I don't know if there is any change on the h323.conf or any other file that I need to do, please let me know, because I have not changed any configuration files. Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Friday, June 18, 2004 3:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 T. Chan wrote: Jeremy I speak for myself, I have been testing with oh323 driver as well, because in my case, your h323 driver is not working, it was working before, but then when I started to upgrade to 0.7.0 version of asterisk and from that point onwards (beginning of January), calls have had no audio. I tried making calls and I was getting no audio at all when the call was connected. Since then, I have not been able to upgrade the asterisk version, because if so, I would not be able to run h323. That is why in my case, I have been trying to explore the other alternative. If you have some idea to it, please let me know, thanks alot So you didn't feel it was important to report your trouble anywhere? I have tested the cvs -head of asterisk with many different types of H.323 gateways and cannot make it fail. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] oh323
Jeremy I did not report that to the bug tracker, I did not even think that was a bug, I just thought may be I did something wrong, and I reported my problem 3 times to this mailing list, trying to get some light to my problem, I did not get any response. This time, at least I got some response, but I don't think it helps much. May be that is why the other gentlemen Michael was trying the other driver as well. Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Friday, June 18, 2004 11:40 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 T. Chan wrote: Jeremy, Yes, I felt that it was important to report my trouble and I did it three times, reporting to the asterisk community, but for some reasons, I was not being responded to at all. I thought my messages were embedded among the hundreds of them and were missed out or everyone was having the same problem and was not able to help. Ok...What bug number? I haven't paid very close attention to Mantis, but I thought I had it setup to email me when someone assigned a bug to me. Jeremy, I have followed all instructions of yours by compiling the correct verson of pwlib and openh323 (by doing make clean opt under each directory), I have then gone into H323 and done a 'make' before going back to /usr/src/asterisk to do a 'make install'. I tried using sjphone, I tried using another asterisk, I tried using cisco to call into it, but I just was not able to get any audio at all, when using the old version, I was able to do so no problem with all the equipment above. I just tried sjphone and chan_h323 and it worked on the very first call. cvs -head. Jeremy, I don't know if there is any change on the h323.conf or any other file that I need to do, please let me know, because I have not changed any configuration files. Look at the h323.conf.sample Jeremy McNamara -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Friday, June 18, 2004 3:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 T. Chan wrote: Jeremy I speak for myself, I have been testing with oh323 driver as well, because in my case, your h323 driver is not working, it was working before, but then when I started to upgrade to 0.7.0 version of asterisk and from that point onwards (beginning of January), calls have had no audio. I tried making calls and I was getting no audio at all when the call was connected. Since then, I have not been able to upgrade the asterisk version, because if so, I would not be able to run h323. That is why in my case, I have been trying to explore the other alternative. If you have some idea to it, please let me know, thanks alot So you didn't feel it was important to report your trouble anywhere? I have tested the cvs -head of asterisk with many different types of H.323 gateways and cannot make it fail. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_h323 no audio both ways
Hi Glen, I have had the same problem for quite awhile, since around February, all cvs codes that I have tried, and with h323, I have been getting no audio. I am forced to stay with mid-Jan version of the cvs because of this. I tried using ulaw, g729, but same results, I have in a few occasions dropped a few lines here to ask for advice, but no response, may be we could try to exchange some ideas. Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Monday, June 14, 2004 6:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_h323 no audio both ways I've compiled chan_h323 with the latest cvs code, but my calls don't pass audio. The call connects just fine, as there are no errors reported on either side, nor in a traffic examination with ethereal. I've tried the following: voip phone - asterisk - asterisk - voip phone voip phone - asterisk - asterisk zap - asterisk - asterisk zap - asterisk - cisco cisco - asterisk I'm using ulaw on all connections. Any clues, ideas, or directions would be appreciated. Thanks, Glen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Canadian DID
Hi, I am dealing with Vontec Communication in Vancouver, they are selling DID and DID usages terminating on your equipment via VOIP. You can contact [EMAIL PROTECTED] TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Linus Surguy Sent: Monday, June 14, 2004 8:53 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Canadian DID Can anyone point me in the direction of a wholesaler of Canadian DID numbers? If they'd be interested in trading them for UK numbering that would be even better! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] oh323 0.6.2
Hi, I had the same problem, this is when I tried to run the latest CVS head. I had to redownload the asterisk version dated 8June to compile. I don't know if this is a bug with asterisk or if this is a compatible issue. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael M. Saunders Sent: Friday, June 11, 2004 5:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] oh323 0.6.2 I am trying to compile the latest channel drivers Can anyone tell me what is wrong ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/root/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1385: too few arguments to function `ast_queue_control' chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1417: too few arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_read': chan_oh323.c:1855: too few arguments to function `ast_dsp_process' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Error 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: H323
Thanks, Andy. I have thus tried to use the other H323 driver written by Michael, I have used the newest PWLIB and OPENH323 libraries and newest OH323 driver. After installing, I was able to get two way audio and all. I have tried this driver before but at the time, there was a false answer supervision problem and I had to abandon it. Now, it seems that this problem has been resolved. However, now I have another problem, I have always configured to write the cdr on MYSQL. However, now with this driver, I tested inbound sip , outbound sip, no problem with MYSQL, I tested inbound sip, and outbound OH323, cdr has been written onto MYSQL, but when I used inbound OH323 and outbound whatever, then CDRs have NOT been written onto MYSQL. Somehow, after using OH323, cdr is not being written onto MYSQL. Please help, Michael, do you know why please? Thanks TC -Original Message- From: Rechenberg, Andrew [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 01, 2004 5:45 PM To: [EMAIL PROTECTED] Cc: T. Chan Subject: RE: [Asterisk-Users] RE: H323 I am having a similar problem with one-way audio from an Avaya hardphone calling a SIP soft phone. Audio from the hardphone is heard on the receiving end (SIP), but audio is not heard on the hardphone. I know audio is being injected into the sound card and being processed by the SIP client (I've tried both X-Lite and Windows Messenger 4.7.2009) because the audio meters show signal coming into the client however nothing is heard on the other end. I am using the following: CVS-HEAD 5/21/04 Pwlib-1.5.2 Openh323-1.12.2 Regards, Andy. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Chan Sent: Tuesday, June 01, 2004 1:25 PM To: Dmitry Mishchenko; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: H323 Dear All, Thanks, but I was already using a pre May 25 CVS version. Does anyone else have any idea please? Thanks TC -Original Message- From: Dmitry Mishchenko [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 01, 2004 6:22 AM To: [EMAIL PROTECTED]; T. Chan Subject: Re: [Asterisk-Users] RE: H323 On Tuesday 01 June 2004 00:56, T. Chan wrote: Dear All, I have used Asterisk for a few months and I have been using a January CVS version, it has been working but has been regularly crashing. I use Asterisk mostly as a softswitch application receiving H323 calls from customers and send to H323 carriers. Since I have been using an older CVS version, the OpenH323 and Pwlib libraries in use have been 1.11.7 and 1.4.11 respectively. I was thinking of using a current asterisk version and see if it is more stable comparing to the version in use. I upgraded to new version, and I understand that with the new version and the H323 code, I need to use the 1.12.2 and 1.5.2 versions of the OpenH323 and Pwlib libraries respectively. I have, therefore, erased the whole Pwlib and Openh323 folders, recreated with the new versions and did the ./configure.make clean.make opt procedures to compile the drivers. I have then compiled all the zaptel, libpri, asterisk as usual, but when I ran the asterisk, I noticed that most calls (calls mostly were sent from Cisco AS5300 and Cisco AS5350) were getting one way audio, the calling party was not able to hear anything even the call was connected, I am not sure if the called party would hear anything, but obviously something is not working properly. I have not exactly the same but rather similar issue with the latest cvs-head. There are recent changes in call of on_start_logical_channel() After moving it to MyH323_ExternalRTPChannel::OnReceivedAckPDU it stopped being called in my configuration. As a result I don't get any audio after call established. And with older approach when on_start_logical_channel was called at MyH323Connection::OnStartLogicalChannel it was working fine. This change was done on May 25 so you may try to use older code from CVS before this date. Jeremy saying the latest version approach is fine, but its not working for me :(. Dmitry Can any of your experts out there help please, thanks? TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004
RE: [Asterisk-Users] Difference between native and 3rd party h323 channel driver ?
