similar to: CallerID Screening Prohibit

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Before I hack out the ',' -> '|' tr in extension.conf parser, any way to escape ',' that I missed? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/ZfuP2TEAILET3McRAjXUAJ0VjuFeABe5jqpSlrBakDC2IMjvrQCfcBYU

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, IAX2 trunking requires zaptel timing. Zaptel timing is provided by Zaptel cards, ztdummy or ztrtc. ztdummy requires usb-uhci and ztrtc can't run on smp systems. So if you only have smp systems with ohci and no zaptel cards (because it's a sip/iax2 gateway) then you're screwed? - -- Regards, Tais M. Hansen

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Seems like Asterisk/chan_sip in some special cases gets it's rtp channels mixed up. I've got a few reports on users hearing someone elses conversation on the line. Could be port problems, but I haven't had time to make any traces or tests yet. Before I start to analyse this periodic problem, I thought I'd just check with the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, When making a call like the one below, I get double DTMF tones on the PSTN side. DTMF tones sent from the PSTN arrives squelched on the SIP side. SIP > Asterisk2 > IAX2 > Asterisk1 > ZAP > PSTN SIP has been configured to use rfc2833 on both the SIP endpoint and the Asterisk. SIP endpoint also suggests a payload value of 101.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm having a problem with the following expression examples. exten => s,1,NoOp($[$[${value} >= 10] & $[${value} < 18]]) exten => s,1,GotoIf($[$[${value} >= 10] & $[${value} < 18]]?3) ${value} is 13 in both examples above. First extension evaluates to 1 while second evaluates to 0 even though it's the same

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Any reason why pickupgroup has been limited to 31? 31 groups are quickly used up when you have multiple companies on the same server. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFBEOMg32si/nlrQ5gRAu3+AJ9FkeGMgb1JaAy2WjY8wBNEsN4WnwCeMFP0

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi. Using inAccess Networks chan_oh323, I'm experiencing some clicks or pops, how can I fix that? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/QMfF2TEAILET3McRAu9zAJwNWtv+QSpka0NGaVk9E/IDHyalhwCgkQME Gynfp5zF0SWZUQEjelp7sBI= =CSqT

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, Asterisk currently uses the timestamps in incoming RTP packets to build outgoing voice frames. Is this true? Would it be possible for me to use i.e. zaprtc as a timing source for the outgoing stream? I.e. in a setup like below I'd like to use zaprtc timing on Ast1 because I don't trust the timestamps coming from

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I try to set up a Linux/Samba box to authenticate users (on Windows 2000 and XP boxes) against a Windows NT4 Primary domain controller but failed with what I tried so far. - - both machines are on the same local network (192.168.17.X) - - the windows box runs NT4. I havn't set up this and I don't know much about it either but I have

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Now that I've been unable to register 2 hardware SIP phones and one software (Kphone), I'm beginning to doubt that chan_sip works at all. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I was just measuring the bitrates of a couple of codecs via iax. I'm getting much higher numbers than expected, so maybe I'm doing something wrong? Measured with iptraf, values displayed are: codec: measured bitrate (bitrate according codec definition) gsm: 52 kbps (13 kpbs) alaw: 154 kbps (?) speex: 57 kpbs (24 kpbs) Seems a little

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, After upgrading the kernel on an Asterisk box, asterisk segfaults on startup. It seems like it's the g729 codec that causes this: #0 0x4015acad in memset () from /lib/libc.so.6 #1 0x4022686a in load_module () at codec_g729b.c:416 #2 0x08054794 in ast_load_resource (resource_name=0x80d1068 "codec_g729b.so") at loader.c:298 #3

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I've been unable to get app_transfer to work, could someone explain how it is supposed to work? Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1 dials ast2 using iax2 and gets instructed to transfer the call to a different extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing happens

Hi, Does Asterisk have a limit to how many NAT'ed SIP clients it supports behind a single IP? I have the weirdest problem ever. I have three SIP endpoints. SNOM phones, if it matters. Their extensions are 200, 201 and 202. Apart from the username/password, the sip entries in sip.conf all have identical configuration. They're all NAT'ed behind the same IP. 200 and 202 registers

Hi again, Well - I didn't see beta8a-2.3.3 in custom dir. Will try. Also I tried to contact Sangoma - they are very fast to answer but main problem is time difference - it's 6 hours between Canada and Europe. Br, dmitry Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK-2600 Glostrup Denmark Phone: +45 70 25 74 74 Fax:???? +45 70 25 73 74 Web: www.comx.dk

when used Posix File System Test Suite(pjd-fstest) to test btrfs, some cases about setfacl failed when noacl mount option used. I simplified used commands in pjd-fstest, and the following steps can reproduce it. ------------------------ # cd btrfs-part/ # mkdir aaa # setfacl -m m::rw aaa <- successed, but not expected by pjd-fstest. ------------------------ I checked ext3, a warning

* Masami Hiramatsu <masami.hiramatsu.pt at hitachi.com> wrote: > Currently the blacklist is maintained by hand in kprobes.c > which is separated from the function definition and is hard > to catch up the kernel update. > To solve this issue, I've tried to implement new > NOKPROBE_SYMBOL() macro for making kprobe blacklist at > build time. Since the NOKPROBE_SYMBOL()

On Tue, 12 Nov 2013 02:18:53 +0900 Masami Hiramatsu <masami.hiramatsu.pt at hitachi.com> wrote: > > > After that we can convert all the rest, probably as part of this series. > > OK, I'll do. :) > BTW, converting all the __kprobes involves many archs, which > kprobes ported. In that case, which mailing-list would better me > to post the series, linux-arch? I

On Fri, Nov 15, 2013 at 5:43 PM, Steven Rostedt <rostedt at goodmis.org> wrote: > On Fri, 15 Nov 2013 04:53:18 +0000 > Masami Hiramatsu <masami.hiramatsu.pt at hitachi.com> wrote: > >> .entry.text is a code area which is used for interrupt/syscall >> entries, and there are many sensitive codes. >> Thus, it is better to prohibit probing on all of such codes

On ISDN lines it's possible to prohibit the presentation of caller id, what if I have a SIP gateway, something like an Audiocodes Mediant 1000. How do I prohibit the caller id presentation on that one? Regards, Kristian -- Kristian Larsson KLL-RIPE Network Engineer Net At Once [AS35706] email: kristian@netatonce.se