Tzafrir Cohen wrote:
I'd like to get your input regarding
http://bugs.digium.com/view.php?id=9645
The patch adds an extra span method:
int (*sync_tick)(struct zt_span *span, int is_master);
The zaptel sync master span will call it to any span (in which it is not
NULL) at each of its ticks.
Latest versions of TE4xxP allows to daisy-chain timing from single source to
other cards. And IMHO it should use single interrupt per all ports joined by
this daisy-chain.
WBR,
Paul.
- Original Message -
From: "Andrew Kohlsmith" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, February 13, 2
As I remember there is special device driver called something like
/dev/zap/timer which have some special ioctl()s to handle timing for Asterisk.
If your driver will provide the same interface - you won!
WBR,
Paul.
- Original Message -
From: Paulo Garcia
To: Asterisk Developers
Hi,
/dev/zap/ used mostly for non-specific tools (like dd
if=/dev/zap/1 of=/tmp/chan1_mon.alaw ;-) ). Real channel drivers and
zaptel-specific tools should use /dev/zap/channel interface instead. You can
easily update chan_ss7 to use /dev/zap/channel mechanism (just look how it
is implemented
Hello,
Shutdown (by ztcfg -s) is required for dynamic spans (ztd-eth, ztd-loc).
WBR,
Paul.
- Original Message -
From: "Kevin P. Fleming" <[EMAIL PROTECTED]>
To: "Asterisk Developers Mailing List"
Sent: Thursday, September 28, 2006 10:25 PM
Subject: Re: [asterisk-dev] shutting down zap
Hello,
Revision 4388 should also fix compiler warning.
WBR,
Paul.
- Original Message -
From: "Vlasis Hatzistavrou" <[EMAIL PROTECTED]>
To: "'Asterisk Developers Mailing List'"
Sent: Thursday, September 28, 2006 8:36 PM
Subject: RE: [asterisk-dev] Checking h323.h presence... no???
> H
Hello,
- Original Message -
From:
Morten Isaksen
To: Asterisk Developers Mailing List
Sent: Wednesday, September 20, 2006 12:31
AM
Subject: Re: [asterisk-dev] Trunk 43282
does not build
And I have some problems with rev 43284.
When I run configure it
Hello,
Dan Austin wrote:
> I see that Mattf just commited some changes to the configure
> script with a comment "Hopes my branch doesn't break", so this
> might not be unexpected
There was try to merge my chan_h323-live. Because I uses different version of
autoconf tools configure should be
r in build on latest trunk
> I did - make clean;make update;./configure;make menuselect
>
> Paul Cadach wrote:
> > Hi,
> >
> > Do menuselect/configure too.
> >
> >
> > WBR,
> > Paul.
> >
> > - Original Message -
>
Hi,
Do menuselect/configure too.
WBR,
Paul.
- Original Message -
From: "Julian Lyndon-Smith" <[EMAIL PROTECTED]>
To: "Asterisk Developers Mailing List"
Sent: Sunday, September 17, 2006 1:49 PM
Subject: [asterisk-dev] error in build on latest trunk
> Looks like something broke:
>
>
Hello,
E1s usually uses HDB3 line coding (or AMI line coding) with optional CRC4
checking, so B8SZ coding is totally wrong.
span=1,1,0,ccs,hdb3,
span=2,2,0,ccs,hdb3,
span=3,3,0,ccs,hdb3,
span=4,4,0,ccs,hdb3,
>From top of /etc/zaptel.conf:
# The framing is one of "d4" or "esf" for T1 or "cas" or "
lines in http://karlsbakk.net/asterisk/pridebug.
> 1.html differ between a normal SETUP and a diverted SETUP, stuff
> asterisk seems unable to parse
>
> thanks
>
> roy
>
> On 29. aug. 2006, at 07.21, Paul Cadach wrote:
>
> > Hello,
> >
> > Your t
Hello,
Your trace shows you don't have RDNIS number in FACILITY message (probably
passing of RDNIS information is prohibited by network).
WBR,
Paul.
- Original Message -
From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
To: "Asterisk Mailing List"
Sent: Monday, August 28, 2006 10:03 PM
Hello,
Russell Bryant wrote:
> > Alternatively, since we have ast_malloc/ast_calloc, should we have ast_free
(not in 1.2, of course)?