Dear Michael I tried using the newest version of your H323 driver, but somehow it seems that it is not hanging up the channels and for some reasons, it is NOT writing my cdr to the mysql database, it was writing properly before. As you can see , the call finished at 2:40:12 but refused to hang up properly until timing out 22 seconds later, please help Jun 2 02:40:12 DEBUG[135181]: chan_oh323.c:2014 oh323_write: OH323/R4096: Pushed 10 bytes into smoother... Jun 2 02:40:12 DEBUG[135181]: channel.c:2560 ast_channel_bridge: Didn't get a frame from channel: OH323/R4096 Jun 2 02:40:12 DEBUG[135181]: channel.c:2630 ast_channel_bridge: Bridge stops bridging channels OH323/R4096 and OH323/L24947 Jun 2 02:40:34 ERROR[135181]: chan_oh323.c:1454 oh323_hangup: OH323/L24947: Failed to hangup channel (timeout). -- Hungup 'OH323/L24947' == Spawn extension (inboundh323, 12124445000, 4) exited non-zero on 'OH323/R4096' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos Sent: Tuesday, June 01, 2004 1:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Difference between native and 3rd party h323 channel driver ? Robert Rozman wrote: Hi, I'm trying to compile h323 channel driver on cvs Asterisk 1.0 but no success (I get a lot of errors - related to pwlib library). I read in docs that there is also 3rd party h323 channel driver (somehow both even share protion of code?). Asterisk-oh323 was the first H.323 channel driver for Asterisk. The included one is a fork of it, which followed a different approach in the internal design and implementation. Currently, both are following totally independent roadmaps. I wonder what are pros and cons of both drivers ? Should I try to compile native driver ? Some features of asterisk-oh323 (OH323 driver): - Jitter buffer (static or dynamic, with configurable limits). - Configurable number of voice frames per RTP packet. - Inbound call rate limiter (experimental, needs more testing). - Configurable limits for inbound, outbound, simultaneous calls at any given time. - RTCP report generation and handling. Normally, you try both of them and keep the one that makes you happy. Thanks in advance, Robert. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: H323
Dear All, Thanks, but I was already using a pre May 25 CVS version. Does anyone else have any idea please? Thanks TC -Original Message- From: Dmitry Mishchenko [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 01, 2004 6:22 AM To: [EMAIL PROTECTED]; T. Chan Subject: Re: [Asterisk-Users] RE: H323 On Tuesday 01 June 2004 00:56, T. Chan wrote: Dear All, I have used Asterisk for a few months and I have been using a January CVS version, it has been working but has been regularly crashing. I use Asterisk mostly as a softswitch application receiving H323 calls from customers and send to H323 carriers. Since I have been using an older CVS version, the OpenH323 and Pwlib libraries in use have been 1.11.7 and 1.4.11 respectively. I was thinking of using a current asterisk version and see if it is more stable comparing to the version in use. I upgraded to new version, and I understand that with the new version and the H323 code, I need to use the 1.12.2 and 1.5.2 versions of the OpenH323 and Pwlib libraries respectively. I have, therefore, erased the whole Pwlib and Openh323 folders, recreated with the new versions and did the ./configure.make clean.make opt procedures to compile the drivers. I have then compiled all the zaptel, libpri, asterisk as usual, but when I ran the asterisk, I noticed that most calls (calls mostly were sent from Cisco AS5300 and Cisco AS5350) were getting one way audio, the calling party was not able to hear anything even the call was connected, I am not sure if the called party would hear anything, but obviously something is not working properly. I have not exactly the same but rather similar issue with the latest cvs-head. There are recent changes in call of on_start_logical_channel() After moving it to MyH323_ExternalRTPChannel::OnReceivedAckPDU it stopped being called in my configuration. As a result I don't get any audio after call established. And with older approach when on_start_logical_channel was called at MyH323Connection::OnStartLogicalChannel it was working fine. This change was done on May 25 so you may try to use older code from CVS before this date. Jeremy saying the latest version approach is fine, but its not working for me :(. Dmitry Can any of your experts out there help please, thanks? TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: H323
Dear All, I have used Asterisk for a few months and I have been using a January CVS version, it has been working but has been regularly crashing. I use Asterisk mostly as a softswitch application receiving H323 calls from customers and send to H323 carriers. Since I have been using an older CVS version, the OpenH323 and Pwlib libraries in use have been 1.11.7 and 1.4.11 respectively. I was thinking of using a current asterisk version and see if it is more stable comparing to the version in use. I upgraded to new version, and I understand that with the new version and the H323 code, I need to use the 1.12.2 and 1.5.2 versions of the OpenH323 and Pwlib libraries respectively. I have, therefore, erased the whole Pwlib and Openh323 folders, recreated with the new versions and did the ./configure.make clean.make opt procedures to compile the drivers. I have then compiled all the zaptel, libpri, asterisk as usual, but when I ran the asterisk, I noticed that most calls (calls mostly were sent from Cisco AS5300 and Cisco AS5350) were getting one way audio, the calling party was not able to hear anything even the call was connected, I am not sure if the called party would hear anything, but obviously something is not working properly. Can any of your experts out there help please, thanks? TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: FreeBSD or Linux
Dear All, I would like to install Asterisk to support my VOIP business, intending to use Asterisk as a VOIP softswitch and/or gateways endpoints. I am considering using either FreeBSD or Linux Redhat. Can someone share the experience as to which OS would provide a better environment for running Asterisk, in terms of running more number of calls more stably. Thank you all. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-oh323, new version 0.5.10
Michael Thanks alot, so native bridging will not be something that you would do anytime soon, eh? Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos Sent: Friday, March 12, 2004 7:51 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk-oh323, new version 0.5.10 T.38 FAX is in the short-term plans for asterisk-oh323. Michael T. Chan wrote: Dear Michael Do you foresee implementing these in the near future, one or the other or both? Thanks Tc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos Sent: Thursday, March 11, 2004 4:49 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk-oh323, new version 0.5.10 Hi TC, T.38 FAX and native bridging are not supported by asterisk-oh323. Michael. T. Chan wrote: Dear Michael, Does your H323 driver run T38 Fax? Also, does your H323 driver have the capability of just proxying signal, and NOT proxying signal and media, just like the canrevite=yes in the sip scenario? Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos Sent: Wednesday, March 10, 2004 7:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk-oh323, new version 0.5.10 Hello all, asterisk-oh323 has been updated. The new version 0.5.10 fixes the incorrect answering of H.323 channels (thanks to the people of the list who helped to trace the problem). Also, I have added support for Gnomemeeting text messages (just for fun). Additionally, the new version contains stability improvements. This will be the last version using the OpenH323/Pwlib v1.12.2/1.5.2. The next version will move on to the latest versions of these libraries. Regards, Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ./M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ./M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-oh323, new version 0.5.10
Dear Michael Do you foresee implementing these in the near future, one or the other or both? Thanks Tc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos Sent: Thursday, March 11, 2004 4:49 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk-oh323, new version 0.5.10 Hi TC, T.38 FAX and native bridging are not supported by asterisk-oh323. Michael. T. Chan wrote: Dear Michael, Does your H323 driver run T38 Fax? Also, does your H323 driver have the capability of just proxying signal, and NOT proxying signal and media, just like the canrevite=yes in the sip scenario? Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos Sent: Wednesday, March 10, 2004 7:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk-oh323, new version 0.5.10 Hello all, asterisk-oh323 has been updated. The new version 0.5.10 fixes the incorrect answering of H.323 channels (thanks to the people of the list who helped to trace the problem). Also, I have added support for Gnomemeeting text messages (just for fun). Additionally, the new version contains stability improvements. This will be the last version using the OpenH323/Pwlib v1.12.2/1.5.2. The next version will move on to the latest versions of these libraries. Regards, Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ./M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-oh323, new version 0.5.10
Dear Michael, Does your H323 driver run T38 Fax? Also, does your H323 driver have the capability of just proxying signal, and NOT proxying signal and media, just like the canrevite=yes in the sip scenario? Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos Sent: Wednesday, March 10, 2004 7:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] asterisk-oh323, new version 0.5.10 Hello all, asterisk-oh323 has been updated. The new version 0.5.10 fixes the incorrect answering of H.323 channels (thanks to the people of the list who helped to trace the problem). Also, I have added support for Gnomemeeting text messages (just for fun). Additionally, the new version contains stability improvements. This will be the last version using the OpenH323/Pwlib v1.12.2/1.5.2. The next version will move on to the latest versions of these libraries. Regards, Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Palm OS5 client
Miguel Can you let me know where I can find the gphone information so that I can give it a try, thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Miguel Cavazos Sent: Thursday, March 04, 2004 5:00 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Palm OS5 client ive been looking for a palm os5 client found gphone there webpage claims to be sip but i just cant make it register against asterisk Miguel On Thu, 2004-03-04 at 07:05, Dean Collins wrote: Does anyone know of a Palm OS5 client that can connect to asterisk? Hopefully I can use gprs to connect back to my home pabx and make local calls while on the road. Also can anyone comment on how well the CE clients work? Cheers, Dean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: codec negotiation prob solved
Dear All, That is my experience with Asterisk too, this codec negotiation is giving us lots of problems. I am using Asterisk mostly for passing-through VOIP traffic. Basically I will have to choose g7231 or g729 and not both. If I choose both, and when calls come in with with both codecs, and the terminating gateway (endpoint) only allows g729, the calls would go through, but if the terminating gateway only allows g7231, then calls would not go through, and if the terminating gateway allows both codecs, call would not go through either. Worst yet, Asterisk seems not to work with t38 fax and I have to allow g711 in order to get fax to go through and that is ONLY between Asterisk, if a cisco calls into my Asterisk with a fax, it just will not work. Anyway, the worst part is if I allow g711 on my Asterisk, ALL calls coming into my Asterisk will get converted to g711 before going out, whether out to a third party equipment (if they allow g711) or my other Asterisk. Conversion takes up too much resource and needless to say bandwidth (for g711), and it is highly a NONO to convert to g711 for all calls, even voice calls. If I allow g723, g729 and g711, calls will NOT just get passthrough, they will get converted. Is there anyway we can debug this problem please? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of SamW Sent: Tuesday, March 02, 2004 3:32 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: codec negotiation prob solved I agree, that * codec negotiation is buggy, there must be some mechanism to give priority to pass through without trying to codec translate. Codec translation need lot of CPU and can deteriorate quality by its nature. Some developer sheding some light on this is buggy codec translation is very appreciated. - SamW -Original Message- From: T. Chan [mailto:[EMAIL PROTECTED] Sent: Friday, February 20, 2004 12:48 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: codec negotiation prob solved I have the same problem, most carriers out there deal with both g723.1 or g729. During passing through via Asterisk, carrier customers will send us calls broadcasting both codecs with one having priority over the other, the way it is supposed to work is that asterisk will try to negotiate the top priority codec first with the terminating endpoint, assuming that the originating endpoint broadcasts g729 as first priority and then g723.1, Asterisk should take g729 and try to negotiate with terminating endpoint and if the terminating takes g729, then the call should be patched and bridged, but if the terminating endpoint takes ONLY g723.1, then Asterisk should then go back and take g723.1 (which is the second priority as per the originating endpoint) and bridge the call through. However, the way Asterisk is doing it is if I allow both g723.1 and g729, then if the originating endpoint broadcasts both codecs and the terminating endpoint only allows g723.1, then the call will not go through and it will say no path from g729 to ., and calls will not go through. Summing up, if originating gateway allows both g723.1 and g729 , Asterisk being the pass-through entity, allows both codecs, and the terminating gateway allows ONLY g723.1, the calls will not go through which is certainly a bug in the asterisk. I wonder if anyone out there has any solution to this problem. TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of dkwok Sent: Friday, February 20, 2004 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: codec negotiation prob solved (Philipp von Klitzing) wrote: FYI - bug 1043 has been fixed on Feb 18: From my log, below, you will see that ast_rtp_bridge is not comparing the codecs properly. Asterisk is currently comparing the integers, and not the bits of the codec. In the below example codec0 = 260. That means Codec0 allows both 256 (g729) and 4 (uLaw). So that would mean that Codec0 and Codec1 do have a Codec Match. Asterisk needs to do a bit compare, and not a int compare in this case. -- SIP/dialnet-8bac answered SIP/chris0-df00 -- Attempting native bridge of SIP/chris0-df00 and SIP/dialnet-8bac Feb 16 11:27:48 WARNING[68889520]: rtp.c:1204 ast_rtp_bridge: codec0 = 260 is not codec1 = 256, cannot native bridge. Feb 16 11:27:50 NOTICE[68889520]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Feb 16 11:27:50 NOTICE[68889520]: rtp.c:293 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256 I have the same problem with codec negotiation, my Voip provider use g729 however I have also connection with Iaxtel which only use GSM. I can only get one or the other codec working when dialing out. My iax.conf setting is below: ; Inter-Asterisk eXchange driver definition [general] port=4569 bandwidth=low disallow=all allow=gsm allow=g729 disallow=g723.1 ; Hm... Proprietary, don't use it... disallow=lpc10
RE: [Asterisk-Users] H323 calls drop on connect
Hi, Todd Did you notice that when you made the calls, were the calls indicated as answered, in both cases? And if so, did the indication answered pop up when the calls were actually picked up and answered or right after the call setup was completed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Todd Wallace Sent: Tuesday, March 02, 2004 4:30 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H323 calls drop on connect I have something new that is happening to me...When I call from a SIP phone and route out OH323, I get a good clear ringing, connect, then it drops me. If I get a telco recorded message, I hear the complete message. If I get a person that answers, I hear about the first 2 seconds, then it drops me. Any ideas where it look? I feel it is in the OH323 config.. Todd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 calls drop on connect
That is not good, isn't it? So, it seems that something is not right there already, why don't you use the other H323 channel driver and give it a try -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Todd Wallace Sent: Tuesday, March 02, 2004 7:21 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] H323 calls drop on connect Right after the call setup was completed... Todd -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Chan Sent: Tuesday, March 02, 2004 4:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] H323 calls drop on connect Hi, Todd Did you notice that when you made the calls, were the calls indicated as answered, in both cases? And if so, did the indication answered pop up when the calls were actually picked up and answered or right after the call setup was completed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Todd Wallace Sent: Tuesday, March 02, 2004 4:30 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H323 calls drop on connect I have something new that is happening to me...When I call from a SIP phone and route out OH323, I get a good clear ringing, connect, then it drops me. If I get a telco recorded message, I hear the complete message. If I get a person that answers, I hear about the first 2 seconds, then it drops me. Any ideas where it look? I feel it is in the OH323 config.. Todd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: simple H323 question
Dear All,
Thanks, but I am not using chan_oh323, I am using chan_h323. The major
reason why I am not using chan_oh323 is because of a bug that Michael is not
yet able to resolve. Every call that goes out via this h323 channel will be
considered connected and picked up (false answer supervision) immediately
after the call setup, even though the call is busy or ring no answer. So is
there anyway to find out codec negotiated for each h323 call via chan_h323
channel?
