Re: [asterisk-users] asterisk 16 and zapata
zaptel.conf should be on /etc not /etc/asterisk On Fri, Sep 12, 2008 at 11:06 PM, Jonn R Taylor [EMAIL PROTECTED]wrote: Should have been 1.6.0rc6. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Jonn R Taylor Sent: Friday, September 12, 2008 4:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 16 and zapata Asterisk 1.6rc4 will only use dahdi. I just went though this on my test system. Jonn -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, September 12, 2008 10:12 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk 16 and zapata On Fri, Sep 12, 2008 at 11:09 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, Sep 12, 2008 at 11:07 AM, hh174 [EMAIL PROTECTED] wrote: Asterisk 1.6 installed with last zaptel... On cli, when typing zap show channels, I get No such command 'zap show channels' (type 'help zap show' for other possible commands) Help doesn't help, of course... I have a zaptel conf on the /etc/asterisk... Any Idea? Olivier Zaptel is not loading. What do you see in /var/log/messages? What is the output of ztcfg -vv? Have you recently upgraded the kernel? If so, you need to rebuild and install zaptel against the new kernel. Thanks, Steve Totaro Another thought since I don't use 1.6 but could the correct syntax be core zap show channels? Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [FreeBSD 6.3/Ports] Make does nothing
On Sat, 13 Sep 2008 00:44:28 +0200, Vincent [EMAIL PROTECTED] wrote: I updated the Ports collection to compile the latest Asterisk, but after running make config, make just returns without doing anything: For those having the same problem: make clean ; make config ; make ; make deinstall ; make reinstall does the trick. I shouldn't have expected csup to remove stale stuff from previous compilings. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MoH with an Aastra 9112i
I've run into this same problem with the Counterpath eyeBeam phones. The only solution I could come up with was to disable re-INVITEs to that the media path pivot never happens when the user is put on hold. If in doubt, send a packet capture of the scenario you're describing. If you don't want to publicise IP information, you can send it to me privately off-list; I have no interest in your network, or disseminating details thereof. Trevor Peirce wrote: Hello, I have some Aastra 9112i's in production that almost function flawlessly. The problem I'm having is when a caller is put on hold they do not hear hold music. If they are on hold for too long (~ a minute?) they are hung up on. All other phones including Aastra 480i and Sipura/Linksys ATAs all seem to be working fine. Is this a quirk anyone else has experienced? Any suggestions on what might need to be tweaked? Thanks, Trevor ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP
Hi, Just to review the test I did: ---SIP extension-- Trunk - | SIPp |---| Asterisk 1 || Asterisk 2 | ------ -- Both Asterisk boxes are virtual machines in VirtualBox and version 1.4.21.2. I generated calls using SIPp, and I monitored the cpu utilization in the Asterisk 1 with top. I compare the cpu utilization when I used IAX and when I used SIP as Trunk protocol. The following are the results (averages) I got with ulaw codecs in both sides: Calls IAX SIP 4 6,0 1,8 10 9,2 4,6 20 19,58,6 30 28,213,5 34 36,916,2 40 38,219,5 50 36,924,3 As you can see, IAX almost doubles the cpu cycles. I repeated the test using gsm as the trunk codec, and in this scenario IAX shows a better performance (Sip extension continues with ulaw): IAX SIP 1,8 25,7 12,841,5 29,247,0 45,769,4 54,278,5 53,383,3 65,787,1 And finally I repeated with iLBC in the trunk, and SIP won again: IAX SIP 8,5 9,4 29,314,5 57,623,6 74,437,3 84,341,5 -- 51,2 -- 67,0 Does this makes sense? Any feedback? Has anybody done similar test for comparison? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming MoH on 1.4
Nope, doesn't seem to work. all I hear is deadair. Original Message Subject: Re: [asterisk-users] Streaming MoH on 1.4 From: "Mark Hamilton" [EMAIL PROTECTED] Date: Sun, September 14, 2008 8:48 pm To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Let me test it out.. Original Message Subject: Re: [asterisk-users] Streaming MoH on 1.4 From: "Klaverstyn, David C" [EMAIL PROTECTED] Date: Sun, September 14, 2008 8:37 pm To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comThis works for me. [SkyFM-80s] mode=custom application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s http://160.79.128.62:6618 [SkyFM-HotHits] mode=custom application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s http://160.79.128.62:6628/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Mark Hamilton Sent: Monday, 15 September 2008 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Streaming MoH on 1.4I'm still searching, but can't find anything anywhere other than the part that just doesn't work which I've tried already. Original Message Subject: [asterisk-users] Streaming MoH on 1.4 From: "Mark Hamilton" [EMAIL PROTECTED] Date: Sun, September 14, 2008 12:56 am To: "Asterisk Mailing" asterisk-users@lists.digium.com Hi, I've looked high and low for any changes that streaming MoH needs on Asterisk 1.4 (.21), followed NerdVittle's article about it (http://nerdvittles.com/index.php?p=92) yet nothing worked. After creating dir stream/ and touch stream.mp3, here's my musiconhold.conf [stream] mode=mp3 directory=/var/lib/asterisk/mohmp3/stream stream = quietmp3:/var/lib/asterisk/mohmp3/stream,http://wbez-sclo.streamguys.us/ ;application=/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s http://wbez-sclo.streamguys.us/ Every write up about this has done either the above, or a variant of the above (like the commented out application line) Nothing works. Using the recommended old version of mpg123 0.59r (which gives that stupid monmp3thread: request to schedule in the past error) When I say it doesn't work, I dial the extension, CLI says started musiconhold, and immediately followed by stopped musiconhold. Sometimes between these two messages there's the request to schedule in the past thing. I know the error/warning is due to ztdummy timing (which I currently don't have right now) but it's still not the big problem. Can somebody please help me out with this? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Streaming MoH on 1.4