>
> I suppose. However, I'm not sure how beneficial it would be other than
> for matching the naming convention of the allocation wrappers. There is
> obviously n
Hello,
Chan Kwang Mien wrote:
> IP Phone A <--> Asterisk IP PBX <---> Analog Phone B
>
> In my tests, echos were generated by IP Phone A when I turned on the
> speaker. As was pointed out, zaptel could not cancel these echos because
> the echo was not received at the zaptel interface, rather o
Hello,
Simone Cittadini wrote:
> Eric "ManxPower" Wieling ha scritto:
>
> > Playback, Background, and most any function that sends audio to the
> > caller will issue an Answer first unless you tell it not to. (see
> > show application)
> >
> There is no Playback, Background or such, only a Dia
Hi,
Anton wrote:
> Jul 3 09:31:45 WARNING[14626]: channel.c:787
> channel_find_locked: Avoided deadlock for '0x822cb08', 10
> retries!
Typical deadlock situation. Please, enable locking traces and check what is hold
the lock (where the lock was made).
WBR,
Paul.
_
Hello,
Johansson Olle E wrote:
> > Are you sure? All channels that supply 'ast_rtp_bridge' as their
> > bridge method should already support it.
>
> That was news to me. Cool if that's the case!
> Never seen or heard about it before.
>
> I stand corrected.
I just acknowledge we have H.323 native
Hello,
Ken Chan wrote:
> Now I have more questions:
> a) Any idea what type of H.323 devices (or softphone) support
> "Empty TCS" feature?
Standard OpenPhone (and other OpenH323-based solutions) supports for Empty TCS.
> b) When are you going to release that native bridge for h.323?
Until ano
Hello,
Ken Chan wrote:
> Currently, when the call is between SIP Phone and Netmeeting (H.323),
> all the RTP packets are going through Asterisk.
Netmeeting isn't support for "Empty TCS (Terminal Capability Set)" feature yet,
so RTP traffic will go through Asterisk
anyway independedly on native b
Hello,
Kevin P. Fleming wrote:
> > A packet is sent every 15s, causing Asterisk to write the same line in
> > the logs every 15s too. The following hack ignores those packets. I do
> > not know whether this should be included in the mainstream sources or
> > not. I will report this to Free Telecom
Hi,
Olle E Johansson wrote:
[skipped]
> During my recent tests with video phones (thanks Grandstream and
> Foniris!) I have found out
> that we have a list of things to do. I have also found out that there
> are a lot of developers out there
> that have done it already - meetme with selectable vid
Hello,
Koopmann, Jan-Peter wrote:
> It is one thing to say "Hey this does not work, it is a bug" and another to
say "Hey this does not work as expected, I created a small enhancement, see the
patch attached" like it was in this case.
[skipped]
You forget to remember about just extending support
Hello,
This is -user question. You should disable
VAD (Voice Activity Detection) on your gateway - Asterisk is not
VAD-friendly.
WBR,Paul.
- Original Message -
From:
wei
lee
To: asterisk-dev@lists.digium.com
Sent: Monday, March 27, 2006 7:20
AM
Subject: [asteri
Hello,
Roy Sigurd Karlsbakk wrote:
[skipped]
> These IRQ storms are only happening on crappy network hardware. My
> testing was one with intel gigabit NICs with large buffers,
> effectively producing < 100 interrupts per second. kernel profiling
> showed time was indeed spent in system calls
The
Hello,
Sergey Kuznetsov wrote:
[skipped]
> > OpenH323 have some deadlock issues I tried to solve about a year ago (you
can find those patches somewhere at
> > bugs.digium.com). I'm not sure my patches are applied to latest openh323
tree, but small fixes to pwlib already have
> > ones...
>
> What w
Hello,
anthony thomas wrote:
> We are testing the t38passthrough branch in our
> gateways and can not fully understand why the RTP
> stream has to go througth the * box disabling native
> RTP bridging.
This feature is not being developed/fully tested to be published. Be patient
and some time it
Hello,
In addition to Kevin P. Fleming message:
[skipped]
> If the final end party does not send ALERTING, and the network does not
> send SESSION PROGRESS with inband audio available, then there is no way
> for Asterisk to know that the call is ringing the endpoint.