Thanks, all
Tommy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michiel Betel
Sent: Saturday, February 28, 2004 2:45 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RE: simple H323 question
Ron McMillan wrote:
One way to do it is to use a sniffer, such as ethereal, to capture the
traffic. You should see it in capability exchange, but also easily see in
RTP packets. There might be better ways. But if you're interested in
pursuing it this way and not sure how to do, please follow up with another
question...
Ron
On Fri, 27 Feb 2004, T. Chan wrote:
Hi, all
I wonder when passing calls through asterisk with H323, is there anyway to
find out what codec the calls are using, anyone can help please, thanks
alot
!
TC
TC, When using chan_oh323 the codec used is stored in the variable
${OH323_CHANCODEC}
Michiel
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[Asterisk-Users] RE: simple H323 question
Hi, all I wonder when passing calls through asterisk with H323, is there anyway to find out what codec the calls are using, anyone can help please, thanks alot ! TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: codec negotiation prob solved
I have the same problem, most carriers out there deal with both g723.1 or g729. During passing through via Asterisk, carrier customers will send us calls broadcasting both codecs with one having priority over the other, the way it is supposed to work is that asterisk will try to negotiate the top priority codec first with the terminating endpoint, assuming that the originating endpoint broadcasts g729 as first priority and then g723.1, Asterisk should take g729 and try to negotiate with terminating endpoint and if the terminating takes g729, then the call should be patched and bridged, but if the terminating endpoint takes ONLY g723.1, then Asterisk should then go back and take g723.1 (which is the second priority as per the originating endpoint) and bridge the call through. However, the way Asterisk is doing it is if I allow both g723.1 and g729, then if the originating endpoint broadcasts both codecs and the terminating endpoint only allows g723.1, then the call will not go through and it will say no path from g729 to ., and calls will not go through. Summing up, if originating gateway allows both g723.1 and g729 , Asterisk being the pass-through entity, allows both codecs, and the terminating gateway allows ONLY g723.1, the calls will not go through which is certainly a bug in the asterisk. I wonder if anyone out there has any solution to this problem. TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of dkwok Sent: Friday, February 20, 2004 8:42 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: codec negotiation prob solved (Philipp von Klitzing) wrote: FYI - bug 1043 has been fixed on Feb 18: From my log, below, you will see that ast_rtp_bridge is not comparing the codecs properly. Asterisk is currently comparing the integers, and not the bits of the codec. In the below example codec0 = 260. That means Codec0 allows both 256 (g729) and 4 (uLaw). So that would mean that Codec0 and Codec1 do have a Codec Match. Asterisk needs to do a bit compare, and not a int compare in this case. -- SIP/dialnet-8bac answered SIP/chris0-df00 -- Attempting native bridge of SIP/chris0-df00 and SIP/dialnet-8bac Feb 16 11:27:48 WARNING[68889520]: rtp.c:1204 ast_rtp_bridge: codec0 = 260 is not codec1 = 256, cannot native bridge. Feb 16 11:27:50 NOTICE[68889520]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Feb 16 11:27:50 NOTICE[68889520]: rtp.c:293 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256 I have the same problem with codec negotiation, my Voip provider use g729 however I have also connection with Iaxtel which only use GSM. I can only get one or the other codec working when dialing out. My iax.conf setting is below: ; Inter-Asterisk eXchange driver definition [general] port=4569 bandwidth=low disallow=all allow=gsm allow=g729 disallow=g723.1 ; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. jitterbuffer=yes dropcount=3 maxjitterbuffer=250 maxexccessbuffer=50 register = dkwok:[EMAIL PROTECTED] tos=lowdelay [iax_home] type=friend context=int-ext auth=md5 user=iax_home secret=cc trunking=yes disallow=all allow=gsm host=dynamic qualify=yes [iaxtel] type=friend disallow=all disallow=g729 allow=gsm trunking=yes context=from-iaxtel [atp] type=friend disallow=all allow=g729 trunking=yes context=atp host=xxx.xxx.xxx.xxx I would like to hear any comment from * developer. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.590 / Virus Database: 373 - Release Date: 2/16/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Faxing in passthrough and codec in passthrough
Dear All, I need your advices, all you experts out there, I have spent alot of time testing but just could not get it to work, so I need your assistance please. I have been trying to passthrough calls with asterisk, that is, receiving calls from customer via VOIP and then directly send the calls out to other VOIP carriers. For example calls to India come in to my asterisk and get routed to Carrier A who is, for example, using a Cisco VOIP gateway to receive my calls and terminate to India. I have 2 problems with passing throughs: 1. Faxing - when calls come into my asterisk, I don't suppose it knows if it is a fax or not, or if there is a way to configure the Asterisk to detect it, how should it be configured? Then the calls to India will go directly back out via VOIP to Carrier A who , for example, uses Cisco VOIP gateways and supports T38 faxing. The problem is that, with this passing through, all faxes become impossible. Another scenario which is NOT a passthrough situation is I have an Asterisk in, for example, New York and another one in, for example, Los Angeles, we tried making faxes from New York from the ZAP channels to Los Angeles, to be terminated via ZAP channels on ISDN circuits. Again, it never works. Ladies and Gentlemen, how should one configure Asterisk to make (a) faxing possible in a passthrough situation with both endpoints cisco and asterisk in the middle and (b) faxing possible with both asterisk as endpoints? 2. Codec negotiation - again in a passthrough situation when both endpoints are non-asterisk, most probably cisco or quintum gateways, I can never allow more than 1 codec, for example, g723 and g729 or with g711. For example, if the terminating endpoint is quintum and allows only g723, the originating endpoint is cisco and it is sending both g723 and g729, I cannot allow on my asterisk both g723 and g729 and let it negotiate because it will not work. I have to ONLY allow g723. However, there might be a quintum2 as a terminating endpoint that uses g729, and if I allow both g723 and g729 on my asterisk, both quintum and quintum2 will not work, if I allow ONLY g723 on the asterisk, only the first quintum will work, and if I allow ONLY g729, only quintum2 will work, I cannot send calls to both quintum and quintum2, all other gateways, softswitches work with allowing all codecs to negotiate, but it does not seem to work with asterisk, I tried h323 and SIP and reaping same results. Can anyone help me to see how I could achieve that please, thanks !! Look forward to your assistance. Thanks TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fast question on extension matching
Dear All, I have a very simple question but could not find any information from the internet. Is there anyway to match code on extensions.conf without having to specify the number of digits? For example, if I want to send 01163 (Philippines to a certain IP address), is there anyway simpler to do than exten = _01163,1,. exten = _01163XXX,1,. exten = _01163XX,1,. exten = _01163X,1,. exten = _01163,1,. exten = _01163XXX,1,. Is there any one line command that could replace having to use XX... to match exact number of digits? Thanks TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fast question on extension matching
Dear Chris, Thanks for your lesson, it sort of works but not perfect. I tried exten = _01163.,1,Application() exten = _011.,1,Application() because I want to send Philippines to a different IP address than the rest of the world, but if I configure that way, even 01163 calls will all go to the second IP address as per 011.,1,Application(). If I take out the 011., then calls WILL go to 01163., if I put the two together it will always go to 011. extension. Any idea please? Thanks again TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Craft Sent: Friday, February 06, 2004 4:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fast question on extension matching T. Chan wrote: Dear All, I have a very simple question but could not find any information from the internet. Is there anyway to match code on extensions.conf without having to specify the number of digits? For example, if I want to send 01163 (Philippines to a certain IP address), is there anyway simpler to do than exten = _01163,1,. exten = _01163XXX,1,. exten = _01163XX,1,. exten = _01163X,1,. exten = _01163,1,. exten = _01163XXX,1,. Is there any one line command that could replace having to use XX... to match exact number of digits? Thanks TC TC, Just do something like: exten = _01163.,1,Application() Cheers, Chris. iax700.824.0300 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can audio streams go client to cleint with IAX?
Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can audio streams go client to cleint with IAX?