You may need to wait up to 45 seconds after asterisk first starts to buffer a bit of the stream. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Monday, 15 September 2008 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Streaming MoH on 1.4 Nope, doesn't seem to work. all I hear is deadair. Original Message Subject: Re: [asterisk-users] Streaming MoH on 1.4 From: Mark Hamilton [EMAIL PROTECTED] Date: Sun, September 14, 2008 8:48 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Let me test it out.. Original Message Subject: Re: [asterisk-users] Streaming MoH on 1.4 From: Klaverstyn, David C [EMAIL PROTECTED] Date: Sun, September 14, 2008 8:37 pm To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com This works for me. [SkyFM-80s] mode=custom application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s http://160.79.128.62:6618 http://160.79.128.62:6618 [SkyFM-HotHits] mode=custom application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s http://160.79.128.62:6628/ http://160.79.128.62:6628/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] On Behalf Of Mark Hamilton Sent: Monday, 15 September 2008 9:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Streaming MoH on 1.4 I'm still searching, but can't find anything anywhere other than the part that just doesn't work which I've tried already. Original Message Subject: [asterisk-users] Streaming MoH on 1.4 From: Mark Hamilton [EMAIL PROTECTED] Date: Sun, September 14, 2008 12:56 am To: Asterisk Mailing asterisk-users@lists.digium.com Hi, I've looked high and low for any changes that streaming MoH needs on Asterisk 1.4 (.21), followed NerdVittle's article about it (http://nerdvittles.com/index.php?p=92 http://nerdvittles.com/index.php?p=92 ) yet nothing worked. After creating dir stream/ and touch stream.mp3, here's my musiconhold.conf [stream] mode=mp3 directory=/var/lib/asterisk/mohmp3/stream stream = quietmp3:/var/lib/asterisk/mohmp3/stream,http://wbez-sclo.streamguys.us/ http://wbez-sclo.streamguys.us/ ;application=/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s http://wbez-sclo.streamguys.us/ http://wbez-sclo.streamguys.us/ Every write up about this has done either the above, or a variant of the above (like the commented out application line) Nothing works. Using the recommended old version of mpg123 0.59r (which gives that stupid monmp3thread: request to schedule in the past error) When I say it doesn't work, I dial the extension, CLI says started musiconhold, and immediately followed by stopped musiconhold. Sometimes between these two messages there's the request to schedule in the past thing. I know the error/warning is due to ztdummy timing (which I currently don't have right now) but it's still not the big problem. Can somebody please help me out with this? Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [U-Boot] [PATCH] 8xx: prevent a machine check in scc_init().
Dear Gary Jennejohn,
In message [EMAIL PROTECTED] you wrote:
Signed-off-by: Gary Jennejohn [EMAIL PROTECTED]
Sorry, but I don't understand what you're doing here, or why.
Why would there be any machine checks in scc_init()? Such problems
have never been repoorted for any systems. Actuallym this part of the
code is one of the oldest in U-Boot - if there were machine check
issues in any configruation, these should be known.
--- a/cpu/mpc8xx/scc.c
+++ b/cpu/mpc8xx/scc.c
@@ -70,6 +70,9 @@ static int scc_recv(struct eth_device* dev);
static int scc_init (struct eth_device* dev, bd_t * bd);
static void scc_halt(struct eth_device* dev);
+/* avoid unnecessary reinitialization of the SCC */
+static int scc_init_completed;
+
int scc_initialize(bd_t *bis)
{
struct eth_device* dev;
@@ -193,6 +196,19 @@ static int scc_init (struct eth_device *dev, bd_t * bis)
volatile immap_t *immr = (immap_t *) CFG_IMMR;
+ /*
+ * This routine is called again and again from eth_init(),
+ * especially when CONFIG_NETCONSOLE is defined and
+ * stdout=nc.
+ * Avoid unneccessary flailing, otherwise we can get a panic here.
+ */
+ if (scc_init_completed) {
+ immr-im_ioport.iop_pcso |= (PC_ENET_CLSN | PC_ENET_RENA);
+ immr-im_cpm.cp_scc[SCC_ENET].scc_gsmrl |=
+ (SCC_GSMRL_ENR | SCC_GSMRL_ENT);
If the initialization has already been done, why would we repeat these
two steps here?
And why exactle these two, and not the others that are performed for a
regular init sequence?
+ return 1;
+ }
+
#if defined(CONFIG_LWMON)
reset_phy();
#endif
I'm not sure if your early return here is actually valid. As you can
see, other actions follow - for example here on the LWMON baord the
PHY reset; I suspect that your change might eventually work on your
current system, but break many others.
Also, I'm not sure if you are4 aware that the eth_init() part may be
called from a interface switching sequence, i. e. when cycling
between several ethenret interfaces. These will have been shut down
before by eth_halt() - skipping the init sequence here seems the
wrong thing to me.
Maybe you could explain what problem you are actually truing to fix,
and especially under which situations you observe machine checks.
Best regards,
Wolfgang Denk
--
DENX Software Engineering GmbH, MD: Wolfgang Denk Detlev Zundel
HRB 165235 Munich, Office: Kirchenstr.5, D-82194 Groebenzell, Germany
Phone: (+49)-8142-66989-10 Fax: (+49)-8142-66989-80 Email: [EMAIL PROTECTED]
The reasonable man adapts himself to the world; the unreasonable one
persists in trying to adapt the world to himself. Therefore all
progress depends on the unreasonable man. - George Bernard Shaw
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