If and only if in-band tone a
Hello,
As decided by me and Tony Mountifield, I'd
updated ztdummy to provide more accurate timing with minimal jitter - bug #6631
at bugs.digium.com (http://bugs.digium.com/view.php?id=6631).
Please, provide feedback if possible.
WBR,Paul.
___
-
Hello,
Olle E Johansson wrote:
> The RTCP branch includes improved support of RTCP, but also a
> reporting facility we do not use currently. Would it be useful to add
> this to a channel variable - or even better a CDR variable - so you
> can add it to CDRs and make reports based on it?
Mostly RT
Hello,
James Harper wrote:
[skipped]
> > Paul Cadach mentioned something about a jitter buffer for TDMoE, but
> > I don't know whether he was talking about an idea or some real code.
> >
> > I'm also thinking about a jitter buffer, but it's tricky. I m
Hi,
Alistair Cunningham wrote:
> Paul,
>
> Thank you for following up. We currently have "switchtype = national".
> Looking at the code, I see in transmit_display():
>
> if(pri->switchtype != PRI_SWITCH_EUROISDN_E1) {
> ie->data[0] = 0xb1;
> ++i;
> }
>
>
Hello,
Dinesh Nair wrote:
> On 02/10/06 22:16 Alexander Lopez said the following:
> > Does the Callback happen after a specifed time period or immediately. We
>
> it should happen after a specified time period and retry for a specific
> number of times. i am not yet sure if infinite retries are re
Hello,
Alistair Cunningham wrote:
[skipped]
>The solution is to ensure your equipment is set to the proper
> switch type, namely NI-1 or NI-2 (as opposed to a Nortel switch type
> like DMS-100).
Current libpri code isn't sends DISPLAY IE for NI-1 or Q.931 switchs, and isn't
uses characte
Hi,
Tony Mountifield wrote:
[skipped]
> When I start up zaptel, with wcfxo, ztd-eth, ztdynamic and zaptel
> loaded, my monitoring machine initially sees one TDMoE packet per
> millisecond, as expected. After between 75 and 90 packets, there is
> suddenly a pause for anything from 80ms to 800ms, af
Hi,
Is location of libh323 included in content of /etc/ld.so.conf file to make
loader to be able to find shared object
files?
WBR,
Paul.
- Original Message -
From: "Hoai-Anh Ngo-Vi" <[EMAIL PROTECTED]>
To:
Sent: Friday, January 27, 2006 7:28 PM
Subject: [asterisk-dev] chan_h323.so
>
Hello,
Looks like your link to Avaya is down.
Probably there is interface down at Avaya side or link is physically
disconnected.
WBR,
Paul.
- Original Message -
From:
Chee
Foong
To: asterisk-dev@lists.digium.com
Sent: Wednesday, October 05, 2005 4:53
PM
Hello,
Q.SIG is based on Q.931, so basic Q.SIG support is already available. Support
for advanced Q.SIG features (including other Q.931-based signallings) like
supplementary services could be developed on per-request basis, just because no
real Q.SIG hardware is available for testing.
Anyway be r
Hello,
Redirection reason is not PRI-specific. For example, all terminal channels (FXS,
Skinny) and some trunking channels (like H.323) is possible to generate real
reason of redirecton. Also, SS7 is redirection-reason capable too.
IMHO at least trunking interfaces (IAX, H.323, PRI, SS7, etc.) sh
Hello,
Michael Beukman wrote:
> I have configured zaptel.conf with this:
>
> span=1,0,0,ccs,hdb3
> clear=1-31
>
> Does this mean I've merged the timeslots into one stream?
You're correct.
> and how then do I go about capturing the "master" channel?
ztcfg -vv shows master channel like this:
Hello,
divertingLegInformation2 support is included with current CVS-HEAD version of
libpri.
WBR,
Paul.
- Original Message -
From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
To: "Asterisk Developers Mailing List"
Sent: Wednesday, June 01, 2005 5:25 PM
Subject: [Asterisk-Dev] Fwd: zap
Hello,
Jeremy McNamara wrote:
> Andreas Czerniak wrote:
> > this problem was solved by an patch by Jose Carlos Garcia Sogo:
> >
> > http://www.beronet.com/bugs/bug_view_page.php?bug_id=046
> >
> > And it only occurs in asterisk stable versions not in the head revision.