Would that be something that Jeremy would work on? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 6:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? If you define possible as is H.323 capable of it, then yes. If you define possible as is asterisk currently capable of it, then no. It wouldn't be too hard, look in channels/h323/ast_h323.cpp to see where Jeremy started on it. You just have to get the openh323 lib to initiate the transfer. - Original Message - From: Marc Fargas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:30 AM Subject: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Is it possible to make audio streams go client to client with H.323 ? (both client being H323) Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de T. Chan Enviado el: lunes, 02 de febrero de 2004 23:56 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Can audio streams go client to cleint with IAX? Dear All, Now, it seems that both IAX and SIP can have the two endpoints communicate the media directly without the media stream passing through the asterisk, can we do the same with H323 too? TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Monday, February 02, 2004 5:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Can audio streams go client to cleint with IAX? yes, IAX does direct transfers - when both ends confirm they can see each other, the asterisk server tells them to talk directly. With the firefly network, we're seeing 90%+ connecting directly. Just to clarify, the audio doesn't separate from the call control. -Adam - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 1:59 AM Subject: [Asterisk-Users] Can audio streams go client to cleint with IAX? With a service like http://www.freshtel.net/?show=home that uses IAX and has servers in Australia, is it possible for the audio streams to take a different path than the call setup and control? In other words can it work like SIP with canreinvite where the two endpoint negotiate audio streams between themselves rather than though the FreshTel server? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Compiling zaptel
Dear all, I have been testing with Asterisk for a bit of time and yesterday I tried to upgrade my kernel to 2.4.20-28, but after I upgraded the kernel, I was not able to compile Zaptel. The kernel runs good and everything intact, I was trying to recompile Asterisk in order to make sure that everything was clean. I have gone into /usr/src/zaptel, done a make clean and then done a make install as what I have always done after updating the asterisk version. However, now I am getting the following error, wct4xxp.c: In function 't4-interrupt' wct4xxp.c: 1357:structure has no member named 'Lock' make: *** [wct4xxp.o] error and then it stopped compiling, can someone please let me know if I am missing something please, greatly appreciated. thanks TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Compiling zaptel
Dear sb I am running redhat 7.3 upgraded to this version of kernel. Yes, I have installed the kernel-sources but not the kernel-util rpms, I don't think I ever did install kernel-util with the original installation of redhat 7.3 or did I? But I was having no problem installing zap at that time though. Do you think I need that? Thanks TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott (7805) Sent: Friday, January 30, 2004 4:10 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Compiling zaptel I take it you are running RedHat 8 (or 9) since this is the most up-to-date kernel. Did you install the kernel-sources and kernel-util rpms as well? You'll need these in order to compile and install zaptel. -sb -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of T. Chan Sent: Friday, January 30, 2004 4:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Compiling zaptel Dear all, I have been testing with Asterisk for a bit of time and yesterday I tried to upgrade my kernel to 2.4.20-28, but after I upgraded the kernel, I was not able to compile Zaptel. The kernel runs good and everything intact, I was trying to recompile Asterisk in order to make sure that everything was clean. I have gone into /usr/src/zaptel, done a make clean and then done a make install as what I have always done after updating the asterisk version. However, now I am getting the following error, wct4xxp.c: In function 't4-interrupt' wct4xxp.c: 1357:structure has no member named 'Lock' make: *** [wct4xxp.o] error and then it stopped compiling, can someone please let me know if I am missing something please, greatly appreciated. thanks TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Compiling zaptel
Hi, all ! I have been testing with Asterisk for a bit of time and yesterday I tried to upgrade my kernel to 2.4.20-28, but after I upgraded the kernel, I was not able to compile Zaptel. The kernel runs good and everything intact, I was trying to recompile Asterisk in order to make sure that everything was clean. I have gone into /usr/src/zaptel, done a make clean and then done a make install as what I have always done after updating the asterisk version. However, now I am getting the following error, wct4xxp.c: In function 't4-interrupt' wct4xxp.c: 1357:structure has no member named 'Lock' make: *** [wct4xxp.o] error and then it stopped compiling, can someone please let me know if I am missing something please, greatly appreciated. thanks Tom --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Compiling zaptel
Dear all, I have been testing with Asterisk for a bit of time and yesterday I tried to upgrade my kernel to 2.4.20-28, but after I upgraded the kernel, I was not able to compile Zaptel. The kernel runs good and everything intact, I was trying to recompile Asterisk in order to make sure that everything was clean. I have gone into /usr/src/zaptel, done a make clean and then done a make install as what I have always done after updating the asterisk version. However, now I am getting the following error, wct4xxp.c: In function 't4-interrupt' wct4xxp.c: 1357:structure has no member named 'Lock' make: *** [wct4xxp.o] error and then it stopped compiling, can someone please let me know if I am missing something please, greatly appreciated. thanks TC --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is it possible to push the media processing off to a gateway for processing?
I think what Todd was referring to was to JUST do the signaling proxy on the Asterisk but not proxying the media. The Asterisk box would ONLY do the signaling handling between the two endpoints and hang over the media stream to go directly between the two endpoints. This is a question I was wondering as well. I was thinking of doing the same thing if possible with H323, I read from somewhere that this is doable with SIP setting canreinvite to YES (or was it NO??) and thereby leaving the media stream to go directly between the endpoints and not proxying the media stream. Please if someone can help, that would be appreciated, thanks for all ! Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Saturday, January 24, 2004 10:53 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Is it possible to push the media processing off to a gateway for processing? If you are considering such a service, you need to develop a more thorough understanding of VoIP protocols and methods for load distribution. To echo what Stephen Critchfield said to someone else just a few hours ago: it's not simple, and you'll probably need a consultant. After you've spent some time designing and discussing, you'll probably be able to do it yourself next time but to try this from scratch is probably a very long trial-and-error effort. The short answer to your question for your research is: use load sharing and SIP redirects to spread load across multiple boxes, and yes, asterisk configs can be pulled from mysql in various ways - dig through the distribution for details. JT Well, I like the features asterisk gives me such as voicemail and IVR, Prompts, etc. I would like to offer an IP Centrex like service, but don't believe that I can handle very large amounts of users on a box. The reason I believe this is that the box would be doing all the media processing/DSP work on the processor and would be bound by the speed and memory of the box as to how many simultaneous sessions it could manage. A gateway has DSP's which are designed to handle this processing. I know they are more expensive, but I could handle large amounts of call volume this way and still keep the features asterisk offers. Another question I also meant to ask was having the ability to read extensions from a database instead of a .conf file. I was curious if anyone has asterisk pulling configs from a database like mysql. Todd -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Friday, January 23, 2004 9:05 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Is it possible to push the media processing off to a gateway for processing? I was wondering if it is possible to have Asterisk push the media processing off to something with DSP's such as a gateway? That way, asterisk just has to handle the call setups and tear downs. Todd Wallace You mean, like what SIP does by default? This is an incomplete question. Please be more specific. If I have a gateway, and I have SIP calls coming in from desktop SIP UA's (hardphones or softphones) then Asterisk can simply re-direct those calls to the gateway. Of course, Asterisk _is_ a gateway, so unless you have specific reasons for doing so, it would make more sense to use Asterisk to tackle those jobs with generic, cheap processing horsepower rather than expensive, proprietary DSP's. If you're just getting Asterisk to handle call setups and teardowns, why not just use a real SIP proxy for that? Or do you not know enough about your question to understand why I would differentiate between the two? (not being nasty here, just wondering if I need to explain more) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or
RE: [Asterisk-Users] LAN card
Dear All, Just an experience to run by all you experts out there. I have started to put more VOIP calls into Asterisk, most are pass-through calls and some are terminating on the Digium card to PSTN. Whenever I get to 10 calls or more, I would start to get choppy sound. I tried to ping other IP addresses from the Asterisk and noticed a big packet loss in the vincinity of 7% to 10%, but when there is no call, pinging the same IP addresses reap no packet loss. It seems that the VOIP packets are causing congestion of some kind on the LAN. I am using 100M, full duplex. I tried an autonegotiated switching hub as well as a more sophisticated managed switching hub and forcing the connection to be 100M Full Duplex, non negotiated. However, I reaped the same result. Question is, do you know if it is better to use Managed switch and forcing the Ethernet connection to be 100M Full Duplex, or to use a normal UnManaged switch and let it negotiate. Also, I am using both a normal PCI LAN card as well as trying to use the onboard Intel 100PRO Lan card, and in both situations, I started to get lose packets when the number of calls increased. My colleagues, can anyone tell me if I am doing something wrong here, or is there something I am forgetting, or I simply need to use a more powerful LAN card due to the demand of VOIP packets. Waiting urgently for advice. Thanks Tom --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LAN card