>
> *Gasps* "Stable" c
Hello,
Read channels/h323/README and follow instructions. You should upgrade your
PWLib/OpenH323 libraries.
WBR,
Paul.
- Original Message -
From: "Ganbold Tsagaankhuu" <[EMAIL PROTECTED]>
To: "Jeremy McNamara" <[EMAIL PROTECTED]>
Cc:
Sent: Friday, May 13, 2005 12:42 PM
Subject: Re: [
Hello,
Michael Manousos wrote:
> > The problems mostly relies on OpenH323 - there is deadlock condition, check
> > #3967. Also, under high load single
cleaner
>
> I have also notice them, but only with the Pandora or above versions.
> That's the main reason that asterisk-oh323 has stuck with the
Hello,
Adam Goryachev wrote:
> On Fri, 2005-05-06 at 15:41 +0600, Paul Cadach wrote:
> > User-space only software which isn't depends on any sort of additional
> > support from kernel level.
>
> Why?, it is still ALL software...
>
> Given that some closed sourc
Hello,
Jerris, Michael MI wrote:
> Major Bugs:
> 4164 g.729 codec one way audio in chan_h323 with Cisco CCM
Patch is available, waiting for commit.
WBR,
Paul.
___
Asterisk-Dev mailing list
Asterisk-Dev@lists.digium.com
http://lists.digium.com/mailma
Hello,
Adam Goryachev wrote:
> On Fri, 2005-05-06 at 14:41 +0600, Paul Cadach wrote:
> > Removing dependencies on zaptel should raise Asterisk as real SOFT switch,
>
> That is just a definition problem how do you define "real soft
> switch"??
User-space only sof
Hello,
Adam Goryachev wrote:
> Either way, I don't see that this should be required for 1.2.0 release.
> I would like to see more work done on exposing the call quality
> (jitter/etc) values for a 1.2.0 release though.
Removing dependencies on zaptel should raise Asterisk as real SOFT switch,
ea
Hello,
Chih-Wei Huang wrote:
> Hello all,
> I encountered another problem about H.323 channel driver.
> The scenario is, A calls Asterisk PBX, and then dials B.
> B then sends ringback tone by early media (via faststart or so).
> However, PBX said
>
> chan_h323.c:703 oh323_write: Asked to transmit
Hello,
Why FastStart handshake for incoming call couldn't be acknowledged on CALL
PROCEEDING message?
WBR,
Paul.
- Original Message -
From: "Paul Cadach" <[EMAIL PROTECTED]>
To: "Asterisk Developers Mailing List"
Sent: Friday, March 25, 2005 5:57 PM
Hello,
Matthew wrote:
> Their original logic was that a user didn't know who they were transferring
> from extension to extension or when a call came back from parking. Changing
> the CallerID to the destination fixed that problem. It makes certain sense,
> but boy did it screw up a lot of things
Hello,
Solaris usually have standard libraries with cutted functionality (I had such
type of problems with Solaris some time
ago, missed function asprintf()). Looks like Asterisk needs to have own
"compatibility" library... :(
WBR,
Paul.
- Original Message -
From: "Brian McCrary" <[EM
Hello,
Olle E. Johansson wrote:
[skip]
> Adding "Accept: application/ms-excel" will not give you output
> in Microsoft Excel format, sorry twisted ;-)
MS Excel (at least 2K and higher) have possibility to convert
specially-formatted HTML data into workbook. I've uses
this feature to make huge Ex
Kevin P. Fleming wrote:
> I've thought about this too, but I think the overhead of every RTP
> stream using the timer separately would just be too much. I think the
> best solution is for rtp.c to create a thread when it loads, use the
> zaptel timer to clock that thread, and have all outbound RTP
Hello,
Matthew Boehm wrote:
[skip]
> If a call comes into our switch as 8005551212, that is the "destination".
> Most of the time, an 800 number gets translated. This becomes the "ringto".