Thanks alot, Ray Well, looking at cat /proc/meminfo, I am getting like 250M memory cached, with 512M total RAM, for all the gateways I have, this is quite consistent. Total Memory usages are always low after reboot and then go up to 450M with time. I was informed that this is normal for Linux. Thanks for your input on Managed switch. However as said, I tried both Managed switch and non-Managed switch but have reaped the same result with packet loss when there are more active calls. Do you have any experience whether I need a good PCI LAN card like 3COM or Intel Express due to the demanding VOIP packets or do you think Intel ONBOARD LAN card should be sufficient? Thanks Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ray Burkholder Sent: Sunday, January 25, 2004 4:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LAN card Take a look at your memory utilization, you should not be paging/caching any memory. Switches are will known not to auto-negotiate properly. All switches, nics, routers, etc should be manually configured for full-duplex. Make sure each connection is set appropriately for 1000/100/10 mpbs, what ever is appropriate for that connection. And yes, you can get full duplex for 10 mpbs connections (in answer to a message a while back on the list). Managed switches are best becuase you can look at them and get an idea of link/packet errors on each port. Obviously you want to completely eliminate errors on each port. Once you've done that, you should be well on your way to a reliable, scalable solution. Quoting T. Chan [EMAIL PROTECTED]: Dear All, Just an experience to run by all you experts out there. I have started to put more VOIP calls into Asterisk, most are pass-through calls and some are terminating on the Digium card to PSTN. Whenever I get to 10 calls or more, I would start to get choppy sound. I tried to ping other IP addresses from the Asterisk and noticed a big packet loss in the vincinity of 7% to 10%, but when there is no call, pinging the same IP addresses reap no packet loss. It seems that the VOIP packets are causing congestion of some kind on the LAN. I am using 100M, full duplex. I tried an autonegotiated switching hub as well as a more sophisticated managed switching hub and forcing the connection to be 100M Full Duplex, non negotiated. However, I reaped the same result. Question is, do you know if it is better to use Managed switch and forcing the Ethernet connection to be 100M Full Duplex, or to use a normal UnManaged switch and let it negotiate. Also, I am using both a normal PCI LAN card as well as trying to use the onboard Intel 100PRO Lan card, and in both situations, I started to get lose packets when the number of calls increased. My colleagues, can anyone tell me if I am doing something wrong here, or is there something I am forgetting, or I simply need to use a more powerful LAN card due to the demand of VOIP packets. Ray Burkholder 704 644 6999 x2002 http://www.oneunified.net [EMAIL PROTECTED] - This mail sent through IMP: http://horde.org/imp/ -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LAN card
Thanks, Ray No, I am not running any other programs other than basic OS and Asterisk. No, I am not using the swapMemory yet, but like you under USED memory, I am using about 450M or above after the computer has been rebooted for a couple of days. I am using a Baystack switch and I will try to look at the statistics, but it is very difficult to believe that something is wrong with all the Baystack switch installed at different locations along with the different Asterisks, when all are having the same problem and symptom. I am just running console and no other programs, basically after installing Redhat, we installed the Asterisk and that is it, and when problem started to happen, CPU usages have been low, I doubt it that it was the CPU usage or memory. When you mentioned Intel 1000XT, is it an ONBOARD or PCI card? What is the maximum number of calls you have experienced with your system? Thanks, Ray. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ray Burkholder Sent: Sunday, January 25, 2004 5:43 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LAN card What else are you running on your server? On my server running asterisk and apache, it has the following: total:used:free: shared: buffers: cached: Mem: 261443584 237064192 243793920 55992320 143912960 Swap: 260104192 11231232 248872960 MemTotal: 255316 kB MemFree: 23808 kB SwapTotal: 254008 kB SwapFree: 243040 kB I've got very little swap usage, even with 256MB total physical. For the switch, have you looked at the statistics? For example on a Cisco: sw2#sho inter f0/1 FastEthernet0/1 is up, line protocol is up Hardware is Fast Ethernet, address is 0005.5e31.5f41 (bia 0005.5e31.5f41) Description: Trunk: r1-skings MTU 1500 bytes, BW 10 Kbit, DLY 100 usec, reliability 251/255, txload 1/255, rxload 1/255 Encapsulation ARPA, loopback not set Keepalive not set Full-duplex, 100Mb/s, 100BaseTX/FX ARP type: ARPA, ARP Timeout 04:00:00 Last input 00:00:40, output 00:00:00, output hang never Last clearing of show interface counters never Queueing strategy: fifo Output queue 0/40, 0 drops; input queue 0/75, 0 drops 30 second input rate 3000 bits/sec, 4 packets/sec 30 second output rate 6000 bits/sec, 8 packets/sec 75312691 packets input, 1770301889 bytes Received 515417 broadcasts, 7622395 runts, 0 giants, 0 throttles 7622399 input errors, 4 CRC, 0 frame, 4 overrun, 92 ignored 0 watchdog, 255441 multicast 0 input packets with dribble condition detected 104212173 packets output, 2775526395 bytes, 0 underruns 0 output errors, 0 collisions, 3 interface resets 0 babbles, 0 late collision, 0 deferred 0 lost carrier, 0 no carrier 0 output buffer failures, 0 output buffers swapped out Looks like I've got some input errors I should be looking into. It should be as close to 0 as possible. An Intel 1000XT are good cards at they do TCP Engine Offload. Or something similar. But voice traffic is measured in kbits/second, which is a very low proportion of 10mbps or even 100mbps. So I'd say take another look at your server and see if an application isn't makeing a mess of your cpu processing. Becuase Asterisk is time senstive, it should really be the only primary process running on your machine. AND, YOU SHOULD NOT BE RUNNING XWINDOWS. The os should have been installed in console mode, and as little as possible relating to X installed. Quoting T. Chan [EMAIL PROTECTED]: Thanks alot, Ray Well, looking at cat /proc/meminfo, I am getting like 250M memory cached, with 512M total RAM, for all the gateways I have, this is quite consistent. Total Memory usages are always low after reboot and then go up to 450M with time. I was informed that this is normal for Linux. Thanks for your input on Managed switch. However as said, I tried both Managed switch and non-Managed switch but have reaped the same result with packet loss when there are more active calls. Do you have any experience whether I need a good PCI LAN card like 3COM or Intel Express due to the demanding VOIP packets or do you think Intel ONBOARD LAN card should be sufficient? Thanks Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ray Burkholder Sent: Sunday, January 25, 2004 4:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] LAN card Take a look at your memory utilization, you should not be paging/caching any memory. Switches are will known not to auto-negotiate properly. All switches, nics, routers, etc should be manually configured for full-duplex. Make sure each connection is set appropriately for 1000/100/10 mpbs, what ever is appropriate for that connection. And yes, you can get full duplex for 10 mpbs connections (in answer to a message a while back on the list). Managed switches are best becuase you can look at them and get
RE: [Asterisk-Users] Has Nufone gone belly-up
Dear All, I will send you a couple more emails, the story is something like for some reasons, Nufone website was not working for a bit of time or something, and this and that. So, there were some customers / potential customers who posted in the asterisk discussion forum titled 'Has Nufone gone belly-up'. There was so many responses and discussions on it (it was meant to be asterisk user discussion forum, helping each other, but my questions were seldom answered, rather so many people was discussing about Nufone bellying up, ridiculous !). Anyway, the point is, what an asshole !! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Monday, January 26, 2004 12:41 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Has Nufone gone belly-up David Liu wrote: I don't use Nufone. But just seeing Jeremy's reply make me want to say something. As an outsider, if the attitude is No messages were ever received from you, thus we never called you back. or How quickly you forget that It makes a customer wonder the attitude the staff at Nufone has. Again I am just speaking from an outsider perspective. To have good service, it is very important to be patient and explain things. Never get angry with your customer and don't show it. Just my advice. Our network and services speak for themselves. If they don't like my attitude after they publicly flame us they can find another provider, I really don't care. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OH323 config file format
Hi, Deepak, how are you? I don't quite understand what you meant by username and password sending calls to a H323 service provider, do you mean you have to register onto their gatekeeper? Or otherwise, you should not need username and password. Meantime, I am trying to setup up SIP calling to a service provider, can you let me know what is the maximum number of calls you have experienced with sending SIP calls to your service provider? Have you experienced any crash? What is the configuration of your computer? Thanks Tom -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Deepakumar JVSent: Wednesday, January 21, 2004 12:38 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] OH323 config file format Hello I am trying to configure my extensions.conf and oh323.conf to termination calls to a H323 service provider. Can anyone send me a sample config files? or tell me where to put the username and password which my service provider has given? also how to put the Dial command in extensions.conf Asterisk rocks. I have a SIP provider configured for all my international calls and it works absolutely fine. Its cool. Thanks in advance Regards Deepak
RE: [Asterisk-Users] RE: current version
I appreciate all your feedbacks, but they seems to have diverted from my original question which was I have been using Asterisk 10 days ago version loaded onto my Redhat 7.3 with kernel 2.4.18-3 running Jeremy's h323 driver. It has been running okay with a bit of problems, like system crashing after certain period of time with 15 simultaneous calls or so. I have tried to load up the current version today again (0.7.1 I guess) and apparently with the new H323 driver as well. I have recompiled the H323 libraries with version Pwlib1.5.2 and Openh3231.12.2 as recommended. However, no call was able to get through at all. I have tried this before when 0.7.0 came out when had the same result, I thought there were bugs, but now I am getting the same thing. I have tried using the same h323.conf configuration as well as trying to change a couple of faststart.parameters, but same result. So, rather I would appreciate any assistance and feedback in related to running Asterisk version 0.7.1 with the most current H323 driver compiled with the current version of OpenH323 libraries. I wonder if any associates out there who have had success running H323 with this, thanks ! Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dan Austin Sent: Monday, January 19, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: current version To be clear I meant using Chan)_h323 with Call Manager where CM is configured with * as a H.323 gateway, not client. CM supports H.323 to direct calls through gateways, and in fact until recently that is all they supported. They now also have MGCP, but only to their IOS platforms, and SIP is coming soon. There are NO sccp-based gateways, from Cisco anyways. Dan -Original Message- From: Eric Wieling [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 6:33 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RE: current version CallManager uses Cisco's own SCCP aka Skinny Protocol, not H323. Asterisk has two SCCP channel drivers available. One is included with Asterisk, one is available for download from somewhere (check the mailing list archives). I don't know if they work with CallManager or now, I *think* they were designed for use with SCCP only phones, but I'm not sure. On Sun, 2004-01-18 at 22:16, Ray Burkholder wrote: Will the other chan_oh323 work? Quoting Brian West [EMAIL PROTECTED]: You can't use chan_h323 with call manager. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Concurrents calls on asterisk with H323