> Both are necessary for billing purposes. Sometimes we have triple
> translation, where an 800 number gets
Hello,
Kai Militzer wrote:
[skip]
> In my opionion it would make great differences in billing. A glitch of a
> few mili-seconds at hangup of a call, may result in a difference of one
> full second between my * CDRs and those of my telco.
Difference in call duration registered at directly connecte
Hi,
Probably you need to manual unlock channels when ones isn't needs anymore by
ast_mutex_unlock(&c1->lock) or
ast_mutex_unlock(&c2->lock), at least after call to ast_bridge_call(). This
should help a little.
Also, check that ast_moh_stop(), ast_deactivate_generator()4,
ast_channel_make_compa
Hello,
- Original Message -
From: "Maurizio Marini" <[EMAIL PROTECTED]>
To:
Sent: Thursday, January 13, 2005 8:52 PM
Subject: [Asterisk-Dev] possible bug in chan_sip
> in chan_sip.c line 1137:
[skip]
> should be (conceptualy):
>while(tmp) {
> if (strcasecmp(tmp
Hello,
Andrew Kohlsmith wrote:
> I found the ITU's documents completely and totally worthless. There was *no*
> actual protocol layout, just highlevel hand-waving in all the documents I
> downloaded.
>
> I found MUCH better info (along with bit-level descriptons) on the CAPI
> website.
Andrew, y
Hi,
> Matt Riddell wrote:
>
> > In that case I'd recommend the quad span E1/T1 card which is available
> > now. (also seeing as you stated you needed 3-4, this will be a lot
> > cheaper)
>
> ...and introduce a single point of failure, as well.
And generates much less interrupts per second, becau
Hi,
Michael Sandee wrote:
> When writing a module which does some magic on unload_module() I ran
> into some trouble...
>
> In contrary to what include/asterisk/module.h says, unload_module() does
> not get called during exit, only when unload is called from the CLI.
Module unloading at exit is e
Hi,
Steve Underwood wrote:
> The *appearance* of DSPs being expensive is an artifact of the way the
> industry runs. DSP cards are priced at an extreme multiple of their
> actual cost, leading to a vicious circle - DSP cards are expensive so
> they have a small market. Why are they expensive? beca
Hello,
- Original Message -
From: "Klaus-Peter Junghanns" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, June 29, 2004 3:18 PM
Subject: Re: [Asterisk-Dev] PRI U2U display messages
> Hi,
>
> you cannot send INFORMATION messages with DISPLAY IEs to the network.
> However you
Hi,
- Original Message -
From: "Storer, Darren" <[EMAIL PROTECTED]>
To: "[EMAIL PROTECTED] Digium. Com" <[EMAIL PROTECTED]>
Sent: Friday, May 07, 2004 4:21 PM
Subject: Re: [Asterisk-Dev] About OpenSS7 integration to Asterisk
[skipped]
> BFGB> "Asterisk soft-HDLC code could not handle SS
Hi,
> > Also, I don't think passing additional 10 kbps for 100 Mbit/s network is so
> > significant.
>
> Ahem.. Thats nice for you to only have to think about using Asterisk on
> Ethernet, however unfortunately some of us are out in the "real
> world" (substitute "third world" if you prefer) and d
Hi,
> I used firefly, in iax mode.
>
> I used ethereal to capture the data.
>
> 33 bytes for 20 ms of audio frame.
>
> 4 bytes of iax header.
Plus 8 bytes - UDP header, 20 bytes - IP header, 14 bytes - EthernetII header. Totally
- 79 bytes. When the same packet
goes through PPP it don't hav
Hi again,
- Original Message -
> I do appreciate your comments on the reorganisation required in
> chan_h323.c Is it possible for you to detail what must be done?
> Or, point me to a doc which states what order things have to happen in?
I had used callgen323 on single-CPU machine to ma
Hi,
- Original Message -
> a)the lack of response from others on this list who have been involved
> with chan_h323.c There is knowledge there that I want to tap into.
> To be honest, I am not keen to "reinvent the wheel". I want to learn
> from others, and learn what trials/tribula
Hi,
Check at the end of right column at page 2 of pointed manual:
"... Two forms of over-voltage protection are provided, one that permits 5V
compliance, and one that does not. For 5V
compliance, a zener-like structure connected to ground turns on when the output rises
to approximately 6.5V. Whe
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