I agree that it should be able to do more than 15 to 20 calls when NOT transcoding, however, I WAS doing pass-through without any transcoding and it was crashing after around 15 to 20 calls, that was the problem, while I was expecting at least hundreds of simultaneous calls ( not channels ) doing pass through, because this is what other softswitches are able to do very reliably. Also, I do not see WHY transcoding should not let us do more than 15-20 calls (in my case, only passing through), I read somewhere that one of our associates here has experienced about 45 calls when transcoding. The question becomes if this is all we can do with one server, what is the point of getting 4 E1s Digium card while one can never be able to use 120 channels transcoding from VOIP to TDM? Can someone shed some light on this please? Thanks ! Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Monday, January 19, 2004 11:30 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Concurrents calls on asterisk with H323 Look for the recent 'capacity testing' thread here. We've had some discussions on it, but so far the bottom line sounds like you won't be able to run more than 20 - 25 decent quality calls before asterisk dies. jesse [snip] Your statement relies completely on assumptions which may be incorrect. Transcoding significantly degrades performance, but without transcoding it may be possible for * to move dozens or hundreds of calls with H.323. See note below. JT Subject: RE: [Asterisk-Users] A question on codec translation. From: Tom Lowe [EMAIL PROTECTED] To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Mon, 12 Jan 2004 08:45:21 -0500 If the incoming and outgoing Codecs are the same, there is no conversion done. It basically becomes a packet relay, what goes in, comes out. I'm not sure of the answer to your second question. However, your question actually begs a question I've been wondering about in the last couple of days: I'm doing H.323 in, H.323 outsimple relay. (This is my customer's requirement...not my preference). What I want to do is ALWAYS use the same codec for the outgoing leg as for the incoming leg. In other words, if the call comes in as G.729, the outgoing call uses G.729 ONLY. If the incoming call is G.711, I want the outgoing to be G.711. I want to avoid any sort of transcoding. Is it possible? Thanks. Tom Lowe (FYI, Dual Xeon 3.06, 120 channels (60 calls) of above scenario, G.729 using less than 10% CPU!) (Remember, no transcoding is being performed) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Latest version of asterisk
Thanks, Matt ! So, am I correct in assuming that there are quite a few (or alot) of us who have had not so good experiences with Asterisk? That Asterisk would crash after it hit a certain number of calls or after a certain period of time with 15-20 calls? I understand that there were others who were able to send a good number of calls through but can anyone tell us if they have had tested and confirmed that Asterisk runs better without or with HT and in terms of number of calls, how many would each one support, in the ballpark? It would also be nice if one could tell us the computer configuration in order to send that many calls without crashing Asterisk. Does it make a difference running the LAN on a ONBOARD LAN card as compared to a PCI Intel or 3COM LAN card, since there is a chance that packets are passing more efficiently on a PCI LAN card? Side question: Is it possible to do passthrough faxing? Like, customers sending me H323 or SIP fax calls and the Asterisk will pass through to another gateway? Anyone successful in doing that? Tommy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of mattf Sent: Monday, January 19, 2004 8:32 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] RE: Latest version of asterisk Hello, I've had Asterisk installed on HT capable machines in both HT mode(with SMP) and non HT mode (with non-SMP) and did not notice any differences functionally between them. The processor load was always less in HT SMP mode than non HT and I have experienced Asterisk deadlocks in both modes so it doesn't really seem to matter if you leave HT on(at least in my experiences). HT basically works by splitting off commands to one of two different virtual processors that both run at about 70% of processor's speed(that's why you may notice compiling to take longer when in HT mode) I have heard of some applications having memory addressing errors with HT but I have not seen any evidence to support that in Asterisk thus far. I'm going to try installing a 4 x T1 card on my Athlon 2xMP server next week and see if Asterisk/Digium performance/compatibility improves over the Intel platform. MATT--- -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 2:54 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Latest version of asterisk T. Chan wrote: Dear All Should one enable HT in the chip when running Asterisk or if we don't, would that offer alot less processing power? T I have read before that HT did not help Asterisk so should be dissabled, but as the chipsets and other hardware get better at using and controlling HT it may help.. Run some tests on your system and see what your conclusions are, then feedback your findings to the list so that others may learn from it.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: Memory problem
Dear all, I have had an experience which I would run by all of you to see if this is normal. I am running a few asterisk servers with 512M RAM memory, and as I have mentioned in previous notes, I have experienced frequent crashes when faced with more than 15-20 simultaneous calls. I have tried to find out if it could be due to (a) Xeon chip running HT, (b) old Kernel version 2.4.18-3, (c) old redhat linux version 7.3, (d) H323 library pwlib and openh323 versions which are 1.5.2 and 1.12.2 respectively among many other parameters. So far, unfortunately, the matter has not been resolved. However, I have noticed that the memory usage on each server has built up with time after the server being rebooted. I have complained about using close to 500M even when there were very few calls on the server but nobody seemed to be able to let me know if they were running at high memory usages except for Jesse who was telling me that his memory usages have always been low. Very recently, I noticed that after I rebooted the servers, the memory usage would start at about 80 M and even after started the Asterisk threads, I was running at about 100 M and even when there were calls, I was running at about 100M-150M, but then after hours it would start to build up to 200M and then 250M and thenfinally close to 500M even after I stopped the Asterisk threads, almost like there is a memory leak somewhere. I wonder if that is normal, if someone can please tell me, or if not normal, what could be the cause to it and how should this be rectified. Thanks alot Tom --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] capacity testing
Title: RE: [Asterisk-Users] capacity testing Jesse, Thanks for your feedback. 1. I am running kernel 2.4.18.3 with linux 7.3, please let me know which version of Redhat are you running on and which kernel are you running, I wonder if that could make a difference too. I am surprised that you can run 25 channels with a PIII 800, while I can only run less than 20 channels with a Xeon 2.4G. Please see if you can run more channels with a better CPU and let me know. 2.I have also tried to use OH323 instead of H323, calls seem to go through but I am just getting the ANSWERED indication even before the calls start to ring, which is not right !! I have compiled the PWLIB 1.5.2 and OH323 1.12.2 and using OH323 version 0.5.7 which is the lastest version, are you having the same experience? Is there anyway you can send me your OH323.conf please? 3. I would love to communicate with you privately to exchange experience. Please send email to [EMAIL PROTECTED], thanks Tom -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Jesse PetersonSent: Friday, January 16, 2004 12:32 PMTo: Asterisk-Users (E-mail)Subject: RE: [Asterisk-Users] capacity testing 1) Yes, I did get that. I've never seen a segmentation fault message, but that should be b/c I've been running the process in the background since it is obviously seg-faulting. I believe you are also correct that most people are not trying to put the load on it that we are. 2) I always see 'safe_asterisk' and 'asterisk -vvvg' running. my monitoring was always done with top, but I've checked w/ ps a couple times and I believe only ever see 1 of each of those processes. I may have to do some tests again to double check that. My CPU problems did not come until the last 10 - 30 seconds before asterisk crashed. This is still odd that our memory processor observations are opposite... the next thing I'm going to try is a dual xeon pIII 800 or 1ghz machine to see what happens. 3) I'm running oh323. It was the one I could get to register w/ my gatekeeper as a gateway - that made it much easier for me to do call routing on both sides. I have also noticed some inconsistencies in the call flows like you mention, but haven't taken the time yet to pinpoint exactly what and when they are happening. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of T. ChanSent: Thursday, January 15, 2004 22:54To: [EMAIL PROTECTED]Cc: Alan ChanSubject: RE: [Asterisk-Users] capacity testing Hi all, and Jesse 1. So, you did get the experience of crashing all of a sudden with the "Disconnected from Asterisk server" error message. I got both this and the segmentation error when crashing. I am running the version of asterisk, libpri and zaptel updated to about 5 days ago, but I have had tested Asterisk for more than a month already and needless to say I have had this experience since Day 1, meaning it has always been a problem even in the previous revisions. Henceforth, I feel that it is an intrinsic Asterisk problem, rather than just the problem with specific versions / revisions. I have posted this problem a few times before, I feel that this is a major problem but surprisingly, I was not getting any feedback at all. I have this feeling that more than 90% of the Asterisk community is using the system for PBX application rather than VOIP, may be, just may be, Asterisk has not been tested with a good number of simultaneous calls. 2. I am using Xeon 2.6G chip, much more powerful than yours, I have not got any problem with CPU usage, at least not during the time that I was watching. The thing is when I start 'safe_asterisk' , I could see when doing a PID, 1 "safe_asterisk" PID session and at least 10 (or more especially when there are more calls) "asterisk -vvvg -c" PID session. Each session takes up about 18M to 20M RAM, when that is why I am seeing all very high memory usage. How many sessions of Asterick do you see running after you loaded it? 3. Are you running H323 (Jeremy) and OH323 (Michael)? I am running Jeremy's and have had this inbound H323 problem. I tried OH323 (Michael) as well, but for some reasons, I am getting this false connect signal, that is, I made an outbound H323 call to a CiscoAS5300 for example, I heard the ring and immediately on my "Asterisk", it showed call answered when it was still ringing. Do you have that experience?? What setting you have if you do not have that experience? 4. Lets talk off list at [EMAIL PROTECTED]. Thanks Tom -Original Message-From: Jesse Peterson [mailto:[EMAIL PROTECTED]On Behalf Of Jesse PetersonSent: Thursday, January 15, 2004 8:21
[Asterisk-Users] RE: current version
Title: RE: [Asterisk-Users] capacity testing Dear All, I have been using Asterisk "10 days ago" version loaded onto my Redhat 7.3 with kernel 2.4.18-3 running Jeremy's h323 driver. It has been running okay with a bit of problems, like system crashing after certain period of time with 15 simultaneous calls or so. I have tried to load up the current version today again (0.7.1 I guess) and apparently with the new H323 driver as well. I have recompiled the H323 libraries with version Pwlib1.5.2 and Openh3231.12.2 as recommended. However, no call was able to get through at all. I have tried this before when 0.7.0 came out when had the same result, I thought there were bugs, but now I am getting the same thing. I have tried using the same h323.conf configuration as well as trying to change a couple of faststart.parameters, but same result. Is there anyone who has had good experience with the new version of Asterisk 0.7.1 with the most current Jeremy H323 driver? Please suggest, thanks Tom
[Asterisk-Users] RE: Latest version of asterisk
Dear All, Based on your experience and knowledge, which Redhat (7.3, 8 or 9) and which kernel is most stable and reliable running the 0.7.1 version of Asterisk? Thanks Tom --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Latest version of asterisk
Dear All Should one enable HT in the chip when running Asterisk or if we don't, would that offer alot less processing power? Tom --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.563 / Virus Database: 355 - Release Date: 1/17/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: PID
Dear All, So are you saying that I should see 1 PID for safe_asterisk and many PIDs for asterisk -vvvg -c or just 1 PID for asterisk -vvvg -c, the problem is I am seeing alot of PIDs for asterisk -vvvg -c on a couple of my systems and only 1 PID for asterisk -vvvg -c, which way is correct and how do I rectify that? THanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of TC Sent: Friday, January 16, 2004 2:27 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: PID safe_asterisk is simply a shell script to restart asterisk if it dies. It does not, in itself, do anything related to telephony. I'd find it extremely strange to find Asterisk running with only one thread, unless you had loaded no resources and no channels, which would make the process effectively useless. usual issue here is one unnamed distro's patched 'ps' cmd thinks you only want to see the parent PID of all running threads on the box, dont that just turn your red hat over ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: PID
Dear All, So are you saying that I should see 1 PID for safe_asterisk and many PIDs for asterisk -vvvg -c or just 1 PID for asterisk -vvvg -c, the problem is I am seeing alot of PIDs for asterisk -vvvg -c on a couple of my systems and only 1 PID for asterisk -vvvg -c on other couple, which way is correct and how do I rectify that? I actually see different PID numbers (second column when I do ps) when I see many PIDs, do threads have different PIDs or I am just seeing different threads, but with different PID numbers? THanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James H. Cloos Jr. Sent: Friday, January 16, 2004 10:08 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: PID TC == TC [EMAIL PROTECTED] writes: TC usual issue here is one unnamed distro's patched 'ps' cmd thinks TC you only want to see the parent PID of all running threads on the TC box, dont that just turn your red hat over ? Get used to it. With NPTL all the treads share the same pid, so top(1) et al will only show one entry for the multi-threaded processes. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: PID
Dear All, So are you saying that I should see 1 PID for safe_asterisk and many PIDs for asterisk -vvvg -c or just 1 PID for asterisk -vvvg -c, the problem is I am seeing alot of PIDs for asterisk -vvvg -c on a couple of my systems and only 1 PID for asterisk -vvvg -c for the other couple, which way is correct and how do I rectify that? When I said seeing multiple PIDs, I mean I actually see like 20 different PID numbers (the second column when I do a ps), not sure if that means 20 different threads or 20 PIDs. How do I take care of wonky threading implementation as you suggested? THanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, January 16, 2004 1:33 PM To: [EMAIL PROTECTED] Cc: Alan Chan Subject: Re: [Asterisk-Users] RE: PID On Thu, Jan 15, 2004 at 09:02:00PM -0500, T. Chan wrote: Does anyone have any idea why there is a difference please? The reason that it is important as well is because each asterisk -vvvg -c is taking up certain memory and with 10 (more when there are calls) or more of these, I am running into memory problem. The fact that you see multiple processes is symptomatic of a wonky threading implementation -- all threads should belong to the same process. The exception to this is chan_h323 which actually fork(2)s a new process. Also, as threads share memory space there is no extra usage with extra threads (except for the overhead of keeping the thread state itself, which is minimal). -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: PID
Thanks alot for your explanation. Can you tell me if there is a way to confirm if I have the nptl in the boxes ? Thanks Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James H. Cloos Jr. Sent: Friday, January 16, 2004 9:23 PM To: T. Chan Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: PID T == T Chan [EMAIL PROTECTED] writes: T So are you saying that I should see 1 PID for T safe_asterisk and many PIDs for asterisk -vvvg -c On old distributions one would expect this. T or [1 PID for safe_asterisk and] just 1 PID for asterisk -vvvg -c The Newest distributions -- ones that have nptl (new posix threads library) -- would do this because with nptl all threads have the same pid. Also, some version of top(1) can recognize multiple threads of the old type and hide the child threads. In that case there is an option in top(1) that will show all of the threads. Hit h in top(1) to see what keys do what. So, either some of your boxen have nptl or some have the version of psutils that hides threads in top(1)'s default settings. -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] capacity testing
Hi, I am a newbie in Asterisk as well, intending to use it in a similar way as you are, communicating with AS5300 as well as other gateways including MAXTNT. I have had similar, but yet different experiences than yours. 1. Asterisk does crash with the number of calls, but in my case, about or less than 20 calls, then I would get either a Segmentation Error and then crashed OR it would just crash saying Disconnected from Asterisk server all of a sudden. 2. I am using Pentium Xeon chip and hence more powerful than yours with 512M RAM, my CPU usage has always been low, however, I have not had a chance to look at the CPU usage just before crashing, but all the time that I was looking, it has been low. Rather the MEMORY has always remained high at 450M usage even with no calls. This is a different experience as compared to yours. 3. I have also noticed that with more calls, and after a certain random period of time, any H323 calls going into the Asterisk would fail, my AS5300 and MAXT TNT would get their calls all rejected from Asterisk. However, Asterisk was still running at the time and I could actually call in and out the zap interface and outbound H323 from Asterisk was not a problem. It seems that something got hung with H323, causing inbound H323 calls into Asterisk to all fail. In this situation, I would have to stop the Asterisk and rerun it to fix the problem. 4. I have not tried the 0.7.0 version, but with existing version, I am not getting reliable and stable system, nothing close to Cisco and Lucent which are rock solid. However, I really love the power and the features of Asterisk, and I remain in good faith to see improvements. Any associate out there who can shed some lights into this? I am rather curious as to why I seem to be using up all memory although I am not running any unnecessary processes, or should I actually disable all modules, other than really necessary ones to support VOIP? Thanks ! Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jesse Peterson Sent: Thursday, January 15, 2004 2:40 PM To: Asterisk-Users (E-mail) Subject: [Asterisk-Users] capacity testing Hello all. I'm new to asterisk and have been using and testing it for about a week now. My initial hope has been to use it as a sip-h323 gateway to tie SIP H323 based ip phones together with my Cisco AS5300 and Lucent MaxTNT/MVAM networks. I am currently running Asterisk 0.5.0 under Redhat 9 on a single PIII 800 with 256megs RAM. I have tried a couple CVS version from the past week (maybe 01/09/04 and 01/14/04) and have not been able to get them to work semi-reliably in my simple 1 or 2 call test cases. v.0.5.0 has supported those ok. Primarily test cases have involved sending ip phone calls via SIP to Asterisk and having Asterisk route the calls using h323 via a gatekeeper to my TNT network which then sends it out the PSTN... and the opposite path, PSTN-TNT-Asterisk-SIP Phone. Another test has been sending a call from a AS5300 using SIP to Asterisk, out H323 to a TNT. Both of those have worked very well with the voice quality being excellent (actually better than a SIP-ISDN T1 hardware solution we've been working with - audiocodes mediant 2k for those interested). This is the test case I describe below as it was the one the allowed me to load Asterisk up with the most calls. Anyway, I know that what I'm doing is not exactly the intended primary use of Asterisk. That said, here's what I found. Voice quality was very good until I had approx. 25 calls up. At that point there were intermittent issues with garbled voice, a little echo, etc. When it reached a little over 30 calls, Asterisk just died (oops). During the test, I was trying to keep an eye on proc. memory util. Memory never seemed to be an issue - even right before the crash the Asterisk process was not using more than 20 - 25MB. Processor utilization was interesting to watch though. I couldn't make any direct/firm correlation, but it seemed like my spikes were coming when Asterisk was doing call setup. Even up to about 25 calls, utilization didn't spike to more the 25% for long, and with ~25 calls seemed to 'idle' around 15%. Above the 25 (when also started noticing voice quality issues), the proc. util. seemed to start going wacky - spikes up to 40, 50, even 60%. Then it went to 99% for a moment, voice quality was horrible if you could hear anything, and Asterisk crashed. I did not find anything in the logs to inidicate any problems, though I've found that to be the case pretty much everytime Asterisk crashes. I saw a list thread in which a developer asked for some gdb output... in it, he said this: Run asterisk with -vvvcg. Do your test (core file generated). Run gdb /usr/sbin/asterisk core_filename From within gdb run bt and send me the output of it. if it is of use, here it is (from asterisk v.0.5.0) - (gdb) bt #0 ast_smoother_feed (s=0xcbf90080, f=0x5de5c4a8) at
[Asterisk-Users] RE: PID
Hi, all ! I have a fast question, I am running a few Asterisk systems, but I just noticed one thing quite peculiar. After I started safe_asterisk, and when I ran PS or TOP, I could see 1 PID safe_asterisk and almost 10 PIDs asterisk -vvvg -c even when there was no call. However, for the other couple, I started safe_asterisk and when I ran PS or TOP, I could see 1 PID safe_asterisk and only 1 PID asterisk -vvvg -c, they are all with Pentium Xeon chip and 512M RAM, no difference in Hardware, and all running the same version of Asterisk on Redhat 7.3. Does anyone have any idea why there is a difference please? The reason that it is important as well is because each asterisk -vvvg -c is taking up certain memory and with 10 (more when there are calls) or more of these, I am running into memory problem. However, in the other case, no matter how many calls I have, I only see 1 PID of asterisk -vvvg -c and seems that I have less of a memory problem. Any feedback to help solve the mystery? Thanks TOm --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.558 / Virus Database: 350 - Release Date: 1/2/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] capacity testing
Title: RE: [Asterisk-Users] capacity testing Hi all, and Jesse 1. So, you did get the experience of crashing all of a sudden with the "Disconnected from Asterisk server" error message. I got both this and the segmentation error when crashing. I am running the version of asterisk, libpri and zaptel updated to about 5 days ago, but I have had tested Asterisk for more than a month already and needless to say I have had this experience since Day 1, meaning it has always been a problem even in the previous revisions. Henceforth, I feel that it is an intrinsic Asterisk problem, rather than just the problem with specific versions / revisions. I have posted this problem a few times before, I feel that this is a major problem but surprisingly, I was not getting any feedback at all. I have this feeling that more than 90% of the Asterisk community is using the system for PBX application rather than VOIP, may be, just may be, Asterisk has not been tested with a good number of simultaneous calls. 2. I am using Xeon 2.6G chip, much more powerful than yours, I have not got any problem with CPU usage, at least not during the time that I was watching. The thing is when I start 'safe_asterisk' , I could see when doing a PID, 1 "safe_asterisk" PID session and at least 10 (or more especially when there are more calls) "asterisk -vvvg -c" PID session. Each session takes up about 18M to 20M RAM, when that is why I am seeing all very high memory usage. How many sessions of Asterick do you see running after you loaded it? 3. Are you running H323 (Jeremy) and OH323 (Michael)? I am running Jeremy's and have had this inbound H323 problem. I tried OH323 (Michael) as well, but for some reasons, I am getting this false connect signal, that is, I made an outbound H323 call to a CiscoAS5300 for example, I heard the ring and immediately on my "Asterisk", it showed call answered when it was still ringing. Do you have that experience?? What setting you have if you do not have that experience? 4. Lets talk off list at [EMAIL PROTECTED]. Thanks Tom -Original Message-From: Jesse Peterson [mailto:[EMAIL PROTECTED]On Behalf Of Jesse PetersonSent: Thursday, January 15, 2004 8:21 PMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] capacity testing Sorry for the malformed mail. My responses are marked with '***' below. jesse == Hi,I am a newbie in Asterisk as well, intending to use it in a similar way asyou are, communicating with AS5300 as well as other gateways includingMAXTNT.I have had similar, but yet different experiences than yours.1. Asterisk does crash with the number of calls, but in my case, about orless than 20 calls, then I would get either a Segmentation Error and thencrashed OR it would just crash saying "Disconnected from Asterisk server"all of a sudden. *** The crashesI experienced were fairly transparent. When I had the console (asterisk -r) running, I saw the 'Disconnected' message you mention.2. I am using Pentium Xeon chip and hence more powerful than yours with 512MRAM, my CPU usage has always been low, however, I have not had a chance tolook at the CPU usage just before crashing, but all the time that I waslooking, it has been low. Rather the MEMORY has always remained high at 450Musage even with no calls. This is a different experience as compared toyours.*** A Xeon of the same speed (800mhz in my case) should certainly perform better - lower, I don't know. I find it a little odd that you experienced basically the opposite of my testing. What version are you running? 3. I have also noticed that with more calls, and after a certain randomperiod of time, any H323 calls going into the Asterisk would fail, my AS5300and MAXT TNT would get their calls all rejected from Asterisk. However,Asterisk was still running at the time and I could actually call in and outthe zap interface and outbound H323 from Asterisk was not a problem. Itseems that something got hung with H323, causing inbound H323 calls intoAsterisk to all fail. In this situation, I would have to stop the Asteriskand rerun it to fix the problem.*** Interesting - I have not experienced that (yet...). 4. I have not tried the 0.7.0 version, but with existing version, I am notgetting reliable and stable system, nothing close to Cisco and Lucent whichare rock solid. However, I really love the power and the features ofAsterisk, and I remain in good faith to see improvements.Any associate out there who can shed some lights into this? I am rathercurious as to why I seem to be using up all memory although I am not runningany unnecessary processes, or should I actually disable all modules, otherthan really necessary ones to support VOIP? *** Since you and I are working in what sounds to be a familiar environment, maybe we should communicate about our test scenarios, etc off list to both help
[Asterisk-Users] crontab
Dear All,I have had a problem that I have posted before, the asterisk kept crashingon me. I have thought that may be before the problem is resolved, I couldtry to implement a cronjob to run /usr/sbin/safe_asterisk, and if Asteriskis not running at that time, it will start it automatically.I have read through documents and I have tried a few things. I have triedadding a file called start_asterisk under /etc/cron.d directory and have put15 * * * * /usr/sbin/safe_asterisk, I have even tried using/var/spool/cron/root and inputted the same command line. I even triedactually putting this command line in the crontab file under /etc/ but Ijust could not get the Asterisk to start automatically. Finally, the crondseemed to be doing something, but when I received mail from crond, it wastelling me a message saying " Asterisk exited on code 127, Asterisk died oncode 127 ". So, I changed the command line to 15 * * * * root/usr/sbin/safe_asterisk. Now as per the mail, it is telling me that"Asterisk exited on code 1, Asterisk died on code 1".Ladies and Gentlemen, can anyone please help and let me know what is the wayto start Asterisk automatically using a cronjob, thanksTommy
[Asterisk-Users] (no subject)
Dear All, I have had a problem that I have posted before, the asterisk kept crashing on me. I have thought that may be before the problem is resolved, I could try to implement a cronjob to run /usr/sbin/safe_asterisk, and if Asterisk is not running at that time, it will start it automatically. I have read through documents and I have tried a few things. I have tried adding a file called start_asterisk under /etc/cron.d directory and have put 15 * * * * /usr/sbin/safe_asterisk, I have even tried using /var/spool/cron/root and inputted the same command line. I even tried actually putting this command line in the crontab file under /etc/ but I just could not get the Asterisk to start automatically. Finally, the crond seemed to be doing something, but when I received mail from crond, it was telling me a message saying " Asterisk exited on code 127, Asterisk died on code 127 ". So, I changed the command line to 15 * * * * root /usr/sbin/safe_asterisk. Now as per the mail, it is telling me that "Asterisk exited on code 1, Asterisk died on code 1". Ladies and Gentlemen, can anyone please help and let me know what is the way to start Asterisk automatically using a cronjob, thanks Tommy
[Asterisk-Users] RE: Asterisk problem
Dear All,
I am a new user of Asterisk interested in setting up a VOIP network based on
Asterisk. I have deployed a few Asterisk servers running on T400P and have
started a few weeks ago to run some LIVE traffic on one of the servers. Most
of my current traffic is via H323 to and from other carriers / customers and
I am using the H323 driver written by Jeremy of Nufone.
I have had problems with this, as it seems that as time goes, for instance
after 5 or 6 hours or sometimes 3 to 4 hours or less, the system would not
be able to take any incoming calls via H323 protocol. All the calls from my
H323 customers (they are mostly carriers sending traffic to me from other
gateways such as cisco AS5300) would be rejected and the system just would
not take the calls for some reasons. I could, however, call into the PRI
(Zap interface) and make an outbound H323 calls to other gateways of my
carriers. That is, incoming H323 calls would not work after a certain period
of operation time but outgoing H323 calls would still be possible.
I tried running the asterisk -vvvgcd in some cases in order to capture any
possible error messages and I was fortunate to be looking at it when it
started to fail once and have received twice this message on the screen:
ThreadID=0x001d4c25 assert.cxx(105) PWLib Assertion fail: Operating System
error, file tlibthrd.cxx, Line 644 Error=24
I have got this twice, one about 10 seconds after the other, and then I
tried calling into the system via H323 and it wouldn't take my calls. Ladies
and gentlemen, do you have any idea what it is and how can I get rid of
this problem because it is doing it every like 2 to 6 hours and is killing
me.
I have then downloaded newer version of the Asterisk about 10 days ago, but
then my Asterisk would start to crash on me, the module would just stop
running by itself and I had to restart the Asterisk. Sometimes, it would
just stop running, but 75% of the time, I would see this error message:
Connected to Asterisk CVS-12/06/03-03:06:28 currently running on localhost
(pid = 23720)
-- Remote UNIX connection
localhost*CLI /usr/sbin/safe_asterisk: line 6: 23720 Segmentation fault
asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
Disconnected from Asterisk server
Executing last minute cleanups
[EMAIL PROTECTED] asterisk]# /usr/sbin/safe_asterisk: line 6: 23752
Segmentation fault asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
It seems that it would try to restart but would fail, and then I had to
restart it manually. This Segmentation error happens only with new version
of Asterisk but not with older version (10 days ago), does anyone here have
any of such experience or know why this is happening? Is this because I am
running T400P instead of TE410P hardware and that the newer version of
software is not 100% compatible with the old hardware? The inbound H323
problem, however, exists for both the newer and older version of Asterisk,
and I wonder if anyone has had this experience before. Is this pertaining to
the hardware as well or is there any other reason/
Ladies and Gentlemen, any feedbacks and advices would be appreciated. Many
Thanks
Tommy Chan
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[Asterisk-Users] Asterisk problem
Dear All,
I am a new user of Asterisk
interested in setting up a VOIP network based on Asterisk. I have deployed a few
Asterisk servers running on T400P and have started a few weeks ago to run some
LIVE traffic on one of the servers. Most of my current traffic is via H323 to
and from other carriers / customers and I am using the H323 driver written by
Jeremy of Nufone.
I have had problems with
this, as it seems that as time goes, for instance after 5 or 6 hours or
sometimes 3 to 4 hours or less, the system would not be able to take any
incoming calls via H323 protocol. All the calls from my H323 customers (they are
mostly carriers sending traffic to me from other gateways such as cisco AS5300)
would be rejected and the system just would not take the calls for some reasons.
I could, however, call into the PRI (Zap interface) and make an outbound H323
calls to other gateways of my carriers. That is, incoming H323 calls would not
work after a certain period of operation time but outgoing H323 calls would
still be possible.
I have then downloaded newer
version of the Asterisk about 10 days ago, but then my Asterisk would start to
crash on me, the module would just stop running by itself and I had to restart
the Asterisk. Sometimes, it would just stop running, but 75% of the time, I
would see this error message:
Connected to Asterisk CVS-12/06/03-03:06:28
currently running on localhost (pid = 23720) -- Remote
UNIX connectionlocalhost*CLI /usr/sbin/safe_asterisk: line 6: 23720
Segmentation fault asterisk ${ASTARGS}
1/dev/${TTY} /dev/${TTY}Asterisk ended with exit status
139Asterisk exited on signal 11.Automatically restarting
Asterisk.
Disconnected from Asterisk
serverExecuting last minute cleanups[EMAIL PROTECTED] asterisk]#
/usr/sbin/safe_asterisk: line 6: 23752 Segmentation
fault asterisk ${ASTARGS} 1/dev/${TTY}
/dev/${TTY}Asterisk ended with exit status 139Asterisk exited on
signal 11.Automatically restarting Asterisk.
It seems that it would
start to restart by would fail, and then I had to restart it manually. This
Segmentation error happens only with new version of Asterisk but not with older
version (10 days ago), does anyone here have any of such experience or know why
this is happening? Is this because I am running T400P instead of TE410P hardware
and that the newer version of software is not 100% compatible with the old
hardware? The inbound H323 problem, however, exists for both the newer and older
version of Asterisk, and I wonder if anyone has had this experience before. Is
this pertaining to the hardware as well or is there any other
reason/
Ladies and Gentlemen, any
feedbacks and advices would be appreciated. Many
Thanks
Tommy Chan
[Asterisk-Users] (no subject)
Dear All, I am going to deploy a VOIP network here in Canada with nodes all over town. This is for long distance services and hence would need a good reliable solution. I have looked into * and am very interested in it with all the value-added features as well as its capability to do H323 and SIP. I understand that a good portion of VOIP operators in the industy is converting to SIP but there are still more H323 VOIP operators out there. That is one reason why I am interested in this solution as it can do both. Most of my customers and carriers are still on H323 and hence I would need to make sure that the * is able to talk with most H323 gateways out there in the market, such as cisco and quintum. There are two things I would like your comments on to make sure that the system will serve my purpose. I need my carrier customers to be able to send calls to my * via H323 VOIP from most H323 gateways and the * to pass through and route the calls to appropriate carriers with cisco or other gateways via VOIP, the process will be H323 mostly and some SIP, working with G723 and / or G729 which are what most VOIP operators are using. In this situation, the inbound and outbound are both via VOIP, either with H323 or SIP with both G723 and / or G729 codec. Is the application, is * compatible with most gateways out there? Another situation is carrier customers will send calls to me to be terminated on the TDM circuits OR I originate calls from the TDM side to be terminated on any gateways on the carrier side. In such, there will be a encoding / decoding process. Would I be able to send calls to most other gateways from my TDM circuits via VOIP H323 or SIP on g729 / G723 codec? Likewise for customers calling into my *, would the * be able to decode G729 and G723 to pass calls to the TDM side? I would love to get some feedbacks and advices (which is greatly appreciated) from you all who have had experiences in doing this before, thanks amillion. Tommy Chan
